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[Gnash-commit] [SCM] Gnash branch, master, updated. e6c2e4bde0225688447d
|
From: |
Benjamin Wolsey |
|
Subject: |
[Gnash-commit] [SCM] Gnash branch, master, updated. e6c2e4bde0225688447d8579f696dff10ed27e03 |
|
Date: |
Sat, 30 Oct 2010 08:17:25 +0000 |
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- Log -----------------------------------------------------------------
http://git.savannah.gnu.org/cgit//commit/?id=e6c2e4bde0225688447d8579f696dff10ed27e03
commit e6c2e4bde0225688447d8579f696dff10ed27e03
Author: Benjamin Wolsey <address@hidden>
Date: Sat Oct 30 10:08:04 2010 +0200
That made another function unused.
diff --git a/libvaapi/VaapiSurfaceGLX.cpp b/libvaapi/VaapiSurfaceGLX.cpp
index bf879f9..6d24e88 100644
--- a/libvaapi/VaapiSurfaceGLX.cpp
+++ b/libvaapi/VaapiSurfaceGLX.cpp
@@ -80,6 +80,8 @@ static inline bool gl_check_error(void)
return gl_do_check_error(1);
}
+#if 0
+
// glGetIntegerv() wrapper
static bool gl_get_param(GLenum param, unsigned int *pval)
{
@@ -104,8 +106,6 @@ struct TextureState {
unsigned int was_bound : 1;
};
-#if 0
-
/// Bind texture, preserve previous texture state
static bool bind_texture(TextureState *ts, GLenum target, GLuint texture)
{
http://git.savannah.gnu.org/cgit//commit/?id=136b669ab9ee350f5543351b6bacf204845dfeb2
commit 136b669ab9ee350f5543351b6bacf204845dfeb2
Author: Benjamin Wolsey <address@hidden>
Date: Sat Oct 30 10:06:07 2010 +0200
Comment out unused functions; no idea what they're for.
diff --git a/libvaapi/VaapiSurfaceGLX.cpp b/libvaapi/VaapiSurfaceGLX.cpp
index be28859..bf879f9 100644
--- a/libvaapi/VaapiSurfaceGLX.cpp
+++ b/libvaapi/VaapiSurfaceGLX.cpp
@@ -104,6 +104,8 @@ struct TextureState {
unsigned int was_bound : 1;
};
+#if 0
+
/// Bind texture, preserve previous texture state
static bool bind_texture(TextureState *ts, GLenum target, GLuint texture)
{
@@ -162,6 +164,8 @@ static void unbind_texture(TextureState *ts)
gl_check_error();
}
+#endif
+
class VaapiSurfaceGLXImpl: public VaapiSurfaceImplBase {
void *surface() const
{ return reinterpret_cast<void *>(VaapiSurfaceImplBase::surface()); }
http://git.savannah.gnu.org/cgit//commit/?id=37954a5dc359733057953c32a03b472675aea35a
commit 37954a5dc359733057953c32a03b472675aea35a
Author: Benjamin Wolsey <address@hidden>
Date: Sat Oct 30 10:03:39 2010 +0200
We always support VP6.
diff --git a/libmedia/ffmpeg/VideoDecoderFfmpeg.cpp
b/libmedia/ffmpeg/VideoDecoderFfmpeg.cpp
index 3349f75..9c5bc17 100644
--- a/libmedia/ffmpeg/VideoDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/VideoDecoderFfmpeg.cpp
@@ -417,10 +417,8 @@ VideoDecoderFfmpeg::flashToFfmpegCodec(videoCodecType
format)
// CODEC_ID_H263I didn't work with Lavc51.50.0
// and NetStream-SquareTest.swf
return CODEC_ID_FLV1;
-#ifdef FFMPEG_VP6
case VIDEO_CODEC_VP6:
return CODEC_ID_VP6F;
-#endif
#ifdef FFMPEG_VP6A
case VIDEO_CODEC_VP6A:
return CODEC_ID_VP6A;
http://git.savannah.gnu.org/cgit//commit/?id=91f3b41c009daf346ffa2078dfc4711177bd6514
commit 91f3b41c009daf346ffa2078dfc4711177bd6514
Author: Benjamin Wolsey <address@hidden>
Date: Sat Oct 30 10:01:20 2010 +0200
Raise required ffmpeg version, drop warnings related to
now-unsupported versions.
diff --git a/macros/ffmpeg.m4 b/macros/ffmpeg.m4
index d8426ca..132cec8 100644
--- a/macros/ffmpeg.m4
+++ b/macros/ffmpeg.m4
@@ -223,24 +223,12 @@ AC_DEFUN([GNASH_PATH_FFMPEG],
dnl AC_EGREP_HEADER(avcodec_decode_audio2, ${avcodec_h}, [avfound=yes],
[avfound=no])
- if test -z "$ffmpeg_num_version" -o "$ffmpeg_num_version" -lt 511100; then
- AC_MSG_WARN([Wrong ffmpeg/libavcodec version! 51.11.0 or greater
required, $ffmpeg_version detected.])
