Merge branch 'master' into windows

engine/LoadScreenUpdate.cpp

@@ -53,8 +53,8 @@ public:
 	DELEGATE_TO_OBJECT_0( int,		heapchk, m_pMemAlloc );
 	DELEGATE_TO_OBJECT_0( bool,		IsDebugHeap, m_pMemAlloc );
 	DELEGATE_TO_OBJECT_2V(			GetActualDbgInfo, const char *&, int &, m_pMemAlloc );
-	DELEGATE_TO_OBJECT_5V(			RegisterAllocation, const char *, int, int, int, unsigned, m_pMemAlloc );
-	DELEGATE_TO_OBJECT_5V(			RegisterDeallocation, const char *, int, int, int, unsigned, m_pMemAlloc );
+	DELEGATE_TO_OBJECT_5V(			RegisterAllocation, const char *, int, size_t, size_t, unsigned, m_pMemAlloc );
+	DELEGATE_TO_OBJECT_5V(			RegisterDeallocation, const char *, int, size_t, size_t, unsigned, m_pMemAlloc );
 	DELEGATE_TO_OBJECT_0( int,		GetVersion, m_pMemAlloc );
 	DELEGATE_TO_OBJECT_0V(			CompactHeap, m_pMemAlloc );
 	DELEGATE_TO_OBJECT_1( MemAllocFailHandler_t, SetAllocFailHandler, MemAllocFailHandler_t, m_pMemAlloc );

engine/ModelInfo.cpp

@@ -208,8 +208,8 @@ protected:
 public:
 	struct ModelFileHandleHash
 	{
-		uint operator()( model_t *p ) const { return Mix32HashFunctor()( (uint32)( p->fnHandle ) ); }
-		uint operator()( FileNameHandle_t fn ) const { return Mix32HashFunctor()( (uint32) fn ); }
+		uint operator()( model_t *p ) const { return PointerHashFunctor()( p->fnHandle ); }
+		uint operator()( FileNameHandle_t fn ) const { return PointerHashFunctor()( fn ); }
 	};
 	struct ModelFileHandleEq
 	{
@@ -532,7 +532,7 @@ const studiohdr_t *CModelInfo::FindModel( const studiohdr_t *pStudioHdr, void **
 //-----------------------------------------------------------------------------
 const studiohdr_t *CModelInfo::FindModel( void *cache ) const
 {
-	return g_pMDLCache->GetStudioHdr( (MDLHandle_t)(int)cache&0xffff );
+	return g_pMDLCache->GetStudioHdr( VoidPtrToMDLHandle( cache ) );
 }
 
 
@@ -541,7 +541,7 @@ const studiohdr_t *CModelInfo::FindModel( void *cache ) const
 //-----------------------------------------------------------------------------
 virtualmodel_t *CModelInfo::GetVirtualModel( const studiohdr_t *pStudioHdr ) const
 {
-	MDLHandle_t handle = (MDLHandle_t)(int)pStudioHdr->virtualModel&0xffff;
+	MDLHandle_t handle = VoidPtrToMDLHandle( pStudioHdr->VirtualModel() );
 	return g_pMDLCache->GetVirtualModelFast( pStudioHdr, handle );
 }
 
@@ -550,13 +550,13 @@ virtualmodel_t *CModelInfo::GetVirtualModel( const studiohdr_t *pStudioHdr ) con
 //-----------------------------------------------------------------------------
 byte *CModelInfo::GetAnimBlock( const studiohdr_t *pStudioHdr, int nBlock ) const
 {
-	MDLHandle_t handle = (MDLHandle_t)(int)pStudioHdr->virtualModel&0xffff;
+	MDLHandle_t handle = VoidPtrToMDLHandle( pStudioHdr->VirtualModel() );
 	return g_pMDLCache->GetAnimBlock( handle, nBlock );
 }
 
 int CModelInfo::GetAutoplayList( const studiohdr_t *pStudioHdr, unsigned short **pAutoplayList ) const
 {
-	MDLHandle_t handle = (MDLHandle_t)(int)pStudioHdr->virtualModel&0xffff;
+	MDLHandle_t handle = VoidPtrToMDLHandle( pStudioHdr->VirtualModel() );
 	return g_pMDLCache->GetAutoplayList( handle, pAutoplayList );
 }
 
@@ -576,22 +576,22 @@ virtualmodel_t *studiohdr_t::GetVirtualModel( void ) const
 {
 	if ( numincludemodels == 0 )
 		return NULL;
-	return g_pMDLCache->GetVirtualModelFast( this, (MDLHandle_t)(int)virtualModel&0xffff );
+	return g_pMDLCache->GetVirtualModelFast( this, VoidPtrToMDLHandle( VirtualModel() ) );
 }
 
 byte *studiohdr_t::GetAnimBlock( int i ) const
 {
-	return g_pMDLCache->GetAnimBlock( (MDLHandle_t)(int)virtualModel&0xffff, i );
+	return g_pMDLCache->GetAnimBlock( VoidPtrToMDLHandle( VirtualModel() ), i );
 }
 
 int	studiohdr_t::GetAutoplayList( unsigned short **pOut ) const
 {
-	return g_pMDLCache->GetAutoplayList( (MDLHandle_t)(int)virtualModel&0xffff, pOut );
+	return g_pMDLCache->GetAutoplayList( VoidPtrToMDLHandle( VirtualModel() ), pOut );
 }
 
 const studiohdr_t *virtualgroup_t::GetStudioHdr( void ) const
 {
-	return g_pMDLCache->GetStudioHdr( (MDLHandle_t)(int)cache&0xffff );
+	return g_pMDLCache->GetStudioHdr( VoidPtrToMDLHandle( cache ) );
 }
 
 

engine/OcclusionSystem.cpp

@@ -1268,7 +1268,9 @@ void CEdgeList::CullSmallOccluders()
 	// Sort the surfaces by screen area, in descending order
 	int nSurfCount = m_Surfaces.Count();
 	s_pSortSurfaces = m_Surfaces.Base();
-	qsort( m_SurfaceSort.Base(), nSurfCount, sizeof(int), SurfCompare );
+
+	if( m_SurfaceSort.Base() )
+		qsort( m_SurfaceSort.Base(), nSurfCount, sizeof(int), SurfCompare );
 
 	// We're going to keep the greater of r_occludermin + All surfaces with a screen area >= r_occluderarea
 	int nMinSurfaces = r_occludermincount.GetInt();
@@ -1282,7 +1284,7 @@ void CEdgeList::CullSmallOccluders()
 
 	bool *bUseSurface = (bool*)stackalloc( nSurfCount * sizeof(bool) );
 	memset( bUseSurface, 0, nSurfCount * sizeof(bool) );
-	
+
 	int i;
 	for ( i = 0; i < nSurfCount; ++i )
 	{

engine/audio/private/snd_mix.cpp

@@ -1962,7 +1962,7 @@ public:
 	{
 		int m_channelNum;
 		int m_vol; // max volume of sound. -1 means "do not cull, ever, do not even do the math"
-		unsigned int m_nameHash; // a unique id for a sound file
+        uintp m_nameHash; // a unique id for a sound file
 	};
 protected:
 	sChannelVolData m_channelInfo[MAX_CHANNELS];
@@ -1994,7 +1994,7 @@ void CChannelCullList::Initialize( CChannelList &list )
 		{
 			m_channelInfo[i].m_vol = ChannelLoudestCurVolume(ch);
 			AssertMsg(m_channelInfo[i].m_vol >= 0, "Sound channel has a negative volume?");
-			m_channelInfo[i].m_nameHash = (unsigned int) ch->sfx;
+			m_channelInfo[i].m_nameHash = (uintp) ch->sfx;
 		}
 		else
 		{
@@ -2029,7 +2029,7 @@ void CChannelCullList::Initialize( CChannelList &list )
 			 ++j )
 		{
 			// j steps through the sorted list until we find ourselves:
-			if (m_channelInfo[j].m_nameHash == (unsigned int)(ch->sfx))
+			if (m_channelInfo[j].m_nameHash == (uintp)(ch->sfx))
 			{
 				// that's another channel playing this sound but louder than me
 				++howManyLouder;

engine/audio/private/snd_mp3_source.cpp

@@ -93,7 +93,7 @@ CAudioSourceMP3::CAudioSourceMP3( CSfxTable *pSfx )
 
 	m_dataStart = 0;
 
-	int file = g_pSndIO->open( pSfx->GetFileName() );
+	intp file = g_pSndIO->open( pSfx->GetFileName() );
 	if ( file != -1 )
 	{
 		m_dataSize = g_pSndIO->size( file );
@@ -239,7 +239,7 @@ void CAudioSourceMP3::GetCacheData( CAudioSourceCachedInfo *info )
 	info->SetSampleRate( m_sampleRate );
 	info->SetDataStart( 0 );
 
-	int file = g_pSndIO->open( m_pSfx->GetFileName() );
+	intp file = g_pSndIO->open( m_pSfx->GetFileName() );
 	if ( !file )
 	{
 		Warning( "Failed to find file for building soundcache [ %s ]\n", m_pSfx->GetFileName() );

engine/audio/private/snd_wave_source.cpp

@@ -723,7 +723,7 @@ bool CAudioSourceWave::GetStartupData( void *dest, int destsize, int& bytesCopie
 	// requesting precache snippet as leader for streaming startup latency
 	if ( destsize )
 	{
-		int file = g_pSndIO->open( m_pSfx->GetFileName() );
+		intp file = g_pSndIO->open( m_pSfx->GetFileName() );
 		if ( !file )
 		{
 			return false;

engine/audio/private/voice.cpp

@@ -37,7 +37,7 @@ void VoiceTweak_EndVoiceTweakMode();
 void EngineTool_OverrideSampleRate( int& rate );
 
 // A fallback codec that should be the most likely to work for local/offline use
-#define VOICE_FALLBACK_CODEC	"vaudio_celt"
+#define VOICE_FALLBACK_CODEC	"vaudio_opus"
 
 // Special entity index used for tweak mode.
 #define TWEAKMODE_ENTITYINDEX				-500
@@ -197,6 +197,9 @@ extern IVoiceRecord* CreateVoiceRecord_AudioQueue(int sampleRate);
 extern IVoiceRecord* CreateVoiceRecord_OpenAL(int sampleRate);
 #endif
 
+#ifdef USE_SDL
+extern IVoiceRecord *CreateVoiceRecord_SDL(int sampleRate);
+#endif
 
 static bool VoiceRecord_Start()
 {
@@ -583,12 +586,13 @@ bool Voice_Init( const char *pCodecName, int nSampleRate )
 
 	bool bSpeex = Q_stricmp( pCodecName, "vaudio_speex" ) == 0;
 	bool bCelt  = Q_stricmp( pCodecName, "vaudio_celt" )  == 0;
+	bool bOpus  = Q_stricmp( pCodecName, "vaudio_opus" )  == 0;
 	bool bSteam = Q_stricmp( pCodecName, "steam" )        == 0;
 	// Miles has not been in use for voice in a long long time.  Not worth the surface to support ancient demos that may
 	// use it (and probably do not work for other reasons)
 	// "vaudio_miles"
 
-	if ( !( bSpeex || bCelt || bSteam ) )
+	if ( !( bSpeex || bCelt || bOpus || bSteam ) )
 	{
 		Msg( "Voice_Init Failed: invalid voice codec %s.\n", pCodecName );
 		return false;
@@ -648,6 +652,8 @@ bool Voice_Init( const char *pCodecName, int nSampleRate )
 	}
 #elif defined( WIN32 )
 	g_pVoiceRecord = CreateVoiceRecord_DSound( Voice_SamplesPerSec() );
+#elif defined( USE_SDL )
+	g_pVoiceRecord = CreateVoiceRecord_SDL( Voice_SamplesPerSec() );
 #else
 	g_pVoiceRecord = CreateVoiceRecord_OpenAL( Voice_SamplesPerSec() );
 #endif
@@ -671,6 +677,12 @@ bool Voice_Init( const char *pCodecName, int nSampleRate )
 		CreateInterfaceFn createCodecFn = NULL;
 		g_hVoiceCodecDLL = FileSystem_LoadModule(pCodecName);
 
+		if( !g_hVoiceCodecDLL || (createCodecFn = Sys_GetFactory(g_hVoiceCodecDLL)) == NULL )
+		{
+			g_hVoiceCodecDLL = FileSystem_LoadModule( VOICE_FALLBACK_CODEC );
+			pCodecName = VOICE_FALLBACK_CODEC;
+		}
+
 		if ( !g_hVoiceCodecDLL || (createCodecFn = Sys_GetFactory(g_hVoiceCodecDLL)) == NULL ||
 		     (g_pEncodeCodec = (IVoiceCodec*)createCodecFn(pCodecName, NULL)) == NULL || !g_pEncodeCodec->Init( quality ) )
 		{
@@ -1075,7 +1087,7 @@ int Voice_GetCompressedData(char *pchDest, int nCount, bool bFinal)
 {
 	// Check g_bVoiceRecordStopping in case g_bUsingSteamVoice changes on us
 	// while waiting for the end of voice data.
-	if ( g_bUsingSteamVoice || g_bVoiceRecordStopping )
+	if ( g_bUsingSteamVoice && g_bVoiceRecordStopping )
 	{
 		uint32 cbCompressedWritten = 0;
 		uint32 cbUncompressedWritten = 0;

engine/audio/private/voice_mixer_controls_openal.cpp

@@ -141,7 +141,7 @@ bool CMixerControls::GetValue_Float(Control iControl, float &value)
 		case MicVolume:
 		{
 			OSStatus theError = noErr;
-			for ( int iChannel = 0; iChannel < 3; iChannel++ )
+			for ( uint iChannel = 0; iChannel < 3; iChannel++ )
 			{
 				// scan the channel list until you find a channel set to non-zero, then use that
 				Float32 theVolume = 0;

engine/audio/private/voice_record_sdl.cpp

@@ -0,0 +1,257 @@
+//========= Copyright 1996-2009, Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $NoKeywords: $
+//
+//=============================================================================//
+// This module implements the voice record and compression functions 
+
+//#include "audio_pch.h"
+//#include "voice.h"
+#include "tier0/platform.h"
+#include "ivoicerecord.h"
+#include "tier0/dbg.h"
+#include "tier0/threadtools.h"
+
+#include <assert.h>
+#include <SDL_audio.h>
+
+#define RECORDING_BUFFER_SECONDS 3
+#define SAMPLE_COUNT 2048
+
+// ------------------------------------------------------------------------------
+// VoiceRecord_SDL
+// ------------------------------------------------------------------------------
+
+struct AudioBuf
+{
+	int Read(char *out, int len)
+	{
+		int nAvalible = (size + (writePtr - readPtr)) % size;
+
+		if( nAvalible == 0 )
+			return 0;
+
+		if( len > nAvalible ) len = nAvalible;
+
+		int diff = (data + size) - readPtr;
+
+
+		if( len > diff )
+		{
+			memcpy(out, readPtr, diff );
+			memcpy(out+diff, data, len-diff );
+		} else memcpy(out, readPtr, len);
+
+		readPtr += len;
+
+		if( readPtr >= data + size )
+			readPtr -= size;
+
+		return len;
+	}
+
+	void Write(char *in, int len)
+	{
+		int diff = (data + size) - writePtr;
+
+		if( len > diff )
+		{
+			memcpy(writePtr, in, diff );
+			memcpy(data, in+diff, len-diff );
+		} else memcpy(writePtr, in, len);
+
+		writePtr += len;
+
+		if (writePtr >= (data + size))
+			writePtr -= size;
+	}
+
+	int size;
+	char *data;
+	char *readPtr;
+	char *writePtr;
+};
+
+class VoiceRecord_SDL : public IVoiceRecord
+{
+protected:
+	virtual				~VoiceRecord_SDL();
+public:
+	VoiceRecord_SDL();
+	virtual void		Release();
+	virtual bool		RecordStart();
+	virtual void		RecordStop();
+
+	// Initialize. The format of the data we expect from the provider is
+	// 8-bit signed mono at the specified sample rate.
+	virtual bool		Init(int sampleRate);
+	virtual void		Idle() {}; // Stub
+	void RenderBuffer( char *pszBuf, int size );
+
+	// Get the most recent N samples.
+	virtual int			GetRecordedData(short *pOut, int nSamplesWanted );
+
+	SDL_AudioSpec m_ReceivedRecordingSpec;
+	int m_BytesPerSample; // Да кому нужна эта ваша инкапсуляция?
+	int m_nSampleRate;
+private:
+	bool				InitalizeInterfaces();	// Initialize the openal capture buffers and other interfaces
+	void				ReleaseInterfaces();	// Release openal buffers and other interfaces
+	void				ClearInterfaces();				// Clear members.
+private:
+
+	SDL_AudioDeviceID m_Device;
+	AudioBuf m_AudioBuffer;
+};
+
+void audioRecordingCallback( void *userdata, uint8 *stream, int len )
+{
+	VoiceRecord_SDL *voice = (VoiceRecord_SDL*)userdata;
+	voice->RenderBuffer( (char*)stream, len );
+}
+
+VoiceRecord_SDL::VoiceRecord_SDL() :
+m_nSampleRate( 0 ) ,m_Device( 0 )
+{
+	m_AudioBuffer.data = NULL;
+	m_AudioBuffer.readPtr = NULL;
+	m_AudioBuffer.writePtr = NULL;
+
+	ClearInterfaces();
+}
+
+VoiceRecord_SDL::~VoiceRecord_SDL()
+{
+	ReleaseInterfaces();
+	ClearInterfaces();
+}
+
+void VoiceRecord_SDL::Release()
+{
+	ReleaseInterfaces();
+	ClearInterfaces();
+	delete this;
+}
+
+bool VoiceRecord_SDL::RecordStart()
+{
+	if ( !m_Device )
+		InitalizeInterfaces();
+
+	if ( !m_Device )
+		return false;
+
+	SDL_PauseAudioDevice( m_Device, SDL_FALSE );
+
+	return true;
+}
+
+
+void VoiceRecord_SDL::RecordStop()
+{
+	// Stop capturing.
+	if ( m_Device )
+		SDL_PauseAudioDevice( m_Device, SDL_TRUE );
+
+	// Release the capture buffer interface and any other resources that are no
+	// longer needed
+	ReleaseInterfaces();
+}
+
+bool VoiceRecord_SDL::InitalizeInterfaces()
+{
+	//Default audio spec
+	SDL_AudioSpec desiredRecordingSpec;
+	SDL_zero(desiredRecordingSpec);
+	desiredRecordingSpec.freq = m_nSampleRate;
+	desiredRecordingSpec.format = AUDIO_S16;
+	desiredRecordingSpec.channels = 1;
+	desiredRecordingSpec.samples = SAMPLE_COUNT;
+	desiredRecordingSpec.callback = audioRecordingCallback;
+	desiredRecordingSpec.userdata = (void*)this;
+
+	//Open recording device
+	m_Device = SDL_OpenAudioDevice( NULL, SDL_TRUE, &desiredRecordingSpec, &m_ReceivedRecordingSpec, 0 );
+
+	if( m_Device != 0 )
+	{
+		//Calculate per sample bytes
+		m_BytesPerSample = m_ReceivedRecordingSpec.channels * ( SDL_AUDIO_BITSIZE( m_ReceivedRecordingSpec.format ) / 8 );
+
+		//Calculate bytes per second
+		int bytesPerSecond = m_ReceivedRecordingSpec.freq * m_BytesPerSample;
+
+		//Allocate and initialize byte buffer
+		m_AudioBuffer.size = RECORDING_BUFFER_SECONDS * bytesPerSecond;
+
+		if( !m_AudioBuffer.data )
+			m_AudioBuffer.data = (char *)malloc( m_AudioBuffer.size );
+
+		m_AudioBuffer.readPtr = m_AudioBuffer.data;
+		m_AudioBuffer.writePtr = m_AudioBuffer.data + SAMPLE_COUNT*m_BytesPerSample*2;
+
+		memset( m_AudioBuffer.data, 0, m_AudioBuffer.size );
+
+		return true;
+	}
+	else
+		return false;
+}
+
+bool VoiceRecord_SDL::Init(int sampleRate)
+{
+	m_nSampleRate = sampleRate;
+	ReleaseInterfaces();
+
+	return true;
+}
+
+
+void VoiceRecord_SDL::ReleaseInterfaces()
+{
+	if( m_Device != 0 )
+		SDL_CloseAudioDevice( m_Device );
+
+	m_Device = 0;
+}
+
+
+void VoiceRecord_SDL::ClearInterfaces()
+{
+	if( m_AudioBuffer.data )
+	{
+		free( m_AudioBuffer.data );
+		m_AudioBuffer.data = NULL;
+		m_AudioBuffer.readPtr = NULL;
+		m_AudioBuffer.writePtr = NULL;
+	}
+	m_Device = 0;
+}
+
+void VoiceRecord_SDL::RenderBuffer( char *pszBuf, int size )
+{
+	m_AudioBuffer.Write( pszBuf, size );
+}
+
+int VoiceRecord_SDL::GetRecordedData(short *pOut, int nSamples )
+{
+	if ( !m_AudioBuffer.data || nSamples == 0 )
+		return 0;
+
+	int cbSamples = nSamples * m_BytesPerSample;
+
+	return m_AudioBuffer.Read( (char*)pOut, cbSamples )/m_BytesPerSample;
+}
+
+IVoiceRecord* CreateVoiceRecord_SDL(int sampleRate)
+{
+	VoiceRecord_SDL *pRecord = new VoiceRecord_SDL;
+	if ( pRecord && pRecord->Init(sampleRate) )
+		return pRecord;
+	else if( pRecord )
+		pRecord->Release();
+
+	return NULL;
+}

engine/baseclient.cpp

@@ -1098,10 +1098,10 @@ void CBaseClient::EndTrace( bf_write &msg )
 	}
 
 	if ( sv_netspike_output.GetInt() & 1 )
-		COM_LogString( SERVER_PACKETS_LOG, logData.String() );
+		COM_LogString( SERVER_PACKETS_LOG, (const char*)logData.String() );
 	if ( sv_netspike_output.GetInt() & 2 )
-		Log( "%s", logData.String() );
-	ETWMark1S( "netspike", logData.String() );
+		Log( "%s", (const char*)logData.String() );
+	ETWMark1S( "netspike", (const char*)logData.String() );
 	m_Trace.m_Records.RemoveAll();
 	m_iTracing = 0;
 }

engine/baseclientstate.cpp

@@ -570,7 +570,7 @@ bool CBaseClientState::PrepareSteamConnectResponse( uint64 unGSSteamID, bool bGS
 		return true;
 	}
 
-#if !defined( NO_STEAM ) && !defined( SWDS  ) 
+#if 0 //!defined( NO_STEAM ) && !defined( SWDS  ) 
 	if ( !Steam3Client().SteamUser() )
 	{
 		COM_ExplainDisconnection( true, "#GameUI_ServerRequireSteam" );
@@ -578,14 +578,14 @@ bool CBaseClientState::PrepareSteamConnectResponse( uint64 unGSSteamID, bool bGS
 		return false;
 	}
 #endif
-	
+
 	netadr_t checkAdr = adr;
 	if ( adr.GetType() == NA_LOOPBACK || adr.IsLocalhost() )
 	{
 		checkAdr.SetIP( net_local_adr.GetIPHostByteOrder() );
 	}
 
-#ifndef SWDS
+#if 0 // #ifndef SWDS
 	// now append the steam3 cookie
 	char steam3Cookie[ STEAM_KEYSIZE ];
 	uint32 steam3CookieLen = 0;
@@ -936,6 +936,7 @@ bool CBaseClientState::ProcessConnectionlessPacket( netpacket_t *packet )
 								int authprotocol = msg.ReadLong();
 								uint64 unGSSteamID = 0;
 								bool bGSSecure = false;
+#if 0
 								if ( authprotocol == PROTOCOL_STEAM )
 								{
 									if ( msg.ReadShort() != 0 )
@@ -963,6 +964,7 @@ bool CBaseClientState::ProcessConnectionlessPacket( netpacket_t *packet )
 										return false;
 									}
 								}
+#endif
 								SendConnectPacket( challenge, authprotocol, unGSSteamID, bGSSecure );
 							}
 							break;

engine/baseserver.cpp

@@ -741,7 +741,7 @@ bool CBaseServer::ProcessConnectionlessPacket(netpacket_t * packet)
 //					break;
 // 				}
 
-				if ( authProtocol == PROTOCOL_STEAM )
+/*				if ( authProtocol == PROTOCOL_STEAM )
 				{
 					int keyLen = msg.ReadShort();
 					if ( keyLen < 0 || keyLen > sizeof(cdkey) )
@@ -753,7 +753,7 @@ bool CBaseServer::ProcessConnectionlessPacket(netpacket_t * packet)
 
 					ConnectClient( packet->from, protocol, challengeNr, clientChallenge, authProtocol, name, password, cdkey, keyLen );	// cd key is actually a raw encrypted key	
 				}
-				else
+				else*/
 				{
 					msg.ReadString( cdkey, sizeof(cdkey) );
 					ConnectClient( packet->from, protocol, challengeNr, clientChallenge, authProtocol, name, password, cdkey, strlen(cdkey) );
@@ -1434,11 +1434,13 @@ bool CBaseServer::CheckChallengeType( CBaseClient * client, int nNewUserID, neta
 		return false;
 	}
 
+#if 0
 	if ( ( nAuthProtocol == PROTOCOL_HASHEDCDKEY ) && (Q_strlen( pchLogonCookie ) <= 0 ||  Q_strlen(pchLogonCookie) != 32 ) )
 	{
 		RejectConnection( adr, clientChallenge, "#GameUI_ServerRejectInvalidCertLen" );
 		return false;
 	}
+#endif
 
 	Assert( !IsReplay() );
 
@@ -1470,23 +1472,24 @@ bool CBaseServer::CheckChallengeType( CBaseClient * client, int nNewUserID, neta
 		client->SetSteamID( CSteamID() ); // set an invalid SteamID
 
 		// Convert raw certificate back into data
-		if ( cbCookie <= 0 || cbCookie >= STEAM_KEYSIZE )
+/*		if ( cbCookie <= 0 || cbCookie >= STEAM_KEYSIZE )
 		{
 			RejectConnection( adr, clientChallenge, "#GameUI_ServerRejectInvalidSteamCertLen" );
 			return false;
-		}
+		}*/
 		netadr_t checkAdr = adr;
 		if ( adr.GetType() == NA_LOOPBACK || adr.IsLocalhost() )
 		{
 			checkAdr.SetIP( net_local_adr.GetIPHostByteOrder() );
 		}
-
+#if 0
 		if ( !Steam3Server().NotifyClientConnect( client, nNewUserID, checkAdr, pchLogonCookie, cbCookie ) 
 			&& !Steam3Server().BLanOnly() ) // the userID isn't alloc'd yet so we need to fill it in manually
 		{
 			RejectConnection( adr, clientChallenge, "#GameUI_ServerRejectSteam" );
 			return false;
 		}
+#endif
 
 		//
 		// Any rejections below this must call SendUserDisconnect

engine/buildcubemaps.cpp

@@ -1042,7 +1042,7 @@ void R_BuildCubemapSamples( int numIterations )
 			}
 		}
 
-		bool bSupportsHDR = g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE;
+		bool bSupportsHDR = true; //g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE;
 
 		for( i = 0; i < pWorldModel->brush.pShared->m_nCubemapSamples; i++ )
 		{
@@ -1080,7 +1080,7 @@ void R_BuildCubemapSamples( int numIterations )
 			return;
 		}
 
-		iBSPPack->SetHDRMode( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE );
+		iBSPPack->SetHDRMode( true /*g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE*/ );
 
 		iBSPPack->LoadBSPFile( g_pFileSystem, cl.m_szLevelFileName );
 

engine/cl_ents_parse.cpp

@@ -451,7 +451,7 @@ void CL_PreserveExistingEntity( int nOldEntity )
 		return;
 	}
 
-	pEnt->OnDataUnchangedInPVS();
+//	pEnt->OnDataUnchangedInPVS();
 }
 
 void CL_CopyExistingEntity( CEntityReadInfo &u )

engine/cl_main.cpp

@@ -72,6 +72,7 @@
 #include "replay_internal.h"
 #endif
 
+#include "language.h"
 #include "igame.h"
 
 // memdbgon must be the last include file in a .cpp file!!!
@@ -892,7 +893,7 @@ CON_COMMAND_F( connect, "Connect to specified server.", FCVAR_DONTRECORD )
 	{
 		ConMsg( "Usage:  connect <server>\n" );
 	}
-	vecArgs.PurgeAndDeleteElements();
+	vecArgs.PurgeAndDeleteElementsArray();
 }
 
 CON_COMMAND_F( redirect, "Redirect client to specified server.", FCVAR_DONTRECORD | FCVAR_SERVER_CAN_EXECUTE )
@@ -2743,14 +2744,18 @@ void CL_InitLanguageCvar()
 	}
 	else
 	{
+		char *szLang = getenv("LANG");
+
 		if ( CommandLine()->CheckParm( "-language" ) )
-		{
 			cl_language.SetValue( CommandLine()->ParmValue( "-language", "english") );
+		else if( szLang )
+		{
+			ELanguage lang = PchLanguageICUCodeToELanguage(szLang, k_Lang_English);
+			const char *szShortLang = GetLanguageShortName(lang);
+			cl_language.SetValue( szShortLang );
 		}
 		else
-		{
 			cl_language.SetValue( "english" );
-		}
 	}
 }
 

engine/cl_parse_event.cpp

@@ -18,7 +18,7 @@ static ConVar cl_showevents	( "cl_showevents", "0", FCVAR_CHEAT, "Print event fi
 // Input  : slot - 
 //			*eventname - 
 //-----------------------------------------------------------------------------
-void CL_DescribeEvent( int slot, CEventInfo *event, const char *eventname )
+void CL_DescribeEvent( intp slot, CEventInfo *event, const char *eventname )
 {
 	int idx = (slot & 31);
 
@@ -81,7 +81,7 @@ void CL_FireEvents( void )
 		return;
 	}
 
-	int i, next;
+	intp i, next;
 	for ( i = cl.events.Head(); i != cl.events.InvalidIndex(); i = next )
 	{
 		next = cl.events.Next( i );

engine/cl_rcon.cpp

@@ -56,12 +56,10 @@ public:
 		// Immediately try to start vprofiling
 		// Also, enable cheats on this client only
 		Cmd_SetRptActive( true );
-		StartVProfData();
 	}
 
 	virtual void OnSocketClosed( SocketHandle_t hSocket, const netadr_t & netAdr, void* pData )
 	{
-		StopVProfData();
 		Cmd_SetRptActive( false );
 		BaseClass::OnSocketClosed( hSocket, netAdr, pData );
 	}
@@ -98,135 +96,6 @@ static void RconAddressChanged_f( IConVar *pConVar, const char *pOldString, floa
 
 static ConVar	rcon_address( "rcon_address", "", FCVAR_SERVER_CANNOT_QUERY|FCVAR_DONTRECORD, "Address of remote server if sending unconnected rcon commands (format x.x.x.x:p) ", RconAddressChanged_f );
 
