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[Gnash-commit] [SCM] Gnash branch, master, updated. release_0_8_9_final-
|
From: |
Gabriele Giacone |
|
Subject: |
[Gnash-commit] [SCM] Gnash branch, master, updated. release_0_8_9_final-1783-g8302351 |
|
Date: |
Mon, 02 Sep 2013 09:42:16 +0000 |
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- Log -----------------------------------------------------------------
http://git.savannah.gnu.org/cgit//commit/?id=830235118f95c57e58bb359bf8d69e6d8f2386f0
commit 830235118f95c57e58bb359bf8d69e6d8f2386f0
Author: Sandro Santilli <address@hidden>
Date: Sat Aug 24 04:51:10 2013 +0200
Fix unused parameter warning.
diff --git a/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
b/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
index 2d0c21c..81e2c49 100644
--- a/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
@@ -19,6 +19,7 @@
//
#include "AudioResamplerFfmpeg.h"
+#include "utility.h"
#include "log.h"
#include <cmath>
@@ -95,6 +96,7 @@ AudioResamplerFfmpeg::resample(boost::uint8_t** input, int
plane_size,
output, 0, MAX_AUDIO_FRAME_SIZE,
input, plane_size, samples);
#else
+ UNUSED( plane_size );
return audio_resample(_context, reinterpret_cast<short*>(*output),
reinterpret_cast<short*>(*input), samples);
#endif
http://git.savannah.gnu.org/cgit//commit/?id=9540d40a56c937455e607a7c981949a55b80d08c
commit 9540d40a56c937455e607a7c981949a55b80d08c
Author: Gabriele Giacone <address@hidden>
Date: Thu Aug 22 19:57:51 2013 +0200
Move MAX_AUDIO_FRAME_SIZE definition to common header.
diff --git a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
index 6a67771..b194f54 100644
--- a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
@@ -30,8 +30,6 @@
//#define GNASH_DEBUG_AUDIO_DECODING
-#define MAX_AUDIO_FRAME_SIZE 192000
-
namespace gnash {
namespace media {
namespace ffmpeg {
diff --git a/libmedia/ffmpeg/ffmpegHeaders.h b/libmedia/ffmpeg/ffmpegHeaders.h
index f8532e2..5cd3785 100644
--- a/libmedia/ffmpeg/ffmpegHeaders.h
+++ b/libmedia/ffmpeg/ffmpegHeaders.h
@@ -38,6 +38,8 @@
#endif
#endif
+#define MAX_AUDIO_FRAME_SIZE 192000
+
#ifdef HAVE_FFMPEG_AVCODEC_H
extern "C" {
# include <ffmpeg/avcodec.h>
http://git.savannah.gnu.org/cgit//commit/?id=d4d62aac67824382eec668abe574669e7d43e980
commit d4d62aac67824382eec668abe574669e7d43e980
Author: Gabriele Giacone <address@hidden>
Date: Thu Aug 22 03:29:20 2013 +0200
Bump minimum supported ffmpeg version ie. debian stable one.
diff --git a/macros/ffmpeg.m4 b/macros/ffmpeg.m4
index dd4b2c8..c3ef987 100644
--- a/macros/ffmpeg.m4
+++ b/macros/ffmpeg.m4
@@ -255,27 +255,14 @@ AC_DEFUN([GNASH_PATH_FFMPEG],
AC_MSG_RESULT($ffmpeg_version ($ffmpeg_num_version))
-dnl AC_EGREP_HEADER(avcodec_decode_audio2, ${avcodec_h}, [avfound=yes],
[avfound=no])
-
- if test -z "$ffmpeg_num_version" -o "$ffmpeg_num_version" -lt 512800; then
- AC_MSG_WARN([Wrong ffmpeg/libavcodec version! 51.28.0 or greater
required, $ffmpeg_version detected.])
- else
- ffmpeg_version_check=ok
- fi
+dnl minimum supported libavcodec version = Debian stable one = currently
Wheezy 7.0 53.35.00
- if test -z "$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!])