+ if test -z "$ffmpeg_num_version" -o "$ffmpeg_num_version" -lt 512800; then
+ AC_MSG_WARN([Wrong ffmpeg/libavcodec version! 51.28.0 or greater
required, $ffmpeg_version detected.])
else
ffmpeg_version_check=ok
fi
- if test ! -z "$ffmpeg_num_version" -a "$ffmpeg_num_version" -gt 512800;
then
- dnl 51.28.0 or higher required
- AC_DEFINE(FFMPEG_AUDIO2, 1, [Define if avcodec_decode_audio2 can be
used.])
- fi
-
- if test -z "$ffmpeg_num_version" -o "$ffmpeg_num_version" -lt 512700; then
- dnl 51.27.0 or higher required
- AC_MSG_WARN([This version of ffmpeg/libavcodec ($ffmpeg_version) is not
able to play VP6 encoded video: 51.27.0 or higher required!])
- else
- AC_DEFINE(FFMPEG_VP6, 1, [Define if ffmpeg can play VP6.])
- fi
-
if test -z "$ffmpeg_num_version" -o "$ffmpeg_num_version" -lt 514900; then
dnl 51.49.0 or higher required
AC_MSG_WARN([This version of ffmpeg/libavcodec ($ffmpeg_version) is not
able to play VP6A encoded video: 51.49.0 or higher required!])
http://git.savannah.gnu.org/cgit//commit/?id=51d84ecd1adb1f6edfe4b91c090d3c432514b761
commit 51d84ecd1adb1f6edfe4b91c090d3c432514b761
Author: Benjamin Wolsey <address@hidden>
Date: Sat Oct 30 10:01:07 2010 +0200
Drop const_cast.
diff --git a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
index c513586..625eb47 100644
--- a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
@@ -540,8 +540,7 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t* input,
// older ffmpeg versions didn't accept a const input..
int tmp = AVCODEC_DECODE_AUDIO(_audioCodecCtx, outPtr, &outSize,
- const_cast<boost::uint8_t*>(input),
- inputSize);
+ input, inputSize);
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug(" avcodec_decode_audio[2](ctx, bufptr, %d, input, %d) "
http://git.savannah.gnu.org/cgit//commit/?id=fc09904ad3796be1622f868d443fa0fa8395eaa3
commit fc09904ad3796be1622f868d443fa0fa8395eaa3
Author: Benjamin Wolsey <address@hidden>
Date: Sat Oct 30 09:53:41 2010 +0200
Tabs to spaces.
diff --git a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
index 3fdb563..c513586 100644
--- a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
@@ -34,12 +34,12 @@
namespace gnash {
namespace media {
namespace ffmpeg {
-
+
AudioDecoderFfmpeg::AudioDecoderFfmpeg(const AudioInfo& info)
- :
- _audioCodec(NULL),
- _audioCodecCtx(NULL),
- _parser(NULL),
+ :
+ _audioCodec(NULL),
+ _audioCodecCtx(NULL),
+ _parser(NULL),
_needsParsing(false)
{
setup(info);
@@ -56,98 +56,98 @@ AudioDecoderFfmpeg::AudioDecoderFfmpeg(const AudioInfo&
info)
}
AudioDecoderFfmpeg::AudioDecoderFfmpeg(SoundInfo& info)
- :
- _audioCodec(NULL),
- _audioCodecCtx(NULL),
- _parser(NULL)
+ :
+ _audioCodec(NULL),
+ _audioCodecCtx(NULL),
+ _parser(NULL)
{
setup(info);
- log_debug(_("AudioDecoderFfmpeg: initialized FFmpeg codec %s (%d)"),
- _audioCodec->name, _audioCodec->id);
+ log_debug(_("AudioDecoderFfmpeg: initialized FFmpeg codec %s (%d)"),
+ _audioCodec->name, _audioCodec->id);
}
AudioDecoderFfmpeg::~AudioDecoderFfmpeg()
{
- if (_audioCodecCtx)
- {
- avcodec_close(_audioCodecCtx);
- av_free(_audioCodecCtx);
- }
- if (_parser) av_parser_close(_parser);
+ if (_audioCodecCtx)
+ {
+ avcodec_close(_audioCodecCtx);
+ av_free(_audioCodecCtx);
+ }
+ if (_parser) av_parser_close(_parser);
}
void AudioDecoderFfmpeg::setup(SoundInfo& info)
{
- // Init the avdecoder-decoder
- avcodec_init();
- avcodec_register_all();// change this to only register need codec?
-
- enum CodecID codec_id;
-
- switch(info.getFormat()) {
- case AUDIO_CODEC_RAW:
- codec_id = CODEC_ID_PCM_U16LE;
- break;
- case AUDIO_CODEC_ADPCM:
- codec_id = CODEC_ID_ADPCM_SWF;
- break;
- case AUDIO_CODEC_MP3:
- codec_id = CODEC_ID_MP3;
+ // Init the avdecoder-decoder
+ avcodec_init();
+ avcodec_register_all();// change this to only register need codec?
