Use main bitstreaming timing logic for TrueHD

mattbeghin · Jun 11, 2018 · a1b40a6 · a1b40a6
1 parent 5ba039a
commit a1b40a6
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Showing 3 changed files with 27 additions and 25 deletions.
diff --git a/decoder/LAVAudio/Bitstream.cpp b/decoder/LAVAudio/Bitstream.cpp
@@ -414,31 +414,17 @@ HRESULT CLAVAudio::DeliverBitstream(AVCodecID codec, const BYTE *buffer, DWORD d
   }
 
   REFERENCE_TIME rtStart = m_rtStart, rtStop = AV_NOPTS_VALUE;
-  // TrueHD timings
-  // Since the SPDIF muxer takes 24 frames and puts them into one IEC61937 frame, we use the cached timestamp from before.
-  if (codec == AV_CODEC_ID_TRUEHD) {
-    // long-term cache is valid
-    if (m_rtBitstreamCache != AV_NOPTS_VALUE)
-      rtStart = m_rtBitstreamCache;
-    // Duration - stop time of the current frame is valid
-    if (rtStopInput != AV_NOPTS_VALUE)
-      rtStop = rtStopInput;
-    else // no actual time of the current frame, use typical TrueHD frame size, 24 * 0.83333ms
-      rtStop = rtStart + (REFERENCE_TIME)(200000 / m_dRate);
-    m_rtStart = rtStop;
-  } else {
-    double dDuration = DBL_SECOND_MULT * (double)m_bsParser.m_dwSamples / m_bsParser.m_dwSampleRate / m_dRate;
-    m_dStartOffset += fmod(dDuration, 1.0);
-
-    // Add rounded duration to rtStop
-    rtStop = rtStart + (REFERENCE_TIME)(dDuration + 0.5);
-    // and unrounded to m_rtStart..
-    m_rtStart += (REFERENCE_TIME)dDuration;
-    // and accumulate error..
-    if (m_dStartOffset > 0.5) {
-      m_rtStart++;
-      m_dStartOffset -= 1.0;
-    }
+  double dDuration = DBL_SECOND_MULT * (double)m_bsParser.m_dwSamples / m_bsParser.m_dwSampleRate / m_dRate;
+  m_dStartOffset += fmod(dDuration, 1.0);
+
+  // Add rounded duration to rtStop
+  rtStop = rtStart + (REFERENCE_TIME)(dDuration + 0.5);
+  // and unrounded to m_rtStart..
+  m_rtStart += (REFERENCE_TIME)dDuration;
+  // and accumulate error..
+  if (m_dStartOffset > 0.5) {
+    m_rtStart++;
+    m_dStartOffset -= 1.0;
   }
 
   REFERENCE_TIME rtJitter = 0;

diff --git a/decoder/LAVAudio/BitstreamParser.cpp b/decoder/LAVAudio/BitstreamParser.cpp
@@ -55,6 +55,8 @@ HRESULT CBitstreamParser::Parse(AVCodecID codec, BYTE *pBuffer, DWORD dwSize, vo
   case AV_CODEC_ID_AC3:
   case AV_CODEC_ID_EAC3:
     return ParseAC3(pBuffer, dwSize, pParserContext);
+  case AV_CODEC_ID_TRUEHD:
+    return ParseTrueHD(pBuffer, dwSize);
   }
   return S_OK;
 }
@@ -86,3 +88,16 @@ HRESULT CBitstreamParser::ParseAC3(BYTE *pBuffer, DWORD dwSize, void *pParserCon
 
   return S_OK;
 }
+
+HRESULT CBitstreamParser::ParseTrueHD(BYTE *pBuffer, DWORD dwSize)
+{
+  if (AV_RB32(pBuffer + 4) == 0xf8726fba)
+  {
+    int nRatebits = pBuffer[8] >> 4;
+
+    m_dwSampleRate = (nRatebits & 8 ? 44100 : 48000) << (nRatebits & 7);
+    m_dwSamples = 24 * (40 << (nRatebits & 7));
+  }
+
+  return S_OK;
+}
diff --git a/decoder/LAVAudio/BitstreamParser.h b/decoder/LAVAudio/BitstreamParser.h
@@ -34,6 +34,7 @@ class CBitstreamParser {
 private:
   HRESULT ParseDTS(BYTE *pBuffer, DWORD dwSize);
   HRESULT ParseAC3(BYTE *pBuffer, DWORD dwSize, void *pParserContext);
+  HRESULT ParseTrueHD(BYTE *pBuffer, DWORD dwSize);
 
 public:
   DWORD m_dwSampleRate = 0;