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[Gnash-commit] gnash ChangeLog libmedia/AudioDecoder.h libmedi...
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From: |
Sandro Santilli |
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Subject: |
[Gnash-commit] gnash ChangeLog libmedia/AudioDecoder.h libmedi... |
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Date: |
Fri, 06 Jun 2008 10:20:00 +0000 |
CVSROOT: /sources/gnash
Module name: gnash
Changes by: Sandro Santilli <strk> 08/06/06 10:19:59
Modified files:
. : ChangeLog
libmedia : AudioDecoder.h
libmedia/ffmpeg: AudioDecoderFfmpeg.cpp AudioDecoderFfmpeg.h
server/asobj : NetStreamFfmpeg.cpp
Log message:
* libmedia/AudioDecoder.h: add a decode(EncodedAudioFrame&)
method.
* libmedia/ffmpeg/AudioDecoderFfmpeg.{cpp,h}: Implement
decode(EncodeAudioFrame), fix resampling code, disable
(log_unimp)
the "parsing" feature (should be dropped)
* server/asobj/NetStreamFfmpeg.cpp: use
AudioDecoder::decode(EncodedAudioFrame) interface, cleaner.
This fixes sound in NetStream (tested with NetStream-SquareTest.swf
modified to load the 2hrs.flv movie - mp3 audio in it).
CVSWeb URLs:
http://cvs.savannah.gnu.org/viewcvs/gnash/ChangeLog?cvsroot=gnash&r1=1.6833&r2=1.6834
http://cvs.savannah.gnu.org/viewcvs/gnash/libmedia/AudioDecoder.h?cvsroot=gnash&r1=1.12&r2=1.13
http://cvs.savannah.gnu.org/viewcvs/gnash/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp?cvsroot=gnash&r1=1.7&r2=1.8
http://cvs.savannah.gnu.org/viewcvs/gnash/libmedia/ffmpeg/AudioDecoderFfmpeg.h?cvsroot=gnash&r1=1.4&r2=1.5
http://cvs.savannah.gnu.org/viewcvs/gnash/server/asobj/NetStreamFfmpeg.cpp?cvsroot=gnash&r1=1.144&r2=1.145
Patches:
Index: ChangeLog
===================================================================
RCS file: /sources/gnash/gnash/ChangeLog,v
retrieving revision 1.6833
retrieving revision 1.6834
diff -u -b -r1.6833 -r1.6834
--- ChangeLog 6 Jun 2008 03:48:27 -0000 1.6833
+++ ChangeLog 6 Jun 2008 10:19:57 -0000 1.6834
@@ -1,5 +1,15 @@
+2008-06-06 Sandro Santilli <address@hidden>
+
+ * libmedia/AudioDecoder.h: add a decode(EncodedAudioFrame&) method.
+ * libmedia/ffmpeg/AudioDecoderFfmpeg.{cpp,h}: Implement
+ decode(EncodeAudioFrame), fix resampling code, disable (log_unimp)
+ the "parsing" feature (should be dropped)
+ * server/asobj/NetStreamFfmpeg.cpp: use
+ AudioDecoder::decode(EncodedAudioFrame) interface, cleaner.
+
2008-06-06 Zou Lunkai <address@hidden>
+ YOUR ADV HERE
2008-06-06 Sandro Santilli <address@hidden>
Index: libmedia/AudioDecoder.h
===================================================================
RCS file: /sources/gnash/gnash/libmedia/AudioDecoder.h,v
retrieving revision 1.12
retrieving revision 1.13
diff -u -b -r1.12 -r1.13
--- libmedia/AudioDecoder.h 3 Jun 2008 16:11:44 -0000 1.12
+++ libmedia/AudioDecoder.h 6 Jun 2008 10:19:59 -0000 1.13
@@ -87,6 +87,23 @@
///
virtual boost::uint8_t* decode(boost::uint8_t* /*input*/,
boost::uint32_t /*inputSize*/, boost::uint32_t& /*outputSize*/,
boost::uint32_t& /*decodedData*/, bool /*parse*/) { return NULL; }
+ /// Decodes an EncodedAudioFrame and returns a pointer to the decoded
data
+ //
+ /// @param input
+ /// The audio data
+ ///
+ /// @param outputSize
+ /// The output size of the video data, is passed by reference.
+ ///
+ /// @return a pointer to the decoded data, or NULL if decoding fails.
+ /// The caller owns the decoded data.