-
-
-//-----------------------------------------------------------------------------
-// Implementation of remote vprof
-//-----------------------------------------------------------------------------
-CRConVProfExport::CRConVProfExport()
-{
-}
-
-void CRConVProfExport::AddListener()
-{
-}
-
-void CRConVProfExport::RemoveListener()
-{
-}
-
-void CRConVProfExport::SetBudgetFlagsFilter( int filter )
-{
-}
-
-int CRConVProfExport::GetNumBudgetGroups()
-{
-	return m_Info.Count();
-}
-
-void CRConVProfExport::GetBudgetGroupInfos( CExportedBudgetGroupInfo *pInfos )
-{
-	memcpy( pInfos, m_Info.Base(), GetNumBudgetGroups() * sizeof(CExportedBudgetGroupInfo) );
-}
-
-void CRConVProfExport::GetBudgetGroupTimes( float times[IVProfExport::MAX_BUDGETGROUP_TIMES] )
-{
-	int nGroups = min( m_Times.Count(), (int)IVProfExport::MAX_BUDGETGROUP_TIMES );
-	memset( times, 0, nGroups * sizeof(float) );
-	nGroups = min( GetNumBudgetGroups(), nGroups );
-	memcpy( times, m_Times.Base(), nGroups * sizeof(float) );
-}
-
-void CRConVProfExport::PauseProfile()
-{
-	// NOTE: This only has effect when testing on a listen server
-	// it shouldn't do anything in the wild. When drawing the budget panel
-	// this will cause the time spent doing so to not be counted
-	VProfExport_Pause();
-}
-
-void CRConVProfExport::ResumeProfile()
-{
-	// NOTE: This only has effect when testing on a listen server
-	// it shouldn't do anything in the wild
-	VProfExport_Resume();
-}		
-
-void CRConVProfExport::CleanupGroupData()
-{
-	int nCount = m_Info.Count();
-	for ( int i = 0; i < nCount; ++i )
-	{
-		delete m_Info[i].m_pName;
-	}
-
-	m_Info.RemoveAll();
-}
-
-void CRConVProfExport::OnRemoteGroupData( const void *data, int len )
-{
-	CUtlBuffer buf( data, len, CUtlBuffer::READ_ONLY );
-	int nFirstGroup = buf.GetInt();
-
-	if ( nFirstGroup == 0 )
-	{
-		CleanupGroupData();
-	}
-	else
-	{
-		Assert( nFirstGroup == m_Info.Count() );
-	}
-
-	// NOTE: See WriteRemoteVProfGroupData in vprof_engine.cpp
-	// to see the encoding of this data
-	int nGroupCount = buf.GetInt();
-	int nBase = m_Info.AddMultipleToTail( nGroupCount );
-	char temp[1024];
-	for ( int i = 0; i < nGroupCount; ++i )
-	{
-		CExportedBudgetGroupInfo *pInfo = &m_Info[nBase + i];
-
-		unsigned char red, green, blue, alpha;
-		red = buf.GetUnsignedChar( );
-		green = buf.GetUnsignedChar( );
-		blue = buf.GetUnsignedChar( );
-		alpha = buf.GetUnsignedChar( );
-		buf.GetString( temp );
-		int nLen = Q_strlen( temp );
-
-		pInfo->m_Color.SetColor( red, green, blue, alpha );
-		char *pBuf = new char[ nLen + 1 ];
-		pInfo->m_pName = pBuf;
-		memcpy( pBuf, temp, nLen+1 );
-		pInfo->m_BudgetFlags = 0;
-	}
-}
-
-void CRConVProfExport::OnRemoteData( const void *data, int len )
-{
-	// NOTE: See WriteRemoteVProfData in vprof_engine.cpp
-	// to see the encoding of this data
-	int nCount = len / sizeof(float);
-	Assert( nCount == m_Info.Count() );
-
-	CUtlBuffer buf( data, len, CUtlBuffer::READ_ONLY );
-	m_Times.SetCount( nCount );
-	memcpy( m_Times.Base(), data, nCount * sizeof(float) );
-}
-
-
-CON_COMMAND( vprof_remote_start, "Request a VProf data stream from the remote server (requires authentication)" )
-{
-	// TODO: Make this work (it might already!)
-//	RCONClient().StartVProfData();
-}
-
-CON_COMMAND( vprof_remote_stop, "Stop an existing remote VProf data request" )
-{
-	// TODO: Make this work (it might already!)
-//	RCONClient().StopVProfData();
-}
-
 #ifdef ENABLE_RPT
 CON_COMMAND_F( rpt_screenshot, "", FCVAR_HIDDEN | FCVAR_DONTRECORD )
 {
@@ -454,22 +323,6 @@ void CRConClient::ParseReceivedData()
 			}
 			break;
 
-		case SERVERDATA_VPROF_DATA:
-			{
-				int nDataSize = m_RecvBuffer.GetInt();
-				m_VProfExport.OnRemoteData( m_RecvBuffer.PeekGet(), nDataSize );
-				m_RecvBuffer.SeekGet( CUtlBuffer::SEEK_CURRENT, nDataSize );
-			}
-			break;
-
-		case SERVERDATA_VPROF_GROUPS:
-			{
-				int nDataSize = m_RecvBuffer.GetInt();
-				m_VProfExport.OnRemoteGroupData( m_RecvBuffer.PeekGet(), nDataSize );
-				m_RecvBuffer.SeekGet( CUtlBuffer::SEEK_CURRENT, nDataSize );
-			}
-			break;
-
 		case SERVERDATA_RESPONSE_STRING:
 			{
 				char pBuf[2048];
@@ -706,45 +559,6 @@ void CRConClient::SendCmd( const char *msg )
 	SendResponse( response );
 }
 
-
-//-----------------------------------------------------------------------------
-// Purpose: Start vprofiling
-//-----------------------------------------------------------------------------
-void CRConClient::StartVProfData()
-{
-	if ( !IsConnected() )
-	{
-		if ( !ConnectSocket() )
-			return;
-	}
-
-	// Override the vprof export to point to our local profiling data
-	OverrideVProfExport( &m_VProfExport );
-
-	CUtlBuffer response;
-	BuildResponse( response, SERVERDATA_VPROF, "", "" );
-	SendResponse( response );
-}
-
-
-//-----------------------------------------------------------------------------
-// Purpose: Stop vprofiling
-//-----------------------------------------------------------------------------
-void CRConClient::StopVProfData()
-{
-	// Reset the vprof export to point to the normal profiling data
-	ResetVProfExport( &m_VProfExport );
-
-	// Don't bother restarting a connection to turn this off
-	if ( !IsConnected() )
-		return;
-
-	CUtlBuffer response;
-	BuildResponse( response, SERVERDATA_REMOVE_VPROF, "", "" );
-	SendResponse( response );
-}
-
-
 //-----------------------------------------------------------------------------
 // Purpose: get data from the server
 //-----------------------------------------------------------------------------

engine/cl_rcon.h

@@ -27,47 +27,6 @@
 // memdbgon must be the last include file in a .cpp file!!!
 #include "tier0/memdbgon.h"
 
-abstract_class IVProfData
-{
-public:
-	virtual void OnRemoteGroupData( const void *data, int len ) = 0;
-	virtual void OnRemoteData( const void *data, int len ) = 0;
-};
-
-
-//-----------------------------------------------------------------------------
-// Used to display client perf data in showbudget
-//-----------------------------------------------------------------------------
-class CRConVProfExport : public IVProfExport, public IVProfData
-{
-	// Inherited from IVProfExport
-public:
-	virtual void AddListener();
-	virtual void RemoveListener();
-	virtual void PauseProfile();
-	virtual void ResumeProfile();
-	virtual void SetBudgetFlagsFilter( int filter );
-	virtual int GetNumBudgetGroups();
-	virtual void GetBudgetGroupInfos( CExportedBudgetGroupInfo *pInfos );
-	virtual void GetBudgetGroupTimes( float times[MAX_BUDGETGROUP_TIMES] );
-
-	// Inherited from IVProfData
-public:
-	virtual void OnRemoteGroupData( const void *data, int len );
-	virtual void OnRemoteData( const void *data, int len );
-
-	// Other public methods
-public:
-	CRConVProfExport();
-
-private:
-	void CleanupGroupData();
-
-	CUtlVector< CExportedBudgetGroupInfo > m_Info;
-	CUtlVector<float> m_Times;	// Times from the most recent snapshot.
-};
-
-
 class CRConClient : public ISocketCreatorListener
 {
 public:
@@ -89,10 +48,6 @@ public:
 	bool IsConnected() const;
 	bool IsAuthenticated() const { return m_bAuthenticated; }
 
-	void RegisterVProfDataCallback( IVProfData *callback );
-	void StopVProfData();
-	void StartVProfData();
-
 	void TakeScreenshot();
 	void GrabConsoleLog();
 
@@ -116,7 +71,6 @@ private:
 	void SaveRemoteScreenshot( const void* pBuffer, int nBufLen );
 	void SaveRemoteConsoleLog( const void* pBuffer, int nBufLen );
 
-	CRConVProfExport m_VProfExport;
 	CSocketCreator	m_Socket;
 	netadr_t		m_Address;
 	int				m_iAuthRequestID;

engine/client.cpp

@@ -1816,14 +1816,14 @@ void CClientState::FinishSignonState_New()
 	//
 	// This is pretty janky, but doesn't really have any cost (and even makes our one-frozen-frame load screen slightly
 	// less likely to trigger OS "not responding" warnings)
-	extern void V_RenderVGuiOnly();
-	V_RenderVGuiOnly();
+//	extern void V_RenderVGuiOnly();
+//	V_RenderVGuiOnly();
 
 	// Before we do anything with the whitelist, make sure we have the proper map pack mounted
 	// this will load the .bsp by setting the world model the string list at the hardcoded index 1.
 	cl.SetModel( 1 );
 
-	V_RenderVGuiOnly();
+	//V_RenderVGuiOnly();
 
 	// Check for a new whitelist. It's good to do it early in the connection process here because if we wait until later,
 	// the client may have loaded some files w/o the proper whitelist restrictions and we'd have to reload them.

engine/cmd.cpp

@@ -579,7 +579,7 @@ void Cmd_Exec_f( const CCommand &args )
 
 	const char *szFile = args[1];
 
-	const char *pPathID = "MOD";
+	const char *pPathID = "*";
 
 	Q_snprintf( fileName, sizeof( fileName ), "//%s/cfg/%s", pPathID, szFile );
 	Q_DefaultExtension( fileName, ".cfg", sizeof( fileName ) );
@@ -624,12 +624,13 @@ void Cmd_Exec_f( const CCommand &args )
 		}
 	}
 
-	char buf[16384] = { 0 };
+	char *buf = new char[16384];
 	int len = 0;
-	char *f = (char *)COM_LoadStackFile( fileName, buf, sizeof( buf ), len );
+	char *f = (char *)COM_LoadStackFile( fileName, buf, 16384, len );
 	if ( !f )
 	{
 		ConMsg( "exec: couldn't exec %s\n", szFile );
+		delete[] buf;
 		return;
 	}
 
@@ -645,7 +646,7 @@ void Cmd_Exec_f( const CCommand &args )
 	ConDMsg( "execing %s\n", szFile );
 
 	// check to make sure we're not going to overflow the cmd_text buffer
-	int hCommand = s_CommandBuffer.GetNextCommandHandle();
+	CommandHandle_t hCommand = s_CommandBuffer.GetNextCommandHandle();
 
 	// Execute each command immediately
 	const char *pszDataPtr = f;
@@ -684,6 +685,9 @@ void Cmd_Exec_f( const CCommand &args )
 			free( f );
 		}
 	}
+
+	delete[] buf;
+
 	// force any queued convar changes to flush before reading/writing them
 	UpdateMaterialSystemConfig();
 }

engine/cmodel.cpp

@@ -2677,6 +2677,9 @@ int     CM_BoxVisible( const Vector& mins, const Vector& maxs, const byte *visbi
                 int cluster = CM_LeafCluster( leafList[i] );
                 int offset = cluster>>3;
 
+		if( offset == -1 )
+			return true;
+
                 if ( offset > vissize )
                 {
                         Sys_Error( "CM_BoxVisible:  cluster %i, offset %i out of bounds %i\n", cluster, offset, vissize );

engine/cmodel_bsp.cpp

@@ -288,7 +288,7 @@ bool CollisionBSPData_Load( const char *pName, CCollisionBSPData *pBSPData )
 	CollisionBSPData_LoadPhysics( pBSPData );
 
 	COM_TimestampedLog( "  CollisionBSPData_LoadDispInfo" );
-    CollisionBSPData_LoadDispInfo( pBSPData );
+	CollisionBSPData_LoadDispInfo( pBSPData );
 
 	return true;
 }
@@ -1110,7 +1110,7 @@ void CollisionBSPData_LoadDispInfo( CCollisionBSPData *pBSPData )
     // get face data
     //
 	int face_lump_to_load = LUMP_FACES;
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&
+	if ( g_pMaterialSystemHardwareConfig->GetHDREnabled() && /*g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&*/
 		CMapLoadHelper::LumpSize( LUMP_FACES_HDR ) > 0 )
 	{
 		face_lump_to_load = LUMP_FACES_HDR;

engine/cmodel_disp.cpp

@@ -228,7 +228,7 @@ public:
 	// Fill out the meshlist for this terrain patch
 	virtual void GetVirtualMesh( void *userData, virtualmeshlist_t *pList )
 	{
-		int index = (int)userData;
+		intp index = (intp)userData;
 		Assert(index >= 0 && index < g_DispCollTreeCount );
 		g_pDispCollTrees[index].GetVirtualMeshList( pList );
 		pList->pHull = NULL;
@@ -243,14 +243,14 @@ public:
 	// returns the bounds for the terrain patch
 	virtual void GetWorldspaceBounds( void *userData, Vector *pMins, Vector *pMaxs )
 	{
-		int index = (int)userData;
+		intp index = (intp)userData;
 		*pMins = g_pDispBounds[index].mins;
 		*pMaxs = g_pDispBounds[index].maxs;
 	}
 	// Query against the AABB tree to find the list of triangles for this patch in a sphere
 	virtual void GetTrianglesInSphere( void *userData, const Vector &center, float radius, virtualmeshtrianglelist_t *pList )
 	{
-		int index = (int)userData;
+		intp index = (intp)userData;
 		pList->triangleCount = g_pDispCollTrees[index].AABBTree_GetTrisInSphere( center, radius, pList->triangleIndices, ARRAYSIZE(pList->triangleIndices) );
 	}
 	void LevelInit( dphysdisp_t *pLump, int lumpSize )
@@ -318,7 +318,7 @@ void CM_CreateDispPhysCollide( dphysdisp_t *pDispLump, int dispLumpSize )
 {
 	g_VirtualTerrain.LevelInit(pDispLump, dispLumpSize);
 	g_TerrainList.SetCount( g_DispCollTreeCount );
-	for ( int i = 0; i < g_DispCollTreeCount; i++ )
+	for ( intp i = 0; i < g_DispCollTreeCount; i++ )
 	{
 		// Don't create a physics collision model for displacements that have been tagged as such.
 		CDispCollTree *pDispTree = &g_pDispCollTrees[i];

engine/colorcorrectionpanel.cpp

@@ -4895,7 +4895,7 @@ void CColorOperationListPanel::PopulateList( )
 			KeyValues *kv = new KeyValues( "operation", "layer", op->GetName() );
 			kv->SetInt( "image", (op->IsEnabled())?1:0 );
 			
-			m_pOperationListPanel->AddItem( kv, (unsigned int)op, false, false );
+			m_pOperationListPanel->AddItem( kv, (uintp)op, false, false );
 		}
 	}
 }

engine/common.cpp

@@ -16,7 +16,11 @@
 #include <characterset.h>
 #include <bitbuf.h>
 #include "common.h"
+#ifdef OSX
+#include <malloc/malloc.h>
+#else
 #include <malloc.h>
+#endif
 #include "traceinit.h"
 #include <filesystem.h>
 #include "filesystem_engine.h"

engine/console.cpp

@@ -479,7 +479,7 @@ Handles cursor positioning, line wrapping, etc
 */
 static bool g_fColorPrintf = false;
 static bool g_bInColorPrint = false;
-extern CThreadLocalInt<> g_bInSpew;
+extern CTHREADLOCALINT g_bInSpew;
 
 void Con_Printf( const char *fmt, ... );
 
@@ -865,6 +865,7 @@ CConPanel::CConPanel( vgui::Panel *parent ) : CBasePanel( parent, "CConPanel" )
 //-----------------------------------------------------------------------------
 CConPanel::~CConPanel( void )
 {
+	g_pConPanel = NULL;
 }
 
 void CConPanel::Con_NPrintf( int idx, const char *msg )

engine/decals.cpp

@@ -14,7 +14,11 @@
 #include "filesystem.h"
 #include "filesystem_engine.h"
 #include "materialsystem/imaterial.h"
+#ifdef OSX
+#include <malloc/malloc.h>
+#else
 #include <malloc.h>
+#endif
 #include "utldict.h"
 
 // memdbgon must be the last include file in a .cpp file!!!

engine/disp.cpp

@@ -547,9 +547,9 @@ bool CDispInfo::Render( CGroupMesh *pGroup, bool bAllowDebugModes )
 			VectorAdd( bbMin, bbMax, vecCenter );
 			vecCenter *= 0.5f;
 
-			int nInt = ( mat_surfaceid.GetInt() != 2 ) ? (int)m_ParentSurfID : (msurface2_t*)m_ParentSurfID - host_state.worldbrush->surfaces2;
+			intp nInt = ( mat_surfaceid.GetInt() != 2 ) ? (intp)m_ParentSurfID : (msurface2_t*)m_ParentSurfID - host_state.worldbrush->surfaces2;
 			char buf[32];
-			Q_snprintf( buf, sizeof( buf ), "%d", nInt );
+			Q_snprintf( buf, sizeof( buf ), "%d", (int)nInt );
 			CDebugOverlay::AddTextOverlay( vecCenter, 0, buf );
 		}
 

engine/disp_interface.cpp

@@ -764,7 +764,7 @@ void DispInfo_BatchDecals( CDispInfo **pVisibleDisps, int nVisibleDisps )
 			// There is only one group at a time.
 			int iGroup = 0;
 
-			int iPool = g_aDispDecalSortPool.Alloc( true );
+			intp iPool = g_aDispDecalSortPool.Alloc( true );
 			g_aDispDecalSortPool[iPool] = decal.m_pDecal;
 			
 			int iSortTree = decal.m_pDecal->m_iSortTree;
@@ -773,7 +773,7 @@ void DispInfo_BatchDecals( CDispInfo **pVisibleDisps, int nVisibleDisps )
 			DecalMaterialBucket_t &materialBucket = g_aDispDecalSortTrees[iSortTree].m_aDecalSortBuckets[iGroup][iTreeType].Element( iSortMaterial );
 			if ( materialBucket.m_nCheckCount == g_nDispDecalSortCheckCount )
 			{	
-				int iHead = materialBucket.m_iHead;
+				intp iHead = materialBucket.m_iHead;
 				g_aDispDecalSortPool.LinkBefore( iHead, iPool );
 			}
 			
@@ -844,7 +844,7 @@ void DispInfo_DrawDecalsGroup( int iGroup, int iTreeType )
 			if ( materialBucketList.Element( iBucket ).m_nCheckCount != g_nDispDecalSortCheckCount )
 				continue;
 			
-			int iHead = materialBucketList.Element( iBucket ).m_iHead;
+			intp iHead = materialBucketList.Element( iBucket ).m_iHead;
 			if ( !g_aDispDecalSortPool.IsValidIndex( iHead ) )
 				continue;
 
@@ -863,7 +863,7 @@ void DispInfo_DrawDecalsGroup( int iGroup, int iTreeType )
 			bool bBatchInit = true;
 			
 			int nCount;
-			int iElement = iHead;
+			intp iElement = iHead;
 			while ( iElement != g_aDispDecalSortPool.InvalidIndex() )
 			{
 				decal_t *pDecal = g_aDispDecalSortPool.Element( iElement );
@@ -1300,7 +1300,7 @@ int DispInfo_ComputeIndex( HDISPINFOARRAY hArray, IDispInfo* pInfo )
 	if( !pArray )
 		return NULL;
 
-	int iElement = ((int)pInfo - (int)(pArray->m_pDispInfos)) / sizeof(CDispInfo);
+	intp iElement = ((intp)pInfo - (intp)(pArray->m_pDispInfos)) / sizeof(CDispInfo);
 
 	Assert( iElement >= 0 && iElement < pArray->m_nDispInfos );
 	return iElement;

engine/download.cpp

@@ -924,14 +924,14 @@ void CDownloadManager::StartNewDownload()
 		m_lastPercent = 0;
 
 		// Start the thread
-		DWORD threadID;
+		uintp threadID;
 		VCRHook_CreateThread(NULL, 0, 
 #ifdef POSIX
 			(void *)
 #endif
-			DownloadThread, m_activeRequest, 0, (unsigned long int *)&threadID );
+			DownloadThread, m_activeRequest, 0, &threadID );
 
-		ThreadDetach( ( ThreadHandle_t )threadID );
+		ReleaseThreadHandle( ( ThreadHandle_t )threadID );
 	}
 	else
 	{
@@ -1070,16 +1070,16 @@ bool CL_IsGamePathValidAndSafeForDownload( const char *pGamePath )
 class CDownloadSystem : public IDownloadSystem
 {
 public:
-	virtual DWORD CreateDownloadThread( RequestContext_t *pContext )
+	virtual uintp CreateDownloadThread( RequestContext_t *pContext )
 	{
-		DWORD nThreadID;
+		uintp nThreadID;
 		VCRHook_CreateThread(NULL, 0,
 #ifdef POSIX
 		 	(void*)
 #endif
-		 	DownloadThread, pContext, 0, (unsigned long int *)&nThreadID );
+		 	DownloadThread, pContext, 0, (uintp *)&nThreadID );
 
-		ThreadDetach( ( ThreadHandle_t )nThreadID );
+		ReleaseThreadHandle( ( ThreadHandle_t )nThreadID );
 		return nThreadID;
 	}
 };

engine/dt_encode.cpp

@@ -39,7 +39,7 @@ void EncodeFloat( const SendProp *pProp, float fVal, bf_write *pOut, int objectI
 	}
 	else // standard clamped-range float
 	{
-		unsigned long ulVal;
+		unsigned int ulVal;
 		int nBits = pProp->m_nBits;
 		if ( flags & SPROP_NOSCALE )
 		{
@@ -109,7 +109,7 @@ static float DecodeFloat(SendProp const *pProp, bf_read *pIn)
 	}
 	else // standard clamped-range float
 	{
-		unsigned long dwInterp = pIn->ReadUBitLong(pProp->m_nBits);
+		unsigned int dwInterp = pIn->ReadUBitLong(pProp->m_nBits);
 		float fVal = (float)dwInterp / ((1 << pProp->m_nBits) - 1);
 		fVal = pProp->m_fLowValue + (pProp->m_fHighValue - pProp->m_fLowValue) * fVal;
 		return fVal;
@@ -281,7 +281,7 @@ void Int_Decode( DecodeInfo *pInfo )
 	{
 		if ( flags & SPROP_UNSIGNED )
 		{
-			pInfo->m_Value.m_Int = (long)pInfo->m_pIn->ReadVarInt32();
+			pInfo->m_Value.m_Int = (int)pInfo->m_pIn->ReadVarInt32();
 		}
 		else
 		{
@@ -295,7 +295,7 @@ void Int_Decode( DecodeInfo *pInfo )
 
 		if( bits != 32 && (flags & SPROP_UNSIGNED) == 0 )
 		{
-			unsigned long highbit = 1ul << (pProp->m_nBits - 1);
+			unsigned int highbit = 1ul << (pProp->m_nBits - 1);
 			if ( pInfo->m_Value.m_Int & highbit )
 			{
 				pInfo->m_Value.m_Int -= highbit; // strip high bit...

engine/dt_localtransfer.cpp

@@ -175,12 +175,12 @@ void BuildPropOffsetToIndexMap( CSendTablePrecalc *pPrecalc, const CStandardSend
 		{
 			const SendProp *pProp = pPrecalc->m_Props[i];
 			
-			int offset = pProp->GetOffset() + (int)pmStack.GetCurStructBase() - 1;
+			intp offset = pProp->GetOffset() + (intp)pmStack.GetCurStructBase() - 1;
 			int elementCount = 1;
 			int elementStride = 0;
 			if ( pProp->GetType() == DPT_Array )
 			{
-				offset = pProp->GetArrayProp()->GetOffset() + (int)pmStack.GetCurStructBase() - 1;
+				offset = pProp->GetArrayProp()->GetOffset() + (intp)pmStack.GetCurStructBase() - 1;
 				elementCount = pProp->m_nElements;
 				elementStride = pProp->m_ElementStride;
 			}

engine/dt_stack.h

@@ -121,7 +121,7 @@ inline unsigned char* UpdateRoutesExplicit_Template( DTStack *pStack, ProxyCalle
 	// Early out.
 	unsigned short iPropProxyIndex = pStack->m_pPrecalc->m_PropProxyIndices[pStack->m_iCurProp];
 	unsigned char **pTest = &pStack->m_pProxies[iPropProxyIndex];
-	if ( *pTest != (unsigned char*)0xFFFFFFFF )
+	if ( *pTest != (unsigned char*)-1 )
 		return *pTest;
 	
 	// Ok.. setup this proxy.
@@ -133,7 +133,7 @@ inline unsigned char* UpdateRoutesExplicit_Template( DTStack *pStack, ProxyCalle
 		CSendTablePrecalc::CProxyPathEntry *pEntry = &pStack->m_pPrecalc->m_ProxyPathEntries[proxyPath.m_iFirstEntry + i];
 		int iProxy = pEntry->m_iProxy;
 		
-		if ( pStack->m_pProxies[iProxy] == (unsigned char*)0xFFFFFFFF )
+		if ( pStack->m_pProxies[iProxy] == (unsigned char*)-1 )
 		{
 			pStack->m_pProxies[iProxy] = ProxyCaller::CallProxy( pStack, pStructBase, pEntry->m_iDatatableProp );
 			if ( !pStack->m_pProxies[iProxy] )

engine/event_system.h

@@ -52,7 +52,7 @@ public:
 	{
 		if ( pData )
 		{
-			delete pData;
+			delete[] pData;
 		}
 	}
 
@@ -65,7 +65,7 @@ public:
 		pSendTable = src.pSendTable;
 		pClientClass = src.pClientClass;
 		filter.AddPlayersFromFilter( &src.filter );
-				
+
 		if ( src.pData )
 		{
 			int size = Bits2Bytes( src.bits );

engine/filesystem_engine.cpp

@@ -8,7 +8,6 @@
 #include "quakedef.h" // for MAX_OSPATH
 #include <stdlib.h>
 #include <assert.h>
-#include <malloc.h>
 #include "filesystem.h"
 #include "bitmap/tgawriter.h"
 #include <tier2/tier2.h>

engine/gl_rsurf.cpp

@@ -4099,7 +4099,9 @@ CBrushBatchRender::brushrender_t *CBrushBatchRender::FindOrCreateRenderBatch( mo
 	surfaceList.Sort( SurfaceCmp );
 	renderT.pPlanes = new cplane_t *[planeList.Count()];
 	renderT.planeCount = planeList.Count();
-	memcpy( renderT.pPlanes, planeList.Base(), sizeof(cplane_t *)*planeList.Count() );
+
+	if( planeList.Base() )
+		memcpy( renderT.pPlanes, planeList.Base(), sizeof(cplane_t *)*planeList.Count() );
 	renderT.pSurfaces = new brushrendersurface_t[surfaceList.Count()];
 	renderT.surfaceCount = surfaceList.Count();
 
@@ -4745,7 +4747,7 @@ struct EnumLeafBoxInfo_t
 	VectorAligned m_vecBoxCenter;
 	VectorAligned m_vecBoxHalfDiagonal;
 	ISpatialLeafEnumerator *m_pIterator;
-	int	m_nContext;
+	intp	m_nContext;
 };
 
 struct EnumLeafSphereInfo_t
@@ -4755,7 +4757,7 @@ struct EnumLeafSphereInfo_t
 	Vector m_vecBoxCenter;
 	Vector m_vecBoxHalfDiagonal;
 	ISpatialLeafEnumerator *m_pIterator;
-	int	m_nContext;
+	intp	m_nContext;
 };
 
 //-----------------------------------------------------------------------------
@@ -5092,7 +5094,7 @@ bool EnumerateLeafInSphere_R( mnode_t *node, EnumLeafSphereInfo_t& info, int nTe
 //-----------------------------------------------------------------------------
 
 static bool EnumerateLeavesAlongRay_R( mnode_t *node, Ray_t const& ray, 
-	float start, float end, ISpatialLeafEnumerator* pEnum, int context )
+	float start, float end, ISpatialLeafEnumerator* pEnum, intp context )
 {
 	// no polygons in solid nodes (don't report these leaves either)
 	if (node->contents == CONTENTS_SOLID)
@@ -5151,7 +5153,7 @@ static bool EnumerateLeavesAlongRay_R( mnode_t *node, Ray_t const& ray,
 //-----------------------------------------------------------------------------
 
 static bool EnumerateLeavesAlongExtrudedRay_R( mnode_t *node, Ray_t const& ray, 
-	float start, float end, ISpatialLeafEnumerator* pEnum, int context )
+	float start, float end, ISpatialLeafEnumerator* pEnum, intp context )
 {
 	// no polygons in solid nodes (don't report these leaves either)
 	if (node->contents == CONTENTS_SOLID)
@@ -5274,10 +5276,10 @@ public:
 	int LeafCount() const;
 
 	// Enumerates the leaves along a ray, box, etc.
-	bool EnumerateLeavesAtPoint( const Vector& pt, ISpatialLeafEnumerator* pEnum, int context );
-	bool EnumerateLeavesInBox( const Vector& mins, const Vector& maxs, ISpatialLeafEnumerator* pEnum, int context );
-	bool EnumerateLeavesInSphere( const Vector& center, float radius, ISpatialLeafEnumerator* pEnum, int context );
-	bool EnumerateLeavesAlongRay( Ray_t const& ray, ISpatialLeafEnumerator* pEnum, int context );
+	bool EnumerateLeavesAtPoint( const Vector& pt, ISpatialLeafEnumerator* pEnum, intp context );
+	bool EnumerateLeavesInBox( const Vector& mins, const Vector& maxs, ISpatialLeafEnumerator* pEnum, intp context );
+	bool EnumerateLeavesInSphere( const Vector& center, float radius, ISpatialLeafEnumerator* pEnum, intp context );
+	bool EnumerateLeavesAlongRay( Ray_t const& ray, ISpatialLeafEnumerator* pEnum, intp context );
 };
 
 //-----------------------------------------------------------------------------
@@ -5302,7 +5304,7 @@ int CEngineBSPTree::LeafCount() const
 //-----------------------------------------------------------------------------
 
 bool CEngineBSPTree::EnumerateLeavesAtPoint( const Vector& pt, 
-									ISpatialLeafEnumerator* pEnum, int context )
+									ISpatialLeafEnumerator* pEnum, intp context )
 {
 	int leaf = CM_PointLeafnum( pt );
 	return pEnum->EnumerateLeaf( leaf, context );
@@ -5313,7 +5315,7 @@ static ConVar opt_EnumerateLeavesFastAlgorithm( "opt_EnumerateLeavesFastAlgorith
 
 
 bool CEngineBSPTree::EnumerateLeavesInBox( const Vector& mins, const Vector& maxs, 
-									ISpatialLeafEnumerator* pEnum, int context )
+									ISpatialLeafEnumerator* pEnum, intp context )
 {
 	if ( !host_state.worldmodel )
 		return false;
@@ -5338,7 +5340,7 @@ bool CEngineBSPTree::EnumerateLeavesInBox( const Vector& mins, const Vector& max
 
 
 bool CEngineBSPTree::EnumerateLeavesInSphere( const Vector& center, float radius, 
-									ISpatialLeafEnumerator* pEnum, int context )
+									ISpatialLeafEnumerator* pEnum, intp context )
 {
 	EnumLeafSphereInfo_t info;
 	info.m_vecCenter = center;
@@ -5352,7 +5354,7 @@ bool CEngineBSPTree::EnumerateLeavesInSphere( const Vector& center, float radius
 }
 
 
-bool CEngineBSPTree::EnumerateLeavesAlongRay( Ray_t const& ray, ISpatialLeafEnumerator* pEnum, int context )
+bool CEngineBSPTree::EnumerateLeavesAlongRay( Ray_t const& ray, ISpatialLeafEnumerator* pEnum, intp context )
 {
 	if (!ray.m_IsSwept)
 	{

engine/hltvdemo.cpp

@@ -22,6 +22,7 @@
 #include "host.h"
 #include "server.h"
 #include "networkstringtableclient.h"
+#include "vcrmode.h"
 