+
+ enum CodecID codec_id;
+
+ switch(info.getFormat()) {
+ case AUDIO_CODEC_RAW:
+ codec_id = CODEC_ID_PCM_U16LE;
+ break;
+ case AUDIO_CODEC_ADPCM:
+ codec_id = CODEC_ID_ADPCM_SWF;
+ break;
+ case AUDIO_CODEC_MP3:
+ codec_id = CODEC_ID_MP3;
_needsParsing=true;
- break;
+ break;
case AUDIO_CODEC_AAC:
codec_id = CODEC_ID_AAC;
_needsParsing=true;
- break;
- default:
- boost::format err = boost::format(
- _("Unsupported audio codec %d")) %
- static_cast<int>(info.getFormat());
- throw MediaException(err.str());
- }
-
- _audioCodec = avcodec_find_decoder(codec_id);
- if (!_audioCodec) {
- audioCodecType codec = info.getFormat();
- boost::format err = boost::format(
- _("libavcodec could not find a decoder for codec %d
(%s)")) %
- static_cast<int>(codec) % codec;
- throw MediaException(err.str());
- }
+ break;
+ default:
+ boost::format err = boost::format(
+ _("Unsupported audio codec %d")) %
+ static_cast<int>(info.getFormat());
+ throw MediaException(err.str());
+ }
+
+ _audioCodec = avcodec_find_decoder(codec_id);
+ if (!_audioCodec) {
+ audioCodecType codec = info.getFormat();
+ boost::format err = boost::format(
+ _("libavcodec could not find a decoder for codec %d (%s)")) %
+ static_cast<int>(codec) % codec;
+ throw MediaException(err.str());
+ }
if ( _needsParsing )
{
- // Init the parser
- _parser = av_parser_init(codec_id);
- if (!_parser) {
+ // Init the parser
+ _parser = av_parser_init(codec_id);
+ if (!_parser) {
throw MediaException(_("AudioDecoderFfmpeg can't initialize "
"MP3 parser"));
- }
+ }
}
- _audioCodecCtx = avcodec_alloc_context();
- if (!_audioCodecCtx) {
- throw MediaException(_("libavcodec couldn't allocate context"));
- }
+ _audioCodecCtx = avcodec_alloc_context();
+ if (!_audioCodecCtx) {
+ throw MediaException(_("libavcodec couldn't allocate context"));
+ }
- int ret = avcodec_open(_audioCodecCtx, _audioCodec);
- if (ret < 0) {
- av_free(_audioCodecCtx);
- _audioCodecCtx=0;
+ int ret = avcodec_open(_audioCodecCtx, _audioCodec);
+ if (ret < 0) {
+ av_free(_audioCodecCtx);
+ _audioCodecCtx=0;
boost::format err = boost::format(
_("AudioDecoderFfmpeg: avcodec_open failed to initialize "
"FFmpeg codec %s (%d)")) % _audioCodec->name % (int)codec_id;
- throw MediaException(err.str());
- }
+ throw MediaException(err.str());
+ }
/// @todo do this only if !_needsParsing ?
- switch (_audioCodecCtx->codec->id)
+ switch (_audioCodecCtx->codec->id)
{
case CODEC_ID_MP3:
break;
- case CODEC_ID_PCM_U16LE:
+ case CODEC_ID_PCM_U16LE:
_audioCodecCtx->channels = (info.isStereo() ? 2 : 1);
_audioCodecCtx->sample_rate = info.getSampleRate();
_audioCodecCtx->sample_fmt = SAMPLE_FMT_S16; // ?! arbitrary ?
@@ -159,93 +159,93 @@ void AudioDecoderFfmpeg::setup(SoundInfo& info)
_audioCodecCtx->sample_rate = info.getSampleRate();
_audioCodecCtx->sample_fmt = SAMPLE_FMT_S16; // ?! arbitrary ?
break;
- }
+ }
}
void AudioDecoderFfmpeg::setup(const AudioInfo& info)
{
- // Init the avdecoder-decoder
- avcodec_init();
- avcodec_register_all();// change this to only register need codec?
+ // Init the avdecoder-decoder
+ avcodec_init();
+ avcodec_register_all();// change this to only register need codec?
- enum CodecID codec_id = CODEC_ID_NONE;
+ enum CodecID codec_id = CODEC_ID_NONE;
- if (info.type == CODEC_TYPE_CUSTOM)
- {
- codec_id = static_cast<CodecID>(info.codec);
+ if (info.type == CODEC_TYPE_CUSTOM)
+ {
+ codec_id = static_cast<CodecID>(info.codec);
_needsParsing=true; // @todo check this !
