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[Gnash-commit] gnash ChangeLog libmedia/ffmpeg/AudioDecoderFfm...
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From: |
Sandro Santilli |
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Subject: |
[Gnash-commit] gnash ChangeLog libmedia/ffmpeg/AudioDecoderFfm... |
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Date: |
Wed, 11 Jun 2008 18:53:29 +0000 |
CVSROOT: /sources/gnash
Module name: gnash
Changes by: Sandro Santilli <strk> 08/06/11 18:53:29
Modified files:
. : ChangeLog
libmedia/ffmpeg: AudioDecoderFfmpeg.cpp
Log message:
* libmedia/ffmpeg/AudioDecoderFfmpeg.cpp (decode): reintroduce
support for the 'parse' parameter. Still not sure if that's
a clean way to decode *sets* of frames, but at least it works
for the time being.
CVSWeb URLs:
http://cvs.savannah.gnu.org/viewcvs/gnash/ChangeLog?cvsroot=gnash&r1=1.6902&r2=1.6903
http://cvs.savannah.gnu.org/viewcvs/gnash/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp?cvsroot=gnash&r1=1.10&r2=1.11
Patches:
Index: ChangeLog
===================================================================
RCS file: /sources/gnash/gnash/ChangeLog,v
retrieving revision 1.6902
retrieving revision 1.6903
diff -u -b -r1.6902 -r1.6903
--- ChangeLog 11 Jun 2008 15:35:49 -0000 1.6902
+++ ChangeLog 11 Jun 2008 18:53:29 -0000 1.6903
@@ -1,5 +1,12 @@
2008-06-11 Sandro Santilli <address@hidden>
+ * libmedia/ffmpeg/AudioDecoderFfmpeg.cpp (decode): reintroduce
+ support for the 'parse' parameter. Still not sure if that's
+ a clean way to decode *sets* of frames, but at least it works
+ for the time being.
+
+2008-06-11 Sandro Santilli <address@hidden>
+
* libbase/Buffer.h: add copy ctor and assignment op.
2008-06-11 Benjamin Wolsey <address@hidden>
Index: libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
===================================================================
RCS file: /sources/gnash/gnash/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp,v
retrieving revision 1.10
retrieving revision 1.11
diff -u -b -r1.10 -r1.11
--- libmedia/ffmpeg/AudioDecoderFfmpeg.cpp 11 Jun 2008 14:40:27 -0000
1.10
+++ libmedia/ffmpeg/AudioDecoderFfmpeg.cpp 11 Jun 2008 18:53:29 -0000
1.11
@@ -20,9 +20,16 @@
#include "AudioDecoderFfmpeg.h"
#include <cmath> // for std::ceil
+#include <algorithm> // for std::copy
//#define GNASH_DEBUG_AUDIO_DECODING
+#ifdef FFMPEG_AUDIO2
+# define AVCODEC_DECODE_AUDIO avcodec_decode_audio2
+#else
+# define AVCODEC_DECODE_AUDIO avcodec_decode_audio
+#endif
+
namespace gnash {
namespace media {
@@ -200,51 +207,98 @@
return 0;
}
-#if 0
- 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;
+ size_t retCapacity = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+ boost::uint8_t* retBuf = new boost::uint8_t[retCapacity];
+ int retBufSize = 0;
-#else
+#ifdef GNASH_DEBUG_AUDIO_DECODING
+ log_debug(" Parsing loop starts, input is %d bytes, retCapacity is %d
bytes", inputSize, retCapacity);
+#endif // GNASH_DEBUG_AUDIO_DECODING
+ decodedBytes = 0; // nothing decoded yet
+ while (decodedBytes < inputSize)
+ {
+ boost::uint8_t* frame=0; // parsed frame (pointer into input)
+ int framesize; // parsed frame size
+
+ int consumed = av_parser_parse(_parser, _audioCodecCtx,
+ &frame, &framesize,
+ input+decodedBytes, inputSize-decodedBytes,
+ 0, 0); // pts & dts
+ if (consumed < 0)
+ {
+ log_error(_("av_parser_parse returned %d. Upgrading
ffmpeg/libavcodec might fix this issue."), consumed);
+ // Setting data position to data size will get the
sound removed
+ // from the active sound list later on.