 // memdbgon must be the last include file in a .cpp file!!!
 #include "tier0/memdbgon.h"

engine/host.cpp

@@ -2750,10 +2750,6 @@ void CheckSpecialCheatVars()
 	if ( !mat_picmip )
 		mat_picmip = g_pCVar->FindVar( "mat_picmip" );
 
-	// In multiplayer, don't allow them to set mat_picmip > 2.	
-	if ( mat_picmip )
-		CheckVarRange_Generic( mat_picmip, -1, 2 );
-	
 	CheckVarRange_r_rootlod();
 	CheckVarRange_r_lod();
 	HandleServerAllowColorCorrection();
@@ -4850,7 +4846,9 @@ void Host_FreeToLowMark( bool server )
 //-----------------------------------------------------------------------------
 void Host_Shutdown(void)
 {
+#ifndef ANDROID
 	extern void ShutdownMixerControls();
+#endif
 
 	if ( host_checkheap )
 	{
@@ -4962,7 +4960,7 @@ void Host_Shutdown(void)
 
 #ifndef SWDS
 	TRACESHUTDOWN( Key_Shutdown() );
-#ifndef _X360
+#if !defined _X360 && !defined ANDROID
 	TRACESHUTDOWN( ShutdownMixerControls() );
 #endif
 #endif

engine/host_saverestore.cpp

@@ -806,7 +806,8 @@ int CSaveRestore::SaveGameSlot( const char *pSaveName, const char *pSaveComment,
 	m_bWaitingForSafeDangerousSave = bIsAutosaveDangerous;
 
 	int iHeaderBufferSize = 64 + tokenSize + pSaveData->GetCurPos();
-	void *pMem = malloc(iHeaderBufferSize);
+	void *pMem = new char[iHeaderBufferSize];
+
 	CUtlBuffer saveHeader( pMem, iHeaderBufferSize );
 
 	// Write the header -- THIS SHOULD NEVER CHANGE STRUCTURE, USE SAVE_HEADER FOR NEW HEADER INFORMATION

engine/l_studio.cpp

@@ -872,7 +872,7 @@ public:
 	virtual void SetupLighting( const Vector &vecCenter );
 	virtual void SuppressEngineLighting( bool bSuppress );
 
-	inline vertexFileHeader_t *CacheVertexData() { return g_pMDLCache->GetVertexData( (MDLHandle_t)(int)m_pStudioHdr->virtualModel&0xffff ); }
+	inline vertexFileHeader_t *CacheVertexData() { return g_pMDLCache->GetVertexData( VoidPtrToMDLHandle( m_pStudioHdr->VirtualModel() ) ); }
 
 	bool Init();
 	void Shutdown();
@@ -3637,12 +3637,12 @@ void CModelRender::ValidateStaticPropColorData( ModelInstanceHandle_t handle )
 	// fetch the header
 	CUtlBuffer utlBuf;
 	char fileName[MAX_PATH];
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() == HDR_TYPE_NONE || g_bBakedPropLightingNoSeparateHDR )
+	if ( !g_pMaterialSystemHardwareConfig->GetHDREnabled() || g_bBakedPropLightingNoSeparateHDR )
 	{
 		Q_snprintf( fileName, sizeof( fileName ), "sp_%d%s.vhv", StaticPropMgr()->GetStaticPropIndex( pProp ), GetPlatformExt() );
 	}
 	else
-	{	
+	{
 		Q_snprintf( fileName, sizeof( fileName ), "sp_hdr_%d%s.vhv", StaticPropMgr()->GetStaticPropIndex( pProp ), GetPlatformExt() );
 	}
 
@@ -3930,13 +3930,13 @@ bool CModelRender::LoadStaticPropColorData( IHandleEntity *pProp, DataCacheHandl
 
 	// each static prop has its own compiled color mesh
 	char fileName[MAX_PATH];
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() == HDR_TYPE_NONE || g_bBakedPropLightingNoSeparateHDR )
+	if ( !g_pMaterialSystemHardwareConfig->GetHDREnabled() || g_bBakedPropLightingNoSeparateHDR )
 	{
-        Q_snprintf( fileName, sizeof( fileName ), "sp_%d%s.vhv", StaticPropMgr()->GetStaticPropIndex( pProp ), GetPlatformExt() );
+	        Q_snprintf( fileName, sizeof( fileName ), "sp_%d%s.vhv", StaticPropMgr()->GetStaticPropIndex( pProp ), GetPlatformExt() );
 	}
 	else
 	{
-        Q_snprintf( fileName, sizeof( fileName ), "sp_hdr_%d%s.vhv", StaticPropMgr()->GetStaticPropIndex( pProp ), GetPlatformExt() );
+        	Q_snprintf( fileName, sizeof( fileName ), "sp_hdr_%d%s.vhv", StaticPropMgr()->GetStaticPropIndex( pProp ), GetPlatformExt() );
 	}
 
 	// mark as invalid, async callback will set upon completion
@@ -4121,7 +4121,7 @@ bool CModelRender::UpdateStaticPropColorData( IHandleEntity *pProp, ModelInstanc
 	if ( !bDebugColor )
 	{
 		// vertexes must be available for lighting calculation
-		vertexFileHeader_t *pVertexHdr = g_pMDLCache->GetVertexData( (MDLHandle_t)(int)pStudioHdr->virtualModel&0xffff );
+		vertexFileHeader_t *pVertexHdr = g_pMDLCache->GetVertexData( VoidPtrToMDLHandle( pStudioHdr->VirtualModel() ) );
 		if ( !pVertexHdr )
 		{
 			// data not available yet

engine/materialproxyfactory.cpp

@@ -6,7 +6,6 @@
 //=============================================================================//
 
 #include <stdlib.h>
-#include <malloc.h>
 #include <string.h>
 #include "materialsystem/imaterialproxy.h"
 #include "materialproxyfactory.h"

engine/matsys_interface.cpp

@@ -251,6 +251,9 @@ static const char *s_pRegistryConVars[] =
 #if defined( OSX )
 	#define MOD_VIDEO_CONFIG_SETTINGS "videoconfig_mac.cfg"
 	#define USE_VIDEOCONFIG_FILE 1
+#elif defined( ANDROID )
+	#define MOD_VIDEO_CONFIG_SETTINGS "videoconfig_android.cfg"
+	#define USE_VIDEOCONFIG_FILE 1
 #elif defined( POSIX )
 	#define MOD_VIDEO_CONFIG_SETTINGS "videoconfig_linux.cfg"
 	#define USE_VIDEOCONFIG_FILE 1

engine/modelloader.cpp

@@ -483,7 +483,8 @@ void CMapLoadHelper::Init( model_t *pMapModel, const char *loadname )
 
 	s_pMap = &g_ModelLoader.m_worldBrushData;
 
-#if 0
+	// nillerusr: Fuck you johns
+
 	// XXX(johns): There are security issues with this system currently. sv_pure doesn't handle unexpected/mismatched
 	//             lumps, so players can create lumps for maps not using them to wallhack/etc.. Currently unused,
 	//             disabling until we have time to make a proper security pass.
@@ -530,7 +531,6 @@ void CMapLoadHelper::Init( model_t *pMapModel, const char *loadname )
 			}
 		}
 	}
-#endif
 }
 
 //-----------------------------------------------------------------------------
@@ -1038,7 +1038,7 @@ bool Map_CheckForHDR( model_t *pModel, const char *pLoadName )
 	}
 	
 	bool bEnableHDR = ( IsX360() && bHasHDR ) ||
-		( bHasHDR && ( mat_hdr_level.GetInt() >= 2 ) &&
+		( bHasHDR &&
 		( g_pMaterialSystemHardwareConfig->GetDXSupportLevel() >= 90 ) );
 	
 	EnableHDR( bEnableHDR );
@@ -1831,7 +1831,7 @@ void *Hunk_AllocNameAlignedClear_( int size, int alignment, const char *pHunkNam
 	Assert(IsPowerOfTwo(alignment));
 	void *pMem = Hunk_AllocName( alignment + size, pHunkName );
 	memset( pMem, 0, size + alignment );
-	pMem = (void *)( ( ( ( unsigned long )pMem ) + (alignment-1) ) & ~(alignment-1) );
+	pMem = (void *)( ( ( ( uintp )pMem ) + (alignment-1) ) & ~(alignment-1) );
 
 	return pMem;
 }
@@ -1856,8 +1856,7 @@ void Mod_LoadFaces( void )
 	int			ti, di;
 
 	int face_lump_to_load = LUMP_FACES;
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&
-		CMapLoadHelper::LumpSize( LUMP_FACES_HDR ) > 0 )
+	if ( g_pMaterialSystemHardwareConfig->GetHDREnabled() && CMapLoadHelper::LumpSize( LUMP_FACES_HDR ) > 0 )
 	{
 		face_lump_to_load = LUMP_FACES_HDR;
 	}
@@ -1870,6 +1869,12 @@ void Mod_LoadFaces( void )
 
 	// align these allocations
 	// If you trip one of these, you need to rethink the alignment of the struct
+#ifdef PLATFORM_64BITS
+    msurface1_t *out1 = Hunk_AllocNameAlignedClear< msurface1_t >( count, alignof(msurface1_t), va( "%s [%s]", lh.GetLoadName(), "surface1" ) );
+    msurface2_t *out2 = Hunk_AllocNameAlignedClear< msurface2_t >( count, alignof(msurface2_t), va( "%s [%s]", lh.GetLoadName(), "surface2" ) );
+
+    msurfacelighting_t *pLighting = Hunk_AllocNameAlignedClear< msurfacelighting_t >( count, alignof(msurfacelighting_t), va( "%s [%s]", lh.GetLoadName(), "surfacelighting" ) );
+#else
 	Assert( sizeof(msurface1_t) == 16 );
 	Assert( sizeof(msurface2_t) == 32 );
 	Assert( sizeof(msurfacelighting_t) == 32 );
@@ -1878,6 +1883,7 @@ void Mod_LoadFaces( void )
 	msurface2_t *out2 = Hunk_AllocNameAlignedClear< msurface2_t >( count, 32, va( "%s [%s]", lh.GetLoadName(), "surface2" ) );
 
 	msurfacelighting_t *pLighting = Hunk_AllocNameAlignedClear< msurfacelighting_t >( count, 32, va( "%s [%s]", lh.GetLoadName(), "surfacelighting" ) );
+#endif
 
 	lh.GetMap()->surfaces1 = out1;
 	lh.GetMap()->surfaces2 = out2;
@@ -2289,8 +2295,7 @@ void Mod_LoadLeafs( void )
 		Mod_LoadLeafs_Version_0( lh );
 		break;
 	case 1:
-		if( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&
-	  		  CMapLoadHelper::LumpSize( LUMP_LEAF_AMBIENT_LIGHTING_HDR ) > 0 )
+		if( g_pMaterialSystemHardwareConfig->GetHDREnabled() && CMapLoadHelper::LumpSize( LUMP_LEAF_AMBIENT_LIGHTING_HDR ) > 0 )
 		{
 			CMapLoadHelper mlh( LUMP_LEAF_AMBIENT_LIGHTING_HDR );
 			CMapLoadHelper mlhTable( LUMP_LEAF_AMBIENT_INDEX_HDR );
@@ -2385,7 +2390,7 @@ void Mod_LoadCubemapSamples( void )
 	lh.GetMap()->m_pCubemapSamples = out;
 	lh.GetMap()->m_nCubemapSamples = count;
 
-	bool bHDR = g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE;
+	bool bHDR =  g_pMaterialSystemHardwareConfig->GetHDREnabled(); //g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE;
 	int nCreateFlags = bHDR ? 0 : TEXTUREFLAGS_SRGB;
 
 	// We have separate HDR versions of the textures.  In order to deal with this,
@@ -2862,7 +2867,7 @@ void Mod_TouchAllData( model_t *pModel, int nServerCount )
 		// skip self, start at children
 		for ( int i=1; i<pVirtualModel->m_group.Count(); ++i )
 		{
-			MDLHandle_t childHandle = (MDLHandle_t)(int)pVirtualModel->m_group[i].cache&0xffff;
+			MDLHandle_t childHandle = (MDLHandle_t)(intp)pVirtualModel->m_group[i].cache&0xffff;
 			model_t *pChildModel = (model_t *)g_pMDLCache->GetUserData( childHandle );
 			if ( pChildModel )
 			{
@@ -4327,7 +4332,7 @@ public:
 		m_pShared = pBrush->brush.pShared;
 		m_count = 0;
 	}
-	bool EnumerateLeaf( int leaf, int )
+	bool EnumerateLeaf( int leaf, intp )
 	{
 		// garymcthack - need to test identity brush models
 		int flags = ( m_pShared->leafs[leaf].leafWaterDataID == -1 ) ? SURFDRAW_ABOVEWATER : SURFDRAW_UNDERWATER;
@@ -4374,7 +4379,7 @@ static void MarkBrushModelWaterSurfaces( model_t* world,
 	model_t* pTemp = host_state.worldmodel;
 	CBrushBSPIterator brushIterator( world, brush );
 	host_state.SetWorldModel( world );
-	g_pToolBSPTree->EnumerateLeavesInBox( mins, maxs, &brushIterator, (int)brush );
+	g_pToolBSPTree->EnumerateLeavesInBox( mins, maxs, &brushIterator, (intp)brush );
 	brushIterator.CheckSurfaces();
 	host_state.SetWorldModel( pTemp );
 }
@@ -4453,8 +4458,7 @@ void CModelLoader::Map_LoadModel( model_t *mod )
 
 	// Until BSP version 19, this must occur after loading texinfo
 	COM_TimestampedLog( "  Mod_LoadLighting" );
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&
-		CMapLoadHelper::LumpSize( LUMP_LIGHTING_HDR ) > 0 )
+	if ( g_pMaterialSystemHardwareConfig->GetHDREnabled() && CMapLoadHelper::LumpSize( LUMP_LIGHTING_HDR ) > 0 )
 	{
 		CMapLoadHelper mlh( LUMP_LIGHTING_HDR );
 		Mod_LoadLighting( mlh );
@@ -4546,8 +4550,7 @@ void CModelLoader::Map_LoadModel( model_t *mod )
 		&m_worldBrushData.m_nAreas );
 
 	COM_TimestampedLog( "  Mod_LoadWorldlights" );
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&
-  		  CMapLoadHelper::LumpSize( LUMP_WORLDLIGHTS_HDR ) > 0 )
+	if ( g_pMaterialSystemHardwareConfig->GetHDREnabled() && CMapLoadHelper::LumpSize( LUMP_WORLDLIGHTS_HDR ) > 0 )
 	{
 		CMapLoadHelper mlh( LUMP_WORLDLIGHTS_HDR );
 		Mod_LoadWorldlights( mlh, true );

engine/net_ws.cpp

@@ -417,7 +417,7 @@ bool NET_StringToAdr ( const char *s, netadr_t *a)
 
 CNetChan *NET_FindNetChannel(int socket, netadr_t &adr)
 {
-	AUTO_LOCK( s_NetChannels );
+	AUTO_LOCK_FM( s_NetChannels );
 
 	int numChannels = s_NetChannels.Count();
 
@@ -798,7 +798,7 @@ INetChannel *NET_CreateNetChannel(int socket, netadr_t *adr, const char * name,
 		// create new channel
 		chan = new CNetChan();
 
-		AUTO_LOCK( s_NetChannels );
+		AUTO_LOCK_FM( s_NetChannels );
 		s_NetChannels.AddToTail( chan );
 	}
 
@@ -817,7 +817,7 @@ void NET_RemoveNetChannel(INetChannel *netchan, bool bDeleteNetChan)
 		return;
 	}
 
-	AUTO_LOCK( s_NetChannels );
+	AUTO_LOCK_FM( s_NetChannels );
 	if ( s_NetChannels.Find( static_cast<CNetChan*>(netchan) ) == s_NetChannels.InvalidIndex() )
 	{
 		DevMsg(1, "NET_CloseNetChannel: unknown channel.\n");
@@ -1389,7 +1389,7 @@ bool NET_GetLoopPacket ( netpacket_t * packet )
 
 	if ( loop->data != loop->defbuffer )
 	{
-		delete loop->data;
+		delete[] loop->data;
 		loop->data = loop->defbuffer;
 	}
 
@@ -1677,7 +1677,7 @@ netpacket_t *NET_GetPacket (int sock, byte *scratch )
 
 void NET_ProcessPending( void )
 {
-	AUTO_LOCK( s_PendingSockets );
+	AUTO_LOCK_FM( s_PendingSockets );
 	for ( int i=0; i<s_PendingSockets.Count();i++ )
 	{
 		pendingsocket_t * psock = &s_PendingSockets[i];
@@ -1712,7 +1712,7 @@ void NET_ProcessPending( void )
 
 		if ( cmd == STREAM_CMD_ACKN )
 		{
-			AUTO_LOCK( s_NetChannels );
+			AUTO_LOCK_FM( s_NetChannels );
 			for ( int j = 0; j < s_NetChannels.Count(); j++ )
 			{
 				CNetChan * chan = s_NetChannels[j];
@@ -1792,7 +1792,7 @@ void NET_ProcessListen(int sock)
 	psock.addr.SetFromSockadr( &sa );
 	psock.time = net_time;
 
-	AUTO_LOCK( s_PendingSockets );
+	AUTO_LOCK_FM( s_PendingSockets );
 	s_PendingSockets.AddToTail( psock );
 
 	// tell client to send challenge number to identify
@@ -1823,7 +1823,7 @@ void NET_ProcessSocket( int sock, IConnectionlessPacketHandler *handler )
 
 	// Scope for the auto_lock
 	{
-		AUTO_LOCK( s_NetChannels );
+		AUTO_LOCK_FM( s_NetChannels );
 
 		// get streaming data from channel sockets
 		int numChannels = s_NetChannels.Count();
@@ -2553,7 +2553,7 @@ void NET_CloseAllSockets (void)
 	}
 
 	// shut down all pending sockets
-	AUTO_LOCK( s_PendingSockets );
+	AUTO_LOCK_FM( s_PendingSockets );
 	for(int j=0; j<s_PendingSockets.Count();j++ )
 	{
 		NET_CloseSocket( s_PendingSockets[j].newsock );
@@ -2836,7 +2836,7 @@ void NET_LogServerStatus( void )
 		return;
 	}
 
-	AUTO_LOCK( s_NetChannels );
+	AUTO_LOCK_FM( s_NetChannels );
 	int numChannels = s_NetChannels.Count();
 
 	if ( numChannels == 0 )
@@ -3376,7 +3376,7 @@ CON_COMMAND( net_channels, "Shows net channel info" )
 		return;
 	}
 
-	AUTO_LOCK( s_NetChannels );
+	AUTO_LOCK_FM( s_NetChannels );
 	for ( int i = 0; i < numChannels; i++ )
 	{
 		NET_PrintChannelStatus( s_NetChannels[i] );
@@ -3391,7 +3391,7 @@ CON_COMMAND( net_start, "Inits multiplayer network sockets" )
 
 CON_COMMAND( net_status, "Shows current network status" )
 {
-	AUTO_LOCK( s_NetChannels );
+	AUTO_LOCK_FM( s_NetChannels );
 	int numChannels = s_NetChannels.Count();
 
 	ConMsg("Net status for host %s:\n", 

engine/packed_entity.cpp

@@ -5,7 +5,11 @@
 // $NoKeywords: $
 //
 //=============================================================================//
+#ifdef OSX
+#include <malloc/malloc.h>
+#else
 #include <malloc.h>
+#endif
 #include <string.h>
 #include <assert.h>
 #include "packed_entity.h"

engine/pr_edict.cpp

@@ -32,7 +32,11 @@
 
 
 #ifdef _DEBUG
+#ifdef OSX
+#include <malloc/malloc.h>
+#else
 #include <malloc.h>
+#endif
 #endif // _DEBUG
 
 static ConVar		sv_useexplicitdelete( "sv_useexplicitdelete", "1", FCVAR_DEVELOPMENTONLY, "Explicitly delete dormant client entities caused by AllowImmediateReuse()." );

engine/pure_server.h

@@ -96,7 +96,7 @@ private:
 		CPureServerWhitelist::CCommand *pBestEntry );
 
 	unsigned short	m_LoadCounter;	// Incremented as we load things so their m_LoadOrder increases.
-	volatile long int m_RefCount;
+	volatile int32 m_RefCount;
 
 	// Commands are applied to files in order.
 	CUtlDict<CCommand*,int>	m_FileCommands;				// file commands

engine/r_decal.cpp

@@ -2001,9 +2001,9 @@ void R_DrawDecalsAllImmediate_GatherDecals( IMatRenderContext *pRenderContext, i
 			if ( g_aDecalSortTrees[iSortTree].m_aDecalSortBuckets[iGroup][iTreeType].Element( iBucket ).m_nCheckCount != nCheckCount )
 				continue;
 
-			int iHead = g_aDecalSortTrees[iSortTree].m_aDecalSortBuckets[iGroup][iTreeType].Element( iBucket ).m_iHead;
+			intp iHead = g_aDecalSortTrees[iSortTree].m_aDecalSortBuckets[iGroup][iTreeType].Element( iBucket ).m_iHead;
 
-			int iElement = iHead;
+			intp iElement = iHead;
 			while ( iElement != g_aDecalSortPool.InvalidIndex() )
 			{
 				decal_t *pDecal = g_aDecalSortPool.Element( iElement );
@@ -2154,11 +2154,11 @@ void R_DrawDecalsAllImmediate( IMatRenderContext *pRenderContext, int iGroup, in
 		{
 			if ( g_aDecalSortTrees[iSortTree].m_aDecalSortBuckets[iGroup][iTreeType].Element( iBucket ).m_nCheckCount != nCheckCount )
 				continue;
-			
-			int iHead = g_aDecalSortTrees[iSortTree].m_aDecalSortBuckets[iGroup][iTreeType].Element( iBucket ).m_iHead;
-			
+
+			intp iHead = g_aDecalSortTrees[iSortTree].m_aDecalSortBuckets[iGroup][iTreeType].Element( iBucket ).m_iHead;
+
 			int nCount;
-			int iElement = iHead;
+			intp iElement = iHead;
 			while ( iElement != g_aDecalSortPool.InvalidIndex() )
 			{
 				decal_t *pDecal = g_aDecalSortPool.Element( iElement );
@@ -2304,7 +2304,7 @@ inline void R_DrawDecalMeshList( DecalMeshList_t &meshList )
 }
 
 #define DECALMARKERS_SWITCHSORTTREE ((decal_t *)0x00000000)
-#define DECALMARKERS_SWITCHBUCKET	((decal_t *)0xFFFFFFFF)
+#define DECALMARKERS_SWITCHBUCKET	((decal_t *)-1)
 //-----------------------------------------------------------------------------
 //
 //-----------------------------------------------------------------------------
@@ -2330,7 +2330,7 @@ void R_DrawDecalsAll_GatherDecals( IMatRenderContext *pRenderContext, int iGroup
 			if ( bucket.m_nCheckCount != nCheckCount )
 				continue;
 
-			int iHead = bucket.m_iHead;
+			intp iHead = bucket.m_iHead;
 			if ( !g_aDecalSortPool.IsValidIndex( iHead ) )
 				continue;
 
@@ -2346,7 +2346,7 @@ void R_DrawDecalsAll_GatherDecals( IMatRenderContext *pRenderContext, int iGroup
 
 			DrawDecals.AddToTail( DECALMARKERS_SWITCHBUCKET );
 
-			int iElement = iHead;
+			intp iElement = iHead;
 			while ( iElement != g_aDecalSortPool.InvalidIndex() )
 			{
 				decal_t *pDecal = g_aDecalSortPool.Element( iElement );
@@ -2647,7 +2647,7 @@ void R_DrawDecalsAll( IMatRenderContext *pRenderContext, int iGroup, int iTreeTy
 			if ( bucket.m_nCheckCount != nCheckCount )
 				continue;
 			
-			int iHead = bucket.m_iHead;
+			intp iHead = bucket.m_iHead;
 			if ( !g_aDecalSortPool.IsValidIndex( iHead ) )
 				continue;
 
@@ -2666,7 +2666,7 @@ void R_DrawDecalsAll( IMatRenderContext *pRenderContext, int iGroup, int iTreeTy
 			bool bBatchInit = true;
 			
 			int nCount;
-			int iElement = iHead;
+			intp iElement = iHead;
 			while ( iElement != g_aDecalSortPool.InvalidIndex() )
 			{
 				decal_t *pDecal = g_aDecalSortPool.Element( iElement );
@@ -3015,7 +3015,7 @@ void DecalSurfaceAdd( SurfaceHandle_t surfID, int iGroup )
 		}
 
 		pDecal->flags &= ~FDECAL_HASUPDATED;
-		int iPool = g_aDecalSortPool.Alloc( true );
+		intp iPool = g_aDecalSortPool.Alloc( true );
 		if ( iPool != g_aDecalSortPool.InvalidIndex() )
 		{
 			g_aDecalSortPool[iPool] = pDecal;
@@ -3024,7 +3024,7 @@ void DecalSurfaceAdd( SurfaceHandle_t surfID, int iGroup )
 			DecalMaterialBucket_t &bucket = sortTree.m_aDecalSortBuckets[iGroup][iTreeType].Element( pDecal->m_iSortMaterial );
 			if ( bucket.m_nCheckCount == nCheckCount )
 			{	
-				int iHead = bucket.m_iHead;
+				intp iHead = bucket.m_iHead;
 				g_aDecalSortPool.LinkBefore( iHead, iPool );
 			}
 			

engine/r_decal.h

@@ -74,7 +74,7 @@ struct DecalMaterialSortData_t
 
 struct DecalMaterialBucket_t
 {
-	int			m_iHead;
+	intp			m_iHead;
 	int			m_nCheckCount;
 };
 
@@ -82,16 +82,16 @@ inline bool DecalSortTreeSortLessFunc( const DecalMaterialSortData_t &decal1, co
 {
 	if ( ( decal1.m_iLightmapPage == -1 ) || ( decal2.m_iLightmapPage == -1 ) )
 	{
-		return ( ( int )decal1.m_pMaterial < ( int )decal2.m_pMaterial );
+		return ( ( intp )decal1.m_pMaterial < ( intp )decal2.m_pMaterial );
 	}
 
-	if ( ( int )decal1.m_pMaterial == ( int )decal2.m_pMaterial )
+	if ( ( intp )decal1.m_pMaterial == ( intp )decal2.m_pMaterial )
 	{
 		return ( decal1.m_iLightmapPage < decal2.m_iLightmapPage );
 	}
 	else
 	{
-		return ( ( int )decal1.m_pMaterial < ( int )decal2.m_pMaterial );
+		return ( ( intp )decal1.m_pMaterial < ( intp )decal2.m_pMaterial );
 	}
 }
 

engine/saverestore_filesystem.cpp

@@ -154,7 +154,7 @@ private:
 	CSaveDirectory	*m_pSaveDirectory;
 	CUtlMap<CUtlSymbol, SaveFile_t> &GetDirectory( void ) { return m_pSaveDirectory->m_Files; }
 	SaveFile_t &GetFile( const int idx ) { return m_pSaveDirectory->m_Files[idx]; }
-	SaveFile_t &GetFile( const FileHandle_t hFile ) { return GetFile( (unsigned int)hFile ); }
+	SaveFile_t &GetFile( const FileHandle_t hFile ) { return GetFile( (uintp)hFile ); }
 
 	FileHandle_t	GetFileHandle( const char *pFileName );
 	int				GetFileIndex( const char *pFileName );
@@ -310,12 +310,12 @@ int CSaveRestoreFileSystem::GetFileIndex( const char *filename )
 
 FileHandle_t CSaveRestoreFileSystem::GetFileHandle( const char *filename )
 {
-	int idx = GetFileIndex( filename );
+	intp idx = GetFileIndex( filename );
 	if ( idx == INVALID_INDEX )
 	{
 		idx = 0;
 	}
-	return (void*)idx;
+	return (FileHandle_t)idx;
 }
 
 //-----------------------------------------------------------------------------
@@ -331,7 +331,7 @@ bool CSaveRestoreFileSystem::FileExists( const char *pFileName, const char *pPat
 //-----------------------------------------------------------------------------
 bool CSaveRestoreFileSystem::HandleIsValid( FileHandle_t hFile )
 {
-	return hFile && GetDirectory().IsValidIndex( (unsigned int)hFile );
+	return hFile && GetDirectory().IsValidIndex( (uintp)hFile );
 }
 
 //-----------------------------------------------------------------------------
@@ -339,7 +339,7 @@ bool CSaveRestoreFileSystem::HandleIsValid( FileHandle_t hFile )
 //-----------------------------------------------------------------------------
 void CSaveRestoreFileSystem::RenameFile( char const *pOldPath, char const *pNewPath, const char *pathID )
 {
-	int idx = GetFileIndex( pOldPath );
+	intp idx = GetFileIndex( pOldPath );
 	if ( idx != INVALID_INDEX )
 	{
 		CUtlSymbol newID = AddString( Q_UnqualifiedFileName( pNewPath ) );
@@ -369,7 +369,7 @@ FileHandle_t CSaveRestoreFileSystem::Open( const char *pFullName, const char *pO
 {
 	SaveFile_t *pFile = NULL;
 	CUtlSymbol id = AddString( Q_UnqualifiedFileName( pFullName ) );
-	int idx = GetDirectory().Find( id );
+	intp idx = GetDirectory().Find( id );
 	if ( idx == INVALID_INDEX )
 	{
 		// Don't create a read-only file
@@ -588,7 +588,7 @@ FSAsyncStatus_t CSaveRestoreFileSystem::AsyncWrite( const char *pFileName, const
 	FileHandle_t hFile = Open( pFileName, "wb" );
 	if ( hFile )
 	{
-		SaveFile_t &file = GetFile( (unsigned int)hFile );
+		SaveFile_t &file = GetFile( (uintp)hFile );
 
 		if( file.eType == WRITE_ONLY )
 		{
@@ -1289,7 +1289,7 @@ public:
 
 				SaveMsg( "DirectoryCopy: AsyncAppend %s, %s\n", szName, pDestFileName );
 				g_pFileSystem->AsyncAppend( pDestFileName, memcpy( new char[MAX_PATH], list[i].szFileName, MAX_PATH), MAX_PATH, true );		// Filename can only be as long as a map name + extension
-				g_pFileSystem->AsyncAppend( pDestFileName, new int(fileSize), sizeof(int), true );
+				g_pFileSystem->AsyncAppend( pDestFileName, memcpy( new char[sizeof(int)], &fileSize, sizeof(int)), sizeof(int), true );
 				g_pFileSystem->AsyncAppendFile( pDestFileName, szName );
 			}
 		}

engine/shadowmgr.cpp

@@ -166,7 +166,7 @@ public:
 	virtual unsigned short InvalidShadowIndex( );
 
 	// Methods of ISpatialLeafEnumerator
-	virtual bool EnumerateLeaf( int leaf, int context );
+	virtual bool EnumerateLeaf( int leaf, intp context );
 
 	// Sets the texture coordinate range for a shadow...
 	virtual void SetShadowTexCoord( ShadowHandle_t handle, float x, float y, float w, float h );
@@ -605,7 +605,7 @@ void CShadowMgr::SetMaterial( Shadow_t& shadow, IMaterial* pMaterial, IMaterial*
 	}
 
 	// Search the sort order handles for an enumeration id match
-	int materialEnum = (int)pMaterial;
+	int materialEnum = (intp)pMaterial;
 	for (unsigned short i = m_SortOrderIds.Head(); i != m_SortOrderIds.InvalidIndex();
 		i = m_SortOrderIds.Next(i) )
 	{
@@ -1536,7 +1536,7 @@ void CShadowMgr::ProjectShadow( ShadowHandle_t handle, const Vector &origin,
 	for ( int i  = 0; i < nLeafCount; ++i )
 	{
 		// NOTE: Scope specifier eliminates virtual function call
-		CShadowMgr::EnumerateLeaf( pLeafList[i], (int)&build );
+		CShadowMgr::EnumerateLeaf( pLeafList[i], (intp)&build );
 	}
 }
 