- }
- else if (info.type == CODEC_TYPE_FLASH)
- {
-
- switch(info.codec)
- {
- case AUDIO_CODEC_RAW:
- codec_id = CODEC_ID_PCM_U16LE;
- break;
-
- case AUDIO_CODEC_ADPCM:
- codec_id = CODEC_ID_ADPCM_SWF;
- break;
-
- case AUDIO_CODEC_MP3:
- codec_id = CODEC_ID_MP3;
+ }
+ else if (info.type == CODEC_TYPE_FLASH)
+ {
+
+ switch(info.codec)
+ {
+ case AUDIO_CODEC_RAW:
+ codec_id = CODEC_ID_PCM_U16LE;
+ break;
+
+ case AUDIO_CODEC_ADPCM:
+ codec_id = CODEC_ID_ADPCM_SWF;
+ break;
+
+ case AUDIO_CODEC_MP3:
+ codec_id = CODEC_ID_MP3;
_needsParsing=true;
- break;
+ break;
- case AUDIO_CODEC_AAC:
- codec_id = CODEC_ID_AAC;
+ case AUDIO_CODEC_AAC:
+ codec_id = CODEC_ID_AAC;
_needsParsing=true;
- break;
+ break;
#ifdef FFMPEG_NELLYMOSER
- // NOTE: bjacques found this failing in decodeFrame
- // (but probably not Ffmpeg's fault, he said)
- // I'd like to take a look at the testcase --strk
- case AUDIO_CODEC_NELLYMOSER:
- codec_id = CODEC_ID_NELLYMOSER;
- // needs parsing ?
- break;
+ // NOTE: bjacques found this failing in decodeFrame
+ // (but probably not Ffmpeg's fault, he said)
+ // I'd like to take a look at the testcase --strk
+ case AUDIO_CODEC_NELLYMOSER:
+ codec_id = CODEC_ID_NELLYMOSER;
+ // needs parsing ?
+ break;
#endif
- default:
- boost::format err = boost::format(
- _("AudioDecoderFfmpeg: unsupported flash audio "
+ default:
+ boost::format err = boost::format(
+ _("AudioDecoderFfmpeg: unsupported flash audio "
"codec %d (%s)")) %
info.codec % (audioCodecType)info.codec;
- throw MediaException(err.str());
- }
- }
- else
- {
- boost::format err = boost::format(
- _("AudioDecoderFfmpeg: unknown codec type %d "
- "(should never happen)")) % info.type;
- throw MediaException(err.str());
- }
-
- _audioCodec = avcodec_find_decoder(codec_id);
- if (!_audioCodec)
- {
- if (info.type == CODEC_TYPE_FLASH) {
- boost::format err = boost::format(
- _("AudioDecoderFfmpeg: libavcodec could not
find a decoder "
+ throw MediaException(err.str());
+ }
+ }
+ else
+ {
+ boost::format err = boost::format(
+ _("AudioDecoderFfmpeg: unknown codec type %d "
+ "(should never happen)")) % info.type;
+ throw MediaException(err.str());
+ }
+
+ _audioCodec = avcodec_find_decoder(codec_id);
+ if (!_audioCodec)
+ {
+ if (info.type == CODEC_TYPE_FLASH) {
+ boost::format err = boost::format(
+ _("AudioDecoderFfmpeg: libavcodec could not find a decoder "
"for codec %d (%s)")) %
info.codec % static_cast<audioCodecType>(info.codec);
- throw MediaException(err.str());
- } else {
- boost::format err = boost::format(
- _("AudioDecoderFfmpeg: libavcodec could not
find a decoder "
+ throw MediaException(err.str());
+ } else {
+ boost::format err = boost::format(
+ _("AudioDecoderFfmpeg: libavcodec could not find a decoder "
"for ffmpeg codec id %s")) % codec_id;
- throw MediaException(err.str());
- }
- }
+ throw MediaException(err.str());
+ }
+ }
- // Init the parser
+ // Init the parser
if (_needsParsing)
{
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug(" Initializing ffmpeg parser");
+ log_debug(" Initializing ffmpeg parser");
#endif // GNASH_DEBUG_AUDIO_DECODING
_parser = av_parser_init(codec_id);
if (!_parser) {
@@ -263,42 +263,42 @@ void AudioDecoderFfmpeg::setup(const AudioInfo& info)
throw MediaException(err.str());
}
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug(" Ffmpeg parser initialized");
+ log_debug(" Ffmpeg parser initialized");
#endif // GNASH_DEBUG_AUDIO_DECODING
}
- // Create an audioCodecCtx from the ffmpeg parser if exists/possible
- _audioCodecCtx = avcodec_alloc_context();
- if (!_audioCodecCtx) {
- throw MediaException(_("AudioDecoderFfmpeg: libavcodec couldn't
"
+ // Create an audioCodecCtx from the ffmpeg parser if exists/possible
+ _audioCodecCtx = avcodec_alloc_context();
+ if (!_audioCodecCtx) {
+ throw MediaException(_("AudioDecoderFfmpeg: libavcodec couldn't "
"allocate context"));
- }
+ }
- if ( info.extra.get() )
- {
- if (dynamic_cast<ExtraAudioInfoFfmpeg*>(info.extra.get())) {
- const ExtraAudioInfoFfmpeg& ei =
+ if ( info.extra.get() )
+ {
+ if (dynamic_cast<ExtraAudioInfoFfmpeg*>(info.extra.get())) {
+ const ExtraAudioInfoFfmpeg& ei =
static_cast<ExtraAudioInfoFfmpeg&>(*info.extra);
- _audioCodecCtx->extradata = ei.data;
- _audioCodecCtx->extradata_size = ei.dataSize;
- } else if (dynamic_cast<ExtraAudioInfoFlv*>(info.extra.get())) {
- ExtraAudioInfoFlv* extra =
+ _audioCodecCtx->extradata = ei.data;
+ _audioCodecCtx->extradata_size = ei.dataSize;
+ } else if (dynamic_cast<ExtraAudioInfoFlv*>(info.extra.get())) {
+ ExtraAudioInfoFlv* extra =
static_cast<ExtraAudioInfoFlv*>(info.extra.get());
- _audioCodecCtx->extradata = extra->data.get();
- _audioCodecCtx->extradata_size = extra->size;
- }
- }
+ _audioCodecCtx->extradata = extra->data.get();
+ _audioCodecCtx->extradata_size = extra->size;
+ }
+ }
// Setup known configurations for the audio codec context
// (should this be done only if ! _needsParsing?)