+ decodedBytes = inputSize;
+ break;
+ }
- 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 GNASH_PARANOIA_LEVEL > 1
+ // the returned frame pointer is inside the input buffer
+ assert(frame >= input+decodedBytes && frame < input+inputSize);
+ // the returned frame size is within the input size
+ assert(framesize <= inputSize);
+ assert(frame+framesize <= input+inputSize);
+ // the number of bytes skipped to get to the frame
+ // is the offset to the parsed frame
+ assert(consumed == framesize);
+ assert(input+decodedBytes == frame);
+#endif
- bufsize = 0;
- while (bufsize == 0 && decodedBytes < inputSize)
- {
- boost::uint8_t* frame;
- int framesize;
+ // all good so far, keep going..
+ // (we might do this immediately, as we'll override
decodedBytes on error anyway)
+ decodedBytes += consumed;
- bytes_decoded = av_parser_parse(_parser, _audioCodecCtx,
&frame, &framesize, input+decodedBytes, inputSize-decodedBytes, 0, 0); //the
last 2 is pts & dts
+#ifdef GNASH_DEBUG_AUDIO_DECODING
+ log_debug(" parsed frame is %d bytes (consumed %d/%d)",
framesize, decodedBytes, inputSize);
+#endif // GNASH_DEBUG_AUDIO_DECODING
- 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
- if (bytes_decoded < 0 || tmp < 0 || bufsize < 0) {
- log_error(_("Error while decoding audio data. Upgrading
ffmpeg/libavcodec might fix this issue."));
+ // Now, decode the frame. We use the ::decodeFrame specialized
function
+ // here so resampling is done appropriately
+ boost::uint32_t outSize = 0;
+ boost::uint8_t* outBuf = decodeFrame(frame, framesize, outSize);
+
+ if (!outBuf)
+ {
// Setting data position to data size will get the
sound removed
// from the active sound list later on.
decodedBytes = inputSize;
break;
}
- decodedBytes += bytes_decoded;
+#ifdef GNASH_DEBUG_AUDIO_DECODING
+ log_debug(" decoded frame is %d bytes, would grow return
buffer size to %d bytes", outSize, retBufSize+(unsigned)outSize);
+#endif // GNASH_DEBUG_AUDIO_DECODING
+
+ //
+ // Now append this data to the buffer we're going to return
+ //
+
+ // if the new data doesn't fit, reallocate the output
+ // TODO: can use the Buffer class here.. if we return it too...
+ if ( retBufSize+(unsigned)outSize > retCapacity )
+ {
+#ifdef GNASH_DEBUG_AUDIO_DECODING
+ log_debug(" output buffer won't be able to hold %d
bytes, capacity is only %d bytes",
+ retBufSize+(unsigned)outSize, retCapacity);
+#endif // GNASH_DEBUG_AUDIO_DECODING
+ boost::uint8_t* tmp = retBuf;
+ retCapacity = std::max(retBufSize+(unsigned)outSize,
retCapacity*2);
+#ifdef GNASH_DEBUG_AUDIO_DECODING
+ log_debug(" reallocating it to hold up to %d bytes",
retCapacity);
+#endif // GNASH_DEBUG_AUDIO_DECODING
+ retBuf = new boost::uint8_t[retCapacity];
+ if ( retBufSize ) std::copy(tmp, tmp+retBufSize,
retBuf);
+ delete [] tmp;
+ }
+ std::copy(outBuf, outBuf+outSize, retBuf+retBufSize);
+ retBufSize+=(unsigned)outSize;
}
-#endif
+
+ outputSize = retBufSize;
+ return retBuf;
}
@@ -270,12 +324,7 @@
// then decoding will eventually reduce it
int outSize = bufsize;
- int tmp =
-#ifdef FFMPEG_AUDIO2
- avcodec_decode_audio2(_audioCodecCtx, outPtr, &outSize, input,
inputSize);
-#else
- avcodec_decode_audio (_audioCodecCtx, outPtr, &outSize, input,
inputSize);
-#endif
+ int tmp = AVCODEC_DECODE_AUDIO(_audioCodecCtx, outPtr, &outSize, input,
inputSize);
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug(" avcodec_decode_audio[2](ctx, bufptr, %d, input, %d)
returned %d and set frame_size to %d",