+ if test -z "$ffmpeg_num_version" -o "$ffmpeg_num_version" -lt 533500; then
+ AC_MSG_WARN([Wrong ffmpeg/libavcodec version! 53.35.0 or greater
required, $ffmpeg_version detected.])
else
- AC_DEFINE(FFMPEG_VP6A, 1, [Define if ffmpeg can play VP6A.])
+ ffmpeg_version_check=ok
fi
- if test -z "$ffmpeg_num_version" -o "$ffmpeg_num_version" -lt 514600; then
- dnl 51.46.0 (r10741) or higher required for CODEC_ID_NELLYMOSER
- AC_MSG_WARN([This version of ffmpeg/libavcodec ($ffmpeg_version) is not
able to decode NELLYMOSER encoded audio: 51.46.0 (r10741) or higher required!])
- else
- AC_DEFINE(FFMPEG_NELLYMOSER, 1, [Define if ffmpeg can decode NELLYMOSER
audio])
- fi
else
AC_MSG_WARN([Could not check ffmpeg version (can't find avcodec.h file)])
# ffmpeg_version_check=ok # this is NOT ok, why would it be ?!
http://git.savannah.gnu.org/cgit//commit/?id=514d488671aa3c987910fb93001defcc60a37528
commit 514d488671aa3c987910fb93001defcc60a37528
Author: Gabriele Giacone <address@hidden>
Date: Thu Aug 15 15:56:58 2013 +0200
Resample formats other than S16.
Introduce libswresample/livavresample.
diff --git a/configure.ac b/configure.ac
index 33a136c..2d9b3b0 100644
--- a/configure.ac
+++ b/configure.ac
@@ -3571,6 +3571,23 @@ if test x"${build_media_ffmpeg}" = x"yes"; then
DEB_INSTALL([libswscale-dev])
PKG_ALTERNATIVE(or reconfigure with --enable-media=gst)
fi
+
+ if test x"${have_ffmpeg_libavutil}" = "xno"; then
+ PKG_ERR([No libavutil development package installed!])
+ PKG_SUGGEST([You can install libavutil from http://ffmpeg.org])
+ DEB_INSTALL([libavutil-dev])
+ PKG_ALTERNATIVE(or reconfigure with --enable-media=gst)
+ fi
+
+ if test x"${ffmpeg_aac_float_planar}" = "xyes" -a \
+ x"${have_ffmpeg_swresample}" = "xno" -a \
+ x"${have_libav_avresample}" = "xno"; then
+ PKG_REC([Please install libswresample/libavresample development package.
If you don't, Gnash will be unable to resample some formats (e.g. AAC on
youtube)])
+ PKG_SUGGEST([You can install libavresample from http://libav.org])
+ PKG_SUGGEST([You can install libswresample from http://ffmpeg.org])
+ DEB_INSTALL([libavresample-dev])
+ PKG_ALTERNATIVE(or reconfigure with --enable-media=gst)
+ fi
fi
if test x$build_cairo = xyes; then
diff --git a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
index e745e1e..6a67771 100644
--- a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
@@ -497,24 +497,16 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
assert(inputSize);
- const size_t bufsize = MAX_AUDIO_FRAME_SIZE;
+ size_t outSize = MAX_AUDIO_FRAME_SIZE;
// 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) {
+ boost::int16_t* outPtr =
reinterpret_cast<boost::int16_t*>(av_malloc(outSize));
+ if (!outPtr) {
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;
-
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug("AudioDecoderFfmpeg: about to decode %d bytes; "
"ctx->channels:%d, ctx->frame_size:%d",
@@ -535,18 +527,18 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
int tmp = avcodec_decode_audio4(_audioCodecCtx, frm, &got_frm, &pkt);
#ifdef GNASH_DEBUG_AUDIO_DECODING
+ const char* fmtname = av_get_sample_fmt_name(_audioCodecCtx->sample_fmt);
log_debug(" decodeFrame | frm->nb_samples: %d | &got_frm: %d | "
"returned %d | inputSize: %d",
frm->nb_samples, got_frm, tmp, inputSize);
#endif
+ int plane_size;
if (tmp >= 0 && got_frm) {
- int ch, plane_size;
- int planar = av_sample_fmt_is_planar(_audioCodecCtx->sample_fmt);