// NOTE: this is done before calling avcodec_open, as that might update
// some of the variables
- switch (codec_id)
+ switch (codec_id)
{
case CODEC_ID_MP3:
break;
- case CODEC_ID_PCM_U16LE:
+ case CODEC_ID_PCM_U16LE:
_audioCodecCtx->channels = (info.stereo ? 2 : 1);
_audioCodecCtx->sample_rate = info.sampleRate;
// was commented out (why?):
@@ -312,23 +312,23 @@ void AudioDecoderFfmpeg::setup(const AudioInfo& info)
// was commented out (why?):
_audioCodecCtx->sample_fmt = SAMPLE_FMT_S16;
break;
- }
+ }
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug(" Opening codec");
+ log_debug(" Opening codec");
#endif // GNASH_DEBUG_AUDIO_DECODING
- int ret = avcodec_open(_audioCodecCtx, _audioCodec);
- if (ret < 0) {
- //avcodec_close(_audioCodecCtx);
- av_free(_audioCodecCtx);
- _audioCodecCtx = 0;
-
- boost::format err = boost::format(
- _("AudioDecoderFfmpeg: avcodec_open failed to
initialize "
- "FFmpeg codec %s (%d)")) % _audioCodec->name %
(int)codec_id;
- throw MediaException(err.str());
- }
+ int ret = avcodec_open(_audioCodecCtx, _audioCodec);
+ if (ret < 0) {
+ //avcodec_close(_audioCodecCtx);
+ av_free(_audioCodecCtx);
+ _audioCodecCtx = 0;
+
+ boost::format err = boost::format(
+ _("AudioDecoderFfmpeg: avcodec_open failed to initialize "
+ "FFmpeg codec %s (%d)")) % _audioCodec->name % (int)codec_id;
+ throw MediaException(err.str());
+ }
}
@@ -338,16 +338,16 @@ AudioDecoderFfmpeg::decode(const boost::uint8_t* input,
outputSize, boost::uint32_t& decodedBytes,
bool parse)
{
- //GNASH_REPORT_FUNCTION;
+ //GNASH_REPORT_FUNCTION;
- if ( ! parse )
- {
+ if ( ! parse )
+ {
if ( _needsParsing )
{
log_error("AudioDecoderFfmpeg::decode called with 'parse' "
"parameter off but we know we need parsing for this codec");
}
- }
+ }
else
{
if ( !_needsParsing )
@@ -360,21 +360,21 @@ AudioDecoderFfmpeg::decode(const boost::uint8_t* input,
}
}
- size_t retCapacity = AVCODEC_MAX_AUDIO_FRAME_SIZE;
- boost::uint8_t* retBuf = new boost::uint8_t[retCapacity];
- int retBufSize = 0;
+ size_t retCapacity = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+ boost::uint8_t* retBuf = new boost::uint8_t[retCapacity];
+ int retBufSize = 0;
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug(" Parsing loop starts, input is %d bytes, retCapacity is %d "
+ log_debug(" Parsing loop starts, input is %d bytes, retCapacity is %d "
"bytes", inputSize, retCapacity);
#endif
- decodedBytes = 0; // nothing decoded yet
- while (decodedBytes < inputSize)
- {
- const boost::uint8_t* frame=0; // parsed frame (pointer into
input)
- int framesize; // parsed frame size
+ decodedBytes = 0; // nothing decoded yet
+ while (decodedBytes < inputSize)
+ {
+ const boost::uint8_t* frame=0; // parsed frame (pointer into input)
+ int framesize; // parsed frame size
- int consumed = parseInput(input+decodedBytes,
+ int consumed = parseInput(input+decodedBytes,
inputSize-decodedBytes,
&frame, &framesize);
if (consumed < 0)
@@ -389,24 +389,24 @@ AudioDecoderFfmpeg::decode(const boost::uint8_t* input,
}
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug(" parsed frame is %d bytes (consumed +%d = %d/%d)",
+ log_debug(" parsed frame is %d bytes (consumed +%d = %d/%d)",
framesize, consumed, decodedBytes+consumed, inputSize);
#endif
#if GNASH_PARANOIA_LEVEL > 1
- if ( frame )
- {
- // the returned frame pointer is inside the input buffer
- assert(frame == input+decodedBytes);
- // the returned frame size is within the input size
- assert(framesize <= inputSize);
- }
+ if ( frame )
+ {
+ // the returned frame pointer is inside the input buffer
+ assert(frame == input+decodedBytes);
+ // the returned frame size is within the input size
+ assert(framesize <= inputSize);
+ }
#endif
- // all good so far, keep going..
