Fix : issue with playback AMR sound, ticket #150;

git-svn-id: https://mpc-hc.svn.sourceforge.net/svnroot/mpc-hc/trunk@2225 10f7b99b-c216-0410-bff0-8a66a9350fd8

2 files changed, 117 insertions, 76 deletions

diff --git a/src/filters/transform/MpaDecFilter/MpaDecFilter.cpp b/src/filters/transform/MpaDecFilter/MpaDecFilter.cpp
index 2f2947c70..f51fb7895 100644
--- a/src/filters/transform/MpaDecFilter/MpaDecFilter.cpp
+++ b/src/filters/transform/MpaDecFilter/MpaDecFilter.cpp
@@ -303,6 +303,9 @@ CMpaDecFilter::CMpaDecFilter(LPUNKNOWN lpunk, HRESULT* phr)
 	m_pPCMData					= NULL;

 	memset (&m_flac, 0, sizeof(m_flac));

 

+	m_pFFBuffer					= NULL;

+	m_nFFBufferSize				= 0;

+

 	CRegKey key;

 	if(ERROR_SUCCESS == key.Open(HKEY_CURRENT_USER, _T("Software\\Gabest\\Filters\\MPEG Audio Decoder"), KEY_READ))

 	{

@@ -321,6 +324,9 @@ CMpaDecFilter::CMpaDecFilter(LPUNKNOWN lpunk, HRESULT* phr)
 

 CMpaDecFilter::~CMpaDecFilter()

 {

+	if (m_pFFBuffer) free(m_pFFBuffer);

+	m_nFFBufferSize	= 0;

+

 	/*

 	CRegKey key;

 	if(ERROR_SUCCESS == key.Create(HKEY_CURRENT_USER, _T("Software\\Gabest\\Filters\\MPEG Audio Decoder")))

@@ -429,16 +435,10 @@ HRESULT CMpaDecFilter::Receive(IMediaSample* pIn)
 	BOOL bNoJitterControl = false;

 	if(subtype == MEDIASUBTYPE_AMR || subtype == MEDIASUBTYPE_SAMR || subtype == MEDIASUBTYPE_SAWB)

 	{

-		//The reason I add this to disable Jitter control for AMR audio stream

-		//is because it seems a lot AMR audio stream haven't report correct output sample rate (my guess)

-		//so if there is jitter control, audio and video will be going very faster than it should (result)

-		//therefor I disable jitter control.

-		//but this is NOT a PERFECT solution. hope more input from other brilliant developers

-		//discuss is welcome  -- tomasen@gmail.com

 		bNoJitterControl = true;

 	}

 

-	if(SUCCEEDED(hr) && abs((int)(m_rtStart - rtStart)) > 1000000 && !bNoJitterControl) // +-100ms jitter is allowed for now

+	if(SUCCEEDED(hr) && _abs64((m_rtStart - rtStart)) > 1000000i64  && !bNoJitterControl) // +-100ms jitter is allowed for now

 	{

 		m_buff.RemoveAll();

 		m_rtStart = rtStart;

@@ -1507,9 +1507,9 @@ HRESULT CMpaDecFilter::Deliver(CAtlArray<float>& pBuff, DWORD nSamplesPerSec, WO
 

 	pOut->SetActualDataLength(pBuff.GetCount()*wfe->wBitsPerSample/8);

 

-WAVEFORMATEX* wfeout = (WAVEFORMATEX*)m_pOutput->CurrentMediaType().Format();

-ASSERT(wfeout->nChannels == wfe->nChannels);

-ASSERT(wfeout->nSamplesPerSec == wfe->nSamplesPerSec);

+	WAVEFORMATEX* wfeout = (WAVEFORMATEX*)m_pOutput->CurrentMediaType().Format();

+	ASSERT(wfeout->nChannels == wfe->nChannels);

+	ASSERT(wfeout->nSamplesPerSec == wfe->nSamplesPerSec);

 

 	float* pDataIn = pBuff.GetData();

 

@@ -2379,75 +2379,113 @@ HRESULT CMpaDecFilter::DeliverFfmpeg(int nCodecId, BYTE* p, int buffsize, int& s
 HRESULT CMpaDecFilter::DeliverFfmpeg(int nCodecId, BYTE* p, int buffsize, int& size)

