[Gnash-commit] [SCM] Gnash branch, master, updated. release_0_8_9_final-2220-g98f4523

[Sandro Santilli]

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http://git.savannah.gnu.org/cgit//commit/?id=98f45235e1a89b9834de5095ae9eb630019e7b8d


commit 98f45235e1a89b9834de5095ae9eb630019e7b8d
Author: Sandro Santilli <address@hidden>
Date:   Wed Sep 23 15:47:15 2015 +0200

    Update item about bug #45722

diff --git a/NEWS b/NEWS
index 266863d..f63b8d7 100644
--- a/NEWS
+++ b/NEWS
@@ -12,7 +12,7 @@ Caveats:
 
 Improvements since 0.8.10 release are:
 
- * Fix loud screech after playing PCM audio with GST backend (#45722)
+ * Fix screeches and crashes playing PCM audio (#45722)
  * Fix corrupted WAV file on --audio-dump (#45887)
  * Fix callback registration issue in ExternalInterface (#37223)
  * Fix possible out-of-bound read in parser (#43865)

http://git.savannah.gnu.org/cgit//commit/?id=09fb39cd5b7d2956e87c46cda58c57359d7d935f


commit 09fb39cd5b7d2956e87c46cda58c57359d7d935f
Author: Nutchanon Wetchasit <address@hidden>
Date:   Wed Sep 23 15:40:56 2015 +0200

    Fix wrong resampling factor and sample count calculation in FFmpeg-based 
audio resampling.

diff --git a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp 
b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
index a0c64f9..2244a6b 100644
--- a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
@@ -544,13 +544,26 @@ AudioDecoderFfmpeg::decodeFrame(const std::uint8_t* input,
 
         // Compute new size based on frame_size and
         // resampling configuration
-        double resampleFactor = (44100.0/_audioCodecCtx->sample_rate) * 
(2.0/_audioCodecCtx->channels);
+
+        // Find out the needed sample rate scaling
+        double resampleFactor = 44100.0/_audioCodecCtx->sample_rate;
+
+        // Compute total number of input samples
+        int inSamples = outSize;
         bool stereo = _audioCodecCtx->channels > 1 ? true : false;
-        int inSamples = stereo ? outSize >> 2 : outSize >> 1;
 
+        if (stereo) inSamples = inSamples >> 1;
+        if (_audioCodecCtx->sample_fmt == AV_SAMPLE_FMT_S16 ||
+            _audioCodecCtx->sample_fmt == AV_SAMPLE_FMT_S16P) {
+            inSamples = inSamples >> 1;
+        }
+
+        // Compute total number of output samples
         int expectedMaxOutSamples = std::ceil(inSamples*resampleFactor);
 
-        // *channels *sampleSize 
+        // Compute output buffer size (in bytes); by multiplying
+        // output samples count with output sample format's frame size,
+        // which is number of bytes per sample (2) times channels (2).
         int resampledFrameSize = expectedMaxOutSamples*2*2;
 
         // Allocate just the required amount of bytes

http://git.savannah.gnu.org/cgit//commit/?id=f71f77ef1dc030ac091b6f3beafe9975d725225f


commit f71f77ef1dc030ac091b6f3beafe9975d725225f
Author: Nutchanon Wetchasit <address@hidden>
Date:   Wed Sep 23 15:40:42 2015 +0200

    Use the original sample rate when utilizing FFmpeg 8-bit PCM codec.

diff --git a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp 
b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
index 0d08a33..a0c64f9 100644
--- a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
@@ -283,8 +283,7 @@ void AudioDecoderFfmpeg::setup(const AudioInfo& info)
                 break;
 
             case AV_CODEC_ID_PCM_U8:
-                // Either FFMPEG or the parser are getting this wrong.
-                _audioCodecCtx->sample_rate = info.sampleRate / 2;
+                _audioCodecCtx->sample_rate = info.sampleRate;
                 _audioCodecCtx->channels = (info.stereo ? 2 : 1);
                 break;
             case AV_CODEC_ID_PCM_S16LE:

http://git.savannah.gnu.org/cgit//commit/?id=1ac2492ce7ca4bcc241cc5b7e0a5542dbd22f429


commit 1ac2492ce7ca4bcc241cc5b7e0a5542dbd22f429
Author: Nutchanon Wetchasit <address@hidden>
Date:   Wed Sep 23 15:39:57 2015 +0200

    Use a correct FFmpeg codec ID for 8-bit PCM sound.

