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| author | dalecurtis@google.com <dalecurtis@google.com@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-05-24 23:54:45 +0000 |
|---|---|---|
| committer | dalecurtis@google.com <dalecurtis@google.com@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-05-24 23:54:45 +0000 |
| commit | 6a1c2ab75d931d7b41124bedef7647d11352bad4 (patch) | |
| tree | aad1d969a5f256d916a4b23858fdfd8045325092 /media | |
| parent | 9eec3554b2f0aee357285f6faa7b7957df2c2e27 (diff) | |
| download | chromium_src-6a1c2ab75d931d7b41124bedef7647d11352bad4.zip chromium_src-6a1c2ab75d931d7b41124bedef7647d11352bad4.tar.gz chromium_src-6a1c2ab75d931d7b41124bedef7647d11352bad4.tar.bz2 | |
Allow sample rate pass through on Linux.
Requiring the native output sample rate to avoid glitching appears
to have just been an issue with a poor tlength choice. xians and
I resolved this prior to turning PulseAudio on by default, but left
the native sample rate requirement in at the time.
Since "native sample rate" can change on the fly with PulseAudio
our approach of specifying FIX_RATE was incorrect. Allowing Pulse
to handle resampling lets us remove the FIX_RATE flag. It also
improves CPU usage in cases where Pulse can configure the output
device to match the requested sample rate.
BUG=229918
TEST=extensive manual checks for glitching.
R=xians@chromium.org
Review URL: https://codereview.chromium.org/15957002
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@202233 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'media')
| -rw-r--r-- | media/audio/pulse/audio_manager_pulse.cc | 6 | ||||
| -rw-r--r-- | media/audio/pulse/pulse_util.cc | 6 |
2 files changed, 8 insertions, 4 deletions
diff --git a/media/audio/pulse/audio_manager_pulse.cc b/media/audio/pulse/audio_manager_pulse.cc index c127d17..fed919a 100644 --- a/media/audio/pulse/audio_manager_pulse.cc +++ b/media/audio/pulse/audio_manager_pulse.cc @@ -141,11 +141,15 @@ AudioParameters AudioManagerPulse::GetPreferredOutputStreamParameters( int buffer_size = kDefaultOutputBufferSize; int bits_per_sample = 16; int input_channels = 0; + int sample_rate; if (input_params.IsValid()) { bits_per_sample = input_params.bits_per_sample(); channel_layout = input_params.channel_layout(); input_channels = input_params.input_channels(); buffer_size = std::min(buffer_size, input_params.frames_per_buffer()); + sample_rate = input_params.sample_rate(); + } else { + sample_rate = GetNativeSampleRate(); } int user_buffer_size = GetUserBufferSize(); @@ -154,7 +158,7 @@ AudioParameters AudioManagerPulse::GetPreferredOutputStreamParameters( return AudioParameters( AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels, - GetNativeSampleRate(), bits_per_sample, buffer_size); + sample_rate, bits_per_sample, buffer_size); } AudioOutputStream* AudioManagerPulse::MakeOutputStream( diff --git a/media/audio/pulse/pulse_util.cc b/media/audio/pulse/pulse_util.cc index 0a4734b..6a4dad1 100644 --- a/media/audio/pulse/pulse_util.cc +++ b/media/audio/pulse/pulse_util.cc @@ -289,9 +289,9 @@ bool CreateOutputStream(pa_threaded_mainloop** mainloop, pa_stream_connect_playback( *stream, NULL, &pa_buffer_attributes, static_cast<pa_stream_flags_t>( - PA_STREAM_FIX_RATE | PA_STREAM_INTERPOLATE_TIMING | - PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE | - PA_STREAM_NOT_MONOTONIC | PA_STREAM_START_CORKED), + PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | + PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_NOT_MONOTONIC | + PA_STREAM_START_CORKED), NULL, NULL) == 0, "pa_stream_connect_playback FAILED "); |