 {

 	HRESULT		hr			= S_OK;

-	int			nPCMLength	= AVCODEC_MAX_AUDIO_FRAME_SIZE;

-	

+	int			nPCMLength	= 0;

 	if (!m_pAVCtx || nCodecId != m_pAVCtx->codec_id)

-		if (!InitFfmpeg (nCodecId))

-		{

-			size = 0;

-			return E_FAIL;

-		}

-

-	size = avcodec_decode_audio2(m_pAVCtx, (int16_t*)m_pPCMData, &nPCMLength, (const uint8_t*)p, buffsize);

-	size = min (size, buffsize);

-

-	if (size>0 && nPCMLength>0)

+	if (!InitFfmpeg (nCodecId))

 	{

-		WAVEFORMATEX*		wfein = (WAVEFORMATEX*)m_pInput->CurrentMediaType().Format();

-		UNUSED_ALWAYS(wfein);

-		CAtlArray<float>	pBuff;

-		int					iSpeakerConfig;

-		int					nRemap;

-		float*				pDataOut;

-                

-		scmap_t* scmap;

-		

-		switch (nCodecId)

-		{

-		case CODEC_ID_EAC3 :

-			scmap = &m_ffmpeg_ac3[FFGetChannelMap(m_pAVCtx)];

-			break;

-		default :

-			scmap = &m_scmap_default[m_pAVCtx->channels-1];

-			break;

-		}

-

-		switch (m_pAVCtx->sample_fmt)

-		{

-		case SAMPLE_FMT_S16 :

-			pBuff.SetCount (nPCMLength / 2);

-			pDataOut = pBuff.GetData();

-

-			for (size_t i=0; i<pBuff.GetCount()/m_pAVCtx->channels; i++)

-			{

-				for(int ch=0; ch<m_pAVCtx->channels; ch++)

-				{

-					*pDataOut = (float)((int16_t*)m_pPCMData) [scmap->ch[ch]+i*m_pAVCtx->channels] / SHRT_MAX;

-					pDataOut++;

-				}

-			}

-			break;

-

-		case SAMPLE_FMT_S32 :

-			pBuff.SetCount (nPCMLength / 4);

-			pDataOut = pBuff.GetData();

-

-			for (size_t i=0; i<pBuff.GetCount()/m_pAVCtx->channels; i++)

-			{

-				for(int ch=0; ch<m_pAVCtx->channels; ch++)

-				{

-					*pDataOut = (float)((int32_t*)m_pPCMData) [scmap->ch[ch]+i*m_pAVCtx->channels] / INT_MAX;

-					pDataOut++;

-				}

-			}

-			break;

-		default :

-			ASSERT(FALSE);

-			break;

-		}

-		hr = Deliver(pBuff, m_pAVCtx->sample_rate, scmap->nChannels, scmap->dwChannelMask);

+		size = 0;

+		return E_FAIL;

 	}

-

+	BYTE* pDataInBuff = p;

+	CAtlArray<float>	pBuffOut;

+	scmap_t* scmap = NULL; 

+	while (buffsize > 0)

+	{

+        nPCMLength	= AVCODEC_MAX_AUDIO_FRAME_SIZE;

+        if (buffsize+FF_INPUT_BUFFER_PADDING_SIZE > m_nFFBufferSize)

+        {

+            m_nFFBufferSize = buffsize+FF_INPUT_BUFFER_PADDING_SIZE;

+            m_pFFBuffer		= (BYTE*)realloc(m_pFFBuffer, m_nFFBufferSize);

+            

+        }

+

+        // Required number of additionally allocated bytes at the end of the input bitstream for decoding.

+        // This is mainly needed because some optimized bitstream readers read

+        // 32 or 64 bit at once and could read over the end.<br>

+        // Note: If the first 23 bits of the additional bytes are not 0, then damaged

+        // MPEG bitstreams could cause overread and segfault.