@@ -1650,7 +1650,7 @@ void CShadowMgr::ProjectFlashlight( ShadowHandle_t handle, const VMatrix& worldT
 	for ( int i = 0; i < nLeafCount; ++i )
 	{
 		// NOTE: Scope specifier eliminates virtual function call
-		CShadowMgr::EnumerateLeaf( pLeafList[i], (int)&build );
+		CShadowMgr::EnumerateLeaf( pLeafList[i], (intp)&build );
 	}
 }
 
@@ -1809,7 +1809,7 @@ void CShadowMgr::ApplyShadowToLeaf( const Shadow_t &shadow, mleaf_t* RESTRICT pL
 //-----------------------------------------------------------------------------
 // Applies a projected texture to all surfaces in the leaf
 //-----------------------------------------------------------------------------
-bool CShadowMgr::EnumerateLeaf( int leaf, int context )
+bool CShadowMgr::EnumerateLeaf( int leaf, intp context )
 {
 	VPROF( "CShadowMgr::EnumerateLeaf" );
 	ShadowBuildInfo_t* pBuild = (ShadowBuildInfo_t*)context;
@@ -2048,34 +2048,34 @@ public:
 class CClipPlane
 {
 public:
-	static inline bool Inside( ShadowVertex_t const& vert )						
+	static inline bool Inside( ShadowVertex_t const& vert )
 	{
-		return DotProduct( vert.m_Position, *m_pNormal ) < m_Dist;
+		return DotProduct( vert.m_Position, m_pNormal ) < m_Dist;
 	}
 
-	static inline float Clip( const Vector& one, const Vector& two )	
+	static inline float Clip( const Vector& one, const Vector& two )
 	{
 		Vector dir;
 		VectorSubtract( two, one, dir );
-		return IntersectRayWithPlane( one, dir, *m_pNormal, m_Dist );
+		return IntersectRayWithPlane( one, dir, m_pNormal, m_Dist );
 	}
 
 	static inline bool IsAbove()	{return false;}
 	static inline bool IsPlane()	{return true;}
 
-	static void SetPlane( const Vector& normal, float dist )
+	static void SetPlane( const Vector normal, float dist )
 	{
-		m_pNormal = &normal;
+		m_pNormal = normal;
 		m_Dist = dist;
 	}
 
 
 private:
-	static const Vector *m_pNormal;
+	static Vector m_pNormal;
 	static float  m_Dist;
 };
 
-const Vector *CClipPlane::m_pNormal;
+Vector CClipPlane::m_pNormal;
 float  CClipPlane::m_Dist;
 
 static inline void ClampTexCoord( ShadowVertex_t *pInVertex, ShadowVertex_t *pOutVertex )

engine/snd_io.cpp

@@ -20,17 +20,17 @@
 class COM_IOReadBinary : public IFileReadBinary
 {
 public:
-	int open( const char *pFileName );
-	int read( void *pOutput, int size, int file );
-	void seek( int file, int pos );
-	unsigned int tell( int file );
-	unsigned int size( int file );
-	void close( int file );
+	intp open( const char *pFileName );
+	int read( void *pOutput, int size, intp file );
+	void seek( intp file, int pos );
+	unsigned int tell( intp file );
+	unsigned int size( intp file );
+	void close( intp file );
 };
 
 
 // prepend sound/ to the filename -- all sounds are loaded from the sound/ directory
-int COM_IOReadBinary::open( const char *pFileName )
+intp COM_IOReadBinary::open( const char *pFileName )
 {
 	char namebuffer[512];
 	FileHandle_t hFile;
@@ -46,10 +46,10 @@ int COM_IOReadBinary::open( const char *pFileName )
 
 	hFile = g_pFileSystem->Open( namebuffer, "rb", "GAME" );
 
-	return (int)hFile;
+	return (intp)hFile;
 }
 
-int COM_IOReadBinary::read( void *pOutput, int size, int file )
+int COM_IOReadBinary::read( void *pOutput, int size, intp file )
 {
 	if ( !file )
 		return 0;
@@ -57,7 +57,7 @@ int COM_IOReadBinary::read( void *pOutput, int size, int file )
 	return g_pFileSystem->Read( pOutput, size, (FileHandle_t)file );
 }
 
-void COM_IOReadBinary::seek( int file, int pos )
+void COM_IOReadBinary::seek( intp file, int pos )
 {
 	if ( !file )
 		return;
@@ -65,7 +65,7 @@ void COM_IOReadBinary::seek( int file, int pos )
 	g_pFileSystem->Seek( (FileHandle_t)file, pos, FILESYSTEM_SEEK_HEAD );
 }
 
-unsigned int COM_IOReadBinary::tell( int file )
+unsigned int COM_IOReadBinary::tell( intp file )
 {
 	if ( !file )
 		return 0;
@@ -73,7 +73,7 @@ unsigned int COM_IOReadBinary::tell( int file )
 	return g_pFileSystem->Tell( (FileHandle_t)file );
 }
 
-unsigned int COM_IOReadBinary::size( int file )
+unsigned int COM_IOReadBinary::size( intp file )
 {
 	if (!file)
 		return 0;
@@ -81,7 +81,7 @@ unsigned int COM_IOReadBinary::size( int file )
 	return g_pFileSystem->Size( (FileHandle_t)file );
 }
 
-void COM_IOReadBinary::close( int file )
+void COM_IOReadBinary::close( intp file )
 {
 	if (!file)
 		return;

engine/sv_main.cpp

@@ -75,6 +75,7 @@
 
 // memdbgon must be the last include file in a .cpp file!!!
 #include "tier0/memdbgon.h"
+#include "tier0/memalloc.h"
 
 extern CNetworkStringTableContainer *networkStringTableContainerServer;
 extern CNetworkStringTableContainer *networkStringTableContainerClient;

engine/sv_remoteaccess.cpp

@@ -284,53 +284,6 @@ void CServerRemoteAccess::WriteDataRequest( CRConServer *pNetworkListener, ra_li
 #endif
 				}
 				break;
-
-#ifdef VPROF_ENABLED
-			case SERVERDATA_VPROF:
-				{
-					char password[25];
-					if ( !GetStringHelper( cmd, password, sizeof(password) ) )
-					{
-						invalidRequest = true;
-						break;
-					}
-					if ( !GetStringHelper( cmd, password, sizeof(password) ) )
-					{
-						invalidRequest = true;
-						break;
-					}
-					if ( IsAuthenticated(listener) )
-					{
-						RegisterVProfDataListener( listener );
-						LogCommand( listener, "Remote VProf started!\n" );
-						RespondString( listener, requestID, "Remote VProf started!\n" );
-					}
-				}
-				break;
-		
-			case SERVERDATA_REMOVE_VPROF:
-				{
-					char password[25];
-					if ( !GetStringHelper( cmd, password, sizeof(password) ) )
-					{
-						invalidRequest = true;
-						break;
-					}
-					if ( !GetStringHelper( cmd, password, sizeof(password) ) )
-					{
-						invalidRequest = true;
-						break;
-					}
-					if ( IsAuthenticated(listener) )
-					{
-						RemoveVProfDataListener( listener );
-						LogCommand( listener, "Remote VProf finished!\n" );
-						RespondString( listener, requestID, "Remote VProf finished!\n" );
-					}
-				}
-				break;
-#endif
-
 			default:
 				Assert(!("Unknown requestType in CServerRemoteAccess::WriteDataRequest()"));
 				cmd.Purge();
@@ -911,16 +864,6 @@ void CServerRemoteAccess::SendResponseToClient( ra_listener_id listenerID, Serve
 	response.Put( pData, nDataLen );
 }
 
-
-//-----------------------------------------------------------------------------
-// Purpose: sends an opaque blob of data from VProf to a remote rcon listener
-//-----------------------------------------------------------------------------
-void CServerRemoteAccess::SendVProfData( ra_listener_id listenerID, bool bGroupData, void *data, int len )
-{
-	Assert( listenerID != m_AdminUIID ); // only RCON clients support this right now
-	SendResponseToClient( listenerID, bGroupData ? SERVERDATA_VPROF_GROUPS : SERVERDATA_VPROF_DATA, data, len );
-}
-
 //-----------------------------------------------------------------------------
 // Purpose: C function for rest of engine to access CServerRemoteAccess class
 //-----------------------------------------------------------------------------

engine/sv_uploadgamestats.cpp

@@ -1099,7 +1099,7 @@ protected:
 	};
 
 public:
-	static unsigned CallbackThreadProc( void *pvParam ) { ((CAsyncUploaderThread*) pvParam)->ThreadProc(); return 0; }
+	static uintp CallbackThreadProc( void *pvParam ) { ((CAsyncUploaderThread*) pvParam)->ThreadProc(); return 0; }
 	void QueueData( char const *szMapName, uint uiBlobVersion, uint uiBlobSize, const void *pvBlob );
 	void TerminateAndSelfDelete();
 };

engine/sys_dll.cpp

@@ -797,7 +797,7 @@ void Sys_ShutdownAuthentication( void )
 //-----------------------------------------------------------------------------
 // Debug library spew output
 //-----------------------------------------------------------------------------
-CThreadLocalInt<> g_bInSpew;
+CTHREADLOCALINT g_bInSpew;
 
 #include "tier1/fmtstr.h"
 

engine/sys_dll2.cpp

@@ -100,6 +100,7 @@
 
 // memdbgon must be the last include file in a .cpp file!!!
 #include "tier0/memdbgon.h"
+#include "tier0/memalloc.h"
 
 //-----------------------------------------------------------------------------
 // Globals
@@ -2260,7 +2261,7 @@ bool EnableLongTickWatcher()
 #ifdef POSIX
 			(void*)
 #endif
-			LongTickWatcherThread, NULL, 0, (unsigned long int *)&nThreadID );
+			LongTickWatcherThread, NULL, 0, (uintp *)&nThreadID );
 
 		bRet = true;
 	}

engine/sys_getmodes.cpp

@@ -2286,7 +2286,9 @@ bool CVideoMode_MaterialSystem::Init( )
     int bitsperpixel = 32;
 
     bool bAllowSmallModes = false;
+#ifndef ANDROID
     if ( CommandLine()->FindParm( "-small" ) )
+#endif
     {
         bAllowSmallModes = true;
     }

engine/sys_mainwind.cpp

@@ -355,7 +355,7 @@ void CGame::HandleMsg_Close( const InputEvent_t &event )
 
 void CGame::DispatchInputEvent( const InputEvent_t &event )
 {
-	switch( event.m_nType )
+	switch( event.m_nType & 0xFFFF )
 	{
 	// Handle button events specially, 
 	// since we have all manner of crazy filtering going on	when dealing with them
@@ -1566,7 +1566,7 @@ void *CGame::GetMainWindowPlatformSpecificHandle( void )
 #ifdef OSX
 	id nsWindow = (id)pInfo.info.cocoa.window;
 	SEL selector = sel_registerName("windowRef");
-	id windowRef = objc_msgSend( nsWindow, selector );
+	id windowRef = ((id(*)(id, SEL))objc_msgSend)( nsWindow, selector );
 	return windowRef;
 #else
 	// Not used on Linux.

engine/tmessage.cpp

@@ -192,7 +192,7 @@ int ParseString( char const *pText, char *buf, size_t bufsize )
 		char const *pStart = pTemp;
 		pTemp = SkipText( pTemp );
 
-		int len =  min( pTemp - pStart + 1, (int)bufsize - 1 );
+        intp len =  min( pTemp - pStart + 1, (ptrdiff_t)bufsize - 1 );
 		Q_strncpy( buf, pStart, len );
 		buf[ len ] = 0;
 		return 1;
@@ -367,7 +367,8 @@ void TextMessageParse( byte *pMemFile, int fileSize )
 
 	client_textmessage_t	textMessages[ MAX_MESSAGES ];
 	
-	int			i, nameHeapSize, textHeapSize, messageSize, nameOffset;
+	int			i, nameHeapSize, textHeapSize, messageSize;
+	intp nameOffset;
 
 	lastNamePos = 0;
 	lineNumber = 0;
@@ -633,4 +634,4 @@ client_textmessage_t *TextMessageGet( const char *pName )
 	}
 
 	return NULL;
-}
+}

engine/vengineserver_impl.cpp

@@ -48,6 +48,7 @@
 #include "replay_internal.h"
 #include "replayserver.h"
 #include "replay/iserverengine.h"
+#include "vcrmode.h"
 
 // memdbgon must be the last include file in a .cpp file!!!
 #include "tier0/memdbgon.h"

engine/vgui_baseui_interface.cpp

@@ -696,10 +696,9 @@ void CEngineVGui::Init()
 	COM_TimestampedLog( "Building Panels (staticGameUIPanel)" );
 
 	staticGameUIPanel = new CEnginePanel( staticPanel, "GameUI Panel" );
-	if (IsAndroid() || CommandLine()->CheckParm("-gameuiproportionality"))
-	{
+
+	if(NeedProportional())
 		staticGameUIPanel->SetProportional(true);
-	}
 
 	staticGameUIPanel->SetBounds( 0, 0, videomode->GetModeUIWidth(), videomode->GetModeUIHeight() );
 	staticGameUIPanel->SetPaintBorderEnabled(false);

engine/view.cpp

@@ -523,7 +523,7 @@ public:
 		int m_nLeafWaterDataID;
 	};
 	
-	bool EnumerateLeaf( int leaf, int context )
+	bool EnumerateLeaf( int leaf, intp context )
 	{
 		BoxIntersectWaterContext_t *pSearchContext = ( BoxIntersectWaterContext_t * )context;
 		mleaf_t *pLeaf = &host_state.worldmodel->brush.pShared->leafs[leaf];
@@ -541,7 +541,7 @@ public:
 		BoxIntersectWaterContext_t context;
 		context.m_bFoundWaterLeaf = false;
 		context.m_nLeafWaterDataID = leafWaterDataID;
-		g_pToolBSPTree->EnumerateLeavesInBox( mins, maxs, this, ( int )&context );
+		g_pToolBSPTree->EnumerateLeavesInBox( mins, maxs, this, ( intp )&context );
 		return context.m_bFoundWaterLeaf;
 	}
 

engine/voice_codecs/celt/opus/opus.h

@@ -0,0 +1,981 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus.h
+ * @brief Opus reference implementation API
+ */
+
+#ifndef OPUS_H
+#define OPUS_H
+
+#include "opus_types.h"
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * @mainpage Opus
+ *
+ * The Opus codec is designed for interactive speech and audio transmission over the Internet.
+ * It is designed by the IETF Codec Working Group and incorporates technology from
+ * Skype's SILK codec and Xiph.Org's CELT codec.
+ *
+ * The Opus codec is designed to handle a wide range of interactive audio applications,
+ * including Voice over IP, videoconferencing, in-game chat, and even remote live music
+ * performances. It can scale from low bit-rate narrowband speech to very high quality
+ * stereo music. Its main features are:
+
+ * @li Sampling rates from 8 to 48 kHz
+ * @li Bit-rates from 6 kb/s to 510 kb/s
+ * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
+ * @li Audio bandwidth from narrowband to full-band
+ * @li Support for speech and music
+ * @li Support for mono and stereo
+ * @li Support for multichannel (up to 255 channels)
+ * @li Frame sizes from 2.5 ms to 60 ms
+ * @li Good loss robustness and packet loss concealment (PLC)
+ * @li Floating point and fixed-point implementation
+ *
+ * Documentation sections:
+ * @li @ref opus_encoder
+ * @li @ref opus_decoder
+ * @li @ref opus_repacketizer
+ * @li @ref opus_multistream
+ * @li @ref opus_libinfo
+ * @li @ref opus_custom
+ */
+
+/** @defgroup opus_encoder Opus Encoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to encode Opus.
+  *
+  * Since Opus is a stateful codec, the encoding process starts with creating an encoder
+  * state. This can be done with:
+  *
+  * @code
+  * int          error;
+  * OpusEncoder *enc;
+  * enc = opus_encoder_create(Fs, channels, application, &error);
+  * @endcode
+  *
+  * From this point, @c enc can be used for encoding an audio stream. An encoder state
+  * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
+  * state @b must @b not be re-initialized for each frame.
+  *
+  * While opus_encoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  *
+  * @code
+  * int          size;
+  * int          error;
+  * OpusEncoder *enc;
+  * size = opus_encoder_get_size(channels);
+  * enc = malloc(size);
+  * error = opus_encoder_init(enc, Fs, channels, application);
+  * @endcode
+  *
+  * where opus_encoder_get_size() returns the required size for the encoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The encoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
+  * interface. All these settings already default to the recommended value, so they should
+  * only be changed when necessary. The most common settings one may want to change are:
+  *
+  * @code
+  * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
+  * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
+  * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
+  * @endcode
+  *
+  * where
+  *
+  * @arg bitrate is in bits per second (b/s)
+  * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
+  * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
+  *
+  * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
+  *
+  * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
+  * @code
+  * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
+  * @endcode
+  *
+  * where
+  * <ul>
+  * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
+  * <li>frame_size is the duration of the frame in samples (per channel)</li>
+  * <li>packet is the byte array to which the compressed data is written</li>
+  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
+  *     Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
+  * </ul>
+  *
+  * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
+  * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
+  * is 2 bytes or less, then the packet does not need to be transmitted (DTX).
+  *
+  * Once the encoder state if no longer needed, it can be destroyed with
+  *
+  * @code
+  * opus_encoder_destroy(enc);
+  * @endcode
+  *
+  * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
+  * then no action is required aside from potentially freeing the memory that was manually
+  * allocated for it (calling free(enc) for the example above)
+  *
+  */
+
+/** Opus encoder state.
+  * This contains the complete state of an Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_encoder_create,opus_encoder_init
+  */
+typedef struct OpusEncoder OpusEncoder;
+
+/** Gets the size of an <code>OpusEncoder</code> structure.
+  * @param[in] channels <tt>int</tt>: Number of channels.
+  *                                   This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
+
+/**
+ */
+
+/** Allocates and initializes an encoder state.
+ * There are three coding modes:
+ *
+ * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
+ *    signals. It enhances the  input signal by high-pass filtering and
+ *    emphasizing formants and harmonics. Optionally  it includes in-band
+ *    forward error correction to protect against packet loss. Use this
+ *    mode for typical VoIP applications. Because of the enhancement,
+ *    even at high bitrates the output may sound different from the input.
+ *
+ * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
+ *    non-voice signals like music. Use this mode for music and mixed
+ *    (music/voice) content, broadcast, and applications requiring less
+ *    than 15 ms of coding delay.
+ *
+ * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
+ *    disables the speech-optimized mode in exchange for slightly reduced delay.
+ *    This mode can only be set on an newly initialized or freshly reset encoder
+ *    because it changes the codec delay.
+ *
+ * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
+ * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                     This must be one of 8000, 12000, 16000,
+ *                                     24000, or 48000.
+ * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+ * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
+ * @note Regardless of the sampling rate and number channels selected, the Opus encoder
+ * can switch to a lower audio bandwidth or number of channels if the bitrate
+ * selected is too low. This also means that it is safe to always use 48 kHz stereo input
+ * and let the encoder optimize the encoding.
+ */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int application,
+    int *error
+);
+
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_encoder_create(),opus_encoder_get_size()
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                      This must be one of 8000, 12000, 16000,
+ *                                      24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+  * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_encoder_init(
+    OpusEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int application
+) OPUS_ARG_NONNULL(1);
+
+/** Encodes an Opus frame.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
+    OpusEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes an Opus frame from floating point input.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range.
+  *          length is frame_size*channels*sizeof(float)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
+    OpusEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
+  * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
+
+/** Perform a CTL function on an Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusEncoder*</tt>: Encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_encoderctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  */
+OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+/**@}*/
+
+/** @defgroup opus_decoder Opus Decoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to decode Opus.
+  *
+  * The decoding process also starts with creating a decoder
+  * state. This can be done with:
+  * @code
+  * int          error;
+  * OpusDecoder *dec;
+  * dec = opus_decoder_create(Fs, channels, &error);
+  * @endcode
+  * where
+  * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
+  * @li channels is the number of channels (1 or 2)
+  * @li error will hold the error code in case of failure (or #OPUS_OK on success)
+  * @li the return value is a newly created decoder state to be used for decoding
+  *
+  * While opus_decoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  * @code
+  * int          size;
+  * int          error;
+  * OpusDecoder *dec;
+  * size = opus_decoder_get_size(channels);
+  * dec = malloc(size);
+  * error = opus_decoder_init(dec, Fs, channels);
+  * @endcode
+  * where opus_decoder_get_size() returns the required size for the decoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The decoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
+  * @code
+  * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
+  * @endcode
+  * where
+  *
+  * @li packet is the byte array containing the compressed data
+  * @li len is the exact number of bytes contained in the packet
+  * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
+  * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
+  *
+  * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
+  * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
+  * buffer is too small to hold the decoded audio.
+  *
+  * Opus is a stateful codec with overlapping blocks and as a result Opus
+  * packets are not coded independently of each other. Packets must be
+  * passed into the decoder serially and in the correct order for a correct
+  * decode. Lost packets can be replaced with loss concealment by calling
+  * the decoder with a null pointer and zero length for the missing packet.
+  *
+  * A single codec state may only be accessed from a single thread at
+  * a time and any required locking must be performed by the caller. Separate
+  * streams must be decoded with separate decoder states and can be decoded
+  * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
+  * defined.
+  *
+  */
+
+/** Opus decoder state.
+  * This contains the complete state of an Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_decoder_create,opus_decoder_init
+  */
+typedef struct OpusDecoder OpusDecoder;
+
+/** Gets the size of an <code>OpusDecoder</code> structure.
+  * @param [in] channels <tt>int</tt>: Number of channels.
+  *                                    This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
+
+/** Allocates and initializes a decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
+  *
+  * Internally Opus stores data at 48000 Hz, so that should be the default
+  * value for Fs. However, the decoder can efficiently decode to buffers
+  * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
+  * data at the full sample rate, or knows the compressed data doesn't
+  * use the full frequency range, it can request decoding at a reduced
+  * rate. Likewise, the decoder is capable of filling in either mono or
+  * interleaved stereo pcm buffers, at the caller's request.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int *error
+);
+
+/** Initializes a previously allocated decoder state.
+  * The state must be at least the size returned by opus_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_decoder_init(
+    OpusDecoder *st,
+    opus_int32 Fs,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Decode an Opus packet.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available, the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an Opus packet with floating point output.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_decoderctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  */
+OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
+  * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
+
+/** Parse an opus packet into one or more frames.
+  * Opus_decode will perform this operation internally so most applications do
+  * not need to use this function.
+  * This function does not copy the frames, the returned pointers are pointers into
+  * the input packet.
+  * @param [in] data <tt>char*</tt>: Opus packet to be parsed
+  * @param [in] len <tt>opus_int32</tt>: size of data
+  * @param [out] out_toc <tt>char*</tt>: TOC pointer
+  * @param [out] frames <tt>char*[48]</tt> encapsulated frames
+  * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
+  * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
+  * @returns number of frames
+  */
+OPUS_EXPORT int opus_packet_parse(
+   const unsigned char *data,
+   opus_int32 len,
+   unsigned char *out_toc,
+   const unsigned char *frames[48],
+   opus_int16 size[48],
+   int *payload_offset
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5);
+
+/** Gets the bandwidth of an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
+  * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
+  * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
+  * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
+  * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples per frame from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet.
+  *                                  This must contain at least one byte of
+  *                                  data.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples per frame.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of channels from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @returns Number of channels
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of frames in an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of frames
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
+  * the signal is already in that range, nothing is done. If there are values
+  * outside of [-1,1], then the signal is clipped as smoothly as possible to
+  * both fit in the range and avoid creating excessive distortion in the
+  * process.
+  * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
+  * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
+  */
+OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
+
+
+/**@}*/
+
+/** @defgroup opus_repacketizer Repacketizer
+  * @{
+  *
+  * The repacketizer can be used to merge multiple Opus packets into a single
+  * packet or alternatively to split Opus packets that have previously been
+  * merged. Splitting valid Opus packets is always guaranteed to succeed,
+  * whereas merging valid packets only succeeds if all frames have the same
+  * mode, bandwidth, and frame size, and when the total duration of the merged
+  * packet is no more than 120 ms. The 120 ms limit comes from the
+  * specification and limits decoder memory requirements at a point where
+  * framing overhead becomes negligible.
+  *
+  * The repacketizer currently only operates on elementary Opus
+  * streams. It will not manipualte multistream packets successfully, except in
+  * the degenerate case where they consist of data from a single stream.
+  *
+  * The repacketizing process starts with creating a repacketizer state, either
+  * by calling opus_repacketizer_create() or by allocating the memory yourself,
+  * e.g.,
+  * @code
+  * OpusRepacketizer *rp;
+  * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
+  * if (rp != NULL)
+  *     opus_repacketizer_init(rp);
+  * @endcode
+  *
+  * Then the application should submit packets with opus_repacketizer_cat(),
+  * extract new packets with opus_repacketizer_out() or
+  * opus_repacketizer_out_range(), and then reset the state for the next set of
+  * input packets via opus_repacketizer_init().
+  *
+  * For example, to split a sequence of packets into individual frames:
+  * @code
+  * unsigned char *data;
+  * int len;
+  * while (get_next_packet(&data, &len))
+  * {
+  *   unsigned char out[1276];
+  *   opus_int32 out_len;
+  *   int nb_frames;
+  *   int err;
+  *   int i;
+  *   err = opus_repacketizer_cat(rp, data, len);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packet(data);
+  *     return err;
+  *   }
+  *   nb_frames = opus_repacketizer_get_nb_frames(rp);
+  *   for (i = 0; i < nb_frames; i++)
+  *   {
+  *     out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packet(data);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *   }
+  *   opus_repacketizer_init(rp);
+  *   release_packet(data);
+  * }
+  * @endcode
+  *
+  * Alternatively, to combine a sequence of frames into packets that each
+  * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
+  * @code
+  * // The maximum number of packets with duration TARGET_DURATION_MS occurs
+  * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
+  * // packets.
+  * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
+  * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
+  * int nb_packets;
+  * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
+  * opus_int32 out_len;
+  * int prev_toc;
+  * nb_packets = 0;
+  * while (get_next_packet(data+nb_packets, len+nb_packets))
+  * {
+  *   int nb_frames;
+  *   int err;
+  *   nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
+  *   if (nb_frames < 1)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return nb_frames;
+  *   }
+  *   nb_frames += opus_repacketizer_get_nb_frames(rp);
+  *   // If adding the next packet would exceed our target, or it has an
+  *   // incompatible TOC sequence, output the packets we already have before
+  *   // submitting it.
+  *   // N.B., The nb_packets > 0 check ensures we've submitted at least one
+  *   // packet since the last call to opus_repacketizer_init(). Otherwise a
+  *   // single packet longer than TARGET_DURATION_MS would cause us to try to
+  *   // output an (invalid) empty packet. It also ensures that prev_toc has
+  *   // been set to a valid value. Additionally, len[nb_packets] > 0 is
+  *   // guaranteed by the call to opus_packet_get_nb_frames() above, so the
+  *   // reference to data[nb_packets][0] should be valid.
+  *   if (nb_packets > 0 && (
+  *       ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
+  *       opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
+  *       TARGET_DURATION_MS*48))
+  *   {
+  *     out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packets(data, nb_packets+1);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *     opus_repacketizer_init(rp);
+  *     release_packets(data, nb_packets);
+  *     data[0] = data[nb_packets];
+  *     len[0] = len[nb_packets];
+  *     nb_packets = 0;
+  *   }
+  *   err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return err;
+  *   }
+  *   prev_toc = data[nb_packets][0];
+  *   nb_packets++;
+  * }
+  * // Output the final, partial packet.
+  * if (nb_packets > 0)
+  * {
+  *   out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *   release_packets(data, nb_packets);
+  *   if (out_len < 0)
+  *     return (int)out_len;
+  *   output_next_packet(out, out_len);
+  * }
+  * @endcode
+  *
+  * An alternate way of merging packets is to simply call opus_repacketizer_cat()
+  * unconditionally until it fails. At that point, the merged packet can be
+  * obtained with opus_repacketizer_out() and the input packet for which
+  * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
+  * repacketizer state.
+  */
+
+typedef struct OpusRepacketizer OpusRepacketizer;
+
+/** Gets the size of an <code>OpusRepacketizer</code> structure.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
+
+/** (Re)initializes a previously allocated repacketizer state.
+  * The state must be at least the size returned by opus_repacketizer_get_size().
+  * This can be used for applications which use their own allocator instead of
+  * malloc().
+  * It must also be called to reset the queue of packets waiting to be
+  * repacketized, which is necessary if the maximum packet duration of 120 ms
+  * is reached or if you wish to submit packets with a different Opus
+  * configuration (coding mode, audio bandwidth, frame size, or channel count).
+  * Failure to do so will prevent a new packet from being added with
+  * opus_repacketizer_cat().
+  * @see opus_repacketizer_create
+  * @see opus_repacketizer_get_size
+  * @see opus_repacketizer_cat
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
+  *                                       (re)initialize.
+  * @returns A pointer to the same repacketizer state that was passed in.
+  */
+OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Allocates memory and initializes the new repacketizer with
+ * opus_repacketizer_init().
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
+
+/** Frees an <code>OpusRepacketizer</code> allocated by
+  * opus_repacketizer_create().
+  * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
+
+/** Add a packet to the current repacketizer state.
+  * This packet must match the configuration of any packets already submitted
+  * for repacketization since the last call to opus_repacketizer_init().
+  * This means that it must have the same coding mode, audio bandwidth, frame
+  * size, and channel count.
+  * This can be checked in advance by examining the top 6 bits of the first
+  * byte of the packet, and ensuring they match the top 6 bits of the first
+  * byte of any previously submitted packet.
+  * The total duration of audio in the repacketizer state also must not exceed
+  * 120 ms, the maximum duration of a single packet, after adding this packet.
+  *
+  * The contents of the current repacketizer state can be extracted into new
+  * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
+  *
+  * In order to add a packet with a different configuration or to add more
+  * audio beyond 120 ms, you must clear the repacketizer state by calling
+  * opus_repacketizer_init().
+  * If a packet is too large to add to the current repacketizer state, no part
+  * of it is added, even if it contains multiple frames, some of which might
+  * fit.
+  * If you wish to be able to add parts of such packets, you should first use
+  * another repacketizer to split the packet into pieces and add them
+  * individually.
+  * @see opus_repacketizer_out_range
+  * @see opus_repacketizer_out
+  * @see opus_repacketizer_init
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
+  *                                       add the packet.
+  * @param[in] data <tt>const unsigned char*</tt>: The packet data.
+  *                                                The application must ensure
+  *                                                this pointer remains valid
+  *                                                until the next call to
+  *                                                opus_repacketizer_init() or
+  *                                                opus_repacketizer_destroy().
+  * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
+  * @returns An error code indicating whether or not the operation succeeded.
+  * @retval #OPUS_OK The packet's contents have been added to the repacketizer
+  *                  state.
+  * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
+  *                              the packet's TOC sequence was not compatible
+  *                              with previously submitted packets (because
+  *                              the coding mode, audio bandwidth, frame size,
+  *                              or channel count did not match), or adding
+  *                              this packet would increase the total amount of
+  *                              audio stored in the repacketizer state to more
+  *                              than 120 ms.
+  */
+OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param begin <tt>int</tt>: The index of the first frame in the current
+  *                            repacketizer state to include in the output.
+  * @param end <tt>int</tt>: One past the index of the last frame in the
+  *                          current repacketizer state to include in the
+  *                          output.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1276</code> for a single frame,
+  *                                    or for multiple frames,
+  *                                    <code>1277*(end-begin)</code>.
+  *                                    However, <code>1*(end-begin)</code> plus
+  *                                    the size of all packet data submitted to
+  *                                    the repacketizer since the last call to
+  *                                    opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
+  *                       frames (begin < 0, begin >= end, or end >
+  *                       opus_repacketizer_get_nb_frames()).
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Return the total number of frames contained in packet data submitted to
+  * the repacketizer state so far via opus_repacketizer_cat() since the last
+  * call to opus_repacketizer_init() or opus_repacketizer_create().
+  * This defines the valid range of packets that can be extracted with
+  * opus_repacketizer_out_range() or opus_repacketizer_out().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
+  *                                       frames.
+  * @returns The total number of frames contained in the packet data submitted
+  *          to the repacketizer state.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * This is a convenience routine that returns all the data submitted so far
+  * in a single packet.
+  * It is equivalent to calling
+  * @code
+  * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
+  *                             data, maxlen)
+  * @endcode
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
+  *                                    However,
+  *                                    <code>1*opus_repacketizer_get_nb_frames(rp)</code>
+  *                                    plus the size of all packet data
+  *                                    submitted to the repacketizer since the
+  *                                    last call to opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
+
+/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
+
+/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
+
+/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
+
+/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */

