Remove voe::TransmitMixer
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 90eaa5a..db3b5e4 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -158,10 +158,8 @@
audio_net_config.queue_delay_ms = 500;
audio_net_config.loss_percent = 5;
- rtc::scoped_refptr<AudioProcessing> audio_processing;
VoiceEngine* voice_engine;
VoEBase* voe_base;
- std::unique_ptr<FakeAudioDevice> fake_audio_device;
VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock());
std::map<uint8_t, MediaType> audio_pt_map;
@@ -177,14 +175,14 @@
task_queue_.SendTask([&]() {
metrics::Reset();
- audio_processing = AudioProcessing::Create();
voice_engine = VoiceEngine::Create();
voe_base = VoEBase::GetInterface(voice_engine);
- fake_audio_device = rtc::MakeUnique<FakeAudioDevice>(
- FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000),
- FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed);
+ rtc::scoped_refptr<FakeAudioDevice> fake_audio_device =
+ new rtc::RefCountedObject<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(),
+ EXPECT_EQ(0, voe_base->Init(fake_audio_device.get(), nullptr,
decoder_factory_));
VoEBase::ChannelConfig config;
config.enable_voice_pacing = true;
@@ -194,7 +192,8 @@
AudioState::Config send_audio_state_config;
send_audio_state_config.voice_engine = voice_engine;
send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
- send_audio_state_config.audio_processing = audio_processing;
+ send_audio_state_config.audio_processing = AudioProcessing::Create();
+ send_audio_state_config.audio_device_module = fake_audio_device;
Call::Config sender_config(event_log_.get());
auto audio_state = AudioState::Create(send_audio_state_config);
@@ -311,8 +310,6 @@
DestroyCalls();
VoiceEngine::Delete(voice_engine);
-
- fake_audio_device.reset();
});
observer.PrintResults();