+ ///
+ virtual boost::uint8_t* decode(const EncodedAudioFrame& /*input*/,
+ boost::uint32_t& /*outputSize*/)
+ {
+ return NULL;
+ }
+
};
} // gnash.media namespace
Index: libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
===================================================================
RCS file: /sources/gnash/gnash/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp,v
retrieving revision 1.7
retrieving revision 1.8
diff -u -b -r1.7 -r1.8
--- libmedia/ffmpeg/AudioDecoderFfmpeg.cpp 5 Jun 2008 13:32:45 -0000
1.7
+++ libmedia/ffmpeg/AudioDecoderFfmpeg.cpp 6 Jun 2008 10:19:59 -0000
1.8
@@ -20,6 +20,8 @@
#include "AudioDecoderFfmpeg.h"
+//#define GNASH_DEBUG_AUDIO_DECODING
+
namespace gnash {
namespace media {
@@ -167,7 +169,7 @@
if (_audioCodecCtx->codec->id != CODEC_ID_MP3) {
_audioCodecCtx->channels = (info->stereo ? 2 : 1);
_audioCodecCtx->sample_rate = info->sampleRate;
- //_audioCodecCtx->sample_fmt = SAMPLE_FMT_S16;
+ _audioCodecCtx->sample_fmt = SAMPLE_FMT_S16; // was commented
out, why ?
}
log_debug(_("AudioDecoderFfmpeg::setup: initialized FFMPEG codec %s
(%d)"),
@@ -176,107 +178,135 @@
return true;
}
-boost::uint8_t* AudioDecoderFfmpeg::decode(boost::uint8_t* input,
boost::uint32_t inputSize, boost::uint32_t& outputSize, boost::uint32_t&
decodedBytes, bool parse)
+boost::uint8_t*
+AudioDecoderFfmpeg::decode(boost::uint8_t* input, boost::uint32_t inputSize,
boost::uint32_t& outputSize, boost::uint32_t& decodedBytes, bool parse)
{
+ //GNASH_REPORT_FUNCTION;
- long bytes_decoded = 0;
- int bufsize = (AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2;
- boost::uint8_t* output = new boost::uint8_t[bufsize];
- boost::uint32_t orgbufsize = bufsize;
- decodedBytes = 0;
-
- if (parse) {
-
- if (!_parser)
+ if ( ! parse )
{
- log_error(_("libavcodec can't parse the current audio
format"));
- return NULL;
+ boost::uint8_t* ret = decodeFrame(input, inputSize, outputSize);
+ if ( ret ) decodedBytes=inputSize;
+ else decodedBytes=0;
+ return ret;
}
+ LOG_ONCE( log_unimpl("Parsing inside FFMPEG AudioDecoder (shouldn't be
needed, core lib should be fixed to not even try)") );
+ // TODO: for each parsed frame, call decodeFrame and
+ // grow the buffer to return...
+ //assert(!parse);
+ decodedBytes=inputSize; // pretend we decoded everything
+ outputSize=0; // and that resulting output is empty
+ return 0;
+}
- bufsize = 0;
- while (bufsize == 0 && decodedBytes < inputSize) {
- boost::uint8_t* frame;
- int framesize;
-
- bytes_decoded = av_parser_parse(_parser,
_audioCodecCtx, &frame, &framesize, input+decodedBytes, inputSize-decodedBytes,
0, 0); //the last 2 is pts & dts
-
- int tmp = 0;
-#ifdef FFMPEG_AUDIO2
- bufsize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
- tmp = avcodec_decode_audio2(_audioCodecCtx,
reinterpret_cast<boost::int16_t*>(output), &bufsize, frame, framesize);
-#else
- tmp = avcodec_decode_audio(_audioCodecCtx,
reinterpret_cast<boost::int16_t*>(output), &bufsize, frame, framesize);
-#endif
+boost::uint8_t*
+AudioDecoderFfmpeg::decode(const EncodedAudioFrame& ef, boost::uint32_t&
outputSize)
+{
+ return decodeFrame(ef.data.get(), ef.dataSize, outputSize);
+}
- if (bytes_decoded < 0 || tmp < 0 || bufsize < 0) {
- log_error(_("Error while decoding audio data.
Upgrading ffmpeg/libavcodec might fix this issue."));
- // Setting data position to data size will get
the sound removed
- // from the active sound list later on.