engine/voice_codecs/celt/opus/opus_custom.h

@@ -0,0 +1,342 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008-2012 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+  @file opus_custom.h
+  @brief Opus-Custom reference implementation API
+ */
+
+#ifndef OPUS_CUSTOM_H
+#define OPUS_CUSTOM_H
+
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef CUSTOM_MODES
+# define OPUS_CUSTOM_EXPORT OPUS_EXPORT
+# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT
+#else
+# define OPUS_CUSTOM_EXPORT
+# ifdef OPUS_BUILD
+#  define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE
+# else
+#  define OPUS_CUSTOM_EXPORT_STATIC
+# endif
+#endif
+
+/** @defgroup opus_custom Opus Custom
+  * @{
+  *  Opus Custom is an optional part of the Opus specification and
+  * reference implementation which uses a distinct API from the regular
+  * API and supports frame sizes that are not normally supported.\ Use
+  * of Opus Custom is discouraged for all but very special applications
+  * for which a frame size different from 2.5, 5, 10, or 20 ms is needed
+  * (for either complexity or latency reasons) and where interoperability
+  * is less important.
+  *
+  * In addition to the interoperability limitations the use of Opus custom
+  * disables a substantial chunk of the codec and generally lowers the
+  * quality available at a given bitrate. Normally when an application needs
+  * a different frame size from the codec it should buffer to match the
+  * sizes but this adds a small amount of delay which may be important
+  * in some very low latency applications. Some transports (especially
+  * constant rate RF transports) may also work best with frames of
+  * particular durations.
+  *
+  * Libopus only supports custom modes if they are enabled at compile time.
+  *
+  * The Opus Custom API is similar to the regular API but the
+  * @ref opus_encoder_create and @ref opus_decoder_create calls take
+  * an additional mode parameter which is a structure produced by
+  * a call to @ref opus_custom_mode_create. Both the encoder and decoder
+  * must create a mode using the same sample rate (fs) and frame size
+  * (frame size) so these parameters must either be signaled out of band
+  * or fixed in a particular implementation.
+  *
+  * Similar to regular Opus the custom modes support on the fly frame size
+  * switching, but the sizes available depend on the particular frame size in
+  * use. For some initial frame sizes on a single on the fly size is available.
+  */
+
+/** Contains the state of an encoder. One encoder state is needed
+    for each stream. It is initialized once at the beginning of the
+    stream. Do *not* re-initialize the state for every frame.
+   @brief Encoder state
+ */
+typedef struct OpusCustomEncoder OpusCustomEncoder;
+
+/** State of the decoder. One decoder state is needed for each stream.
+    It is initialized once at the beginning of the stream. Do *not*
+    re-initialize the state for every frame.
+   @brief Decoder state
+ */
+typedef struct OpusCustomDecoder OpusCustomDecoder;
+
+/** The mode contains all the information necessary to create an
+    encoder. Both the encoder and decoder need to be initialized
+    with exactly the same mode, otherwise the output will be
+    corrupted.
+   @brief Mode configuration
+ */
+typedef struct OpusCustomMode OpusCustomMode;
+
+/** Creates a new mode struct. This will be passed to an encoder or
+  * decoder. The mode MUST NOT BE DESTROYED until the encoders and
+  * decoders that use it are destroyed as well.
+  * @param [in] Fs <tt>int</tt>: Sampling rate (8000 to 96000 Hz)
+  * @param [in] frame_size <tt>int</tt>: Number of samples (per channel) to encode in each
+  *        packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes)
+  * @param [out] error <tt>int*</tt>: Returned error code (if NULL, no error will be returned)
+  * @return A newly created mode
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error);
+
+/** Destroys a mode struct. Only call this after all encoders and
+  * decoders using this mode are destroyed as well.
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Mode to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C)
+
+/* Encoder */
+/** Gets the size of an OpusCustomEncoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+# ifdef CUSTOM_MODES
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_encoder_create(),opus_custom_encoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_init(
+    OpusCustomEncoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+# endif
+#endif
+
+
+/** Creates a new encoder state. Each stream needs its own encoder
+  * state (can't be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created encoder state.
+*/
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+
+/** Destroys a an encoder state.
+  * @param[in] st <tt>OpusCustomEncoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: PCM audio in float format, with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range. There must be exactly
+  *          frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float(
+    OpusCustomEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian).
+  *          There must be exactly frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+ */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode(
+    OpusCustomEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_encoderctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C)
+/* Decoder */
+
+/** Gets the size of an OpusCustomDecoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Initializes a previously allocated decoder state
+  * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_decoder_create(),opus_custom_decoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init(
+    OpusCustomDecoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+#endif
+
+
+/** Creates a new decoder state. Each stream needs its own decoder state (can't
+  * be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode</tt>: Contains all the information about the characteristics of the
+  *          stream (must be the same characteristics as used for the encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created decoder state.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+/** Destroys a an decoder state.
+  * @param[in] st <tt>OpusCustomDecoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st);
+
+/** Decode an opus custom frame with floating point output
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    float *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an opus custom frame
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    opus_int16 *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_genericctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_CUSTOM_H */

engine/voice_codecs/celt/opus/opus_defines.h

@@ -0,0 +1,799 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_defines.h
+ * @brief Opus reference implementation constants
+ */
+
+#ifndef OPUS_DEFINES_H
+#define OPUS_DEFINES_H
+
+#include "opus_types.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @defgroup opus_errorcodes Error codes
+ * @{
+ */
+/** No error @hideinitializer*/
+#define OPUS_OK                0
+/** One or more invalid/out of range arguments @hideinitializer*/
+#define OPUS_BAD_ARG          -1
+/** Not enough bytes allocated in the buffer @hideinitializer*/
+#define OPUS_BUFFER_TOO_SMALL -2
+/** An internal error was detected @hideinitializer*/
+#define OPUS_INTERNAL_ERROR   -3
+/** The compressed data passed is corrupted @hideinitializer*/
+#define OPUS_INVALID_PACKET   -4
+/** Invalid/unsupported request number @hideinitializer*/
+#define OPUS_UNIMPLEMENTED    -5
+/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
+#define OPUS_INVALID_STATE    -6
+/** Memory allocation has failed @hideinitializer*/
+#define OPUS_ALLOC_FAIL       -7
+/**@}*/
+
+/** @cond OPUS_INTERNAL_DOC */
+/**Export control for opus functions */
+
+#ifndef OPUS_EXPORT
+# if defined(WIN32)
+#  if defined(OPUS_BUILD) && defined(DLL_EXPORT)
+#   define OPUS_EXPORT __declspec(dllexport)
+#  else
+#   define OPUS_EXPORT
+#  endif
+# elif defined(__GNUC__) && defined(OPUS_BUILD)
+#  define OPUS_EXPORT __attribute__ ((visibility ("default")))
+# else
+#  define OPUS_EXPORT
+# endif
+#endif
+
+# if !defined(OPUS_GNUC_PREREQ)
+#  if defined(__GNUC__)&&defined(__GNUC_MINOR__)
+#   define OPUS_GNUC_PREREQ(_maj,_min) \
+ ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
+#  else
+#   define OPUS_GNUC_PREREQ(_maj,_min) 0
+#  endif
+# endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(3,0)
+#  define OPUS_RESTRICT __restrict__
+# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
+#  define OPUS_RESTRICT __restrict
+# else
+#  define OPUS_RESTRICT
+# endif
+#else
+# define OPUS_RESTRICT restrict
+#endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(2,7)
+#  define OPUS_INLINE __inline__
+# elif (defined(_MSC_VER))
+#  define OPUS_INLINE __inline
+# else
+#  define OPUS_INLINE
+# endif
+#else
+# define OPUS_INLINE inline
+#endif
+
+/**Warning attributes for opus functions
+  * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
+  * some paranoid null checks. */
+#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
+#else
+# define OPUS_WARN_UNUSED_RESULT
+#endif
+#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_ARG_NONNULL(_x)  __attribute__ ((__nonnull__(_x)))
+#else
+# define OPUS_ARG_NONNULL(_x)
+#endif
+
+/** These are the actual Encoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+#define OPUS_SET_APPLICATION_REQUEST         4000
+#define OPUS_GET_APPLICATION_REQUEST         4001
+#define OPUS_SET_BITRATE_REQUEST             4002
+#define OPUS_GET_BITRATE_REQUEST             4003
+#define OPUS_SET_MAX_BANDWIDTH_REQUEST       4004
+#define OPUS_GET_MAX_BANDWIDTH_REQUEST       4005
+#define OPUS_SET_VBR_REQUEST                 4006
+#define OPUS_GET_VBR_REQUEST                 4007
+#define OPUS_SET_BANDWIDTH_REQUEST           4008
+#define OPUS_GET_BANDWIDTH_REQUEST           4009
+#define OPUS_SET_COMPLEXITY_REQUEST          4010
+#define OPUS_GET_COMPLEXITY_REQUEST          4011
+#define OPUS_SET_INBAND_FEC_REQUEST          4012
+#define OPUS_GET_INBAND_FEC_REQUEST          4013
+#define OPUS_SET_PACKET_LOSS_PERC_REQUEST    4014
+#define OPUS_GET_PACKET_LOSS_PERC_REQUEST    4015
+#define OPUS_SET_DTX_REQUEST                 4016
+#define OPUS_GET_DTX_REQUEST                 4017
+#define OPUS_SET_VBR_CONSTRAINT_REQUEST      4020
+#define OPUS_GET_VBR_CONSTRAINT_REQUEST      4021
+#define OPUS_SET_FORCE_CHANNELS_REQUEST      4022
+#define OPUS_GET_FORCE_CHANNELS_REQUEST      4023
+#define OPUS_SET_SIGNAL_REQUEST              4024
+#define OPUS_GET_SIGNAL_REQUEST              4025
+#define OPUS_GET_LOOKAHEAD_REQUEST           4027
+/* #define OPUS_RESET_STATE 4028 */
+#define OPUS_GET_SAMPLE_RATE_REQUEST         4029
+#define OPUS_GET_FINAL_RANGE_REQUEST         4031
+#define OPUS_GET_PITCH_REQUEST               4033
+#define OPUS_SET_GAIN_REQUEST                4034
+#define OPUS_GET_GAIN_REQUEST                4045 /* Should have been 4035 */
+#define OPUS_SET_LSB_DEPTH_REQUEST           4036
+#define OPUS_GET_LSB_DEPTH_REQUEST           4037
+#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
+#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
+#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
+#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
+#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
+/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
+#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046
+#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047
+#define OPUS_GET_IN_DTX_REQUEST              4049
+
+/** Defines for the presence of extended APIs. */
+#define OPUS_HAVE_OPUS_PROJECTION_H
+
+/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
+#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
+#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
+#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
+#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
+/** @endcond */
+
+/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
+  * @see opus_genericctls, opus_encoderctls
+  * @{
+  */
+/* Values for the various encoder CTLs */
+#define OPUS_AUTO                           -1000 /**<Auto/default setting @hideinitializer*/
+#define OPUS_BITRATE_MAX                       -1 /**<Maximum bitrate @hideinitializer*/
+
+/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
+ * @hideinitializer */
+#define OPUS_APPLICATION_VOIP                2048
+/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
+ * @hideinitializer */
+#define OPUS_APPLICATION_AUDIO               2049
+/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
+ * @hideinitializer */
+#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
+
+#define OPUS_SIGNAL_VOICE                    3001 /**< Signal being encoded is voice */
+#define OPUS_SIGNAL_MUSIC                    3002 /**< Signal being encoded is music */
+#define OPUS_BANDWIDTH_NARROWBAND            1101 /**< 4 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_MEDIUMBAND            1102 /**< 6 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_WIDEBAND              1103 /**< 8 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_SUPERWIDEBAND         1104 /**<12 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_FULLBAND              1105 /**<20 kHz bandpass @hideinitializer*/
+
+#define OPUS_FRAMESIZE_ARG                   5000 /**< Select frame size from the argument (default) */
+#define OPUS_FRAMESIZE_2_5_MS                5001 /**< Use 2.5 ms frames */
+#define OPUS_FRAMESIZE_5_MS                  5002 /**< Use 5 ms frames */
+#define OPUS_FRAMESIZE_10_MS                 5003 /**< Use 10 ms frames */
+#define OPUS_FRAMESIZE_20_MS                 5004 /**< Use 20 ms frames */
+#define OPUS_FRAMESIZE_40_MS                 5005 /**< Use 40 ms frames */
+#define OPUS_FRAMESIZE_60_MS                 5006 /**< Use 60 ms frames */
+#define OPUS_FRAMESIZE_80_MS                 5007 /**< Use 80 ms frames */
+#define OPUS_FRAMESIZE_100_MS                5008 /**< Use 100 ms frames */
+#define OPUS_FRAMESIZE_120_MS                5009 /**< Use 120 ms frames */
+
+/**@}*/
+
+
+/** @defgroup opus_encoderctls Encoder related CTLs
+  *
+  * These are convenience macros for use with the \c opus_encode_ctl
+  * interface. They are used to generate the appropriate series of
+  * arguments for that call, passing the correct type, size and so
+  * on as expected for each particular request.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
+  * if (ret != OPUS_OK) return ret;
+  *
+  * opus_int32 rate;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * @endcode
+  *
+  * @see opus_genericctls, opus_encoder
+  * @{
+  */
+
+/** Configures the encoder's computational complexity.
+  * The supported range is 0-10 inclusive with 10 representing the highest complexity.
+  * @see OPUS_GET_COMPLEXITY
+  * @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
+  *
+  * @hideinitializer */
+#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
+/** Gets the encoder's complexity configuration.
+  * @see OPUS_SET_COMPLEXITY
+  * @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
+  *                                      inclusive.
+  * @hideinitializer */
+#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the bitrate in the encoder.
+  * Rates from 500 to 512000 bits per second are meaningful, as well as the
+  * special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
+  * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
+  * rate as it can, which is useful for controlling the rate by adjusting the
+  * output buffer size.
+  * @see OPUS_GET_BITRATE
+  * @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
+  *                                   is determined based on the number of
+  *                                   channels and the input sampling rate.
+  * @hideinitializer */
+#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
+/** Gets the encoder's bitrate configuration.
+  * @see OPUS_SET_BITRATE
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
+  *                                      The default is determined based on the
+  *                                      number of channels and the input
+  *                                      sampling rate.
+  * @hideinitializer */
+#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables variable bitrate (VBR) in the encoder.
+  * The configured bitrate may not be met exactly because frames must
+  * be an integer number of bytes in length.
+  * @see OPUS_GET_VBR
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
+  *               cause noticeable quality degradation.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
+  *               #OPUS_SET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
+/** Determine if variable bitrate (VBR) is enabled in the encoder.
+  * @see OPUS_SET_VBR
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
+  *               #OPUS_GET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables constrained VBR in the encoder.
+  * This setting is ignored when the encoder is in CBR mode.
+  * @warning Only the MDCT mode of Opus currently heeds the constraint.
+  *  Speech mode ignores it completely, hybrid mode may fail to obey it
+  *  if the LPC layer uses more bitrate than the constraint would have
+  *  permitted.
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @see OPUS_SET_VBR
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
+  *               frame of buffering delay assuming a transport with a
+  *               serialization speed of the nominal bitrate.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
+/** Determine if constrained VBR is enabled in the encoder.
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @see OPUS_GET_VBR
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default).</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures mono/stereo forcing in the encoder.
+  * This can force the encoder to produce packets encoded as either mono or
+  * stereo, regardless of the format of the input audio. This is useful when
+  * the caller knows that the input signal is currently a mono source embedded
+  * in a stereo stream.
+  * @see OPUS_GET_FORCE_CHANNELS
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
+/** Gets the encoder's forced channel configuration.
+  * @see OPUS_SET_FORCE_CHANNELS
+  * @param[out] x <tt>opus_int32 *</tt>:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the maximum bandpass that the encoder will select automatically.
+  * Applications should normally use this instead of #OPUS_SET_BANDWIDTH
+  * (leaving that set to the default, #OPUS_AUTO). This allows the
+  * application to set an upper bound based on the type of input it is
+  * providing, but still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_MAX_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Gets the encoder's configured maximum allowed bandpass.
+  * @see OPUS_SET_MAX_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Sets the encoder's bandpass to a specific value.
+  * This prevents the encoder from automatically selecting the bandpass based
+  * on the available bitrate. If an application knows the bandpass of the input
+  * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
+  * instead, which still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Configures the type of signal being encoded.
+  * This is a hint which helps the encoder's mode selection.
+  * @see OPUS_GET_SIGNAL
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal type.
+  * @see OPUS_SET_SIGNAL
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Configures the encoder's intended application.
+  * The initial value is a mandatory argument to the encoder_create function.
+  * @see OPUS_GET_APPLICATION
+  * @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured application.
+  * @see OPUS_SET_APPLICATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the total samples of delay added by the entire codec.
+  * This can be queried by the encoder and then the provided number of samples can be
+  * skipped on from the start of the decoder's output to provide time aligned input
+  * and output. From the perspective of a decoding application the real data begins this many
+  * samples late.
+  *
+  * The decoder contribution to this delay is identical for all decoders, but the
+  * encoder portion of the delay may vary from implementation to implementation,
+  * version to version, or even depend on the encoder's initial configuration.
+  * Applications needing delay compensation should call this CTL rather than
+  * hard-coding a value.
+  * @param[out] x <tt>opus_int32 *</tt>:   Number of lookahead samples
+  * @hideinitializer */
+#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of inband forward error correction (FEC).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_INBAND_FEC
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable inband FEC (default).</dd>
+  * <dt>1</dt><dd>Enable inband FEC.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of inband forward error correction.
+  * @see OPUS_SET_INBAND_FEC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Inband FEC disabled (default).</dd>
+  * <dt>1</dt><dd>Inband FEC enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's expected packet loss percentage.
+  * Higher values trigger progressively more loss resistant behavior in the encoder
+  * at the expense of quality at a given bitrate in the absence of packet loss, but
+  * greater quality under loss.
+  * @see OPUS_GET_PACKET_LOSS_PERC
+  * @param[in] x <tt>opus_int32</tt>:   Loss percentage in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured packet loss percentage.
+  * @see OPUS_SET_PACKET_LOSS_PERC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
+  *                                      in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of discontinuous transmission (DTX).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_DTX
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable DTX (default).</dd>
+  * <dt>1</dt><dd>Enabled DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of discontinuous transmission.
+  * @see OPUS_SET_DTX
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>DTX disabled (default).</dd>
+  * <dt>1</dt><dd>DTX enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
+/** Configures the depth of signal being encoded.
+  *
+  * This is a hint which helps the encoder identify silence and near-silence.
+  * It represents the number of significant bits of linear intensity below
+  * which the signal contains ignorable quantization or other noise.
+  *
+  * For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
+  * for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
+  * for 16-bit linear pcm input with opus_encode_float().
+  *
+  * When using opus_encode() instead of opus_encode_float(), or when libopus
+  * is compiled for fixed-point, the encoder uses the minimum of the value
+  * set here and the value 16.
+  *
+  * @see OPUS_GET_LSB_DEPTH
+  * @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
+  *                                   (default: 24).
+  * @hideinitializer */
+#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal depth.
+  * @see OPUS_SET_LSB_DEPTH
+  * @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
+  *                                      24 (default: 24).
+  * @hideinitializer */
+#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of variable duration frames.
+  * When variable duration is enabled, the encoder is free to use a shorter frame
+  * size than the one requested in the opus_encode*() call.
+  * It is then the user's responsibility
+  * to verify how much audio was encoded by checking the ToC byte of the encoded
+  * packet. The part of the audio that was not encoded needs to be resent to the
+  * encoder for the next call. Do not use this option unless you <b>really</b>
+  * know what you are doing.
+  * @see OPUS_GET_EXPERT_FRAME_DURATION
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured use of variable duration frames.
+  * @see OPUS_SET_EXPERT_FRAME_DURATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables almost all use of prediction, making frames almost
+  * completely independent. This reduces quality.
+  * @see OPUS_GET_PREDICTION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable prediction (default).</dd>
+  * <dt>1</dt><dd>Disable prediction.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured prediction status.
+  * @see OPUS_SET_PREDICTION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Prediction enabled (default).</dd>
+  * <dt>1</dt><dd>Prediction disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_genericctls Generic CTLs
+  *
+  * These macros are used with the \c opus_decoder_ctl and
+  * \c opus_encoder_ctl calls to generate a particular
+  * request.
+  *
+  * When called on an \c OpusDecoder they apply to that
+  * particular decoder instance. When called on an
+  * \c OpusEncoder they apply to the corresponding setting
+  * on that encoder instance, if present.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * opus_int32 pitch;
+  * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
+  * if (ret == OPUS_OK) return ret;
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
+  *
+  * opus_int32 enc_bw, dec_bw;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
+  * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
+  * if (enc_bw != dec_bw) {
+  *   printf("packet bandwidth mismatch!\n");
+  * }
+  * @endcode
+  *
+  * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
+  * @{
+  */
+
+/** Resets the codec state to be equivalent to a freshly initialized state.
+  * This should be called when switching streams in order to prevent
+  * the back to back decoding from giving different results from
+  * one at a time decoding.
+  * @hideinitializer */
+#define OPUS_RESET_STATE 4028
+
+/** Gets the final state of the codec's entropy coder.
+  * This is used for testing purposes,
+  * The encoder and decoder state should be identical after coding a payload
+  * (assuming no data corruption or software bugs)
+  *
+  * @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
+  *
+  * @hideinitializer */
+#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
+
+/** Gets the encoder's configured bandpass or the decoder's last bandpass.
+  * @see OPUS_SET_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the sampling rate the encoder or decoder was initialized with.
+  * This simply returns the <code>Fs</code> value passed to opus_encoder_init()
+  * or opus_decoder_init().
+  * @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
+  * @hideinitializer
+  */
+#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables the use of phase inversion for intensity stereo,
+  * improving the quality of mono downmixes, but slightly reducing normal
+  * stereo quality. Disabling phase inversion in the decoder does not comply
+  * with RFC 6716, although it does not cause any interoperability issue and
+  * is expected to become part of the Opus standard once RFC 6716 is updated
+  * by draft-ietf-codec-opus-update.
+  * @see OPUS_GET_PHASE_INVERSION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable phase inversion (default).</dd>
+  * <dt>1</dt><dd>Disable phase inversion.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PHASE_INVERSION_DISABLED(x) OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured phase inversion status.
+  * @see OPUS_SET_PHASE_INVERSION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Stereo phase inversion enabled (default).</dd>
+  * <dt>1</dt><dd>Stereo phase inversion disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+/** Gets the DTX state of the encoder.
+  * Returns whether the last encoded frame was either a comfort noise update
+  * during DTX or not encoded because of DTX.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>The encoder is not in DTX.</dd>
+  * <dt>1</dt><dd>The encoder is in DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_IN_DTX(x) OPUS_GET_IN_DTX_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_decoderctls Decoder related CTLs
+  * @see opus_genericctls, opus_encoderctls, opus_decoder
+  * @{
+  */
+
+/** Configures decoder gain adjustment.
+  * Scales the decoded output by a factor specified in Q8 dB units.
+  * This has a maximum range of -32768 to 32767 inclusive, and returns
+  * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
+  * This setting survives decoder reset.
+  *
+  * gain = pow(10, x/(20.0*256))
+  *
+  * @param[in] x <tt>opus_int32</tt>:   Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
+/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
+  * @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
+  * @hideinitializer */
+#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the pitch of the last decoded frame, if available.
+  * This can be used for any post-processing algorithm requiring the use of pitch,
+  * e.g. time stretching/shortening. If the last frame was not voiced, or if the
+  * pitch was not coded in the frame, then zero is returned.
+  *
+  * This CTL is only implemented for decoder instances.
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
+  *
+  * @hideinitializer */
+#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_libinfo Opus library information functions
+  * @{
+  */
+
+/** Converts an opus error code into a human readable string.
+  *
+  * @param[in] error <tt>int</tt>: Error number
+  * @returns Error string
+  */
+OPUS_EXPORT const char *opus_strerror(int error);
+
+/** Gets the libopus version string.
+  *
+  * Applications may look for the substring "-fixed" in the version string to
+  * determine whether they have a fixed-point or floating-point build at
+  * runtime.
+  *
+  * @returns Version string
+  */
+OPUS_EXPORT const char *opus_get_version_string(void);
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_DEFINES_H */

engine/voice_codecs/celt/opus/opus_multistream.h

@@ -0,0 +1,660 @@
+/* Copyright (c) 2011 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_multistream.h
+ * @brief Opus reference implementation multistream API
+ */
+
+#ifndef OPUS_MULTISTREAM_H
+#define OPUS_MULTISTREAM_H
+
+#include "opus.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** Macros to trigger compilation errors when the wrong types are provided to a
+  * CTL. */
+/**@{*/
+#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
+#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
+/**@}*/
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
+#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
+/**@}*/
+
+/** @endcond */
+
+/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
+  * @ref opus_decoderctls may be applied to a multistream encoder or decoder as
+  * well.
+  * In addition, you may retrieve the encoder or decoder state for an specific
+  * stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
+  * #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
+  */
+/**@{*/
+
+/** Gets the encoder state for an individual stream of a multistream encoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the encoder.
+  * @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
+  *                                       encoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
+
+/** Gets the decoder state for an individual stream of a multistream decoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the decoder.
+  * @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
+  *                                       decoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
+
+/**@}*/
+
+/** @defgroup opus_multistream Opus Multistream API
+  * @{
+  *
+  * The multistream API allows individual Opus streams to be combined into a
+  * single packet, enabling support for up to 255 channels. Unlike an
+  * elementary Opus stream, the encoder and decoder must negotiate the channel
+  * configuration before the decoder can successfully interpret the data in the
+  * packets produced by the encoder. Some basic information, such as packet
+  * duration, can be computed without any special negotiation.
+  *
+  * The format for multistream Opus packets is defined in
+  * <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
+  * and is based on the self-delimited Opus framing described in Appendix B of
+  * <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
+  * Normal Opus packets are just a degenerate case of multistream Opus packets,
+  * and can be encoded or decoded with the multistream API by setting
+  * <code>streams</code> to <code>1</code> when initializing the encoder or
+  * decoder.
+  *
+  * Multistream Opus streams can contain up to 255 elementary Opus streams.
+  * These may be either "uncoupled" or "coupled", indicating that the decoder
+  * is configured to decode them to either 1 or 2 channels, respectively.
+  * The streams are ordered so that all coupled streams appear at the
+  * beginning.
+  *
+  * A <code>mapping</code> table defines which decoded channel <code>i</code>
+  * should be used for each input/output (I/O) channel <code>j</code>. This table is
+  * typically provided as an unsigned char array.
+  * Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
+  * If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
+  * encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
+  * is even, or  as the right channel of stream <code>(i/2)</code> if
+  * <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
+  * mono in stream <code>(i - coupled_streams)</code>, unless it has the special
+  * value 255, in which case it is omitted from the encoding entirely (the
+  * decoder will reproduce it as silence). Each value <code>i</code> must either
+  * be the special value 255 or be less than <code>streams + coupled_streams</code>.
+  *
+  * The output channels specified by the encoder
+  * should use the
+  * <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
+  * channel ordering</a>. A decoder may wish to apply an additional permutation
+  * to the mapping the encoder used to achieve a different output channel
+  * order (e.g. for outputing in WAV order).
+  *
+  * Each multistream packet contains an Opus packet for each stream, and all of
+  * the Opus packets in a single multistream packet must have the same
+  * duration. Therefore the duration of a multistream packet can be extracted
+  * from the TOC sequence of the first stream, which is located at the
+  * beginning of the packet, just like an elementary Opus stream:
+  *
+  * @code
+  * int nb_samples;
+  * int nb_frames;
+  * nb_frames = opus_packet_get_nb_frames(data, len);
+  * if (nb_frames < 1)
+  *   return nb_frames;
+  * nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
+  * @endcode
+  *
+  * The general encoding and decoding process proceeds exactly the same as in
+  * the normal @ref opus_encoder and @ref opus_decoder APIs.
+  * See their documentation for an overview of how to use the corresponding
+  * multistream functions.
+  */
+
+/** Opus multistream encoder state.
+  * This contains the complete state of a multistream Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_init
+  */
+typedef struct OpusMSEncoder OpusMSEncoder;
+
+/** Opus multistream decoder state.
+  * This contains the complete state of a multistream Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_decoder_init
+  */
+typedef struct OpusMSDecoder OpusMSDecoder;
+
+/**\name Multistream encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusMSEncoder structure.
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
+      int channels,
+      int mapping_family
+);
+
+
+/** Allocates and initializes a multistream encoder state.
+  * Call opus_multistream_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
+
+/** Initialize a previously allocated multistream encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+OPUS_EXPORT int opus_multistream_surround_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6) OPUS_ARG_NONNULL(7);
+
+/** Encodes a multistream Opus frame.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
+    OpusMSEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a multistream Opus frame from floating point input.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
+      OpusMSEncoder *st,
+      const float *pcm,
+      int frame_size,
+      unsigned char *data,
+      opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusMSEncoder</code> allocated by
+  * opus_multistream_encoder_create().
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
+
+/** Perform a CTL function on a multistream Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+/**\name Multistream decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusMSDecoder</code> structure.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+/** Allocates and initializes a multistream decoder state.
+  * Call opus_multistream_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+/** Intialize a previously allocated decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_deocder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_decoder_init(
+      OpusMSDecoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+/** Decode a multistream Opus packet.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode a multistream Opus packet with floating point output.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on a multistream Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusMSDecoder</code> allocated by
+  * opus_multistream_decoder_create().
+  * @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_MULTISTREAM_H */