int data_size = av_samples_get_buffer_size( &plane_size,
_audioCodecCtx->channels, frm->nb_samples,
_audioCodecCtx->sample_fmt, 1);
- if (outSize < data_size) {
+ if (static_cast<int>(outSize) < data_size) {
log_error(_("output buffer size is too small for the current frame
"
"(%d < %d)"), outSize, data_size);
return NULL;
@@ -554,19 +546,22 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
memcpy(outPtr, frm->extended_data[0], plane_size);
+#if !(defined(HAVE_SWRESAMPLE_H) || defined(HAVE_AVRESAMPLE_H))
+ int planar = av_sample_fmt_is_planar(_audioCodecCtx->sample_fmt);
if (planar && _audioCodecCtx->channels > 1) {
uint8_t *out = ((uint8_t *)outPtr) + plane_size;
- for (ch = 1; ch < _audioCodecCtx->channels; ch++) {
+ for (int ch = 1; ch < _audioCodecCtx->channels; ch++) {
memcpy(out, frm->extended_data[ch], plane_size);
out += plane_size;
}
}
+#endif
+
outSize = data_size;
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug(" decodeFrame | planar: %d | _audioCodecCtx->sample_fmt: %d
| "
- "av_get_sample_fmt_name: %s | outSize: %d",
- planar, _audioCodecCtx->sample_fmt,
- av_get_sample_fmt_name(_audioCodecCtx->sample_fmt), outSize);
+ log_debug(" decodeFrame | fmt: %d | fmt_name: %s | planar: %d | "
+ "plane_size: %d | outSize: %d",
+ _audioCodecCtx->sample_fmt, fmtname, planar, plane_size, outSize);
#endif
} else {
if (tmp < 0)
@@ -577,7 +572,6 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t* input,
log_error(_("Upgrading ffmpeg/libavcodec might fix this issue."));
outputSize = 0;
av_freep(&frm);
- av_free(output);
return NULL;
}
@@ -600,26 +594,27 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
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(" decodeFrame | Calling the resampler, resampleFactor: %d | "
+ "in %d hz %d ch %d bytes %d samples, %s fmt", resampleFactor,
+ _audioCodecCtx->sample_rate, _audioCodecCtx->channels, outSize,
+ inSamples, fmtname);
+ log_debug(" decodeFrame | out 44100 hz 2 ch %d bytes",
+ resampledFrameSize);
#endif
- int outSamples = _resampler.resample(outPtr, // input
- reinterpret_cast<boost::int16_t*>(resampledOutput), // output
- inSamples); // input..
+ int outSamples = _resampler.resample(frm->extended_data, // input
+ plane_size, // input
+ frm->nb_samples, // input
+ &resampledOutput); // output
+
+ // make sure to set outPtr *after* we use it as input to the resampler
+ outPtr = reinterpret_cast<boost::int16_t*>(resampledOutput);
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug("resampler returned %d samples ", outSamples);
#endif
- // make sure to set outPtr *after* we use it as input to the resampler
- outPtr = reinterpret_cast<boost::int16_t*>(resampledOutput);
-
- av_free(output);
+ av_freep(&frm);
if (expectedMaxOutSamples < outSamples) {
log_error(_(" --- Computation of resampled samples (%d) < then the
actual returned samples (%d)"),
@@ -646,10 +641,9 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
}
else {
boost::uint8_t* newOutput = new boost::uint8_t[outSize];
- std::memcpy(newOutput, output, outSize);
+ std::memcpy(newOutput, outPtr, outSize);
outPtr = reinterpret_cast<boost::int16_t*>(newOutput);
av_freep(&frm);
- av_free(output);
}
outputSize = outSize;
diff --git a/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
b/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
index 98ac8b5..2d0c21c 100644
--- a/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
@@ -33,48 +33,73 @@ AudioResamplerFfmpeg::AudioResamplerFfmpeg()
