* Use the new avcodec_decode_audio3() call, avcodec_decode_audio2()

gives the deprecated warning... We need to cache an AVPacket for this.
 * Check the allocation of fOutputBuffer.
 * When seeking, we need to flush the already decoded stuff
   in fOutputBuffer, and throw away the last chunk buffer as well.
 * Handle an incomplete input format at least to the point of not
   crashing with a divide error (mp3_reader would give us such an
   incomplete format for example).
 * _DecodeAudio():
   - Fixed some edge cases in the audio decoding loop: avcodec_decode_audio3()
     can return a 0 length, which means no error, but no decoded frames
     either. ffplay throws away the chunk in this case, do the same.
  - Convert some invalid situations that were printf()s into debugger()s.
  - Add much more comments to explain how everything works.
 * Fixed the occasional coding style issue.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@38400 a95241bf-73f2-0310-859d-f6bbb57e9c96
This commit is contained in:
Stephan Aßmus
2010-08-27 13:21:03 +00:00
parent 156a187d0d
commit d5479a3bd0
2 changed files with 84 additions and 54 deletions
@@ -112,9 +112,9 @@ AVCodecDecoder::~AVCodecDecoder()
#ifdef DO_PROFILING
if (profileCounter > 0) {
printf("[%c] profile: d1 = %lld, d2 = %lld (%Ld)\n",
fIsAudio?('a'):('v'), decodingTime / profileCounter, conversionTime / profileCounter,
fFrame);
printf("[%c] profile: d1 = %lld, d2 = %lld (%Ld)\n",
fIsAudio?('a'):('v'), decodingTime / profileCounter,
conversionTime / profileCounter, fFrame);
}
#endif
@@ -157,8 +157,11 @@ AVCodecDecoder::Setup(media_format* ioEncodedFormat, const void* infoBuffer,
fIsAudio = (ioEncodedFormat->type == B_MEDIA_ENCODED_AUDIO);
TRACE("[%c] AVCodecDecoder::Setup()\n", fIsAudio?('a'):('v'));
if (fIsAudio && !fOutputBuffer)
fOutputBuffer = new char[AVCODEC_MAX_AUDIO_FRAME_SIZE];
if (fIsAudio && fOutputBuffer == NULL) {
fOutputBuffer = new(std::nothrow) char[AVCODEC_MAX_AUDIO_FRAME_SIZE];
if (fOutputBuffer == NULL)
return B_NO_MEMORY;
}
#ifdef TRACE_AV_CODEC
char buffer[1024];
@@ -271,9 +274,8 @@ AVCodecDecoder::Seek(uint32 seekTo, int64 seekFrame, int64* frame,
// Reset the FFmpeg codec to flush buffers, so we keep the sync
#if 1
if (fCodecInitDone) {
fCodecInitDone = false;
avcodec_close(fContext);
fCodecInitDone = (avcodec_open(fContext, fCodec) >= 0);
fCodecInitDone = avcodec_open(fContext, fCodec) >= 0;
}
#else
// For example, this doesn't work on the H.264 codec. :-/
@@ -298,6 +300,13 @@ AVCodecDecoder::Seek(uint32 seekTo, int64 seekFrame, int64* frame,
} else
return B_BAD_VALUE;
// Flush internal buffers as well.
fChunkBuffer = NULL;
fChunkBufferOffset = 0;
fChunkBufferSize = 0;
fOutputBufferOffset = 0;
fOutputBufferSize = 0;
fFrame = *frame;
fStartTime = *time;
TRACE("so new frame is %Ld at time %.6f\n", *frame, *time / 1000000.0);
@@ -365,11 +374,17 @@ AVCodecDecoder::_NegotiateAudioOutputFormat(media_format* inOutFormat)
= fInputFormat.u.encoded_audio.output.channel_count;
outputAudioFormat.format = fInputFormat.u.encoded_audio.output.format;
// Check that format is not still a wild card!
