[haiku-commits] r38400 - haiku/trunk/src/add-ons/media/plugins/ffmpeg
- From: superstippi@xxxxxx
- To: haiku-commits@xxxxxxxxxxxxx
- Date: Fri, 27 Aug 2010 15:21:03 +0200 (CEST)
Author: stippi
Date: 2010-08-27 15:21:03 +0200 (Fri, 27 Aug 2010)
New Revision: 38400
Changeset: http://dev.haiku-os.org/changeset/38400
Modified:
haiku/trunk/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.cpp
haiku/trunk/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.h
Log:
* 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.
Modified: haiku/trunk/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.cpp
===================================================================
--- haiku/trunk/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.cpp
2010-08-27 12:40:05 UTC (rev 38399)
+++ haiku/trunk/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.cpp
2010-08-27 13:21:03 UTC (rev 38400)
@@ -112,9 +112,9 @@
#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 @@
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 @@
// 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 @@
} 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 @@
= 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 @@
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 @@
fOutputBufferOffset = 0;
fOutputBufferSize = 0;
+ av_init_packet(&fAudioTempPacket);
+
inOutFormat->require_flags = 0;
inOutFormat->deny_flags = B_MEDIA_MAUI_UNDEFINED_FLAGS;
@@ -547,43 +564,53 @@
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 @@
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;
}
Modified: haiku/trunk/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.h
===================================================================
--- haiku/trunk/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.h
2010-08-27 12:40:05 UTC (rev 38399)
+++ haiku/trunk/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.h
2010-08-27 13:21:03 UTC (rev 38400)
@@ -105,6 +105,7 @@
int32 fOutputBufferOffset;
int32 fOutputBufferSize;
+ AVPacket fAudioTempPacket;
};
#endif // AVCODEC_DECODER_H
Other related posts:
- » [haiku-commits] r38400 - haiku/trunk/src/add-ons/media/plugins/ffmpeg - superstippi
