Index: media/ffmpeg/ffmpeg_common.cc |
diff --git a/media/ffmpeg/ffmpeg_common.cc b/media/ffmpeg/ffmpeg_common.cc |
index 3d3a8b89957d36be73c1480b1a63788dee3049d9..b65ceddeb0fd17185f15843196e3feda31fe6d26 100644 |
--- a/media/ffmpeg/ffmpeg_common.cc |
+++ b/media/ffmpeg/ffmpeg_common.cc |
@@ -14,6 +14,7 @@ |
#include "media/base/decoder_buffer.h" |
#include "media/base/video_decoder_config.h" |
#include "media/base/video_util.h" |
+#include "media/media_features.h" |
namespace media { |
@@ -67,6 +68,12 @@ static AudioCodec CodecIDToAudioCodec(AVCodecID codec_id) { |
switch (codec_id) { |
case AV_CODEC_ID_AAC: |
return kCodecAAC; |
+#if BUILDFLAG(ENABLE_AC3_EAC3_AUDIO_DEMUXING) |
+ case AV_CODEC_ID_AC3: |
+ return kCodecAC3; |
+ case AV_CODEC_ID_EAC3: |
+ return kCodecEAC3; |
+#endif |
case AV_CODEC_ID_MP3: |
return kCodecMP3; |
case AV_CODEC_ID_VORBIS: |
@@ -309,18 +316,37 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context, |
codec_context->channel_layout, codec_context->channels); |
int sample_rate = codec_context->sample_rate; |
- if (codec == kCodecOpus) { |
- // |codec_context->sample_fmt| is not set by FFmpeg because Opus decoding is |
- // not enabled in FFmpeg. It doesn't matter what value is set here, so long |
- // as it's valid, the true sample format is selected inside the decoder. |
- sample_format = kSampleFormatF32; |
- |
- // Always use 48kHz for OPUS. Technically we should match to the highest |
- // supported hardware sample rate among [8, 12, 16, 24, 48] kHz, but we |
- // don't know the hardware sample rate at this point and those rates are |
- // rarely used for output. See the "Input Sample Rate" section of the spec: |
- // http://tools.ietf.org/html/draft-terriberry-oggopus-01#page-11 |
- sample_rate = 48000; |
+ switch (codec) { |
+ case kCodecOpus: |
+ // |codec_context->sample_fmt| is not set by FFmpeg because Opus decoding |
+ // is not enabled in FFmpeg. It doesn't matter what value is set here, so |
+ // long as it's valid, the true sample format is selected inside the |
+ // decoder. |
+ sample_format = kSampleFormatF32; |
+ |
+ // Always use 48kHz for OPUS. Technically we should match to the highest |
+ // supported hardware sample rate among [8, 12, 16, 24, 48] kHz, but we |
+ // don't know the hardware sample rate at this point and those rates are |
+ // rarely used for output. See the "Input Sample Rate" section of the |
+ // spec: http://tools.ietf.org/html/draft-terriberry-oggopus-01#page-11 |
+ sample_rate = 48000; |
+ break; |
+ |
+ // For AC3/EAC3 we enable only demuxing, but not decoding, so FFmpeg does |
+ // not fill |sample_fmt|. |
+ case kCodecAC3: |
+ case kCodecEAC3: |
+#if BUILDFLAG(ENABLE_AC3_EAC3_AUDIO_DEMUXING) |
+ // The spec for AC3/EAC3 audio is ETSI TS 102 366. According to sections |
+ // F.3.1 and F.5.1 in that spec the sample_format for AC3/EAC3 must be 16. |
+ sample_format = kSampleFormatS16; |
+#else |
+ NOTREACHED(); |
+#endif |
+ break; |
+ |
+ default: |
+ break; |
} |
base::TimeDelta seek_preroll; |
@@ -353,9 +379,19 @@ bool AVCodecContextToAudioDecoderConfig(const AVCodecContext* codec_context, |
seek_preroll, |
codec_context->delay); |
- if (codec != kCodecOpus) { |
- DCHECK_EQ(av_get_bytes_per_sample(codec_context->sample_fmt) * 8, |
- config->bits_per_channel()); |
+ // Verify that AudioConfig.bits_per_channel was calculated correctly for |
+ // codecs that have |sample_fmt| set by FFmpeg. |
+ switch (codec) { |
+ case kCodecOpus: |
+#if BUILDFLAG(ENABLE_AC3_EAC3_AUDIO_DEMUXING) |
+ case kCodecAC3: |
+ case kCodecEAC3: |
+#endif |
+ break; |
+ default: |
+ DCHECK_EQ(av_get_bytes_per_sample(codec_context->sample_fmt) * 8, |
+ config->bits_per_channel()); |
+ break; |
} |
return true; |