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Moving/renaming webrtc/common.h.

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This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.

- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.

BUG=webrtc:5879

Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
This commit is contained in:
solenberg
2016-09-07 07:34:41 -07:00
committed by Commit bot
parent 4016a0b2e8
commit 88499ecaca
44 changed files with 96 additions and 221 deletions
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6
webrtc/call/call_perf_tests.cc
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@ -159,9 +159,9 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
audio_rtp_speed);
EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
Config voe_config;
voe_config.Set<VoicePacing>(new VoicePacing(true));
int send_channel_id = voe_base->CreateChannel(voe_config);
VoEBase::ChannelConfig config;
config.enable_voice_pacing = true;
int send_channel_id = voe_base->CreateChannel(config);
int recv_channel_id = voe_base->CreateChannel();
AudioState::Config send_audio_state_config;