From 688335507909215843b37f201787da80d8b68266 Mon Sep 17 00:00:00 2001 From: tnagel Date: Wed, 27 Aug 2014 04:01:40 -0700 Subject: Revert of Remove the last piece of deprecated synchronous IO code. (patchset #5 of https://codereview.chromium.org/460373002/) Reason for revert: Build failure on Google Chrome ChromeOS: http://build.chromium.org/p/chromium.chrome/buildstatus?builder=Google%20Chrome%20ChromeOS&number=71164 Original issue's description: > Remove the last piece of deprecated synchronous IO code. > > BUG=337096 > TEST=bots > > Committed: https://chromium.googlesource.com/chromium/src/+/c4a64ffdd0d511c66d774341fa0a318af7911193 TBR=dalecurtis@chromium.org,ajm@chromium.org,rkc@chromium.org,avi@chromium.org,palmer@chromium.org,xians@chromium.org NOTREECHECKS=true NOTRY=true BUG=337096 Review URL: https://codereview.chromium.org/509893002 Cr-Commit-Position: refs/heads/master@{#292133} --- media/audio/alsa/audio_manager_alsa.cc | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'media/audio/alsa') diff --git a/media/audio/alsa/audio_manager_alsa.cc b/media/audio/alsa/audio_manager_alsa.cc index 7624834..beb60ba 100644 --- a/media/audio/alsa/audio_manager_alsa.cc +++ b/media/audio/alsa/audio_manager_alsa.cc @@ -311,6 +311,7 @@ AudioParameters AudioManagerAlsa::GetPreferredOutputStreamParameters( int sample_rate = kDefaultSampleRate; int buffer_size = kDefaultOutputBufferSize; int bits_per_sample = 16; + int input_channels = 0; if (input_params.IsValid()) { // Some clients, such as WebRTC, have a more limited use case and work // acceptably with a smaller buffer size. The check below allows clients @@ -320,6 +321,7 @@ AudioParameters AudioManagerAlsa::GetPreferredOutputStreamParameters( sample_rate = input_params.sample_rate(); bits_per_sample = input_params.bits_per_sample(); channel_layout = input_params.channel_layout(); + input_channels = input_params.input_channels(); buffer_size = std::min(input_params.frames_per_buffer(), buffer_size); } @@ -328,7 +330,7 @@ AudioParameters AudioManagerAlsa::GetPreferredOutputStreamParameters( buffer_size = user_buffer_size; return AudioParameters( - AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, + AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels, sample_rate, bits_per_sample, buffer_size, AudioParameters::NO_EFFECTS); } -- cgit v1.1