- // (we might do this immediately, as we'll override decodedBytes
+ // all good so far, keep going..
+ // (we might do this immediately, as we'll override decodedBytes
// on error anyway)
- decodedBytes += consumed;
+ decodedBytes += consumed;
if ( ! framesize )
{
@@ -442,60 +442,60 @@ AudioDecoderFfmpeg::decode(const boost::uint8_t* input,
}
- // Now, decode the frame. We use the ::decodeFrame specialized
function
- // here so resampling is done appropriately
- boost::uint32_t outSize = 0;
- boost::scoped_array<boost::uint8_t> outBuf(
+ // Now, decode the frame. We use the ::decodeFrame specialized function
+ // here so resampling is done appropriately
+ boost::uint32_t outSize = 0;
+ boost::scoped_array<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;
- }
+ if (!outBuf)
+ {
+ // Setting data position to data size will get the sound removed
+ // from the active sound list later on.
+ decodedBytes = inputSize;
+ break;
+ }
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug(" decoded frame is %d bytes, would grow return "
+ log_debug(" decoded frame is %d bytes, would grow return "
"buffer size to %d bytes", outSize,
retBufSize+static_cast<unsigned int>(outSize));
#endif
- //
- // Now append this data to the buffer we're going to return
- //
+ //
+ // 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 )
- {
+ // 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, "
+ log_debug(" output buffer won't be able to hold %d bytes, "
"capacity is only %d bytes",
retBufSize+(unsigned)outSize, retCapacity);
#endif
boost::uint8_t* tmp = retBuf;
- retCapacity =
std::max(retBufSize+static_cast<size_t>(outSize),
+ retCapacity = std::max(retBufSize+static_cast<size_t>(outSize),
retCapacity * 2);
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug(" reallocating it to hold up to %d bytes",
+ 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;
+ if ( retBufSize ) std::copy(tmp, tmp+retBufSize, retBuf);
+ delete [] tmp;
}
- std::copy(outBuf.get(), outBuf.get()+outSize,
retBuf+retBufSize);
- retBufSize += static_cast<unsigned int>(outSize);
- }
+ std::copy(outBuf.get(), outBuf.get()+outSize, retBuf+retBufSize);
+ retBufSize += static_cast<unsigned int>(outSize);
+ }
-
- outputSize = retBufSize;
- return retBuf;
+
+ outputSize = retBufSize;
+ return retBuf;
}
@@ -503,34 +503,34 @@ boost::uint8_t*
AudioDecoderFfmpeg::decode(const EncodedAudioFrame& ef,
boost::uint32_t& outputSize)
{
- return decodeFrame(ef.data.get(), ef.dataSize, outputSize);
+ return decodeFrame(ef.data.get(), ef.dataSize, outputSize);
}
boost::uint8_t*
AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t* input,
boost::uint32_t inputSize, boost::uint32_t& outputSize)
{
- //GNASH_REPORT_FUNCTION;
+ //GNASH_REPORT_FUNCTION;
assert(inputSize);
const size_t bufsize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
- // TODO: make this a private member, to reuse (see NetStreamFfmpeg in
0.8.3)
+ // TODO: make this a private member, to reuse (see NetStreamFfmpeg in
0.8.3)
boost::uint8_t* output;
- output = reinterpret_cast<boost::uint8_t*>(av_malloc(bufsize));
- if (!output) {
- log_error(_("failed to allocate audio buffer."));
- outputSize = 0;
- return NULL;
- }
+ output = reinterpret_cast<boost::uint8_t*>(av_malloc(bufsize));
+ if (!output) {
+ log_error(_("failed to allocate audio buffer."));
+ outputSize = 0;
+ return NULL;
+ }
boost::int16_t* outPtr = reinterpret_cast<boost::int16_t*>(output);
- // We initialize output size to the full size
- // then decoding will eventually reduce it
- int outSize = bufsize;
+ // We initialize output size to the full size
+ // then decoding will eventually reduce it
+ int outSize = bufsize;
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug("AudioDecoderFfmpeg: about to decode %d bytes; "
@@ -539,105 +539,105 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
#endif
// older ffmpeg versions didn't accept a const input..