engine/voice_codecs/celt/opus/opus_projection.h

@@ -0,0 +1,568 @@
+/* Copyright (c) 2017 Google Inc.
+   Written by Andrew Allen */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_projection.h
+ * @brief Opus projection reference API
+ */
+
+#ifndef OPUS_PROJECTION_H
+#define OPUS_PROJECTION_H
+
+#include "opus_multistream.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.c
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST    6001
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST    6003
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST         6005
+/**@}*/
+
+
+/** @endcond */
+
+/** @defgroup opus_projection_ctls Projection specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_projection_encoder_ctl() and opus_projection_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls,
+  * @ref opus_decoderctls, and @ref opus_multistream_ctls may be applied to a
+  * projection encoder or decoder as well.
+  */
+/**@{*/
+
+/** Gets the gain (in dB. S7.8-format) of the demixing matrix from the encoder.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the gain (in dB. S7.8-format)
+  *                                      of the demixing matrix.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Gets the size in bytes of the demixing matrix from the encoder.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the size in bytes of the
+  *                                      demixing matrix.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Copies the demixing matrix to the supplied pointer location.
+  * @param[out] x <tt>unsigned char *</tt>: Returns the demixing matrix to the
+  *                                         supplied pointer location.
+  * @param y <tt>opus_int32</tt>: The size in bytes of the reserved memory at the
+  *                              pointer location.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX(x,y) OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST, x, __opus_check_int(y)
+
+
+/**@}*/
+
+/** Opus projection encoder state.
+ * This contains the complete state of a projection Opus encoder.
+ * It is position independent and can be freely copied.
+ * @see opus_projection_ambisonics_encoder_create
+ */
+typedef struct OpusProjectionEncoder OpusProjectionEncoder;
+
+
+/** Opus projection decoder state.
+  * This contains the complete state of a projection Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_projection_decoder_create
+  * @see opus_projection_decoder_init
+  */
+typedef struct OpusProjectionDecoder OpusProjectionDecoder;
+
+
+/**\name Projection encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusProjectionEncoder structure.
+  * @param channels <tt>int</tt>: The total number of input channels to encode.
+  *                               This must be no more than 255.
+  * @param mapping_family <tt>int</tt>: The mapping family to use for selecting
+  *                                     the appropriate projection.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_ambisonics_encoder_get_size(
+    int channels,
+    int mapping_family
+);
+
+
+/** Allocates and initializes a projection encoder state.
+  * Call opus_projection_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param mapping_family <tt>int</tt>: The mapping family to use for selecting
+  *                                     the appropriate projection.
+  * @param[out] streams <tt>int *</tt>: The total number of streams that will
+  *                                     be encoded from the input.
+  * @param[out] coupled_streams <tt>int *</tt>: Number of coupled (2 channel)
+  *                                 streams that will be encoded from the input.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionEncoder *opus_projection_ambisonics_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int mapping_family,
+    int *streams,
+    int *coupled_streams,
+    int application,
+    int *error
+) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5);
+
+
+/** Initialize a previously allocated projection encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_projection_ambisonics_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_projection_ambisonics_encoder_create
+  * @see opus_projection_ambisonics_encoder_get_size
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_projection_ambisonics_encoder_init(
+    OpusProjectionEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int mapping_family,
+    int *streams,
+    int *coupled_streams,
+    int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
+
+
+/** Encodes a projection Opus frame.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode(
+    OpusProjectionEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+
+/** Encodes a projection Opus frame from floating point input.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode_float(
+    OpusProjectionEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+
+/** Frees an <code>OpusProjectionEncoder</code> allocated by
+  * opus_projection_ambisonics_encoder_create().
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to be freed.
+  */
+OPUS_EXPORT void opus_projection_encoder_destroy(OpusProjectionEncoder *st);
+
+
+/** Perform a CTL function on a projection Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, @ref opus_multistream_ctls, or
+  *                @ref opus_projection_ctls
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  * @see opus_projection_ctls
+  */
+OPUS_EXPORT int opus_projection_encoder_ctl(OpusProjectionEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+/**@}*/
+
+/**\name Projection decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusProjectionDecoder</code> structure.
+  * @param channels <tt>int</tt>: The total number of output channels.
+  *                               This must be no more than 255.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_decoder_get_size(
+    int channels,
+    int streams,
+    int coupled_streams
+);
+
+
+/** Allocates and initializes a projection decoder state.
+  * Call opus_projection_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
+  *                         that mapping from coded channels to output channels,
+  *                         as described in @ref opus_projection and
+  *                         @ref opus_projection_ctls.
+  * @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
+  *                                                  demixing matrix, as
+  *                                                  described in @ref
+  *                                                  opus_projection_ctls.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionDecoder *opus_projection_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int streams,
+    int coupled_streams,
+    unsigned char *demixing_matrix,
+    opus_int32 demixing_matrix_size,
+    int *error
+) OPUS_ARG_NONNULL(5);
+
+
+/** Intialize a previously allocated projection decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_projection_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_projection_decoder_create
+  * @see opus_projection_deocder_get_size
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
+  *                         that mapping from coded channels to output channels,
+  *                         as described in @ref opus_projection and
+  *                         @ref opus_projection_ctls.
+  * @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
+  *                                                  demixing matrix, as
+  *                                                  described in @ref
+  *                                                  opus_projection_ctls.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_projection_decoder_init(
+    OpusProjectionDecoder *st,
+    opus_int32 Fs,
+    int channels,
+    int streams,
+    int coupled_streams,
+    unsigned char *demixing_matrix,
+    opus_int32 demixing_matrix_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+
+/** Decode a projection Opus packet.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode(
+    OpusProjectionDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+
+/** Decode a projection Opus packet with floating point output.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode_float(
+    OpusProjectionDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+
+/** Perform a CTL function on a projection Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, @ref opus_multistream_ctls, or
+  *                @ref opus_projection_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  * @see opus_projection_ctls
+  */
+OPUS_EXPORT int opus_projection_decoder_ctl(OpusProjectionDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+/** Frees an <code>OpusProjectionDecoder</code> allocated by
+  * opus_projection_decoder_create().
+  * @param st <tt>OpusProjectionDecoder</tt>: Projection decoder state to be freed.
+  */
+OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st);
+
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_PROJECTION_H */

engine/voice_codecs/celt/opus/opus_types.h

@@ -0,0 +1,166 @@
+/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
+/* Modified by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+/* opus_types.h based on ogg_types.h from libogg */
+
+/**
+   @file opus_types.h
+   @brief Opus reference implementation types
+*/
+#ifndef OPUS_TYPES_H
+#define OPUS_TYPES_H
+
+#define opus_int         int                     /* used for counters etc; at least 16 bits */
+#define opus_int64       long long
+#define opus_int8        signed char
+
+#define opus_uint        unsigned int            /* used for counters etc; at least 16 bits */
+#define opus_uint64      unsigned long long
+#define opus_uint8       unsigned char
+
+/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
+#if (defined(__STDC__) && __STDC__ && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
+#include <stdint.h>
+#  undef opus_int64
+#  undef opus_int8
+#  undef opus_uint64
+#  undef opus_uint8
+   typedef int8_t opus_int8;
+   typedef uint8_t opus_uint8;
+   typedef int16_t opus_int16;
+   typedef uint16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef uint32_t opus_uint32;
+   typedef int64_t opus_int64;
+   typedef uint64_t opus_uint64;
+#elif defined(_WIN32)
+
+#  if defined(__CYGWIN__)
+#    include <_G_config.h>
+     typedef _G_int32_t opus_int32;
+     typedef _G_uint32_t opus_uint32;
+     typedef _G_int16 opus_int16;
+     typedef _G_uint16 opus_uint16;
+#  elif defined(__MINGW32__)
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+#  elif defined(__MWERKS__)
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+#  else
+     /* MSVC/Borland */
+     typedef __int32 opus_int32;
+     typedef unsigned __int32 opus_uint32;
+     typedef __int16 opus_int16;
+     typedef unsigned __int16 opus_uint16;
+#  endif
+
+#elif defined(__MACOS__)
+
+#  include <sys/types.h>
+   typedef SInt16 opus_int16;
+   typedef UInt16 opus_uint16;
+   typedef SInt32 opus_int32;
+   typedef UInt32 opus_uint32;
+
+#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
+
+#  include <sys/types.h>
+   typedef int16_t opus_int16;
+   typedef u_int16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined(__BEOS__)
+
+   /* Be */
+#  include <inttypes.h>
+   typedef int16 opus_int16;
+   typedef u_int16 opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined (__EMX__)
+
+   /* OS/2 GCC */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined (DJGPP)
+
+   /* DJGPP */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(R5900)
+
+   /* PS2 EE */
+   typedef int opus_int32;
+   typedef unsigned opus_uint32;
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+
+#elif defined(__SYMBIAN32__)
+
+   /* Symbian GCC */
+   typedef signed short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef signed int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef long opus_int32;
+   typedef unsigned long opus_uint32;
+
+#elif defined(CONFIG_TI_C6X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#else
+
+   /* Give up, take a reasonable guess */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#endif
+
+#endif  /* OPUS_TYPES_H */

engine/voice_codecs/celt/voiceencoder_celt.cpp

@@ -41,13 +41,13 @@ damage. */
 #include "ivoicecodec.h"
 #include "iframeencoder.h"
 
-#ifdef POSIX
+/*#ifdef POSIX
 #include "source/osx/config.h"
 #else
 #include "source/msvc/config.h"
-#endif
+#endif*/
 #include <stdio.h>
-#include "celt.h"
+#include <celt/celt.h>
 
 
 // NOTE: This has to be the last file included!
@@ -67,21 +67,10 @@ struct celt_versions
 
 celt_versions g_CeltVersion[CELT_VERSION] = 
 {
-	{
-		44100, 256, 120 
-	},
-
-	{
-		22050, 120, 60
-	},
-
-	{
-		22050, 256, 60
-	},
-
-	{
-		22050, 512, 64
-	},
+	{44100, 256, 120},
+	{22050, 120, 60},
+	{22050, 256, 60},
+	{22050, 512, 64},
 };
 
 class VoiceEncoder_Celt : public IFrameEncoder

engine/voice_codecs/celt/wscript

@@ -0,0 +1,54 @@
+#! /usr/bin/env python
+# encoding: utf-8
+
+from waflib import Utils
+import os
+
+top = '.'
+PROJECT_NAME = 'vaudio_celt'
+
+def options(opt):
+	# stub
+	return
+
+def configure(conf):
+	conf.define('SPEEX_EXPORTS',1)
+#	conf.define('NO_HOOK_MALLOC',1)
+
+def build(bld):
+	source = [
+		'voiceencoder_celt.cpp',
+		'../frame_encoder/voice_codec_frame.cpp',
+		'../../../tier1/interface.cpp'
+	]
+
+	includes = [
+		'.',
+		'../frame_encoder',
+		'../../../public',
+		'../../../public/tier1',
+		'../../../public/tier0',
+		'../../',
+		'../../../common',
+		'../../audio/public'
+	]
+
+	defines = []
+
+	libs = ['tier0','tier1','vstdlib','CELT']
+
+	install_path = bld.env.LIBDIR
+
+	bld.shlib(
+		source   = source,
+		target   = PROJECT_NAME,
+		name     = PROJECT_NAME,
+		features = 'c cxx',
+		includes = includes,
+		defines  = defines,
+		use      = libs,
+		install_path = install_path,
+		subsystem = bld.env.MSVC_SUBSYSTEM,
+		idx      = bld.get_taskgen_count()
+	)
+

engine/voice_codecs/opus/opus/opus.h

@@ -0,0 +1,981 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus.h
+ * @brief Opus reference implementation API
+ */
+
+#ifndef OPUS_H
+#define OPUS_H
+
+#include "opus_types.h"
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * @mainpage Opus
+ *
+ * The Opus codec is designed for interactive speech and audio transmission over the Internet.
+ * It is designed by the IETF Codec Working Group and incorporates technology from
+ * Skype's SILK codec and Xiph.Org's CELT codec.
+ *
+ * The Opus codec is designed to handle a wide range of interactive audio applications,
+ * including Voice over IP, videoconferencing, in-game chat, and even remote live music
+ * performances. It can scale from low bit-rate narrowband speech to very high quality
+ * stereo music. Its main features are:
+
+ * @li Sampling rates from 8 to 48 kHz
+ * @li Bit-rates from 6 kb/s to 510 kb/s
+ * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
+ * @li Audio bandwidth from narrowband to full-band
+ * @li Support for speech and music
+ * @li Support for mono and stereo
+ * @li Support for multichannel (up to 255 channels)
+ * @li Frame sizes from 2.5 ms to 60 ms
+ * @li Good loss robustness and packet loss concealment (PLC)
+ * @li Floating point and fixed-point implementation
+ *
+ * Documentation sections:
+ * @li @ref opus_encoder
+ * @li @ref opus_decoder
+ * @li @ref opus_repacketizer
+ * @li @ref opus_multistream
+ * @li @ref opus_libinfo
+ * @li @ref opus_custom
+ */
+
+/** @defgroup opus_encoder Opus Encoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to encode Opus.
+  *
+  * Since Opus is a stateful codec, the encoding process starts with creating an encoder
+  * state. This can be done with:
+  *
+  * @code
+  * int          error;
+  * OpusEncoder *enc;
+  * enc = opus_encoder_create(Fs, channels, application, &error);
+  * @endcode
+  *
+  * From this point, @c enc can be used for encoding an audio stream. An encoder state
+  * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
+  * state @b must @b not be re-initialized for each frame.
+  *
+  * While opus_encoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  *
+  * @code
+  * int          size;
+  * int          error;
+  * OpusEncoder *enc;
+  * size = opus_encoder_get_size(channels);
+  * enc = malloc(size);
+  * error = opus_encoder_init(enc, Fs, channels, application);
+  * @endcode
+  *
+  * where opus_encoder_get_size() returns the required size for the encoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The encoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
+  * interface. All these settings already default to the recommended value, so they should
+  * only be changed when necessary. The most common settings one may want to change are:
+  *
+  * @code
+  * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
+  * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
+  * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
+  * @endcode
+  *
+  * where
+  *
+  * @arg bitrate is in bits per second (b/s)
+  * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
+  * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
+  *
+  * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
+  *
+  * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
+  * @code
+  * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
+  * @endcode
+  *
+  * where
+  * <ul>
+  * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
+  * <li>frame_size is the duration of the frame in samples (per channel)</li>
+  * <li>packet is the byte array to which the compressed data is written</li>
+  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
+  *     Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
+  * </ul>
+  *
+  * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
+  * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
+  * is 2 bytes or less, then the packet does not need to be transmitted (DTX).
+  *
+  * Once the encoder state if no longer needed, it can be destroyed with
+  *
+  * @code
+  * opus_encoder_destroy(enc);
+  * @endcode
+  *
+  * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
+  * then no action is required aside from potentially freeing the memory that was manually
+  * allocated for it (calling free(enc) for the example above)
+  *
+  */
+
+/** Opus encoder state.
+  * This contains the complete state of an Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_encoder_create,opus_encoder_init
+  */
+typedef struct OpusEncoder OpusEncoder;
+
+/** Gets the size of an <code>OpusEncoder</code> structure.
+  * @param[in] channels <tt>int</tt>: Number of channels.
+  *                                   This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
+
+/**
+ */
+
+/** Allocates and initializes an encoder state.
+ * There are three coding modes:
+ *
+ * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
+ *    signals. It enhances the  input signal by high-pass filtering and
+ *    emphasizing formants and harmonics. Optionally  it includes in-band
+ *    forward error correction to protect against packet loss. Use this
+ *    mode for typical VoIP applications. Because of the enhancement,
+ *    even at high bitrates the output may sound different from the input.
+ *
+ * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
+ *    non-voice signals like music. Use this mode for music and mixed
+ *    (music/voice) content, broadcast, and applications requiring less
+ *    than 15 ms of coding delay.
+ *
+ * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
+ *    disables the speech-optimized mode in exchange for slightly reduced delay.
+ *    This mode can only be set on an newly initialized or freshly reset encoder
+ *    because it changes the codec delay.
+ *
+ * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
+ * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                     This must be one of 8000, 12000, 16000,
+ *                                     24000, or 48000.
+ * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+ * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
+ * @note Regardless of the sampling rate and number channels selected, the Opus encoder
+ * can switch to a lower audio bandwidth or number of channels if the bitrate
+ * selected is too low. This also means that it is safe to always use 48 kHz stereo input
+ * and let the encoder optimize the encoding.
+ */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int application,
+    int *error
+);
+
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_encoder_create(),opus_encoder_get_size()
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                      This must be one of 8000, 12000, 16000,
+ *                                      24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+  * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_encoder_init(
+    OpusEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int application
+) OPUS_ARG_NONNULL(1);
+
+/** Encodes an Opus frame.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
+    OpusEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes an Opus frame from floating point input.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range.
+  *          length is frame_size*channels*sizeof(float)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
+    OpusEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
+  * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
+
+/** Perform a CTL function on an Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusEncoder*</tt>: Encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_encoderctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  */
+OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+/**@}*/
+
+/** @defgroup opus_decoder Opus Decoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to decode Opus.
+  *
+  * The decoding process also starts with creating a decoder
+  * state. This can be done with:
+  * @code
+  * int          error;
+  * OpusDecoder *dec;
+  * dec = opus_decoder_create(Fs, channels, &error);
+  * @endcode
+  * where
+  * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
+  * @li channels is the number of channels (1 or 2)
+  * @li error will hold the error code in case of failure (or #OPUS_OK on success)
+  * @li the return value is a newly created decoder state to be used for decoding
+  *
+  * While opus_decoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  * @code
+  * int          size;
+  * int          error;
+  * OpusDecoder *dec;
+  * size = opus_decoder_get_size(channels);
+  * dec = malloc(size);
+  * error = opus_decoder_init(dec, Fs, channels);
+  * @endcode
+  * where opus_decoder_get_size() returns the required size for the decoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The decoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
+  * @code
+  * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
+  * @endcode
+  * where
+  *
+  * @li packet is the byte array containing the compressed data
+  * @li len is the exact number of bytes contained in the packet
+  * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
+  * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
+  *
+  * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
+  * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
+  * buffer is too small to hold the decoded audio.
+  *
+  * Opus is a stateful codec with overlapping blocks and as a result Opus
+  * packets are not coded independently of each other. Packets must be
+  * passed into the decoder serially and in the correct order for a correct
+  * decode. Lost packets can be replaced with loss concealment by calling
+  * the decoder with a null pointer and zero length for the missing packet.
+  *
+  * A single codec state may only be accessed from a single thread at
+  * a time and any required locking must be performed by the caller. Separate
+  * streams must be decoded with separate decoder states and can be decoded
+  * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
+  * defined.
+  *
+  */
+
+/** Opus decoder state.
+  * This contains the complete state of an Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_decoder_create,opus_decoder_init
+  */
+typedef struct OpusDecoder OpusDecoder;
+
+/** Gets the size of an <code>OpusDecoder</code> structure.
+  * @param [in] channels <tt>int</tt>: Number of channels.
+  *                                    This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
+
+/** Allocates and initializes a decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
+  *
+  * Internally Opus stores data at 48000 Hz, so that should be the default
+  * value for Fs. However, the decoder can efficiently decode to buffers
+  * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
+  * data at the full sample rate, or knows the compressed data doesn't
+  * use the full frequency range, it can request decoding at a reduced
+  * rate. Likewise, the decoder is capable of filling in either mono or
+  * interleaved stereo pcm buffers, at the caller's request.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int *error
+);
+
+/** Initializes a previously allocated decoder state.
+  * The state must be at least the size returned by opus_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_decoder_init(
+    OpusDecoder *st,
+    opus_int32 Fs,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Decode an Opus packet.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available, the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an Opus packet with floating point output.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_decoderctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  */
+OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
+  * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
+
+/** Parse an opus packet into one or more frames.
+  * Opus_decode will perform this operation internally so most applications do
+  * not need to use this function.
+  * This function does not copy the frames, the returned pointers are pointers into
+  * the input packet.
+  * @param [in] data <tt>char*</tt>: Opus packet to be parsed
+  * @param [in] len <tt>opus_int32</tt>: size of data
+  * @param [out] out_toc <tt>char*</tt>: TOC pointer
+  * @param [out] frames <tt>char*[48]</tt> encapsulated frames
+  * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
+  * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
+  * @returns number of frames
+  */
+OPUS_EXPORT int opus_packet_parse(
+   const unsigned char *data,
+   opus_int32 len,
+   unsigned char *out_toc,
+   const unsigned char *frames[48],
+   opus_int16 size[48],
+   int *payload_offset
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5);
+
+/** Gets the bandwidth of an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
+  * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
+  * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
+  * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
+  * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples per frame from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet.
+  *                                  This must contain at least one byte of
+  *                                  data.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples per frame.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of channels from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @returns Number of channels
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of frames in an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of frames
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
+  * the signal is already in that range, nothing is done. If there are values
+  * outside of [-1,1], then the signal is clipped as smoothly as possible to
+  * both fit in the range and avoid creating excessive distortion in the
+  * process.
+  * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
+  * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
+  */
+OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
+
+
+/**@}*/
+
+/** @defgroup opus_repacketizer Repacketizer
+  * @{
+  *
+  * The repacketizer can be used to merge multiple Opus packets into a single
+  * packet or alternatively to split Opus packets that have previously been
+  * merged. Splitting valid Opus packets is always guaranteed to succeed,
+  * whereas merging valid packets only succeeds if all frames have the same
+  * mode, bandwidth, and frame size, and when the total duration of the merged
+  * packet is no more than 120 ms. The 120 ms limit comes from the
+  * specification and limits decoder memory requirements at a point where
+  * framing overhead becomes negligible.
+  *
+  * The repacketizer currently only operates on elementary Opus
+  * streams. It will not manipualte multistream packets successfully, except in
+  * the degenerate case where they consist of data from a single stream.
+  *
+  * The repacketizing process starts with creating a repacketizer state, either
+  * by calling opus_repacketizer_create() or by allocating the memory yourself,
+  * e.g.,
+  * @code
+  * OpusRepacketizer *rp;
+  * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
+  * if (rp != NULL)
+  *     opus_repacketizer_init(rp);
+  * @endcode
+  *
+  * Then the application should submit packets with opus_repacketizer_cat(),
+  * extract new packets with opus_repacketizer_out() or
+  * opus_repacketizer_out_range(), and then reset the state for the next set of
+  * input packets via opus_repacketizer_init().
+  *
+  * For example, to split a sequence of packets into individual frames:
+  * @code
+  * unsigned char *data;
+  * int len;
+  * while (get_next_packet(&data, &len))
+  * {
+  *   unsigned char out[1276];
+  *   opus_int32 out_len;
+  *   int nb_frames;
+  *   int err;
+  *   int i;
+  *   err = opus_repacketizer_cat(rp, data, len);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packet(data);
+  *     return err;
+  *   }
+  *   nb_frames = opus_repacketizer_get_nb_frames(rp);
+  *   for (i = 0; i < nb_frames; i++)
+  *   {
+  *     out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packet(data);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *   }
+  *   opus_repacketizer_init(rp);
+  *   release_packet(data);
+  * }
+  * @endcode
+  *
+  * Alternatively, to combine a sequence of frames into packets that each
+  * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
+  * @code
+  * // The maximum number of packets with duration TARGET_DURATION_MS occurs
+  * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
+  * // packets.
+  * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
+  * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
+  * int nb_packets;
+  * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
+  * opus_int32 out_len;
+  * int prev_toc;
+  * nb_packets = 0;
+  * while (get_next_packet(data+nb_packets, len+nb_packets))
+  * {
+  *   int nb_frames;
+  *   int err;
+  *   nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
+  *   if (nb_frames < 1)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return nb_frames;
+  *   }
+  *   nb_frames += opus_repacketizer_get_nb_frames(rp);
+  *   // If adding the next packet would exceed our target, or it has an
+  *   // incompatible TOC sequence, output the packets we already have before
+  *   // submitting it.
+  *   // N.B., The nb_packets > 0 check ensures we've submitted at least one
+  *   // packet since the last call to opus_repacketizer_init(). Otherwise a
+  *   // single packet longer than TARGET_DURATION_MS would cause us to try to
+  *   // output an (invalid) empty packet. It also ensures that prev_toc has
+  *   // been set to a valid value. Additionally, len[nb_packets] > 0 is
+  *   // guaranteed by the call to opus_packet_get_nb_frames() above, so the
+  *   // reference to data[nb_packets][0] should be valid.
+  *   if (nb_packets > 0 && (
+  *       ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
+  *       opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
+  *       TARGET_DURATION_MS*48))
+  *   {
+  *     out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packets(data, nb_packets+1);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *     opus_repacketizer_init(rp);
+  *     release_packets(data, nb_packets);
+  *     data[0] = data[nb_packets];
+  *     len[0] = len[nb_packets];
+  *     nb_packets = 0;
+  *   }
+  *   err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return err;
+  *   }
+  *   prev_toc = data[nb_packets][0];
+  *   nb_packets++;
+  * }
+  * // Output the final, partial packet.
+  * if (nb_packets > 0)
+  * {
+  *   out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *   release_packets(data, nb_packets);
+  *   if (out_len < 0)
+  *     return (int)out_len;
+  *   output_next_packet(out, out_len);
+  * }
+  * @endcode
+  *
+  * An alternate way of merging packets is to simply call opus_repacketizer_cat()
+  * unconditionally until it fails. At that point, the merged packet can be
+  * obtained with opus_repacketizer_out() and the input packet for which
+  * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
+  * repacketizer state.
+  */
+
+typedef struct OpusRepacketizer OpusRepacketizer;
+
+/** Gets the size of an <code>OpusRepacketizer</code> structure.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
+
+/** (Re)initializes a previously allocated repacketizer state.
+  * The state must be at least the size returned by opus_repacketizer_get_size().
+  * This can be used for applications which use their own allocator instead of
+  * malloc().
+  * It must also be called to reset the queue of packets waiting to be
+  * repacketized, which is necessary if the maximum packet duration of 120 ms
+  * is reached or if you wish to submit packets with a different Opus
+  * configuration (coding mode, audio bandwidth, frame size, or channel count).
+  * Failure to do so will prevent a new packet from being added with
+  * opus_repacketizer_cat().
+  * @see opus_repacketizer_create
+  * @see opus_repacketizer_get_size
+  * @see opus_repacketizer_cat
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
+  *                                       (re)initialize.
+  * @returns A pointer to the same repacketizer state that was passed in.
+  */
+OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Allocates memory and initializes the new repacketizer with
+ * opus_repacketizer_init().
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
+
+/** Frees an <code>OpusRepacketizer</code> allocated by
+  * opus_repacketizer_create().
+  * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
+
+/** Add a packet to the current repacketizer state.
+  * This packet must match the configuration of any packets already submitted
+  * for repacketization since the last call to opus_repacketizer_init().
+  * This means that it must have the same coding mode, audio bandwidth, frame
+  * size, and channel count.
+  * This can be checked in advance by examining the top 6 bits of the first
+  * byte of the packet, and ensuring they match the top 6 bits of the first
+  * byte of any previously submitted packet.
+  * The total duration of audio in the repacketizer state also must not exceed
+  * 120 ms, the maximum duration of a single packet, after adding this packet.
+  *
+  * The contents of the current repacketizer state can be extracted into new
+  * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
+  *
+  * In order to add a packet with a different configuration or to add more
+  * audio beyond 120 ms, you must clear the repacketizer state by calling
+  * opus_repacketizer_init().
+  * If a packet is too large to add to the current repacketizer state, no part
+  * of it is added, even if it contains multiple frames, some of which might
+  * fit.
+  * If you wish to be able to add parts of such packets, you should first use
+  * another repacketizer to split the packet into pieces and add them
+  * individually.
+  * @see opus_repacketizer_out_range
+  * @see opus_repacketizer_out
+  * @see opus_repacketizer_init
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
+  *                                       add the packet.
+  * @param[in] data <tt>const unsigned char*</tt>: The packet data.
+  *                                                The application must ensure
+  *                                                this pointer remains valid
+  *                                                until the next call to
+  *                                                opus_repacketizer_init() or
+  *                                                opus_repacketizer_destroy().
+  * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
+  * @returns An error code indicating whether or not the operation succeeded.
+  * @retval #OPUS_OK The packet's contents have been added to the repacketizer
+  *                  state.
+  * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
+  *                              the packet's TOC sequence was not compatible
+  *                              with previously submitted packets (because
+  *                              the coding mode, audio bandwidth, frame size,
+  *                              or channel count did not match), or adding
+  *                              this packet would increase the total amount of
+  *                              audio stored in the repacketizer state to more
+  *                              than 120 ms.
+  */
+OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param begin <tt>int</tt>: The index of the first frame in the current
+  *                            repacketizer state to include in the output.
+  * @param end <tt>int</tt>: One past the index of the last frame in the
+  *                          current repacketizer state to include in the
+  *                          output.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1276</code> for a single frame,
+  *                                    or for multiple frames,
+  *                                    <code>1277*(end-begin)</code>.
+  *                                    However, <code>1*(end-begin)</code> plus
+  *                                    the size of all packet data submitted to
+  *                                    the repacketizer since the last call to
+  *                                    opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
+  *                       frames (begin < 0, begin >= end, or end >
+  *                       opus_repacketizer_get_nb_frames()).
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Return the total number of frames contained in packet data submitted to
+  * the repacketizer state so far via opus_repacketizer_cat() since the last
+  * call to opus_repacketizer_init() or opus_repacketizer_create().
+  * This defines the valid range of packets that can be extracted with
+  * opus_repacketizer_out_range() or opus_repacketizer_out().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
+  *                                       frames.
+  * @returns The total number of frames contained in the packet data submitted
+  *          to the repacketizer state.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * This is a convenience routine that returns all the data submitted so far
+  * in a single packet.
+  * It is equivalent to calling
+  * @code
+  * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
+  *                             data, maxlen)
+  * @endcode
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
+  *                                    However,
+  *                                    <code>1*opus_repacketizer_get_nb_frames(rp)</code>
+  *                                    plus the size of all packet data
+  *                                    submitted to the repacketizer since the
+  *                                    last call to opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
+
+/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
+
+/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
+
+/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
+
+/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */

engine/voice_codecs/opus/opus/opus_custom.h

@@ -0,0 +1,342 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008-2012 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+  @file opus_custom.h
+  @brief Opus-Custom reference implementation API
+ */
+
+#ifndef OPUS_CUSTOM_H
+#define OPUS_CUSTOM_H
+
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef CUSTOM_MODES
+# define OPUS_CUSTOM_EXPORT OPUS_EXPORT
+# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT
+#else
+# define OPUS_CUSTOM_EXPORT
+# ifdef OPUS_BUILD
+#  define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE
+# else
+#  define OPUS_CUSTOM_EXPORT_STATIC
+# endif
+#endif
+
+/** @defgroup opus_custom Opus Custom
+  * @{
+  *  Opus Custom is an optional part of the Opus specification and
+  * reference implementation which uses a distinct API from the regular
+  * API and supports frame sizes that are not normally supported.\ Use
+  * of Opus Custom is discouraged for all but very special applications
+  * for which a frame size different from 2.5, 5, 10, or 20 ms is needed
+  * (for either complexity or latency reasons) and where interoperability
+  * is less important.
+  *
+  * In addition to the interoperability limitations the use of Opus custom
+  * disables a substantial chunk of the codec and generally lowers the
+  * quality available at a given bitrate. Normally when an application needs
+  * a different frame size from the codec it should buffer to match the
+  * sizes but this adds a small amount of delay which may be important
+  * in some very low latency applications. Some transports (especially
+  * constant rate RF transports) may also work best with frames of
+  * particular durations.
+  *
+  * Libopus only supports custom modes if they are enabled at compile time.
+  *
+  * The Opus Custom API is similar to the regular API but the
+  * @ref opus_encoder_create and @ref opus_decoder_create calls take
+  * an additional mode parameter which is a structure produced by
+  * a call to @ref opus_custom_mode_create. Both the encoder and decoder
+  * must create a mode using the same sample rate (fs) and frame size
+  * (frame size) so these parameters must either be signaled out of band
+  * or fixed in a particular implementation.
+  *
+  * Similar to regular Opus the custom modes support on the fly frame size
+  * switching, but the sizes available depend on the particular frame size in
+  * use. For some initial frame sizes on a single on the fly size is available.
+  */
+
+/** Contains the state of an encoder. One encoder state is needed
+    for each stream. It is initialized once at the beginning of the
+    stream. Do *not* re-initialize the state for every frame.
+   @brief Encoder state
+ */
+typedef struct OpusCustomEncoder OpusCustomEncoder;
+
+/** State of the decoder. One decoder state is needed for each stream.
+    It is initialized once at the beginning of the stream. Do *not*
+    re-initialize the state for every frame.
+   @brief Decoder state
+ */
+typedef struct OpusCustomDecoder OpusCustomDecoder;
+
+/** The mode contains all the information necessary to create an
+    encoder. Both the encoder and decoder need to be initialized
+    with exactly the same mode, otherwise the output will be
+    corrupted.
+   @brief Mode configuration
+ */
+typedef struct OpusCustomMode OpusCustomMode;
+
+/** Creates a new mode struct. This will be passed to an encoder or
+  * decoder. The mode MUST NOT BE DESTROYED until the encoders and
+  * decoders that use it are destroyed as well.
+  * @param [in] Fs <tt>int</tt>: Sampling rate (8000 to 96000 Hz)
+  * @param [in] frame_size <tt>int</tt>: Number of samples (per channel) to encode in each
+  *        packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes)
+  * @param [out] error <tt>int*</tt>: Returned error code (if NULL, no error will be returned)
+  * @return A newly created mode
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error);
+
+/** Destroys a mode struct. Only call this after all encoders and
+  * decoders using this mode are destroyed as well.
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Mode to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C)
+
+/* Encoder */
+/** Gets the size of an OpusCustomEncoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+# ifdef CUSTOM_MODES
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_encoder_create(),opus_custom_encoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_init(
+    OpusCustomEncoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+# endif
+#endif
+
+
+/** Creates a new encoder state. Each stream needs its own encoder
+  * state (can't be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created encoder state.
+*/
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+
+/** Destroys a an encoder state.
+  * @param[in] st <tt>OpusCustomEncoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: PCM audio in float format, with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range. There must be exactly
+  *          frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float(
+    OpusCustomEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian).
+  *          There must be exactly frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+ */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode(
+    OpusCustomEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_encoderctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C)
+/* Decoder */
+
+/** Gets the size of an OpusCustomDecoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Initializes a previously allocated decoder state
+  * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_decoder_create(),opus_custom_decoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init(
+    OpusCustomDecoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+#endif
+
+
+/** Creates a new decoder state. Each stream needs its own decoder state (can't
+  * be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode</tt>: Contains all the information about the characteristics of the
+  *          stream (must be the same characteristics as used for the encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created decoder state.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+/** Destroys a an decoder state.
+  * @param[in] st <tt>OpusCustomDecoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st);
+
+/** Decode an opus custom frame with floating point output
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    float *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an opus custom frame
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    opus_int16 *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_genericctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_CUSTOM_H */