if (outputAudioFormat.format == 0)
if (outputAudioFormat.format == 0) {
TRACE(" format still a wild-card, assuming B_AUDIO_SHORT.\n");
outputAudioFormat.format = media_raw_audio_format::B_AUDIO_SHORT;
}
// Check that channel count is not still a wild card!
if (outputAudioFormat.channel_count == 0) {
TRACE(" channel_count still a wild-card, assuming stereo.\n");
outputAudioFormat.channel_count = 2;
}
outputAudioFormat.buffer_size
= 1024 * fInputFormat.u.encoded_audio.output.channel_count;
outputAudioFormat.buffer_size = 1024 * outputAudioFormat.channel_count;
inOutFormat->type = B_MEDIA_RAW_AUDIO;
inOutFormat->u.raw_audio = outputAudioFormat;
@@ -377,7 +392,7 @@ AVCodecDecoder::_NegotiateAudioOutputFormat(media_format* inOutFormat)
fContext->frame_size = (int)fInputFormat.u.encoded_audio.frame_size;
fContext->sample_rate
= (int)fInputFormat.u.encoded_audio.output.frame_rate;
fContext->channels = fInputFormat.u.encoded_audio.output.channel_count;
fContext->channels = outputAudioFormat.channel_count;
fContext->block_align = fBlockAlign;
fContext->extradata = (uint8_t*)fExtraData;
fContext->extradata_size = fExtraDataSize;
@@ -426,6 +441,8 @@ AVCodecDecoder::_NegotiateAudioOutputFormat(media_format* inOutFormat)
fOutputBufferOffset = 0;
fOutputBufferSize = 0;
av_init_packet(&fAudioTempPacket);
inOutFormat->require_flags = 0;
inOutFormat->deny_flags = B_MEDIA_MAUI_UNDEFINED_FLAGS;
@@ -547,43 +564,53 @@ AVCodecDecoder::_NegotiateVideoOutputFormat(media_format* inOutFormat)
status_t
AVCodecDecoder::_DecodeAudio(void* outBuffer, int64* outFrameCount,
AVCodecDecoder::_DecodeAudio(void* _buffer, int64* outFrameCount,
media_header* mediaHeader, media_decode_info* info)
{
TRACE_AUDIO("AVCodecDecoder::_DecodeAudio()\n");
// TRACE_AUDIO(" audio start_time %.6f\n",
// mediaHeader->start_time / 1000000.0);
TRACE_AUDIO("AVCodecDecoder::_DecodeAudio(audio start_time %.6fs)\n",
mediaHeader->start_time / 1000000.0);
char* output_buffer = (char*)outBuffer;
*outFrameCount = 0;
uint8* buffer = reinterpret_cast<uint8*>(_buffer);
while (*outFrameCount < fOutputFrameCount) {
// Check conditions which would hint at broken code below.
if (fOutputBufferSize < 0) {
TRACE_AUDIO(" ############ fOutputBufferSize %ld\n",
fOutputBufferSize);
debugger("Decoding read past the end of the output buffer!");
fOutputBufferSize = 0;
}
if (fChunkBufferSize < 0) {
TRACE_AUDIO(" ############ fChunkBufferSize %ld\n",
fChunkBufferSize);
debugger("Decoding read past the end of the chunk buffer!");
fChunkBufferSize = 0;
}
if (fOutputBufferSize > 0) {
// We still have decoded audio frames from the last
// invokation, which start at fOutputBuffer + fOutputBufferOffset
// and are of fOutputBufferSize. Copy those into the buffer,
// but not more than it can hold.