diff --git a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp 
b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
index c98e2d8..0d08a33 100644
--- a/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioDecoderFfmpeg.cpp
@@ -181,9 +181,17 @@ void AudioDecoderFfmpeg::setup(const AudioInfo& info)
             case AUDIO_CODEC_UNCOMPRESSED:
             case AUDIO_CODEC_RAW:
                 if (info.sampleSize == 2) {
+                    // Flash's 16-bit UNCOMPRESSED and RAW audio comes as
+                    // a signed PCM format. UNCOMPRESSED audio always use
+                    // little-endian byte order. RAW audio uses encoder's
+                    // native byte order; as majority of encoders run
+                    // on little-endian machine, we use little-endian for it.
                     codec_id = AV_CODEC_ID_PCM_S16LE;
                 } else {
-                    codec_id = AV_CODEC_ID_PCM_S8;
+                    // Flash's 8-bit UNCOMPRESSED and RAW audio comes as an
+                    // unsigned PCM format. No difference between
+                    // UNCOMPRESSED and RAW.
+                    codec_id = AV_CODEC_ID_PCM_U8;
                 }
                 break;
 
@@ -274,7 +282,7 @@ void AudioDecoderFfmpeg::setup(const AudioInfo& info)
             case AV_CODEC_ID_MP3:
                 break;
 
-            case AV_CODEC_ID_PCM_S8:
+            case AV_CODEC_ID_PCM_U8:
                 // Either FFMPEG or the parser are getting this wrong.
                 _audioCodecCtx->sample_rate = info.sampleRate / 2;
                 _audioCodecCtx->channels = (info.stereo ? 2 : 1);
diff --git a/libmedia/ffmpeg/ffmpegHeaders.h b/libmedia/ffmpeg/ffmpegHeaders.h
index 53b5b18..2e4944e 100644
--- a/libmedia/ffmpeg/ffmpegHeaders.h
+++ b/libmedia/ffmpeg/ffmpegHeaders.h
@@ -107,6 +107,7 @@ extern "C" {
 #define AV_CODEC_ID_NELLYMOSER CODEC_ID_NELLYMOSER
 #define AV_CODEC_ID_NONE CODEC_ID_NONE
 #define AV_CODEC_ID_PCM_S8 CODEC_ID_PCM_S8
+#define AV_CODEC_ID_PCM_U8 CODEC_ID_PCM_U8
 #define AV_CODEC_ID_PCM_S16LE CODEC_ID_PCM_S16LE
 #define AV_CODEC_ID_PCM_U16LE CODEC_ID_PCM_U16LE
 #define AV_CODEC_ID_VP6A CODEC_ID_VP6A

http://git.savannah.gnu.org/cgit//commit/?id=058c338618aa3dcdaa92834c601c6b96f4bc2b9f


commit 058c338618aa3dcdaa92834c601c6b96f4bc2b9f
Author: Nutchanon Wetchasit <address@hidden>
Date:   Wed Sep 23 15:39:42 2015 +0200

    Don't skip resampling non-16 bit audio when swresample.h and avresample.h 
are both unavailable.

diff --git a/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp 
b/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
index bd2f864..87cc558 100644
--- a/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
@@ -49,9 +49,7 @@ AudioResamplerFfmpeg::~AudioResamplerFfmpeg() {
 bool
 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

http://git.savannah.gnu.org/cgit//commit/?id=f3e1688fd5c9a2bd57970996c5c200aa387ce844


commit f3e1688fd5c9a2bd57970996c5c200aa387ce844
Author: Nutchanon Wetchasit <address@hidden>
Date:   Wed Sep 23 15:39:29 2015 +0200

    Use source sample format when calling FFmpeg/libAV audio resampler.

diff --git a/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp 
b/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
index 0b0dc6c..bd2f864 100644
--- a/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
+++ b/libmedia/ffmpeg/AudioResamplerFfmpeg.cpp
@@ -61,7 +61,7 @@ AudioResamplerFfmpeg::init(AVCodecContext* ctx) {
 #else
             _context = av_audio_resample_init(2, ctx->channels,
                 44100, ctx->sample_rate,
-                AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16,
+                AV_SAMPLE_FMT_S16, ctx->sample_fmt,
                 16, 10, 0, 0.8);
 #endif
 #if defined(HAVE_SWRESAMPLE_H) || defined(HAVE_AVRESAMPLE_H)

-----------------------------------------------------------------------

Summary of changes:
 NEWS                                     |    2 +-
 libmedia/ffmpeg/AudioDecoderFfmpeg.cpp   |   34 +++++++++++++++++++++++------
 libmedia/ffmpeg/AudioResamplerFfmpeg.cpp |    4 +--
 libmedia/ffmpeg/ffmpegHeaders.h          |    1 +
 4 files changed, 30 insertions(+), 11 deletions(-)


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