engine/voice_codecs/opus/opus/opus_defines.h

@@ -0,0 +1,799 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_defines.h
+ * @brief Opus reference implementation constants
+ */
+
+#ifndef OPUS_DEFINES_H
+#define OPUS_DEFINES_H
+
+#include "opus_types.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @defgroup opus_errorcodes Error codes
+ * @{
+ */
+/** No error @hideinitializer*/
+#define OPUS_OK                0
+/** One or more invalid/out of range arguments @hideinitializer*/
+#define OPUS_BAD_ARG          -1
+/** Not enough bytes allocated in the buffer @hideinitializer*/
+#define OPUS_BUFFER_TOO_SMALL -2
+/** An internal error was detected @hideinitializer*/
+#define OPUS_INTERNAL_ERROR   -3
+/** The compressed data passed is corrupted @hideinitializer*/
+#define OPUS_INVALID_PACKET   -4
+/** Invalid/unsupported request number @hideinitializer*/
+#define OPUS_UNIMPLEMENTED    -5
+/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
+#define OPUS_INVALID_STATE    -6
+/** Memory allocation has failed @hideinitializer*/
+#define OPUS_ALLOC_FAIL       -7
+/**@}*/
+
+/** @cond OPUS_INTERNAL_DOC */
+/**Export control for opus functions */
+
+#ifndef OPUS_EXPORT
+# if defined(WIN32)
+#  if defined(OPUS_BUILD) && defined(DLL_EXPORT)
+#   define OPUS_EXPORT __declspec(dllexport)
+#  else
+#   define OPUS_EXPORT
+#  endif
+# elif defined(__GNUC__) && defined(OPUS_BUILD)
+#  define OPUS_EXPORT __attribute__ ((visibility ("default")))
+# else
+#  define OPUS_EXPORT
+# endif
+#endif
+
+# if !defined(OPUS_GNUC_PREREQ)
+#  if defined(__GNUC__)&&defined(__GNUC_MINOR__)
+#   define OPUS_GNUC_PREREQ(_maj,_min) \
+ ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
+#  else
+#   define OPUS_GNUC_PREREQ(_maj,_min) 0
+#  endif
+# endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(3,0)
+#  define OPUS_RESTRICT __restrict__
+# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
+#  define OPUS_RESTRICT __restrict
+# else
+#  define OPUS_RESTRICT
+# endif
+#else
+# define OPUS_RESTRICT restrict
+#endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(2,7)
+#  define OPUS_INLINE __inline__
+# elif (defined(_MSC_VER))
+#  define OPUS_INLINE __inline
+# else
+#  define OPUS_INLINE
+# endif
+#else
+# define OPUS_INLINE inline
+#endif
+
+/**Warning attributes for opus functions
+  * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
+  * some paranoid null checks. */
+#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
+#else
+# define OPUS_WARN_UNUSED_RESULT
+#endif
+#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_ARG_NONNULL(_x)  __attribute__ ((__nonnull__(_x)))
+#else
+# define OPUS_ARG_NONNULL(_x)
+#endif
+
+/** These are the actual Encoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+#define OPUS_SET_APPLICATION_REQUEST         4000
+#define OPUS_GET_APPLICATION_REQUEST         4001
+#define OPUS_SET_BITRATE_REQUEST             4002
+#define OPUS_GET_BITRATE_REQUEST             4003
+#define OPUS_SET_MAX_BANDWIDTH_REQUEST       4004
+#define OPUS_GET_MAX_BANDWIDTH_REQUEST       4005
+#define OPUS_SET_VBR_REQUEST                 4006
+#define OPUS_GET_VBR_REQUEST                 4007
+#define OPUS_SET_BANDWIDTH_REQUEST           4008
+#define OPUS_GET_BANDWIDTH_REQUEST           4009
+#define OPUS_SET_COMPLEXITY_REQUEST          4010
+#define OPUS_GET_COMPLEXITY_REQUEST          4011
+#define OPUS_SET_INBAND_FEC_REQUEST          4012
+#define OPUS_GET_INBAND_FEC_REQUEST          4013
+#define OPUS_SET_PACKET_LOSS_PERC_REQUEST    4014
+#define OPUS_GET_PACKET_LOSS_PERC_REQUEST    4015
+#define OPUS_SET_DTX_REQUEST                 4016
+#define OPUS_GET_DTX_REQUEST                 4017
+#define OPUS_SET_VBR_CONSTRAINT_REQUEST      4020
+#define OPUS_GET_VBR_CONSTRAINT_REQUEST      4021
+#define OPUS_SET_FORCE_CHANNELS_REQUEST      4022
+#define OPUS_GET_FORCE_CHANNELS_REQUEST      4023
+#define OPUS_SET_SIGNAL_REQUEST              4024
+#define OPUS_GET_SIGNAL_REQUEST              4025
+#define OPUS_GET_LOOKAHEAD_REQUEST           4027
+/* #define OPUS_RESET_STATE 4028 */
+#define OPUS_GET_SAMPLE_RATE_REQUEST         4029
+#define OPUS_GET_FINAL_RANGE_REQUEST         4031
+#define OPUS_GET_PITCH_REQUEST               4033
+#define OPUS_SET_GAIN_REQUEST                4034
+#define OPUS_GET_GAIN_REQUEST                4045 /* Should have been 4035 */
+#define OPUS_SET_LSB_DEPTH_REQUEST           4036
+#define OPUS_GET_LSB_DEPTH_REQUEST           4037
+#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
+#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
+#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
+#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
+#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
+/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
+#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046
+#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047
+#define OPUS_GET_IN_DTX_REQUEST              4049
+
+/** Defines for the presence of extended APIs. */
+#define OPUS_HAVE_OPUS_PROJECTION_H
+
+/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
+#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
+#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
+#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
+#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
+/** @endcond */
+
+/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
+  * @see opus_genericctls, opus_encoderctls
+  * @{
+  */
+/* Values for the various encoder CTLs */
+#define OPUS_AUTO                           -1000 /**<Auto/default setting @hideinitializer*/
+#define OPUS_BITRATE_MAX                       -1 /**<Maximum bitrate @hideinitializer*/
+
+/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
+ * @hideinitializer */
+#define OPUS_APPLICATION_VOIP                2048
+/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
+ * @hideinitializer */
+#define OPUS_APPLICATION_AUDIO               2049
+/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
+ * @hideinitializer */
+#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
+
+#define OPUS_SIGNAL_VOICE                    3001 /**< Signal being encoded is voice */
+#define OPUS_SIGNAL_MUSIC                    3002 /**< Signal being encoded is music */
+#define OPUS_BANDWIDTH_NARROWBAND            1101 /**< 4 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_MEDIUMBAND            1102 /**< 6 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_WIDEBAND              1103 /**< 8 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_SUPERWIDEBAND         1104 /**<12 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_FULLBAND              1105 /**<20 kHz bandpass @hideinitializer*/
+
+#define OPUS_FRAMESIZE_ARG                   5000 /**< Select frame size from the argument (default) */
+#define OPUS_FRAMESIZE_2_5_MS                5001 /**< Use 2.5 ms frames */
+#define OPUS_FRAMESIZE_5_MS                  5002 /**< Use 5 ms frames */
+#define OPUS_FRAMESIZE_10_MS                 5003 /**< Use 10 ms frames */
+#define OPUS_FRAMESIZE_20_MS                 5004 /**< Use 20 ms frames */
+#define OPUS_FRAMESIZE_40_MS                 5005 /**< Use 40 ms frames */
+#define OPUS_FRAMESIZE_60_MS                 5006 /**< Use 60 ms frames */
+#define OPUS_FRAMESIZE_80_MS                 5007 /**< Use 80 ms frames */
+#define OPUS_FRAMESIZE_100_MS                5008 /**< Use 100 ms frames */
+#define OPUS_FRAMESIZE_120_MS                5009 /**< Use 120 ms frames */
+
+/**@}*/
+
+
+/** @defgroup opus_encoderctls Encoder related CTLs
+  *
+  * These are convenience macros for use with the \c opus_encode_ctl
+  * interface. They are used to generate the appropriate series of
+  * arguments for that call, passing the correct type, size and so
+  * on as expected for each particular request.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
+  * if (ret != OPUS_OK) return ret;
+  *
+  * opus_int32 rate;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * @endcode
+  *
+  * @see opus_genericctls, opus_encoder
+  * @{
+  */
+
+/** Configures the encoder's computational complexity.
+  * The supported range is 0-10 inclusive with 10 representing the highest complexity.
+  * @see OPUS_GET_COMPLEXITY
+  * @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
+  *
+  * @hideinitializer */
+#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
+/** Gets the encoder's complexity configuration.
+  * @see OPUS_SET_COMPLEXITY
+  * @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
+  *                                      inclusive.
+  * @hideinitializer */
+#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the bitrate in the encoder.
+  * Rates from 500 to 512000 bits per second are meaningful, as well as the
+  * special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
+  * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
+  * rate as it can, which is useful for controlling the rate by adjusting the
+  * output buffer size.
+  * @see OPUS_GET_BITRATE
+  * @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
+  *                                   is determined based on the number of
+  *                                   channels and the input sampling rate.
+  * @hideinitializer */
+#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
+/** Gets the encoder's bitrate configuration.
+  * @see OPUS_SET_BITRATE
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
+  *                                      The default is determined based on the
+  *                                      number of channels and the input
+  *                                      sampling rate.
+  * @hideinitializer */
+#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables variable bitrate (VBR) in the encoder.
+  * The configured bitrate may not be met exactly because frames must
+  * be an integer number of bytes in length.
+  * @see OPUS_GET_VBR
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
+  *               cause noticeable quality degradation.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
+  *               #OPUS_SET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
+/** Determine if variable bitrate (VBR) is enabled in the encoder.
+  * @see OPUS_SET_VBR
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
+  *               #OPUS_GET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables constrained VBR in the encoder.
+  * This setting is ignored when the encoder is in CBR mode.
+  * @warning Only the MDCT mode of Opus currently heeds the constraint.
+  *  Speech mode ignores it completely, hybrid mode may fail to obey it
+  *  if the LPC layer uses more bitrate than the constraint would have
+  *  permitted.
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @see OPUS_SET_VBR
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
+  *               frame of buffering delay assuming a transport with a
+  *               serialization speed of the nominal bitrate.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
+/** Determine if constrained VBR is enabled in the encoder.
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @see OPUS_GET_VBR
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default).</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures mono/stereo forcing in the encoder.
+  * This can force the encoder to produce packets encoded as either mono or
+  * stereo, regardless of the format of the input audio. This is useful when
+  * the caller knows that the input signal is currently a mono source embedded
+  * in a stereo stream.
+  * @see OPUS_GET_FORCE_CHANNELS
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
+/** Gets the encoder's forced channel configuration.
+  * @see OPUS_SET_FORCE_CHANNELS
+  * @param[out] x <tt>opus_int32 *</tt>:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the maximum bandpass that the encoder will select automatically.
+  * Applications should normally use this instead of #OPUS_SET_BANDWIDTH
+  * (leaving that set to the default, #OPUS_AUTO). This allows the
+  * application to set an upper bound based on the type of input it is
+  * providing, but still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_MAX_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Gets the encoder's configured maximum allowed bandpass.
+  * @see OPUS_SET_MAX_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Sets the encoder's bandpass to a specific value.
+  * This prevents the encoder from automatically selecting the bandpass based
+  * on the available bitrate. If an application knows the bandpass of the input
+  * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
+  * instead, which still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Configures the type of signal being encoded.
+  * This is a hint which helps the encoder's mode selection.
+  * @see OPUS_GET_SIGNAL
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal type.
+  * @see OPUS_SET_SIGNAL
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Configures the encoder's intended application.
+  * The initial value is a mandatory argument to the encoder_create function.
+  * @see OPUS_GET_APPLICATION
+  * @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured application.
+  * @see OPUS_SET_APPLICATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the total samples of delay added by the entire codec.
+  * This can be queried by the encoder and then the provided number of samples can be
+  * skipped on from the start of the decoder's output to provide time aligned input
+  * and output. From the perspective of a decoding application the real data begins this many
+  * samples late.
+  *
+  * The decoder contribution to this delay is identical for all decoders, but the
+  * encoder portion of the delay may vary from implementation to implementation,
+  * version to version, or even depend on the encoder's initial configuration.
+  * Applications needing delay compensation should call this CTL rather than
+  * hard-coding a value.
+  * @param[out] x <tt>opus_int32 *</tt>:   Number of lookahead samples
+  * @hideinitializer */
+#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of inband forward error correction (FEC).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_INBAND_FEC
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable inband FEC (default).</dd>
+  * <dt>1</dt><dd>Enable inband FEC.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of inband forward error correction.
+  * @see OPUS_SET_INBAND_FEC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Inband FEC disabled (default).</dd>
+  * <dt>1</dt><dd>Inband FEC enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's expected packet loss percentage.
+  * Higher values trigger progressively more loss resistant behavior in the encoder
+  * at the expense of quality at a given bitrate in the absence of packet loss, but
+  * greater quality under loss.
+  * @see OPUS_GET_PACKET_LOSS_PERC
+  * @param[in] x <tt>opus_int32</tt>:   Loss percentage in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured packet loss percentage.
+  * @see OPUS_SET_PACKET_LOSS_PERC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
+  *                                      in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of discontinuous transmission (DTX).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_DTX
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable DTX (default).</dd>
+  * <dt>1</dt><dd>Enabled DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of discontinuous transmission.
+  * @see OPUS_SET_DTX
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>DTX disabled (default).</dd>
+  * <dt>1</dt><dd>DTX enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
+/** Configures the depth of signal being encoded.
+  *
+  * This is a hint which helps the encoder identify silence and near-silence.
+  * It represents the number of significant bits of linear intensity below
+  * which the signal contains ignorable quantization or other noise.
+  *
+  * For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
+  * for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
+  * for 16-bit linear pcm input with opus_encode_float().
+  *
+  * When using opus_encode() instead of opus_encode_float(), or when libopus
+  * is compiled for fixed-point, the encoder uses the minimum of the value
+  * set here and the value 16.
+  *
+  * @see OPUS_GET_LSB_DEPTH
+  * @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
+  *                                   (default: 24).
+  * @hideinitializer */
+#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal depth.
+  * @see OPUS_SET_LSB_DEPTH
+  * @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
+  *                                      24 (default: 24).
+  * @hideinitializer */
+#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of variable duration frames.
+  * When variable duration is enabled, the encoder is free to use a shorter frame
+  * size than the one requested in the opus_encode*() call.
+  * It is then the user's responsibility
+  * to verify how much audio was encoded by checking the ToC byte of the encoded
+  * packet. The part of the audio that was not encoded needs to be resent to the
+  * encoder for the next call. Do not use this option unless you <b>really</b>
+  * know what you are doing.
+  * @see OPUS_GET_EXPERT_FRAME_DURATION
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured use of variable duration frames.
+  * @see OPUS_SET_EXPERT_FRAME_DURATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables almost all use of prediction, making frames almost
+  * completely independent. This reduces quality.
+  * @see OPUS_GET_PREDICTION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable prediction (default).</dd>
+  * <dt>1</dt><dd>Disable prediction.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured prediction status.
+  * @see OPUS_SET_PREDICTION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Prediction enabled (default).</dd>
+  * <dt>1</dt><dd>Prediction disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_genericctls Generic CTLs
+  *
+  * These macros are used with the \c opus_decoder_ctl and
+  * \c opus_encoder_ctl calls to generate a particular
+  * request.
+  *
+  * When called on an \c OpusDecoder they apply to that
+  * particular decoder instance. When called on an
+  * \c OpusEncoder they apply to the corresponding setting
+  * on that encoder instance, if present.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * opus_int32 pitch;
+  * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
+  * if (ret == OPUS_OK) return ret;
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
+  *
+  * opus_int32 enc_bw, dec_bw;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
+  * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
+  * if (enc_bw != dec_bw) {
+  *   printf("packet bandwidth mismatch!\n");
+  * }
+  * @endcode
+  *
+  * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
+  * @{
+  */
+
+/** Resets the codec state to be equivalent to a freshly initialized state.
+  * This should be called when switching streams in order to prevent
+  * the back to back decoding from giving different results from
+  * one at a time decoding.
+  * @hideinitializer */
+#define OPUS_RESET_STATE 4028
+
+/** Gets the final state of the codec's entropy coder.
+  * This is used for testing purposes,
+  * The encoder and decoder state should be identical after coding a payload
+  * (assuming no data corruption or software bugs)
+  *
+  * @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
+  *
+  * @hideinitializer */
+#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
+
+/** Gets the encoder's configured bandpass or the decoder's last bandpass.
+  * @see OPUS_SET_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the sampling rate the encoder or decoder was initialized with.
+  * This simply returns the <code>Fs</code> value passed to opus_encoder_init()
+  * or opus_decoder_init().
+  * @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
+  * @hideinitializer
+  */
+#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables the use of phase inversion for intensity stereo,
+  * improving the quality of mono downmixes, but slightly reducing normal
+  * stereo quality. Disabling phase inversion in the decoder does not comply
+  * with RFC 6716, although it does not cause any interoperability issue and
+  * is expected to become part of the Opus standard once RFC 6716 is updated
+  * by draft-ietf-codec-opus-update.
+  * @see OPUS_GET_PHASE_INVERSION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable phase inversion (default).</dd>
+  * <dt>1</dt><dd>Disable phase inversion.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PHASE_INVERSION_DISABLED(x) OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured phase inversion status.
+  * @see OPUS_SET_PHASE_INVERSION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Stereo phase inversion enabled (default).</dd>
+  * <dt>1</dt><dd>Stereo phase inversion disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+/** Gets the DTX state of the encoder.
+  * Returns whether the last encoded frame was either a comfort noise update
+  * during DTX or not encoded because of DTX.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>The encoder is not in DTX.</dd>
+  * <dt>1</dt><dd>The encoder is in DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_IN_DTX(x) OPUS_GET_IN_DTX_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_decoderctls Decoder related CTLs
+  * @see opus_genericctls, opus_encoderctls, opus_decoder
+  * @{
+  */
+
+/** Configures decoder gain adjustment.
+  * Scales the decoded output by a factor specified in Q8 dB units.
+  * This has a maximum range of -32768 to 32767 inclusive, and returns
+  * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
+  * This setting survives decoder reset.
+  *
+  * gain = pow(10, x/(20.0*256))
+  *
+  * @param[in] x <tt>opus_int32</tt>:   Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
+/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
+  * @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
+  * @hideinitializer */
+#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the pitch of the last decoded frame, if available.
+  * This can be used for any post-processing algorithm requiring the use of pitch,
+  * e.g. time stretching/shortening. If the last frame was not voiced, or if the
+  * pitch was not coded in the frame, then zero is returned.
+  *
+  * This CTL is only implemented for decoder instances.
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
+  *
+  * @hideinitializer */
+#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_libinfo Opus library information functions
+  * @{
+  */
+
+/** Converts an opus error code into a human readable string.
+  *
+  * @param[in] error <tt>int</tt>: Error number
+  * @returns Error string
+  */
+OPUS_EXPORT const char *opus_strerror(int error);
+
+/** Gets the libopus version string.
+  *
+  * Applications may look for the substring "-fixed" in the version string to
+  * determine whether they have a fixed-point or floating-point build at
+  * runtime.
+  *
+  * @returns Version string
+  */
+OPUS_EXPORT const char *opus_get_version_string(void);
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_DEFINES_H */

engine/voice_codecs/opus/opus/opus_multistream.h

@@ -0,0 +1,660 @@
+/* Copyright (c) 2011 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_multistream.h
+ * @brief Opus reference implementation multistream API
+ */
+
+#ifndef OPUS_MULTISTREAM_H
+#define OPUS_MULTISTREAM_H
+
+#include "opus.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** Macros to trigger compilation errors when the wrong types are provided to a
+  * CTL. */
+/**@{*/
+#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
+#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
+/**@}*/
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
+#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
+/**@}*/
+
+/** @endcond */
+
+/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
+  * @ref opus_decoderctls may be applied to a multistream encoder or decoder as
+  * well.
+  * In addition, you may retrieve the encoder or decoder state for an specific
+  * stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
+  * #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
+  */
+/**@{*/
+
+/** Gets the encoder state for an individual stream of a multistream encoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the encoder.
+  * @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
+  *                                       encoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
+
+/** Gets the decoder state for an individual stream of a multistream decoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the decoder.
+  * @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
+  *                                       decoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
+
+/**@}*/
+
+/** @defgroup opus_multistream Opus Multistream API
+  * @{
+  *
+  * The multistream API allows individual Opus streams to be combined into a
+  * single packet, enabling support for up to 255 channels. Unlike an
+  * elementary Opus stream, the encoder and decoder must negotiate the channel
+  * configuration before the decoder can successfully interpret the data in the
+  * packets produced by the encoder. Some basic information, such as packet
+  * duration, can be computed without any special negotiation.
+  *
+  * The format for multistream Opus packets is defined in
+  * <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
+  * and is based on the self-delimited Opus framing described in Appendix B of
+  * <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
+  * Normal Opus packets are just a degenerate case of multistream Opus packets,
+  * and can be encoded or decoded with the multistream API by setting
+  * <code>streams</code> to <code>1</code> when initializing the encoder or
+  * decoder.
+  *
+  * Multistream Opus streams can contain up to 255 elementary Opus streams.
+  * These may be either "uncoupled" or "coupled", indicating that the decoder
+  * is configured to decode them to either 1 or 2 channels, respectively.
+  * The streams are ordered so that all coupled streams appear at the
+  * beginning.
+  *
+  * A <code>mapping</code> table defines which decoded channel <code>i</code>
+  * should be used for each input/output (I/O) channel <code>j</code>. This table is
+  * typically provided as an unsigned char array.
+  * Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
+  * If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
+  * encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
+  * is even, or  as the right channel of stream <code>(i/2)</code> if
+  * <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
+  * mono in stream <code>(i - coupled_streams)</code>, unless it has the special
+  * value 255, in which case it is omitted from the encoding entirely (the
+  * decoder will reproduce it as silence). Each value <code>i</code> must either
+  * be the special value 255 or be less than <code>streams + coupled_streams</code>.
+  *
+  * The output channels specified by the encoder
+  * should use the
+  * <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
+  * channel ordering</a>. A decoder may wish to apply an additional permutation
+  * to the mapping the encoder used to achieve a different output channel
+  * order (e.g. for outputing in WAV order).
+  *
+  * Each multistream packet contains an Opus packet for each stream, and all of
+  * the Opus packets in a single multistream packet must have the same
+  * duration. Therefore the duration of a multistream packet can be extracted
+  * from the TOC sequence of the first stream, which is located at the
+  * beginning of the packet, just like an elementary Opus stream:
+  *
+  * @code
+  * int nb_samples;
+  * int nb_frames;
+  * nb_frames = opus_packet_get_nb_frames(data, len);
+  * if (nb_frames < 1)
+  *   return nb_frames;
+  * nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
+  * @endcode
+  *
+  * The general encoding and decoding process proceeds exactly the same as in
+  * the normal @ref opus_encoder and @ref opus_decoder APIs.
+  * See their documentation for an overview of how to use the corresponding
+  * multistream functions.
+  */
+
+/** Opus multistream encoder state.
+  * This contains the complete state of a multistream Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_init
+  */
+typedef struct OpusMSEncoder OpusMSEncoder;
+
+/** Opus multistream decoder state.
+  * This contains the complete state of a multistream Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_decoder_init
+  */
+typedef struct OpusMSDecoder OpusMSDecoder;
+
+/**\name Multistream encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusMSEncoder structure.
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
+      int channels,
+      int mapping_family
+);
+
+
+/** Allocates and initializes a multistream encoder state.
+  * Call opus_multistream_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
+
+/** Initialize a previously allocated multistream encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+OPUS_EXPORT int opus_multistream_surround_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6) OPUS_ARG_NONNULL(7);
+
+/** Encodes a multistream Opus frame.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
+    OpusMSEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a multistream Opus frame from floating point input.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
+      OpusMSEncoder *st,
+      const float *pcm,
+      int frame_size,
+      unsigned char *data,
+      opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusMSEncoder</code> allocated by
+  * opus_multistream_encoder_create().
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
+
+/** Perform a CTL function on a multistream Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+/**\name Multistream decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusMSDecoder</code> structure.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+/** Allocates and initializes a multistream decoder state.
+  * Call opus_multistream_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+/** Intialize a previously allocated decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_deocder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_decoder_init(
+      OpusMSDecoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+/** Decode a multistream Opus packet.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode a multistream Opus packet with floating point output.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on a multistream Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusMSDecoder</code> allocated by
+  * opus_multistream_decoder_create().
+  * @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_MULTISTREAM_H */

engine/voice_codecs/opus/opus/opus_projection.h

@@ -0,0 +1,568 @@
+/* Copyright (c) 2017 Google Inc.
+   Written by Andrew Allen */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_projection.h
+ * @brief Opus projection reference API
+ */
+
+#ifndef OPUS_PROJECTION_H
+#define OPUS_PROJECTION_H
+
+#include "opus_multistream.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.c
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST    6001
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST    6003
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST         6005
+/**@}*/
+
+
+/** @endcond */
+
+/** @defgroup opus_projection_ctls Projection specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_projection_encoder_ctl() and opus_projection_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls,
+  * @ref opus_decoderctls, and @ref opus_multistream_ctls may be applied to a
+  * projection encoder or decoder as well.
+  */
+/**@{*/
+
+/** Gets the gain (in dB. S7.8-format) of the demixing matrix from the encoder.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the gain (in dB. S7.8-format)
+  *                                      of the demixing matrix.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Gets the size in bytes of the demixing matrix from the encoder.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the size in bytes of the
+  *                                      demixing matrix.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Copies the demixing matrix to the supplied pointer location.
+  * @param[out] x <tt>unsigned char *</tt>: Returns the demixing matrix to the
+  *                                         supplied pointer location.
+  * @param y <tt>opus_int32</tt>: The size in bytes of the reserved memory at the
+  *                              pointer location.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX(x,y) OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST, x, __opus_check_int(y)
+
+
+/**@}*/
+
+/** Opus projection encoder state.
+ * This contains the complete state of a projection Opus encoder.
+ * It is position independent and can be freely copied.
+ * @see opus_projection_ambisonics_encoder_create
+ */
+typedef struct OpusProjectionEncoder OpusProjectionEncoder;
+
+
+/** Opus projection decoder state.
+  * This contains the complete state of a projection Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_projection_decoder_create
+  * @see opus_projection_decoder_init
+  */
+typedef struct OpusProjectionDecoder OpusProjectionDecoder;
+
+
+/**\name Projection encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusProjectionEncoder structure.
+  * @param channels <tt>int</tt>: The total number of input channels to encode.
+  *                               This must be no more than 255.
+  * @param mapping_family <tt>int</tt>: The mapping family to use for selecting
+  *                                     the appropriate projection.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_ambisonics_encoder_get_size(
+    int channels,
+    int mapping_family
+);
+
+
+/** Allocates and initializes a projection encoder state.
+  * Call opus_projection_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param mapping_family <tt>int</tt>: The mapping family to use for selecting
+  *                                     the appropriate projection.
+  * @param[out] streams <tt>int *</tt>: The total number of streams that will
+  *                                     be encoded from the input.
+  * @param[out] coupled_streams <tt>int *</tt>: Number of coupled (2 channel)
+  *                                 streams that will be encoded from the input.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionEncoder *opus_projection_ambisonics_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int mapping_family,
+    int *streams,
+    int *coupled_streams,
+    int application,
+    int *error
+) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5);
+
+
+/** Initialize a previously allocated projection encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_projection_ambisonics_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_projection_ambisonics_encoder_create
+  * @see opus_projection_ambisonics_encoder_get_size
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_projection_ambisonics_encoder_init(
+    OpusProjectionEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int mapping_family,
+    int *streams,
+    int *coupled_streams,
+    int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
+
+
+/** Encodes a projection Opus frame.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode(
+    OpusProjectionEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+
+/** Encodes a projection Opus frame from floating point input.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode_float(
+    OpusProjectionEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+
+/** Frees an <code>OpusProjectionEncoder</code> allocated by
+  * opus_projection_ambisonics_encoder_create().
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to be freed.
+  */
+OPUS_EXPORT void opus_projection_encoder_destroy(OpusProjectionEncoder *st);
+
+
+/** Perform a CTL function on a projection Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, @ref opus_multistream_ctls, or
+  *                @ref opus_projection_ctls
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  * @see opus_projection_ctls
+  */
+OPUS_EXPORT int opus_projection_encoder_ctl(OpusProjectionEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+/**@}*/
+
+/**\name Projection decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusProjectionDecoder</code> structure.
+  * @param channels <tt>int</tt>: The total number of output channels.
+  *                               This must be no more than 255.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_decoder_get_size(
+    int channels,
+    int streams,
+    int coupled_streams
+);
+
+
+/** Allocates and initializes a projection decoder state.
+  * Call opus_projection_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
+  *                         that mapping from coded channels to output channels,
+  *                         as described in @ref opus_projection and
+  *                         @ref opus_projection_ctls.
+  * @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
+  *                                                  demixing matrix, as
+  *                                                  described in @ref
+  *                                                  opus_projection_ctls.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionDecoder *opus_projection_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int streams,
+    int coupled_streams,
+    unsigned char *demixing_matrix,
+    opus_int32 demixing_matrix_size,
+    int *error
+) OPUS_ARG_NONNULL(5);
+
+
+/** Intialize a previously allocated projection decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_projection_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_projection_decoder_create
+  * @see opus_projection_deocder_get_size
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
+  *                         that mapping from coded channels to output channels,
+  *                         as described in @ref opus_projection and
+  *                         @ref opus_projection_ctls.
+  * @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
+  *                                                  demixing matrix, as
+  *                                                  described in @ref
+  *                                                  opus_projection_ctls.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_projection_decoder_init(
+    OpusProjectionDecoder *st,
+    opus_int32 Fs,
+    int channels,
+    int streams,
+    int coupled_streams,
+    unsigned char *demixing_matrix,
+    opus_int32 demixing_matrix_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+
+/** Decode a projection Opus packet.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode(
+    OpusProjectionDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+
+/** Decode a projection Opus packet with floating point output.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode_float(
+    OpusProjectionDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+
+/** Perform a CTL function on a projection Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, @ref opus_multistream_ctls, or
+  *                @ref opus_projection_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  * @see opus_projection_ctls
+  */
+OPUS_EXPORT int opus_projection_decoder_ctl(OpusProjectionDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+/** Frees an <code>OpusProjectionDecoder</code> allocated by
+  * opus_projection_decoder_create().
+  * @param st <tt>OpusProjectionDecoder</tt>: Projection decoder state to be freed.
+  */
+OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st);
+
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_PROJECTION_H */

