From 63fe8e1f389cb2e89ff78c0b78362a6741006482 Mon Sep 17 00:00:00 2001 From: "dwkang@webrtc.org" Date: Mon, 23 Sep 2013 05:42:22 +0000 Subject: [PATCH] Enable SetInitialPlayoutDelay on Android. Background: In Chrome mirroring which uses 500ms buffering mode, audio video mismatch happens in the begining because of the lack of the api. BUG=b/10538425 TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release -f *InitialPlayoutDelayTest*' R=henrika@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2177004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4807 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../audio_coding/main/test/initial_delay_unittest.cc | 12 ++++++------ webrtc/voice_engine/voe_video_sync_impl.cc | 1 - 2 files changed, 6 insertions(+), 7 deletions(-) diff --git a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc index a73180802..d1a977602 100644 --- a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc +++ b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc @@ -119,38 +119,38 @@ class InitialPlayoutDelayTest : public ::testing::Test { Channel* channel_a2b_; }; -TEST_F( InitialPlayoutDelayTest, DISABLED_ON_ANDROID(NbMono)) { +TEST_F( InitialPlayoutDelayTest, NbMono) { CodecInst codec; AudioCodingModule::Codec("L16", &codec, 8000, 1); Run(codec, 3000); } -TEST_F( InitialPlayoutDelayTest, DISABLED_ON_ANDROID(WbMono)) { +TEST_F( InitialPlayoutDelayTest, WbMono) { CodecInst codec; AudioCodingModule::Codec("L16", &codec, 16000, 1); Run(codec, 3000); } -TEST_F( InitialPlayoutDelayTest, DISABLED_ON_ANDROID(SwbMono)) { +TEST_F( InitialPlayoutDelayTest, SwbMono) { CodecInst codec; AudioCodingModule::Codec("L16", &codec, 32000, 1); Run(codec, 2000); // NetEq buffer is not sufficiently large for 3 sec of // PCM16 super-wideband. } -TEST_F( InitialPlayoutDelayTest, DISABLED_ON_ANDROID(NbStereo)) { +TEST_F( InitialPlayoutDelayTest, NbStereo) { CodecInst codec; AudioCodingModule::Codec("L16", &codec, 8000, 2); Run(codec, 3000); } -TEST_F( InitialPlayoutDelayTest, DISABLED_ON_ANDROID(WbStereo)) { +TEST_F( InitialPlayoutDelayTest, WbStereo) { CodecInst codec; AudioCodingModule::Codec("L16", &codec, 16000, 2); Run(codec, 3000); } -TEST_F( InitialPlayoutDelayTest, DISABLED_ON_ANDROID(SwbStereo)) { +TEST_F( InitialPlayoutDelayTest, SwbStereo) { CodecInst codec; AudioCodingModule::Codec("L16", &codec, 32000, 2); Run(codec, 2000); // NetEq buffer is not sufficiently large for 3 sec of diff --git a/webrtc/voice_engine/voe_video_sync_impl.cc b/webrtc/voice_engine/voe_video_sync_impl.cc index cd377eb82..4645e2529 100644 --- a/webrtc/voice_engine/voe_video_sync_impl.cc +++ b/webrtc/voice_engine/voe_video_sync_impl.cc @@ -145,7 +145,6 @@ int VoEVideoSyncImpl::SetInitialPlayoutDelay(int channel, int delay_ms) WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), "SetInitialPlayoutDelay(channel=%d, delay_ms=%d)", channel, delay_ms); - ANDROID_NOT_SUPPORTED(_shared->statistics()); IPHONE_NOT_SUPPORTED(_shared->statistics()); if (!_shared->statistics().Initialized())