:_context(NULL)
{
}
-
-AudioResamplerFfmpeg::~AudioResamplerFfmpeg()
-{
- if ( _context ) {
- audio_resample_close( _context );
- }
+AudioResamplerFfmpeg::~AudioResamplerFfmpeg() {
+ if (_context) {
+#ifdef HAVE_SWRESAMPLE_H
+ swr_free(&_context);
+#elif HAVE_AVRESAMPLE_H
+ avresample_close(_context);
+ avresample_free(&_context);
+#else
+ audio_resample_close(_context);
+#endif
+ }
}
bool
-AudioResamplerFfmpeg::init( AVCodecContext* ctx )
-{
- if ( (ctx->sample_rate != 44100) || (ctx->channels != 2) ) {
- if ( ! _context ) {
-#if LIBAVCODEC_VERSION_MAJOR >= 53
- _context = av_audio_resample_init(
- 2, ctx->channels, 44100, ctx->sample_rate,
- AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16,
- 16, 10, 0, 0.8
+AudioResamplerFfmpeg::init(AVCodecContext* ctx) {
+ if ((ctx->sample_rate != 44100) ||
+#if defined(HAVE_SWRESAMPLE_H) || defined(HAVE_AVRESAMPLE_H)
+ (ctx->sample_fmt != AV_SAMPLE_FMT_S16) ||
+#endif
+ (ctx->channels != 2)) {
+ if (! _context) {
+#ifdef HAVE_SWRESAMPLE_H
+ _context = swr_alloc();
+#elif HAVE_AVRESAMPLE_H
+ _context = avresample_alloc_context();
#else
- _context = audio_resample_init(
- 2, ctx->channels, 44100, ctx->sample_rate
+ _context = av_audio_resample_init(2, ctx->channels,
+ 44100, ctx->sample_rate,
+ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16,
+ 16, 10, 0, 0.8);
+#endif
+#if defined(HAVE_SWRESAMPLE_H) || defined(HAVE_AVRESAMPLE_H)
+ av_opt_set_int(_context, "in_channel_layout", ctx->channel_layout,
0);
+ av_opt_set_int(_context, "out_channel_layout",
AV_CH_LAYOUT_STEREO, 0);
+ av_opt_set_int(_context, "in_sample_rate", ctx->sample_rate, 0);
+ av_opt_set_int(_context, "out_sample_rate", 44100, 0);
+ av_opt_set_int(_context, "in_sample_fmt", ctx->sample_fmt, 0);
+ av_opt_set_int(_context, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
#endif
- );
+#ifdef HAVE_SWRESAMPLE_H
+ swr_init(_context);
+#elif HAVE_AVRESAMPLE_H
+ avresample_open(_context);
+#endif
+ }
+ return true;
}
-
- return true;
- }
-
- return false;
+ return false;
}
int
-AudioResamplerFfmpeg::resample(
- boost::int16_t* input,
- boost::int16_t* output,
- int samples
- )
-{
- return audio_resample( _context, output, input, samples );
+AudioResamplerFfmpeg::resample(boost::uint8_t** input, int plane_size,
+ int samples, boost::uint8_t** output) {
+#ifdef HAVE_SWRESAMPLE_H
+ return swr_convert(_context,
+ output, MAX_AUDIO_FRAME_SIZE,
+ const_cast<const uint8_t**>(input), samples);
+#elif HAVE_AVRESAMPLE_H
+ return avresample_convert(_context,
+ output, 0, MAX_AUDIO_FRAME_SIZE,
+ input, plane_size, samples);
+#else
+ return audio_resample(_context, reinterpret_cast<short*>(*output),
+ reinterpret_cast<short*>(*input), samples);
+#endif
}
-
} // gnash.media.ffmpeg namespace
} // gnash.media namespace
} // namespace gnash
diff --git a/libmedia/ffmpeg/AudioResamplerFfmpeg.h
b/libmedia/ffmpeg/AudioResamplerFfmpeg.h
index e77aec4..3dc2748 100644
--- a/libmedia/ffmpeg/AudioResamplerFfmpeg.h
+++ b/libmedia/ffmpeg/AudioResamplerFfmpeg.h
@@ -56,21 +56,29 @@ public:
/// @param input
/// A pointer to the audio data that needs resampling
///
- /// @param output
- /// A pointer to where the resampled output should be placed
+ /// @param plane_size
+ /// Input plane/line size
///
/// @param samples
/// Number of samples in the audio
///
+ /// @param output
+ /// A pointer to where the resampled output should be placed
+ ///
/// @return the number of samples in the output data.