- int tmp = AVCODEC_DECODE_AUDIO(_audioCodecCtx, outPtr, &outSize,
+ int tmp = AVCODEC_DECODE_AUDIO(_audioCodecCtx, outPtr, &outSize,
const_cast<boost::uint8_t*>(input),
inputSize);
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug(" avcodec_decode_audio[2](ctx, bufptr, %d, input, %d) "
+ log_debug(" avcodec_decode_audio[2](ctx, bufptr, %d, input, %d) "
"returned %d; set frame_size=%d",
- bufsize, inputSize, tmp, outSize);
+ bufsize, inputSize, tmp, outSize);
#endif
- if (tmp < 0) {
- log_error(_("avcodec_decode_audio returned %d. Upgrading "
+ if (tmp < 0) {
+ log_error(_("avcodec_decode_audio returned %d. Upgrading "
"ffmpeg/libavcodec might fix this issue."), tmp);
- outputSize = 0;
+ outputSize = 0;
av_free(output);
- }
+ }
- if (outSize < 2) {
- log_error(_("outputSize:%d after decoding %d bytes of input
audio "
+ if (outSize < 2) {
+ log_error(_("outputSize:%d after decoding %d bytes of input audio "
"data. Upgrading ffmpeg/libavcodec might fix this issue."),
- outputSize, inputSize);
- outputSize = 0;
+ outputSize, inputSize);
+ outputSize = 0;
av_free(output);
- }
+ }
- // Resampling is needed.
- if (_resampler.init(_audioCodecCtx)) {
- // Resampling is needed.
+ // Resampling is needed.
+ 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 inSamples = stereo ? outSize >> 2 : outSize >> 1;
+ // 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 inSamples = stereo ? outSize >> 2 : outSize >> 1;
- int expectedMaxOutSamples = std::ceil(inSamples*resampleFactor);
+ int expectedMaxOutSamples = std::ceil(inSamples*resampleFactor);
- // *channels *sampleSize
- int resampledFrameSize = expectedMaxOutSamples*2*2;
+ // *channels *sampleSize
+ int resampledFrameSize = expectedMaxOutSamples*2*2;
- // Allocate just the required amount of bytes
- boost::uint8_t* resampledOutput = new
boost::uint8_t[resampledFrameSize];
+ // Allocate just the required amount of bytes
+ boost::uint8_t* resampledOutput = new
boost::uint8_t[resampledFrameSize];
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug("Calling the resampler; resampleFactor:%d; "
- "ouput to 44100hz, 2channels, %dbytes; "
- "input is %dhz, %dchannels, %dbytes, %dsamples",
- resampleFactor,
- resampledFrameSize, _audioCodecCtx->sample_rate,
- _audioCodecCtx->channels, outSize, inSamples);
+ log_debug("Calling the resampler; resampleFactor:%d; "
+ "ouput to 44100hz, 2channels, %dbytes; "
+ "input is %dhz, %dchannels, %dbytes, %dsamples",
+ resampleFactor,
+ resampledFrameSize, _audioCodecCtx->sample_rate,
+ _audioCodecCtx->channels, outSize, inSamples);
#endif
- int outSamples = _resampler.resample(outPtr, // input
- reinterpret_cast<boost::int16_t*>(resampledOutput), //
output
- inSamples); // input..
+ int outSamples = _resampler.resample(outPtr, // input
+ reinterpret_cast<boost::int16_t*>(resampledOutput), // output
+ inSamples); // input..
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug("resampler returned %d samples ", outSamples);
+ log_debug("resampler returned %d samples ", outSamples);
#endif
- // make sure to set outPtr *after* we use it as input to the
resampler
+ // make sure to set outPtr *after* we use it as input to the resampler
outPtr = reinterpret_cast<boost::int16_t*>(resampledOutput);
av_free(output);
- if (expectedMaxOutSamples < outSamples) {
- log_error(" --- Computation of resampled samples (%d) <
then the actual returned samples (%d)",
- expectedMaxOutSamples, outSamples);
-
- 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", inSamples);
-
- log_debug(" output sample rate (assuming): %d", 44100);
- log_debug(" output channels (assuming): %d", 2);
- log_debug(" output samples: %d", outSamples);
-
- /// Memory errors...
- abort();
- }
-
- // Use the actual number of samples returned, multiplied
- // to get size in bytes (not two-byte samples) and for
- // stereo?
- outSize = outSamples * 2 * 2;
-
- }
- else {
- boost::uint8_t* newOutput = new boost::uint8_t[outSize];
- std::memcpy(newOutput, output, outSize);
- outPtr = reinterpret_cast<boost::int16_t*>(newOutput);
+ if (expectedMaxOutSamples < outSamples) {
+ log_error(" --- Computation of resampled samples (%d) < then the
actual returned samples (%d)",
+ expectedMaxOutSamples, outSamples);
+
+ 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", inSamples);
+
+ log_debug(" output sample rate (assuming): %d", 44100);
+ log_debug(" output channels (assuming): %d", 2);
+ log_debug(" output samples: %d", outSamples);
+
+ /// Memory errors...
+ abort();
+ }
+
+ // Use the actual number of samples returned, multiplied
+ // to get size in bytes (not two-byte samples) and for
+ // stereo?