- decodedBytes = inputSize;
- break;
- }
+boost::uint8_t*
+AudioDecoderFfmpeg::decodeFrame(boost::uint8_t* input, boost::uint32_t
inputSize, boost::uint32_t& outputSize)
+{
+ //GNASH_REPORT_FUNCTION;
- decodedBytes += bytes_decoded;
- }
+ //static const unsigned int bufsize = (AVCODEC_MAX_AUDIO_FRAME_SIZE *
3) / 2;
+ static const unsigned int bufsize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
- } else {
+ // TODO: make this a private member, to reuse (see NetStreamFfmpeg in
0.8.3)
+ boost::uint8_t* output = new boost::uint8_t[bufsize];
+ boost::int16_t* outPtr = reinterpret_cast<boost::int16_t*>(output);
- int tmp = 0;
+ // We initialize output size to the full size
+ // then decoding will eventually reduce it
+ int outSize = bufsize;
+ int tmp =
#ifdef FFMPEG_AUDIO2
- tmp = avcodec_decode_audio2(_audioCodecCtx,
reinterpret_cast<boost::int16_t*>(output), &bufsize, input, inputSize);
+ avcodec_decode_audio2(_audioCodecCtx, outPtr, &outSize, input,
inputSize);
#else
- tmp = avcodec_decode_audio(_audioCodecCtx,
reinterpret_cast<boost::int16_t*>(output), &bufsize, input, inputSize);
+ avcodec_decode_audio (_audioCodecCtx, outPtr, &outSize, input,
inputSize);
#endif
+#ifdef GNASH_DEBUG_AUDIO_DECODING
+ log_debug(" avcodec_decode_audio[2](ctx, bufptr, %d, input, %d)
returned %d and set frame_size to %d",
+ bufsize, inputSize, tmp, outSize);
+#endif
-
- if (bytes_decoded < 0 || tmp < 0 || bufsize < 0) {
- log_error(_("Error while decoding audio data. Upgrading
ffmpeg/libavcodec might fix this issue."));
- // Setting data position to data size will get the
sound removed
- // from the active sound list later on.
- decodedBytes = 0;
+ if (tmp < 0)
+ {
+ log_error(_("avcodec_decode_audio returned %d. Upgrading
ffmpeg/libavcodec might fix this issue."), tmp);
outputSize = 0;
delete [] output;
return NULL;
}
- decodedBytes = inputSize;
- }
-
- // Error handling
- if (bufsize < 1) {
- log_error(_("Error while decoding audio data."));
- delete [] output;
- decodedBytes = 0;
+ if (outSize < 1)
+ {
+ log_error(_("outputSize:%d after decoding audio data. Upgrading
ffmpeg/libavcodec might fix this issue."),
+ outputSize);
outputSize = 0;
+ delete [] output;
return NULL;
}
// Resampling is needed.
- if (_resampler.init(_audioCodecCtx)) {
+ if (_resampler.init(_audioCodecCtx))
+ {
+ // Resampling is needed.
+
+ // Compute new size based on frame_size and
+ // resampling configuration
+ double resampleFactor = (44100.0/_audioCodecCtx->sample_rate) *
(2.0/_audioCodecCtx->channels);
bool stereo = _audioCodecCtx->channels > 1 ? true : false;
- int samples = stereo ? bufsize >> 2 : bufsize >> 1;
+ int samples = stereo ? outSize >> 2 : outSize >> 1;
- boost::uint8_t* tmp = new boost::uint8_t[orgbufsize];
+ int resampledFrameSize = int(ceil(outSize*resampleFactor));
- samples =
_resampler.resample(reinterpret_cast<boost::int16_t*>(output),
-
reinterpret_cast<boost::int16_t*>(tmp),
- samples);
- outputSize = samples *2 *2; // the resampled audio has
samplesize 2, and is stereo
- boost::uint8_t* ret = new boost::uint8_t[outputSize];
- memcpy(ret, tmp, outputSize);
- delete [] tmp;
- delete [] output;
- return ret;
- } else {
- outputSize = bufsize;
- boost::uint8_t* ret = new boost::uint8_t[outputSize];
- memcpy(ret, output, outputSize);
+ // Allocate just the required amount of bytes
+ boost::uint8_t* resampledOutput = new boost::uint8_t[bufsize];
// be safe for now, but should be resampledFrameSize;
+
+#ifdef GNASH_DEBUG_AUDIO_DECODING
+ log_debug("Calling the resampler; "
+ "ouput to 44100hz, 2channels; "
+ "input is %dhz, %dchannels",
+ _audioCodecCtx->sample_rate, _audioCodecCtx->channels);
+#endif
+
+ samples = _resampler.resample(outPtr, // input
+ reinterpret_cast<boost::int16_t*>(resampledOutput), //
output
+ samples); // input..