///
- DSOEXPORT int resample(
- boost::int16_t* input, boost::int16_t* output, int samples
- );
+ DSOEXPORT int resample(boost::uint8_t** input, int plane_size,
+ int samples, boost::uint8_t** output);
private:
- // The container of the resample format information.
- ReSampleContext* _context;
+ // The container of the resample format information.
+#ifdef HAVE_SWRESAMPLE_H
+ SwrContext* _context;
+#elif HAVE_AVRESAMPLE_H
+ AVAudioResampleContext* _context;
+#else
+ ReSampleContext* _context;
+#endif
};
} // gnash.media.ffmpeg namespace
diff --git a/libmedia/ffmpeg/ffmpegHeaders.h b/libmedia/ffmpeg/ffmpegHeaders.h
index a05cee7..f8532e2 100644
--- a/libmedia/ffmpeg/ffmpegHeaders.h
+++ b/libmedia/ffmpeg/ffmpegHeaders.h
@@ -100,4 +100,50 @@ extern "C" {
#define CODECID CodecID
#endif
+#ifdef HAVE_SWRESAMPLE_H
+extern "C" {
+#include <swresample.h>
+}
+#endif
+
+#ifdef HAVE_FFMPEG_SWRESAMPLE_H
+extern "C" {
+#include <ffmpeg/swresample.h>
+}
+#define HAVE_SWRESAMPLE_H 1
+#endif
+
+#ifdef HAVE_LIBSWRESAMPLE_SWRESAMPLE_H
+extern "C" {
+#include <libswresample/swresample.h>
+}
+#define HAVE_SWRESAMPLE_H 1
+#endif
+
+#ifdef HAVE_AVRESAMPLE_H
+extern "C" {
+#include <avresample.h>
+}
+#endif
+
+#ifdef HAVE_LIBAV_AVRESAMPLE_H
+extern "C" {
+#include <libav/avresample.h>
+}
+#define HAVE_AVRESAMPLE_H 1
+#endif
+
+#ifdef HAVE_LIBAVRESAMPLE_AVRESAMPLE_H
+extern "C" {
+#include <libavresample/avresample.h>
+}
+#define HAVE_AVRESAMPLE_H 1
+#endif
+
+#ifdef HAVE_LIBAVUTIL_OPT_H
+extern "C" {
+#include <libavutil/opt.h>
+}
+#endif
+
#endif // GNASH_MEDIA_FFMPEG_HEADERS_H
diff --git a/macros/ffmpeg.m4 b/macros/ffmpeg.m4
index 7e042d9..dd4b2c8 100644
--- a/macros/ffmpeg.m4
+++ b/macros/ffmpeg.m4
@@ -281,6 +281,21 @@ dnl AC_EGREP_HEADER(avcodec_decode_audio2, ${avcodec_h},
[avfound=yes], [avfou
# ffmpeg_version_check=ok # this is NOT ok, why would it be ?!
fi
+dnl Check if installed ffmpeg/libav already switched aac decoding from S16 to
+dnl float planar format (FLTP). If recent enough, we'll recommend to install
+dnl a dedicate resampling library to get aac properly decoded.
+dnl Commit in question (3d3cf6745e2a5dc9c377244454c3186d75b177fa) corresponds
+dnl to libavcodec ffmpeg version 54.77.100 / libav version 54.33.0.
