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| author | dalecurtis@google.com <dalecurtis@google.com@0039d316-1c4b-4281-b951-d872f2087c98> | 2014-05-31 22:24:48 +0000 |
|---|---|---|
| committer | dalecurtis@google.com <dalecurtis@google.com@0039d316-1c4b-4281-b951-d872f2087c98> | 2014-05-31 22:24:48 +0000 |
| commit | 28a10808e26f47156c2e592c7f5c4d97119971e1 (patch) | |
| tree | f4c5818d12671f4ece3e3d65a0b781241d2930ef /media | |
| parent | f4b492e99cb4ebf162ace826b55a62a4602ed252 (diff) | |
| download | chromium_src-28a10808e26f47156c2e592c7f5c4d97119971e1.zip chromium_src-28a10808e26f47156c2e592c7f5c4d97119971e1.tar.gz chromium_src-28a10808e26f47156c2e592c7f5c4d97119971e1.tar.bz2 | |
Always use the source channel layout with AudioBufferConverter.
There are no web sites currently using this feature, but it causes
all multichannel users to waste resources unnecessarily upmixing,
generating, and transferring empty channel data.
Further this breaks fancy OSX multichannel users which have non
standard channel layouts. Mostly because we don't want to deal
with the plethora of ways OSX wants to map channels ourselves.
Instead we can require that MSE users which want to change channel
counts, must specify the maximum number of channels in the first
initialization segment.
This also fixes use cases where clients routed multichannel streams
into WebAudio for processing instead of actual playout.
BUG=266674,379288
TEST=Only 2 channels are created for stereo sources when a 7.1
device is connected.
R=rileya@chromium.org, wolenetz@chromium.org
Review URL: https://codereview.chromium.org/304233006
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@274068 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'media')
| -rw-r--r-- | media/filters/audio_renderer_impl.cc | 20 | ||||
| -rw-r--r-- | media/filters/pipeline_integration_test.cc | 5 |
2 files changed, 15 insertions, 10 deletions
diff --git a/media/filters/audio_renderer_impl.cc b/media/filters/audio_renderer_impl.cc index 87c4a86..d07826a 100644 --- a/media/filters/audio_renderer_impl.cc +++ b/media/filters/audio_renderer_impl.cc @@ -283,13 +283,19 @@ void AudioRendererImpl::Initialize(DemuxerStream* stream, } else { // TODO(rileya): Support hardware config changes const AudioParameters& hw_params = hardware_config_->GetOutputConfig(); - audio_parameters_.Reset(hw_params.format(), - hw_params.channel_layout(), - hw_params.channels(), - hw_params.input_channels(), - hw_params.sample_rate(), - hw_params.bits_per_sample(), - hardware_config_->GetHighLatencyBufferSize()); + audio_parameters_.Reset( + hw_params.format(), + // Always use the source's channel layout and channel count to avoid + // premature downmixing (http://crbug.com/379288), platform specific + // issues around channel layouts (http://crbug.com/266674), and + // unnecessary upmixing overhead. + stream->audio_decoder_config().channel_layout(), + ChannelLayoutToChannelCount( + stream->audio_decoder_config().channel_layout()), + hw_params.input_channels(), + hw_params.sample_rate(), + hw_params.bits_per_sample(), + hardware_config_->GetHighLatencyBufferSize()); } audio_clock_.reset(new AudioClock(audio_parameters_.sample_rate())); diff --git a/media/filters/pipeline_integration_test.cc b/media/filters/pipeline_integration_test.cc index 8e42b8c..a0f505d 100644 --- a/media/filters/pipeline_integration_test.cc +++ b/media/filters/pipeline_integration_test.cc @@ -963,9 +963,8 @@ TEST_P(PipelineIntegrationTest, MediaSource_MP3) { EXPECT_TRUE(WaitUntilOnEnded()); - // Verify that codec delay was stripped, if it wasn't the hash would be: - // "5.16,1.25,7.78,4.29,8.98,2.76," - EXPECT_EQ("5.81,2.71,8.97,4.32,7.83,1.12,", GetAudioHash()); + // Verify that codec delay was stripped. + EXPECT_EQ("1.01,2.71,4.18,4.32,3.04,1.12,", GetAudioHash()); } TEST_P(PipelineIntegrationTest, MediaSource_MP3_TimestampOffset) { |