+        memcpy(m_pFFBuffer, pDataInBuff, buffsize);

+        memset(m_pFFBuffer+buffsize,0,FF_INPUT_BUFFER_PADDING_SIZE);

+

+	    int used_byte = avcodec_decode_audio2(m_pAVCtx, (int16_t*)m_pPCMData, &nPCMLength, (const uint8_t*)m_pFFBuffer, buffsize);

+        

+        if(used_byte < 0 ) { size = used_byte;  return S_OK; }

+        if(used_byte == 0 && nPCMLength <= 0 ) {size = used_byte; return S_OK; }

+        size += used_byte;//

+

+	    if ( nPCMLength>0)

+	    {

+		    WAVEFORMATEX*		wfein = (WAVEFORMATEX*)m_pInput->CurrentMediaType().Format();

+		    CAtlArray<float>	pBuff;

+		    int					iSpeakerConfig;

+		    int					nRemap;

+		    float*				pDataOut;

+                    

+		    nRemap = FFGetChannelMap (m_pAVCtx);

+		    if (nRemap >=0)

+		    {

+					   

+    		

+		        switch (nCodecId)

+		        {

+		            case CODEC_ID_EAC3 :

+			            scmap = &m_ffmpeg_ac3[FFGetChannelMap(m_pAVCtx)];

+			            break;

+		            default :

+			            scmap = &m_scmap_default[m_pAVCtx->channels-1];

+			            break;

+		        }

+

+			    switch (m_pAVCtx->sample_fmt)

+			    {

+			    case SAMPLE_FMT_S16 :

+				    pBuff.SetCount (nPCMLength / 2);

+				    pDataOut = pBuff.GetData();

+

+				    for (size_t i=0; i<pBuff.GetCount()/m_pAVCtx->channels; i++)

+				    {

+					    for(int ch=0; ch<m_pAVCtx->channels; ch++)

+					    {

+						    *pDataOut = (float)((int16_t*)m_pPCMData) [scmap->ch[ch]+i*m_pAVCtx->channels] / SHRT_MAX;

+						    pDataOut++;

+					    }

+				    }

+				    break;

+

+			    case SAMPLE_FMT_S32 :

+				    pBuff.SetCount (nPCMLength / 4);

+				    pDataOut = pBuff.GetData();

+

+				    for (size_t i=0; i<pBuff.GetCount()/m_pAVCtx->channels; i++)

+				    {

+					    for(int ch=0; ch<m_pAVCtx->channels; ch++)

+					    {

+					    	*pDataOut = (float)((int32_t*)m_pPCMData) [scmap->ch[ch]+i*m_pAVCtx->channels] / INT_MAX;

+							pDataOut++;

+					    }

+				    }

+				    break;

+			    default :

+				    ASSERT(FALSE);

+				    break;

+			    }

+

+                if(pBuff.GetCount() > 0){

+                    int idx_start = pBuffOut.GetCount();

+                    pBuffOut.SetCount( idx_start + pBuff.GetCount()  );

+                    for(int i = 0; i< pBuff.GetCount(); i++){

+                        pBuffOut[idx_start+i] = pBuff[i];

+                    }

+                }

+		   

+		    }

+        }

+

+        buffsize	-= used_byte;

+        pDataInBuff += used_byte;

+    }

+    if(pBuffOut.GetCount() > 0 && scmap)

+        hr = Deliver(pBuffOut, m_pAVCtx->sample_rate, scmap->nChannels, scmap->dwChannelMask);

 	return hr;

 }

 #endif

diff --git a/src/filters/transform/MpaDecFilter/MpaDecFilter.h b/src/filters/transform/MpaDecFilter/MpaDecFilter.h
index a7de54e8a..999979bff 100644
--- a/src/filters/transform/MpaDecFilter/MpaDecFilter.h
+++ b/src/filters/transform/MpaDecFilter/MpaDecFilter.h
@@ -143,6 +143,9 @@ protected:
 	HRESULT DeliverFfmpeg(int nCodecId, BYTE* p, int buffsize, int& size);

 	static void		LogLibAVCodec(void* par,int level,const char *fmt,va_list valist);

 

+	BYTE*	m_pFFBuffer;

+	int		m_nFFBufferSize;

+

 protected:

 	CCritSec m_csProps;

 	MPCSampleFormat m_iSampleFormat;