+dnl | ffmpeg | 1.0 (54.59.100) S16 | 1.1 (54.86.100) FLTP |
+dnl | libav | 0.8 (53.35.0) S16 | 9 (54.35.0) FLTP |
+
+ if test ${ffmpeg_major_version} -ge 55 -o \
+ \( ${ffmpeg_major_version} -eq 54 -a ${ffmpeg_micro_version} -eq 0 \) -o \
+ \( ${ffmpeg_major_version} -eq 54 -a ${ffmpeg_minor_version} -ge 77 -a \
+ ${ffmpeg_micro_version} -ge 100 \); then
+ ffmpeg_aac_float_planar=yes
+ fi
+
LIBAVCODEC_IDENT=${ffmpeg_version}
FFMPEG_CFLAGS="-D__STDC_CONSTANT_MACROS ${ac_cv_path_ffmpeg_incl}"
@@ -319,6 +334,46 @@ dnl AC_EGREP_HEADER(avcodec_decode_audio2, ${avcodec_h},
[avfound=yes], [avfou
ffmpeg_version_check=
fi
+ AC_MSG_CHECKING([for libavutil/opt.h])
+ have_ffmpeg_libavutil=no
+ if test -f "${ffmpeg_top_incl}/libavutil/opt.h"; then
+ have_ffmpeg_libavutil=yes
+ AC_DEFINE(HAVE_LIBAVUTIL_OPT_H, 1, [Define if libavutil/opt.h is found])
+ fi
+ AC_MSG_RESULT($have_ffmpeg_libavutil)
+
+ AC_MSG_CHECKING([for swresample.h])
+ have_ffmpeg_swresample=no
+ if test -f "${ffmpeg_top_incl}/ffmpeg/swresample.h"; then
+ have_ffmpeg_swresample=yes
+ AC_DEFINE(HAVE_FFMPEG_SWRESAMPLE_H, 1, [Define if swresample.h is found])
+ fi
+ if test -f "${ffmpeg_top_incl}/libswresample/swresample.h"; then
+ have_ffmpeg_swresample=yes
+ AC_DEFINE(HAVE_LIBSWRESAMPLE_SWRESAMPLE_H, 1, [Define if swresample.h is
found])
+ fi
+ if test -f "${ffmpeg_top_incl}/swresample.h"; then
+ have_ffmpeg_swresample=yes
+ AC_DEFINE(HAVE_SWRESAMPLE_H, 1, [Define if swresample.h is found])
+ fi
+ AC_MSG_RESULT($have_ffmpeg_swresample)
+
+ AC_MSG_CHECKING([for avresample.h])
+ have_libav_avresample=no
+ if test -f "${ffmpeg_top_incl}/libav/avresample.h"; then
+ have_libav_avresample=yes
+ AC_DEFINE(HAVE_LIBAV_AVRESAMPLE_H, 1, [Define if avresample.h is found])
+ fi
+ if test -f "${ffmpeg_top_incl}/libavresample/avresample.h"; then
+ have_libav_avresample=yes
+ AC_DEFINE(HAVE_LIBAVRESAMPLE_AVRESAMPLE_H, 1, [Define if avresample.h is
found])
+ fi
+ if test -f "${ffmpeg_top_incl}/avresample.h"; then
+ have_libav_avresample=yes
+ AC_DEFINE(HAVE_AVRESAMPLE_H, 1, [Define if avresample.h is found])
+ fi
+ AC_MSG_RESULT($have_libav_avresample)
+
AC_MSG_CHECKING([for libavcodec/vaapi.h])
have_ffmpeg_vaapi="no"
if test -f "${ffmpeg_top_incl}/ffmpeg/vaapi.h"; then
@@ -625,6 +680,51 @@ dnl AC_EGREP_HEADER(avcodec_decode_audio2, ${avcodec_h},
[avfound=yes], [avfou
fi
dnl End of SWSCALE library looking }
+ dnl Look for the {SW,AV}RESAMPLE libraries {
+ dnl
+ AC_MSG_CHECKING([for libswresample library])
+ if test x"$PKG_CONFIG" != x -a x${cross_compiling} = xno; then
+ $PKG_CONFIG --exists libswresample && libswresample=`$PKG_CONFIG