+
+ // make sure to set outPtr *after* we use it as input to the
resampler
+ outPtr = reinterpret_cast<boost::int16_t*>(resampledOutput);
delete [] output;
- return ret;
+
+ if (resampledFrameSize < samples*2*2)
+ {
+ log_error(" --- Computation of resampled frame size
(%d) < then the one based on samples (%d)",
+ resampledFrameSize, samples*2*2);
+
+ log_debug(" input frame size: %d", outSize);
+ log_debug(" input sample rate: %d",
_audioCodecCtx->sample_rate);
+ log_debug(" input channels: %d",
_audioCodecCtx->channels);
+ log_debug(" input samples: %d", samples);
+
+ log_debug(" output sample rate (assuming): %d", 44100);
+ log_debug(" output channels (assuming): %d", 2);
+ log_debug(" output samples: %d", samples);
+
+ abort(); // the call to resample() likely corrupted
memory...
}
+
+ // we let the consistency check run before we override outSize
+ // to make debugging values correct
+ outSize = resampledFrameSize;
+
+ }
+
+ outputSize = outSize;
+ return reinterpret_cast<uint8_t*>(outPtr);
}
Index: libmedia/ffmpeg/AudioDecoderFfmpeg.h
===================================================================
RCS file: /sources/gnash/gnash/libmedia/ffmpeg/AudioDecoderFfmpeg.h,v
retrieving revision 1.4
retrieving revision 1.5
diff -u -b -r1.4 -r1.5
--- libmedia/ffmpeg/AudioDecoderFfmpeg.h 23 May 2008 05:58:09 -0000
1.4
+++ libmedia/ffmpeg/AudioDecoderFfmpeg.h 6 Jun 2008 10:19:59 -0000
1.5
@@ -54,8 +54,12 @@
boost::uint8_t* decode(boost::uint8_t* input, boost::uint32_t
inputSize, boost::uint32_t& outputSize, boost::uint32_t& decodedBytes, bool
parse);
+ boost::uint8_t* decode(const EncodedAudioFrame& af, boost::uint32_t&
outputSize);
+
private:
+ boost::uint8_t* decodeFrame(boost::uint8_t* input, boost::uint32_t
inputSize, boost::uint32_t& outputSize);
+
AVCodec* _audioCodec;
AVCodecContext* _audioCodecCtx;
AVCodecParserContext* _parser;
Index: server/asobj/NetStreamFfmpeg.cpp
===================================================================
RCS file: /sources/gnash/gnash/server/asobj/NetStreamFfmpeg.cpp,v
retrieving revision 1.144
retrieving revision 1.145
diff -u -b -r1.144 -r1.145
--- server/asobj/NetStreamFfmpeg.cpp 5 Jun 2008 13:51:55 -0000 1.144
+++ server/asobj/NetStreamFfmpeg.cpp 6 Jun 2008 10:19:59 -0000 1.145
@@ -507,15 +507,15 @@
}
media::raw_mediadata_t* raw = new media::raw_mediadata_t();
- boost::uint32_t decodedData=0;
- bool parseAudio = true; // I don't get this...
- raw->m_data = _audioDecoder->decode(frame->data.get(), frame->dataSize,
raw->m_size, decodedData, parseAudio);
+ raw->m_data = _audioDecoder->decode(*frame, raw->m_size);
- if ( decodedData != frame->dataSize )
- {
- log_error("FIXME: NetStreamFfmpeg::decodeNextAudioFrame: not
all data in EncodedAudioFrame was decoded, just %d/%d bytes.",
- decodedData,frame->dataSize);
- }
+#ifdef GNASH_DEBUG_DECODING
+ log_debug("NetStreamFfmpeg::decodeNextAudioFrame: "
+ "%d bytes of encoded audio "
+ "decoded to %d bytes",
+ frame->dataSize,
+ raw->m_size);
+#endif // GNASH_DEBUG_DECODING
//raw->m_stream_index = m_audio_index; // no idea what this is needed
for
raw->m_ptr = raw->m_data; // no idea what this is needed for