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Move ADM initialization into WebRtcVoiceEngine

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Bug: webrtc:4690
Change-Id: I3b8950fdb13835964c5bf41162731eff5048bf1a
Reviewed-on: https://webrtc-review.googlesource.com/23820
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20823}
This commit is contained in:
Fredrik Solenberg
2017-11-21 20:33:05 +01:00
committed by Commit Bot
parent 37e489c985
commit d319534143
20 changed files with 114 additions and 280 deletions
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7
call/call_perf_tests.cc
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@ -176,6 +176,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
fake_audio_device = rtc::MakeUnique<FakeAudioDevice>(
FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000),
FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed);
EXPECT_EQ(0, fake_audio_device->Init());
EXPECT_EQ(0, voe_base->Init(fake_audio_device.get(), audio_processing.get(),
decoder_factory_));
VoEBase::ChannelConfig config;
@ -189,9 +190,11 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
send_audio_state_config.audio_processing = audio_processing;
Call::Config sender_config(event_log_.get());
sender_config.audio_state = AudioState::Create(send_audio_state_config);
auto audio_state = AudioState::Create(send_audio_state_config);
fake_audio_device->RegisterAudioCallback(audio_state->audio_transport());
sender_config.audio_state = audio_state;
Call::Config receiver_config(event_log_.get());
receiver_config.audio_state = sender_config.audio_state;
receiver_config.audio_state = audio_state;
CreateCalls(sender_config, receiver_config);
std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_),