--libs-only-l libswresample`
+ else
+ libswresample=""
+ fi
+ if test x"${libswresample}" = x; then
+ if test -f ${top_lib_dir}/libswresample.a -o -f
${top_lib_dir}/libswresample.${shlibext}; then
+ ac_cv_path_ffmpeg_lib="${ac_cv_path_ffmpeg_lib} -lswresample"
+ AC_MSG_RESULT(yes)
+ else
+ AC_MSG_RESULT(no)
+ if test x${cross_compiling} = xno; then
+ AC_CHECK_LIB(swresample, swresample,
[ac_cv_path_ffmpeg_lib="${ac_cv_path_ffmpeg_lib} -lswresample"])
+ fi
+ fi
+ else
+ AC_MSG_RESULT(${libswresample})
+ ac_cv_path_ffmpeg_lib="${ac_cv_path_ffmpeg_lib} ${libswresample}"
+ fi
+
+ AC_MSG_CHECKING([for libavresample library])
+ if test x"$PKG_CONFIG" != x -a x${cross_compiling} = xno; then
+ $PKG_CONFIG --exists libavresample && libavresample=`$PKG_CONFIG
--libs-only-l libavresample`
+ else
+ libavresample=""
+ fi
+ if test x"${libavresample}" = x; then
+ if test -f ${top_lib_dir}/libavresample.a -o -f
${top_lib_dir}/libavresample.${shlibext}; then
+ ac_cv_path_ffmpeg_lib="${ac_cv_path_ffmpeg_lib} -lavresample"
+ AC_MSG_RESULT(yes)
+ else
+ AC_MSG_RESULT(no)
+ if test x${cross_compiling} = xno; then
+ AC_CHECK_LIB(avresample, avresample,
[ac_cv_path_ffmpeg_lib="${ac_cv_path_ffmpeg_lib} -lavresample"])
+ fi
+ fi
+ else
+ AC_MSG_RESULT(${libavresample})
+ ac_cv_path_ffmpeg_lib="${ac_cv_path_ffmpeg_lib} ${libavresample}"
+ fi
+ dnl End of {SW,AV}RESAMPLE libraries looking }
+
fi
dnl End of all optional library tests }
http://git.savannah.gnu.org/cgit//commit/?id=2a76028d8b2529db4dc3aca7efd8c0fa029ad18a
commit 2a76028d8b2529db4dc3aca7efd8c0fa029ad18a
Author: Gabriele Giacone <address@hidden>
Date: Thu Aug 15 14:06:41 2013 +0200
AudioDecoderFfmpeg.cpp: use avcodec_decode_audio4.
Replace _audio3 and _audio2 with _audio4.
Suppress warnings about deprecated _audio3.
diff --git a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
index f45a42d..e745e1e 100644
--- a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
@@ -30,12 +30,6 @@
//#define GNASH_DEBUG_AUDIO_DECODING
-#if LIBAVCODEC_VERSION_MAJOR >= 53
-#define AVCODEC_DECODE_AUDIO avcodec_decode_audio3
-#else
-#define AVCODEC_DECODE_AUDIO avcodec_decode_audio2
-#endif
-
#define MAX_AUDIO_FRAME_SIZE 192000
namespace gnash {
@@ -523,45 +517,66 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t*
input,
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug("AudioDecoderFfmpeg: about to decode %d bytes; "
- "ctx->channels:%d, avctx->frame_size:%d",
+ "ctx->channels:%d, ctx->frame_size:%d",
inputSize, _audioCodecCtx->channels, _audioCodecCtx->frame_size);
#endif
// older ffmpeg versions didn't accept a const input..