engine/voice_codecs/opus/opus/opus_types.h

@@ -0,0 +1,166 @@
+/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
+/* Modified by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+/* opus_types.h based on ogg_types.h from libogg */
+
+/**
+   @file opus_types.h
+   @brief Opus reference implementation types
+*/
+#ifndef OPUS_TYPES_H
+#define OPUS_TYPES_H
+
+#define opus_int         int                     /* used for counters etc; at least 16 bits */
+#define opus_int64       long long
+#define opus_int8        signed char
+
+#define opus_uint        unsigned int            /* used for counters etc; at least 16 bits */
+#define opus_uint64      unsigned long long
+#define opus_uint8       unsigned char
+
+/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
+#if (defined(__STDC__) && __STDC__ && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
+#include <stdint.h>
+#  undef opus_int64
+#  undef opus_int8
+#  undef opus_uint64
+#  undef opus_uint8
+   typedef int8_t opus_int8;
+   typedef uint8_t opus_uint8;
+   typedef int16_t opus_int16;
+   typedef uint16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef uint32_t opus_uint32;
+   typedef int64_t opus_int64;
+   typedef uint64_t opus_uint64;
+#elif defined(_WIN32)
+
+#  if defined(__CYGWIN__)
+#    include <_G_config.h>
+     typedef _G_int32_t opus_int32;
+     typedef _G_uint32_t opus_uint32;
+     typedef _G_int16 opus_int16;
+     typedef _G_uint16 opus_uint16;
+#  elif defined(__MINGW32__)
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+#  elif defined(__MWERKS__)
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+#  else
+     /* MSVC/Borland */
+     typedef __int32 opus_int32;
+     typedef unsigned __int32 opus_uint32;
+     typedef __int16 opus_int16;
+     typedef unsigned __int16 opus_uint16;
+#  endif
+
+#elif defined(__MACOS__)
+
+#  include <sys/types.h>
+   typedef SInt16 opus_int16;
+   typedef UInt16 opus_uint16;
+   typedef SInt32 opus_int32;
+   typedef UInt32 opus_uint32;
+
+#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
+
+#  include <sys/types.h>
+   typedef int16_t opus_int16;
+   typedef u_int16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined(__BEOS__)
+
+   /* Be */
+#  include <inttypes.h>
+   typedef int16 opus_int16;
+   typedef u_int16 opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined (__EMX__)
+
+   /* OS/2 GCC */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined (DJGPP)
+
+   /* DJGPP */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(R5900)
+
+   /* PS2 EE */
+   typedef int opus_int32;
+   typedef unsigned opus_uint32;
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+
+#elif defined(__SYMBIAN32__)
+
+   /* Symbian GCC */
+   typedef signed short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef signed int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef long opus_int32;
+   typedef unsigned long opus_uint32;
+
+#elif defined(CONFIG_TI_C6X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#else
+
+   /* Give up, take a reasonable guess */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#endif
+
+#endif  /* OPUS_TYPES_H */

engine/voice_codecs/opus/voiceencoder_opus.cpp

@@ -0,0 +1,184 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $NoKeywords: $
+//
+//=============================================================================//
+#include "ivoicecodec.h"
+#include "iframeencoder.h"
+
+#include <stdio.h>
+#include <opus/opus.h>
+#include <opus/opus_custom.h>
+
+
+// NOTE: This has to be the last file included!
+#include "tier0/memdbgon.h"
+#include "tier0/dbg.h"
+
+#define CHANNELS 1
+
+struct opus_options
+{
+	int iSampleRate;
+	int iRawFrameSize;
+	int iPacketSize;
+};
+
+opus_options g_OpusOpts[] =
+{
+        {44100, 256, 120},
+        {22050, 120, 60},
+        {22050, 256, 60},
+        {22050, 512, 64},
+};
+
+class VoiceEncoder_Opus : public IFrameEncoder
+{
+public:
+	VoiceEncoder_Opus();
+	virtual ~VoiceEncoder_Opus();
+
+	// Interfaces IFrameDecoder
+
+	bool Init(int quality, int &rawFrameSize, int &encodedFrameSize);
+	void Release();
+	void DecodeFrame(const char *pCompressed, char *pDecompressedBytes);
+	void EncodeFrame(const char *pUncompressedBytes, char *pCompressed);
+	bool ResetState();
+
+private:
+
+	bool	InitStates();
+	void	TermStates();
+
+	OpusCustomEncoder *m_EncoderState;	// Celt internal encoder state
+	OpusCustomDecoder *m_DecoderState; // Celt internal decoder state
+	OpusCustomMode	*m_Mode;
+
+	int m_iVersion;
+};
+
+extern IVoiceCodec* CreateVoiceCodec_Frame(IFrameEncoder *pEncoder);
+
+void* CreateCeltVoiceCodec()
+{
+	IFrameEncoder *pEncoder = new VoiceEncoder_Opus;
+	return CreateVoiceCodec_Frame( pEncoder );
+}
+
+EXPOSE_INTERFACE_FN(CreateCeltVoiceCodec, IVoiceCodec, "vaudio_opus")
+
+//////////////////////////////////////////////////////////////////////
+// Construction/Destruction
+//////////////////////////////////////////////////////////////////////
+
+VoiceEncoder_Opus::VoiceEncoder_Opus()
+{
+	m_EncoderState = NULL;
+	m_DecoderState = NULL;
+	m_Mode = NULL;
+	m_iVersion = 0;
+}
+
+VoiceEncoder_Opus::~VoiceEncoder_Opus()
+{
+	TermStates();
+}
+
+bool VoiceEncoder_Opus::Init( int quality, int &rawFrameSize, int &encodedFrameSize)
+{
+	m_iVersion = quality;
+
+	rawFrameSize = g_OpusOpts[m_iVersion].iRawFrameSize * BYTES_PER_SAMPLE;
+
+	int iError = 0;
+
+	m_Mode = opus_custom_mode_create( g_OpusOpts[m_iVersion].iSampleRate, g_OpusOpts[m_iVersion].iRawFrameSize, &iError );
+
+	m_EncoderState = opus_custom_encoder_create( m_Mode, CHANNELS, NULL);
+	m_DecoderState = opus_custom_decoder_create( m_Mode, CHANNELS, NULL);
+
+	if ( !InitStates() )
+		return false;
+
+	encodedFrameSize = g_OpusOpts[m_iVersion].iPacketSize;
+
+	return true;
+}
+
+void VoiceEncoder_Opus::Release()
+{
+	delete this;
+}
+
+void VoiceEncoder_Opus::EncodeFrame(const char *pUncompressedBytes, char *pCompressed)
+{
+	unsigned char output[1024];
+
+	opus_custom_encode( m_EncoderState, (opus_int16*)pUncompressedBytes, g_OpusOpts[m_iVersion].iRawFrameSize, output, g_OpusOpts[m_iVersion].iPacketSize );
+
+	for ( int i = 0; i < g_OpusOpts[m_iVersion].iPacketSize; i++ )
+	{
+		*pCompressed = (char)output[i];
+		pCompressed++;
+	}
+}
+
+void VoiceEncoder_Opus::DecodeFrame(const char *pCompressed, char *pDecompressedBytes)
+{
+	unsigned char output[1024];
+	char *out = (char *)pCompressed;
+
+	if ( !pCompressed )
+	{
+		int decoded = opus_custom_decode( m_DecoderState, NULL, g_OpusOpts[m_iVersion].iPacketSize, (opus_int16 *)pDecompressedBytes, g_OpusOpts[m_iVersion].iRawFrameSize );
+		return;
+	}
+
+	for ( int i = 0; i < g_OpusOpts[m_iVersion].iPacketSize; i++ )
+	{
+		output[i] = ( unsigned char ) ( ( *out < 0 ) ? (*out + 256) : *out );
+		out++;
+	}
+
+	//celt_decoder_ctl( m_DecoderState, CELT_RESET_STATE_REQUEST, NULL );
+	int decoded = opus_custom_decode( m_DecoderState, output, g_OpusOpts[m_iVersion].iPacketSize, (opus_int16 *)pDecompressedBytes, g_OpusOpts[m_iVersion].iRawFrameSize );
+}
+
+bool VoiceEncoder_Opus::ResetState()
+{
+	opus_custom_encoder_ctl(m_EncoderState, OPUS_RESET_STATE );
+	opus_custom_decoder_ctl(m_DecoderState, OPUS_RESET_STATE );
+
+	return true;
+}
+
+bool VoiceEncoder_Opus::InitStates()
+{
+	if ( !m_EncoderState || !m_DecoderState )
+		return false;
+
+	opus_custom_encoder_ctl(m_EncoderState, OPUS_RESET_STATE );
+	opus_custom_decoder_ctl(m_DecoderState, OPUS_RESET_STATE );
+
+	return true;
+}
+
+void VoiceEncoder_Opus::TermStates()
+{
+	if( m_EncoderState )
+	{
+		opus_custom_encoder_destroy( m_EncoderState );
+		m_EncoderState = NULL;
+	}
+
+	if( m_DecoderState )
+	{
+		opus_custom_decoder_destroy( m_DecoderState );
+		m_DecoderState = NULL;
+	}
+
+	opus_custom_mode_destroy( m_Mode );
+}

engine/voice_codecs/opus/wscript

@@ -0,0 +1,53 @@
+#! /usr/bin/env python
+# encoding: utf-8
+
+from waflib import Utils
+import os
+
+top = '.'
+PROJECT_NAME = 'vaudio_opus'
+
+def options(opt):
+	return
+
+def configure(conf):
+	conf.define('OPUS_EXPORTS',1)
+
+def build(bld):
+	source = [
+		'voiceencoder_opus.cpp',
+		'../frame_encoder/voice_codec_frame.cpp',
+		'../../../tier1/interface.cpp'
+	]
+
+	includes = [
+		'.',
+		'../frame_encoder',
+		'../../../public',
+		'../../../public/tier1',
+		'../../../public/tier0',
+		'../../',
+		'../../../common',
+		'../../audio/public',
+		'celt'
+	]
+
+	defines = []
+
+	libs = ['tier0','tier1','vstdlib','OPUS']
+
+	install_path = bld.env.LIBDIR
+
+	bld.shlib(
+		source   = source,
+		target   = PROJECT_NAME,
+		name     = PROJECT_NAME,
+		features = 'c cxx',
+		includes = includes,
+		defines  = defines,
+		use      = libs,
+		install_path = install_path,
+		subsystem = bld.env.MSVC_SUBSYSTEM,
+		idx      = bld.get_taskgen_count()
+	)
+

engine/voice_codecs/speex/VoiceEncoder_Speex.h

@@ -45,7 +45,7 @@ damage. */
 
 #include "iframeencoder.h"
 
-#include <speex.h>
+#include <speex/speex.h>
 
 class VoiceEncoder_Speex : public IFrameEncoder
 {

engine/voice_codecs/speex/source/AUTHORS

@@ -1,15 +0,0 @@
-Jean-Marc Valin <jean-marc.valin@hermes.usherb.ca>
-   All the code except the following
-
-David Rowe <david@voicetronix.com.au> via VoiceTronix
-   lsp.c lsp.h 
-   Also ideas and feedback
-
-John Francis Edwards:
-   wave_out.[ch], some #ifdefs for windows port and MSVC project files
-
-Atsuhiko Yamanaka <ymnk@jcraft.com>:
-   Patch to speexenc.c to add Vorbis comment format
-
-Radim Kolar <hsn@cybermail.net>:
-   Patch to speexenc.c for supporting more input formats

engine/voice_codecs/speex/source/COPYING

@@ -1,26 +0,0 @@
-Redistribution and use in source and binary forms, with or without
-modification, are permitted provided that the following conditions
-are met:
-
-- Redistributions of source code must retain the above copyright
-notice, this list of conditions and the following disclaimer.
-
-- Redistributions in binary form must reproduce the above copyright
-notice, this list of conditions and the following disclaimer in the
-documentation and/or other materials provided with the distribution.
-
-- Neither the name of the Xiph.org Foundation nor the names of its
-contributors may be used to endorse or promote products derived from
-this software without specific prior written permission.
-
-THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

engine/voice_codecs/speex/source/ChangeLog

@@ -1,6 +0,0 @@
-2002/03/27 Jean-Marc Valin:
-Working encoder and decoder for both narrowband and wideband.
-
-2002/02/27 Jean-Marc Valin:
-Got the basic encoder working as a demo with quantization only on some 
-parameters.

engine/voice_codecs/speex/source/INSTALL

@@ -1,8 +0,0 @@
-Installing Speex is as easy as:
-
-% ./configure [--prefix=<install-path>]
-% make
-% make install
-
-Note that if you are using the code from CVS, you will need to run "autogen.sh"
-instead of "configure".

engine/voice_codecs/speex/source/Makefile.am

@@ -1,11 +0,0 @@
-## Process this file with automake to produce Makefile.in. -*-Makefile-*-
-
-# Disable automatic dependency tracking if using other tools than gcc and gmake
-#AUTOMAKE_OPTIONS = no-dependencies
-
-EXTRA_DIST = Speex.spec Speex.spec.in
-
-SUBDIRS = libspeex @src@ doc win32
-
-rpm: dist
-	rpmbuild -ta ${PACKAGE}-${VERSION}.tar.gz

engine/voice_codecs/speex/source/Makefile.in

@@ -1,347 +0,0 @@
-# Makefile.in generated automatically by automake 1.4-p6 from Makefile.am
-
-# Copyright (C) 1994, 1995-8, 1999, 2001 Free Software Foundation, Inc.
-# This Makefile.in is free software; the Free Software Foundation
-# gives unlimited permission to copy and/or distribute it,
-# with or without modifications, as long as this notice is preserved.
-
-# This program is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
-# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
-# PARTICULAR PURPOSE.
-
-# Disable automatic dependency tracking if using other tools than gcc and gmake
-#AUTOMAKE_OPTIONS = no-dependencies
-
-
-SHELL = @SHELL@
-
-srcdir = @srcdir@
-top_srcdir = @top_srcdir@
-VPATH = @srcdir@
-prefix = @prefix@
-exec_prefix = @exec_prefix@
-
-bindir = @bindir@
-sbindir = @sbindir@
-libexecdir = @libexecdir@
-datadir = @datadir@
-sysconfdir = @sysconfdir@
-sharedstatedir = @sharedstatedir@
-localstatedir = @localstatedir@
-libdir = @libdir@
-infodir = @infodir@
-mandir = @mandir@
-includedir = @includedir@
-oldincludedir = /usr/include
-
-DESTDIR =
-
-pkgdatadir = $(datadir)/@PACKAGE@
-pkglibdir = $(libdir)/@PACKAGE@
-pkgincludedir = $(includedir)/@PACKAGE@
-
-top_builddir = .
-
-ACLOCAL = @ACLOCAL@
-AUTOCONF = @AUTOCONF@
-AUTOMAKE = @AUTOMAKE@
-AUTOHEADER = @AUTOHEADER@
-
-INSTALL = @INSTALL@
-INSTALL_PROGRAM = @INSTALL_PROGRAM@ $(AM_INSTALL_PROGRAM_FLAGS)
-INSTALL_DATA = @INSTALL_DATA@
-INSTALL_SCRIPT = @INSTALL_SCRIPT@
-transform = @program_transform_name@
-
-NORMAL_INSTALL = :
-PRE_INSTALL = :
-POST_INSTALL = :
-NORMAL_UNINSTALL = :
-PRE_UNINSTALL = :
-POST_UNINSTALL = :
-host_alias = @host_alias@
-host_triplet = @host@
-AS = @AS@
-CC = @CC@
-DLLTOOL = @DLLTOOL@
-ECHO = @ECHO@
-EXEEXT = @EXEEXT@
-LIBTOOL = @LIBTOOL@
-LN_S = @LN_S@
-MAINT = @MAINT@
-MAKEINFO = @MAKEINFO@
-OBJDUMP = @OBJDUMP@
-OBJEXT = @OBJEXT@
-OGG_INCLUDES = @OGG_INCLUDES@
-OGG_LDFLAGS = @OGG_LDFLAGS@
-OGG_LIBS = @OGG_LIBS@
-PACKAGE = @PACKAGE@
-RANLIB = @RANLIB@
-SPEEX_LT_AGE = @SPEEX_LT_AGE@
-SPEEX_LT_CURRENT = @SPEEX_LT_CURRENT@
-SPEEX_LT_REVISION = @SPEEX_LT_REVISION@
-STRIP = @STRIP@
-VERSION = @VERSION@
-src = @src@
-
-EXTRA_DIST = Speex.spec Speex.spec.in
-
-SUBDIRS = libspeex @src@ doc win32
-ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
-mkinstalldirs = $(SHELL) $(top_srcdir)/mkinstalldirs
-CONFIG_CLEAN_FILES =  Speex.spec
-DIST_COMMON =  README AUTHORS COPYING ChangeLog INSTALL Makefile.am \
-Makefile.in NEWS Speex.spec.in TODO acinclude.m4 aclocal.m4 \
-config.guess config.sub configure configure.in install-sh ltmain.sh \
-missing mkinstalldirs
-
-
-DISTFILES = $(DIST_COMMON) $(SOURCES) $(HEADERS) $(TEXINFOS) $(EXTRA_DIST)
-
-TAR = gtar
-GZIP_ENV = --best
-all: all-redirect
-.SUFFIXES:
-$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ Makefile.am $(top_srcdir)/configure.in $(ACLOCAL_M4) 
-	cd $(top_srcdir) && $(AUTOMAKE) --gnu --include-deps Makefile
-
-Makefile: $(srcdir)/Makefile.in  $(top_builddir)/config.status
-	cd $(top_builddir) \
-	  && CONFIG_FILES=$@ CONFIG_HEADERS= $(SHELL) ./config.status
-
-$(ACLOCAL_M4): @MAINTAINER_MODE_TRUE@ configure.in  acinclude.m4
-	cd $(srcdir) && $(ACLOCAL)
-
-config.status: $(srcdir)/configure $(CONFIG_STATUS_DEPENDENCIES)
-	$(SHELL) ./config.status --recheck
-$(srcdir)/configure: @MAINTAINER_MODE_TRUE@$(srcdir)/configure.in $(ACLOCAL_M4) $(CONFIGURE_DEPENDENCIES)
-	cd $(srcdir) && $(AUTOCONF)
-Speex.spec: $(top_builddir)/config.status Speex.spec.in
-	cd $(top_builddir) && CONFIG_FILES=$@ CONFIG_HEADERS= $(SHELL) ./config.status
-
-# This directory's subdirectories are mostly independent; you can cd
-# into them and run `make' without going through this Makefile.
-# To change the values of `make' variables: instead of editing Makefiles,
-# (1) if the variable is set in `config.status', edit `config.status'
-#     (which will cause the Makefiles to be regenerated when you run `make');
-# (2) otherwise, pass the desired values on the `make' command line.
-
-@SET_MAKE@
-
-all-recursive install-data-recursive install-exec-recursive \
-installdirs-recursive install-recursive uninstall-recursive  \
-check-recursive installcheck-recursive info-recursive dvi-recursive:
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-	dot_seen=no; \
-	target=`echo $@ | sed s/-recursive//`; \
-	list='$(SUBDIRS)'; for subdir in $$list; do \
-	  echo "Making $$target in $$subdir"; \
-	  if test "$$subdir" = "."; then \
-	    dot_seen=yes; \
-	    local_target="$$target-am"; \
-	  else \
-	    local_target="$$target"; \
-	  fi; \
-	  (cd $$subdir && $(MAKE) $(AM_MAKEFLAGS) $$local_target) \
-	   || case "$$amf" in *=*) exit 1;; *k*) fail=yes;; *) exit 1;; esac; \
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-	if test "$$dot_seen" = "no"; then \
-	  $(MAKE) $(AM_MAKEFLAGS) "$$target-am" || exit 1; \
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engine/voice_codecs/speex/source/NEWS

@@ -1 +0,0 @@
-2002/02/13: Creation of the "Speex" project

engine/voice_codecs/speex/source/README

@@ -1,9 +0,0 @@
-See INSTALL file for instruction on how to install Speex.
-
-The Speex is a patent-free, Open Source/Free Software voice codec. Unlike other codecs like MP3 and Ogg Vorbis, Speex is designed to compress voice at bitrates in the 2-45 kbps range. Possible applications include VoIP, internet audio streaming, archiving of speech data (e.g. voice mail), and audio books. In some sense, it is meant to be complementary to the Ogg Vorbis codec.
-
-To use the Speex command line tools:
-
-% speexenc [options] input_file.wav compressed_file.spx
-
-% speexdec [options] compressed_file.spx output_file.wav

engine/voice_codecs/speex/source/Speex.spec

@@ -1,68 +0,0 @@
-%define name     speex
-%define ver      1.0.1
-%define rel      1
-
-Summary: An open-source, patent-free speech codec
-Name: %name
-Version: %ver
-Release: %rel
-Copyright: BSD
-Group: Application/Devel
-Source: http://www.speex.org/download/%{name}-%{ver}.tar.gz
-URL: http://www.speex.org/
-Vendor: Speex
-Packager: Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca)
-BuildRoot: /var/tmp/%{name}-build-root
-Docdir: /usr/share/doc
-
-%description
-Speex is a patent-free audio codec designed especially for voice (unlike 
-Vorbis which targets general audio) signals and providing good narrowband 
-and wideband quality. This project aims to be complementary to the Vorbis
-codec.
-
-%package devel
-Summary:	Speex development files
-Group:		Development/Libraries
-Requires:	%{name} = %{version}
-
-%description devel
-Speex development files.
-
-%changelog
-* Thu Oct 03 2002 Jean-Marc Valin 
-- Added devel package inspired from PLD spec file
-
-* Tue Jul 30 2002 Fredrik Rambris <boost@users.sourceforge.net> 0.5.2
-- Added buildroot and docdir and ldconfig. Makes it builadble by non-roots
-  and also doesn't write to actual library paths when building.
-
-%prep
-%setup
-
-%build
-export CFLAGS='-O3 -DRELEASE'
-./configure --prefix=/usr --enable-shared --enable-static
-make
-
-%install
-rm -rf $RPM_BUILD_ROOT
-make DESTDIR=$RPM_BUILD_ROOT install
-
-%post -p /sbin/ldconfig
-%postun -p /sbin/ldconfig
-
-%files
-%defattr(644,root,root,755)
-%doc COPYING AUTHORS ChangeLog NEWS README
-%doc doc/manual.pdf
-/usr/share/man/man1/speexenc.1*
-/usr/share/man/man1/speexdec.1*
-%attr(755,root,root) %{_bindir}/speex*
-%attr(755,root,root) %{_libdir}/libspeex*.so*
-
-%files devel
-%defattr(644,root,root,755)
-%attr(755,root,root) %{_libdir}/libspeex*.la
-%{_includedir}/speex*.h
-%{_libdir}/libspeex*.a

engine/voice_codecs/speex/source/Speex.spec.in

@@ -1,68 +0,0 @@
-%define name     @PACKAGE@
-%define ver      @VERSION@
-%define rel      1
-
-Summary: An open-source, patent-free speech codec
-Name: %name
-Version: %ver
-Release: %rel
-Copyright: BSD
-Group: Application/Devel
-Source: http://www.speex.org/download/%{name}-%{ver}.tar.gz
-URL: http://www.speex.org/
-Vendor: Speex
-Packager: Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca)
-BuildRoot: /var/tmp/%{name}-build-root
-Docdir: /usr/share/doc
-
-%description
-Speex is a patent-free audio codec designed especially for voice (unlike 
-Vorbis which targets general audio) signals and providing good narrowband 
-and wideband quality. This project aims to be complementary to the Vorbis
-codec.
-
-%package devel
-Summary:	Speex development files
-Group:		Development/Libraries
-Requires:	%{name} = %{version}
-
-%description devel
-Speex development files.
-
-%changelog
-* Thu Oct 03 2002 Jean-Marc Valin 
-- Added devel package inspired from PLD spec file
-
-* Tue Jul 30 2002 Fredrik Rambris <boost@users.sourceforge.net> 0.5.2
-- Added buildroot and docdir and ldconfig. Makes it builadble by non-roots
-  and also doesn't write to actual library paths when building.
-
-%prep
-%setup
-
-%build
-export CFLAGS='-O3 -DRELEASE'
-./configure --prefix=/usr --enable-shared --enable-static
-make
-
-%install
-rm -rf $RPM_BUILD_ROOT
-make DESTDIR=$RPM_BUILD_ROOT install
-
-%post -p /sbin/ldconfig
-%postun -p /sbin/ldconfig
-
-%files
-%defattr(644,root,root,755)
-%doc COPYING AUTHORS ChangeLog NEWS README
-%doc doc/manual.pdf
-/usr/share/man/man1/speexenc.1*
-/usr/share/man/man1/speexdec.1*
-%attr(755,root,root) %{_bindir}/speex*
-%attr(755,root,root) %{_libdir}/libspeex*.so*
-
-%files devel
-%defattr(644,root,root,755)
-%attr(755,root,root) %{_libdir}/libspeex*.la
-%{_includedir}/speex*.h
-%{_libdir}/libspeex*.a

engine/voice_codecs/speex/source/TODO

@@ -1,18 +0,0 @@
-Features
--Add maximum/minimum bit-rate control for VBR
--Get the encoder to use the rate of packet loss (more conservative pitch gains)
-
-Long-term quality improvements
--Improve perceptual enhancement (including wideband)
-
-Standards
-*Complete Speex RTP profile
--MIME type registration
--MS ACM wrapper
-
-*required for 1.0
-
-ideas:
-Peelable stream (double codebook, higher bands, stereo)
-LPC from spectral domain
-Better psycho-acoustics? Masking curve from Vorbis

engine/voice_codecs/speex/source/acinclude.m4

@@ -1,84 +0,0 @@
-AC_DEFUN(AC_FIND_FILE,
-[
-$3=NONE
-for i in $2;
-do
-  for j in $1;
-  do
-    if test -r "$i/$j"; then
-      $3=$i
-      break 2
-    fi
-  done
-done
-])
-
-AC_DEFUN(AC_PATH_LIBOGG,
-[
-OGG_LIBS="-logg"
-
-AC_MSG_CHECKING([for libogg])
-
-ac_ogg_includes=NONE ac_ogg_libraries=NONE ac_ogg_bindir=NONE
-ogg_libraries=""
-ogg_includes=""
-AC_ARG_WITH(ogg-dir,
-    [  --with-ogg-dir=DIR       where the root of OGG is installed ],
-    [  ac_ogg_includes="$withval"/include
-       ac_ogg_libraries="$withval"/lib
-    ])
-
-AC_ARG_WITH(ogg-includes,
-    [  --with-ogg-includes=DIR  where the OGG includes are. ],
-    [  
-       ac_ogg_includes="$withval"
-    ])
-    
-ogg_libs_given=no
-
-AC_ARG_WITH(ogg-libraries,
-    [  --with-ogg-libraries=DIR where the OGG library is installed.],
-    [  ac_ogg_libraries="$withval"
-       ogg_libs_given=yes
-    ])
-
-ogg_incdirs="/usr/include /usr/lib/ogg/include /opt/include /usr/local/ogg/include /usr/include/ogg /usr/include /usr/local/include"
-if test ! "$ac_ogg_includes" = "NONE"; then
-  ogg_incdirs="$ac_ogg_includes $ac_ogg_includes/.. $ogg_incdirs"
-fi
-AC_FIND_FILE(ogg/ogg.h, $ogg_incdirs, ogg_incdir)
-echo "Ogg includes in $ogg_incdir"
-
-
-ogg_libdirs="$ac_ogg_libraries /usr/lib/ogg/lib /usr/lib /opt/lib /usr/local/ogg/lib /usr/local/lib /usr/lib/ogg /usr/local/lib"
-test -n "$OGGDIR" && ogg_libdirs="$OGGDIR/lib $OGGDIR $ogg_libdirs"
-if test ! "$ac_ogg_libraries" = "NONE"; then
-  ogg_libdirs="$ac_ogg_libraries $ogg_libdirs"
-fi
-
-test=NONE
-ogg_libdir=NONE
-for dir in $ogg_libdirs; do
-  try="ls -1 $dir/libogg*"
-  if test=`eval $try 2> /dev/null`; then ogg_libdir=$dir; break; else echo "tried $dir" >&AC_FD_CC ; fi
-done
-
-echo "Ogg libraries in $ogg_libdir"
-
-if test "$ogg_libdir" = "NONE" || test "$ogg_incdir" = "NONE"; then
-   have_libogg=no
-else
-   have_libogg=yes
-   AC_DEFINE(HAVE_LIBOGG)
-fi
-
-OGG_INCLUDES="-I$ogg_incdir"
-OGG_LDFLAGS="-L$ogg_libdir"
-
-
-AC_SUBST(OGG_LIBS)
-AC_SUBST(OGG_INCLUDES)
-AC_SUBST(OGG_LDFLAGS)
-
-])
-

engine/voice_codecs/speex/source/configure.in

@@ -1,69 +0,0 @@
-dnl Process this file with autoconf to produce a configure script. -*-m4-*-
-
-AC_INIT(libspeex/speex.h)
-
-SPEEX_MAJOR_VERSION=1
-SPEEX_MINOR_VERSION=0
-SPEEX_MICRO_VERSION=1
-SPEEX_VERSION=1.0.1
-
-SPEEX_LT_CURRENT=2
-SPEEX_LT_REVISION=0
-SPEEX_LT_AGE=1
-
-AC_SUBST(SPEEX_LT_CURRENT)
-AC_SUBST(SPEEX_LT_REVISION)
-AC_SUBST(SPEEX_LT_AGE)
-
-# For automake.
-VERSION=$SPEEX_VERSION
-PACKAGE=speex
-
-AM_INIT_AUTOMAKE($PACKAGE, $VERSION, no-define)
-AM_MAINTAINER_MODE
-
-AC_CANONICAL_HOST
-AM_PROG_LIBTOOL
-
-AC_C_BIGENDIAN
-AC_CHECK_HEADERS(sys/soundcard.h)
-
-AC_ARG_ENABLE(ogg,
-      [  --enable-ogg=[yes/no]    Turn on or off the use of ogg
-libraries [default=yes]],
-      [case "${enableval}" in
-        yes) useogg=true ;;
-        no)  useogg=false ;;
-        *) AC_MSG_ERROR(bad value ${enableval} for --enable-ogg) ;;
-      esac],[useogg=true])
-
-if test x$useogg = xtrue; then
-AC_PATH_LIBOGG
-fi
-
-if test "$have_libogg" = yes; then
-   src=src
-else
-   src=
-fi
-AC_SUBST(src)
-
-AC_CHECK_LIB(m, sin)
-AC_CHECK_LIB(gnugetopt, getopt_long)
-
-AC_DEFINE_UNQUOTED(VERSION, "${VERSION}")
-
-AC_ARG_ENABLE(sse, [  --enable-sse             enable SSE support], [if test "$enableval" = yes; then AC_DEFINE(_USE_SSE) fi])
-
-dnl Output the makefiles and version.h.
-
-AC_OUTPUT([Makefile libspeex/Makefile src/Makefile doc/Makefile Speex.spec
-           win32/Makefile win32/libspeex/Makefile win32/speexenc/Makefile
-           win32/speexdec/Makefile ])
-
-if test "x$src" = "x"; then 
-echo "You don't seem to have libogg installed. Only the Speex library (libspeex) will be built (no encoder/decoder executable)"
-echo "You can download libogg from http://www.ogg.org/ogg/index.html"
-fi
-
-echo "Type \"make; make install\" to compile and install Speex";