+ outSize = outSamples * 2 * 2;
+
+ }
+ else {
+ boost::uint8_t* newOutput = new boost::uint8_t[outSize];
+ std::memcpy(newOutput, output, outSize);
+ outPtr = reinterpret_cast<boost::int16_t*>(newOutput);
av_free(output);
- }
+ }
- outputSize = outSize;
- return reinterpret_cast<uint8_t*>(outPtr);
+ outputSize = outSize;
+ return reinterpret_cast<uint8_t*>(outPtr);
}
int
http://git.savannah.gnu.org/cgit//commit/?id=ddfb00078eafa27116261f8c30750731d5080c84
commit ddfb00078eafa27116261f8c30750731d5080c84
Author: Benjamin Wolsey <address@hidden>
Date: Sat Oct 30 09:52:51 2010 +0200
Drop support for avcodec_decode_audio (Debian stable has the newer
function).
diff --git a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
index 12cb899..3fdb563 100644
--- a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
@@ -29,11 +29,7 @@
//#define GNASH_DEBUG_AUDIO_DECODING
-#ifdef FFMPEG_AUDIO2
-# define AVCODEC_DECODE_AUDIO avcodec_decode_audio2
-#else
-# define AVCODEC_DECODE_AUDIO avcodec_decode_audio
-#endif
+#define AVCODEC_DECODE_AUDIO avcodec_decode_audio2
namespace gnash {
namespace media {
http://git.savannah.gnu.org/cgit//commit/?id=b6015e75fcbd38ee61f85973015e343769168b04
commit b6015e75fcbd38ee61f85973015e343769168b04
Author: Benjamin Wolsey <address@hidden>
Date: Sat Oct 30 09:52:03 2010 +0200
We always use aligned memory, so drop the alternative.
diff --git a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
index cead0c7..12cb899 100644
--- a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
@@ -29,10 +29,6 @@
//#define GNASH_DEBUG_AUDIO_DECODING
-/* SIMD versions of DSPContext.float_to_int16_interleave() needs input
- and output buffers aligned to 16-byte boundaries */
-#define NEEDS_ALIGNED_MEMORY 1
-
#ifdef FFMPEG_AUDIO2
# define AVCODEC_DECODE_AUDIO avcodec_decode_audio2
#else
@@ -527,20 +523,12 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
// TODO: make this a private member, to reuse (see NetStreamFfmpeg in
0.8.3)
boost::uint8_t* output;
- if (NEEDS_ALIGNED_MEMORY)
- {
output = reinterpret_cast<boost::uint8_t*>(av_malloc(bufsize));
- if (output == NULL)
- {
+ if (!output) {
log_error(_("failed to allocate audio buffer."));
outputSize = 0;
return NULL;
}
- }
- else
- {
- output = new boost::uint8_t[bufsize];
- }
boost::int16_t* outPtr = reinterpret_cast<boost::int16_t*>(output);
@@ -565,36 +553,25 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
bufsize, inputSize, tmp, outSize);
#endif
- if (tmp < 0)
- {
+ if (tmp < 0) {
log_error(_("avcodec_decode_audio returned %d. Upgrading "
"ffmpeg/libavcodec might fix this issue."), tmp);
outputSize = 0;
- if (NEEDS_ALIGNED_MEMORY)
- av_free(output);
- else
- delete [] output;
- return NULL;
+ av_free(output);
}
- if (outSize < 2)
- {
+ if (outSize < 2) {
log_error(_("outputSize:%d after decoding %d bytes of input
audio "
"data. Upgrading ffmpeg/libavcodec might fix this issue."),
outputSize, inputSize);
outputSize = 0;
- if (NEEDS_ALIGNED_MEMORY)
- av_free(output);
- else
- delete [] output;
- return NULL;
+ av_free(output);
}
// Resampling is needed.
- if (_resampler.init(_audioCodecCtx))
- {
+ if (_resampler.init(_audioCodecCtx)) {
// Resampling is needed.
// Compute new size based on frame_size and
@@ -629,15 +606,11 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
#endif
// make sure to set outPtr *after* we use it as input to the
resampler
- outPtr = reinterpret_cast<boost::int16_t*>(resampledOutput);
+ outPtr = reinterpret_cast<boost::int16_t*>(resampledOutput);
- if (NEEDS_ALIGNED_MEMORY)
- av_free(output);
- else
- delete [] output;
+ av_free(output);
- if (expectedMaxOutSamples < outSamples)
- {
+ if (expectedMaxOutSamples < outSamples) {
log_error(" --- Computation of resampled samples (%d) <
then the actual returned samples (%d)",
expectedMaxOutSamples, outSamples);
@@ -660,10 +633,9 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
outSize = outSamples * 2 * 2;
}
- else if (NEEDS_ALIGNED_MEMORY)
- {
+ else {
boost::uint8_t* newOutput = new boost::uint8_t[outSize];
- memcpy(newOutput, output, outSize);
+ std::memcpy(newOutput, output, outSize);
outPtr = reinterpret_cast<boost::int16_t*>(newOutput);
av_free(output);
}
-----------------------------------------------------------------------
Summary of changes:
libmedia/ffmpeg/AudioDecoderFfmpeg.cpp | 641 +++++++++++++++-----------------
libmedia/ffmpeg/VideoDecoderFfmpeg.cpp | 2 -
libvaapi/VaapiSurfaceGLX.cpp | 4 +
macros/ffmpeg.m4 | 16 +-
4 files changed, 310 insertions(+), 353 deletions(-)
hooks/post-receive
--
Gnash
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Benjamin Wolsey <=