-#if LIBAVCODEC_VERSION_MAJOR >= 53
AVPacket pkt;
+ int got_frm = 0;
av_init_packet(&pkt);
- pkt.data = (uint8_t*) input;
+ pkt.data = const_cast<uint8_t*>(input);
pkt.size = inputSize;
-#endif
- int tmp = AVCODEC_DECODE_AUDIO(_audioCodecCtx, outPtr, &outSize,
-#if LIBAVCODEC_VERSION_MAJOR >= 53
- &pkt);
-#else
- input, inputSize);
-#endif
+ AVFrame *frm = avcodec_alloc_frame();
+ if (!frm) {
+ log_error(_("failed to allocate frame."));
+ return NULL;
+ }
+ int tmp = avcodec_decode_audio4(_audioCodecCtx, frm, &got_frm, &pkt);
#ifdef GNASH_DEBUG_AUDIO_DECODING
- log_debug(" avcodec_decode_audio[2](ctx, bufptr, %d, input, %d) "
- "returned %d; set frame_size=%d",
- bufsize, inputSize, tmp, outSize);
+ log_debug(" decodeFrame | frm->nb_samples: %d | &got_frm: %d | "
+ "returned %d | inputSize: %d",
+ frm->nb_samples, got_frm, tmp, inputSize);
#endif
- if (tmp < 0) {
- log_error(_("avcodec_decode_audio returned %d. Upgrading "
- "ffmpeg/libavcodec might fix this issue."), tmp);
- outputSize = 0;
+ if (tmp >= 0 && got_frm) {
+ int ch, plane_size;
+ int planar = av_sample_fmt_is_planar(_audioCodecCtx->sample_fmt);
+ int data_size = av_samples_get_buffer_size( &plane_size,
+ _audioCodecCtx->channels, frm->nb_samples,
+ _audioCodecCtx->sample_fmt, 1);
+ if (outSize < data_size) {
+ log_error(_("output buffer size is too small for the current frame
"
+ "(%d < %d)"), outSize, data_size);
+ return NULL;
+ }
- av_free(output);
- return NULL;
- }
+ memcpy(outPtr, frm->extended_data[0], plane_size);
- if (outSize < 2) {
- log_error(_("outputSize:%d after decoding %d bytes of input audio "
- "data. Upgrading ffmpeg/libavcodec might fix this issue."),
- outputSize, inputSize);
+ if (planar && _audioCodecCtx->channels > 1) {
+ uint8_t *out = ((uint8_t *)outPtr) + plane_size;
+ for (ch = 1; ch < _audioCodecCtx->channels; ch++) {
+ memcpy(out, frm->extended_data[ch], plane_size);
+ out += plane_size;
+ }
+ }
+ outSize = data_size;
+#ifdef GNASH_DEBUG_AUDIO_DECODING
+ log_debug(" decodeFrame | planar: %d | _audioCodecCtx->sample_fmt: %d
| "
+ "av_get_sample_fmt_name: %s | outSize: %d",
+ planar, _audioCodecCtx->sample_fmt,
+ av_get_sample_fmt_name(_audioCodecCtx->sample_fmt), outSize);
+#endif
+ } else {
+ if (tmp < 0)
+ log_error(_("avcodec_decode_audio returned %d."), tmp);
+ if (outSize < 2)
+ log_error(_("outputSize:%d after decoding %d bytes of input audio "
+ "data."), outputSize, inputSize);
+ log_error(_("Upgrading ffmpeg/libavcodec might fix this issue."));
outputSize = 0;
-
+ av_freep(&frm);
av_free(output);
return NULL;
}
@@ -633,6 +648,7 @@ AudioDecoderFfmpeg::decodeFrame(const boost::uint8_t* input,
boost::uint8_t* newOutput = new boost::uint8_t[outSize];
std::memcpy(newOutput, output, outSize);
outPtr = reinterpret_cast<boost::int16_t*>(newOutput);
+ av_freep(&frm);
av_free(output);
}
-----------------------------------------------------------------------
Summary of changes:
configure.ac | 17 ++++
libmedia/ffmpeg/AudioDecoderFfmpeg.cpp | 128 ++++++++++++++++--------------
libmedia/ffmpeg/AudioResamplerFfmpeg.cpp | 89 ++++++++++++++-------
libmedia/ffmpeg/AudioResamplerFfmpeg.h | 22 ++++--
libmedia/ffmpeg/ffmpegHeaders.h | 48 +++++++++++
macros/ffmpeg.m4 | 121 ++++++++++++++++++++++++----
6 files changed, 310 insertions(+), 115 deletions(-)
hooks/post-receive
--
Gnash
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