int32 frames = min_c(fOutputFrameCount - *outFrameCount,
fOutputBufferSize / fOutputFrameSize);
memcpy(output_buffer, fOutputBuffer + fOutputBufferOffset,
frames * fOutputFrameSize);
fOutputBufferOffset += frames * fOutputFrameSize;
fOutputBufferSize -= frames * fOutputFrameSize;
output_buffer += frames * fOutputFrameSize;
if (frames == 0)
debugger("fOutputBufferSize not multiple of frame size!");
size_t remainingSize = frames * fOutputFrameSize;
memcpy(buffer, fOutputBuffer + fOutputBufferOffset, remainingSize);
fOutputBufferOffset += remainingSize;
fOutputBufferSize -= remainingSize;
buffer += remainingSize;
*outFrameCount += frames;
fStartTime += (bigtime_t)((1000000LL * frames) / fOutputFrameRate);
continue;
}
if (fChunkBufferSize == 0) {
// Time to read the next chunk buffer. We use a separate
// media_header, since the chunk header may not belong to
// the start of the decoded audio frames we return. For
// example we may have used frames from a previous invokation,
// or we may have to read several chunks until we fill up the
// output buffer.
media_header chunkMediaHeader;
status_t err;
err = GetNextChunk(&fChunkBuffer, &fChunkBufferSize,
status_t err = GetNextChunk(&fChunkBuffer, &fChunkBufferSize,
&chunkMediaHeader);
if (err == B_LAST_BUFFER_ERROR) {
TRACE_AUDIO(" Last Chunk with chunk size %ld\n",
@@ -600,38 +627,40 @@ AVCodecDecoder::_DecodeAudio(void* outBuffer, int64* outFrameCount,
fStartTime = chunkMediaHeader.start_time;
if (*outFrameCount == 0)
mediaHeader->start_time = chunkMediaHeader.start_time;
continue;
}
if (fOutputBufferSize == 0) {
int len;
int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
len = avcodec_decode_audio2(fContext, (short *)fOutputBuffer,
&out_size, (uint8_t*)fChunkBuffer + fChunkBufferOffset,
fChunkBufferSize);
if (len < 0) {
if (!fAudioDecodeError) {
printf("########### audio decode error, "
"fChunkBufferSize %ld, fChunkBufferOffset %ld\n",
fChunkBufferSize, fChunkBufferOffset);
fAudioDecodeError = true;
}
out_size = 0;
len = 0;
fChunkBufferOffset = 0;
fChunkBufferSize = 0;
} else
fAudioDecodeError = false;
fChunkBufferOffset += len;
fChunkBufferSize -= len;
fOutputBufferOffset = 0;
fOutputBufferSize = out_size;
fAudioTempPacket.data = (uint8_t*)fChunkBuffer + fChunkBufferOffset;
fAudioTempPacket.size = fChunkBufferSize;
// Initialize decodedBytes to the output buffer size.
int decodedBytes = AVCODEC_MAX_AUDIO_FRAME_SIZE;
int usedBytes = avcodec_decode_audio3(fContext,
(int16*)fOutputBuffer, &decodedBytes, &fAudioTempPacket);
if (usedBytes < 0 && !fAudioDecodeError) {
// Failure
printf("########### audio decode error, "
"fChunkBufferSize %ld, fChunkBufferOffset %ld\n",
fChunkBufferSize, fChunkBufferOffset);
fAudioDecodeError = true;
}
if (usedBytes <= 0) {
// Error or failure to produce decompressed output.
// Skip the chunk buffer data entirely.
usedBytes = fChunkBufferSize;
decodedBytes = 0;
} else {
// Success
fAudioDecodeError = false;
}
//printf(" chunk size: %d, decoded: %d, used: %d\n",
//fAudioTempPacket.size, decodedBytes, usedBytes);
fChunkBufferOffset += usedBytes;
fChunkBufferSize -= usedBytes;
fOutputBufferOffset = 0;
fOutputBufferSize = decodedBytes;
}
TRACE_AUDIO(" frame count: %lld\n", *outFrameCount);
fFrame += *outFrameCount;
// TRACE("Played %Ld frames at time %Ld\n",*outFrameCount, mediaHeader->start_time);
TRACE_AUDIO(" frame count: %lld current: %lld\n", *outFrameCount, fFrame);
return B_OK;
}
@@ -105,6 +105,7 @@ private:
int32 fOutputBufferOffset;
int32 fOutputBufferSize;
AVPacket fAudioTempPacket;
};
#endif // AVCODEC_DECODER_H