Move ADM initialization into WebRtcVoiceEngine
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}diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 4fdb68c..d6c2dbe 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -75,7 +75,6 @@
audio_mixer_(new rtc::RefCountedObject<MockAudioMixer>()) {
using testing::Invoke;
- EXPECT_CALL(voice_engine_, audio_device_module());
EXPECT_CALL(voice_engine_, audio_transport());
AudioState::Config config;
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 8e42029..c67eb9b 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -148,7 +148,6 @@
audio_encoder_(nullptr) {
using testing::Invoke;
- EXPECT_CALL(voice_engine_, audio_device_module());
EXPECT_CALL(voice_engine_, audio_transport());
AudioState::Config config;
diff --git a/audio/audio_state.cc b/audio/audio_state.cc
index a1dd956..5a30c53 100644
--- a/audio/audio_state.cc
+++ b/audio/audio_state.cc
@@ -29,13 +29,6 @@
config_.audio_mixer) {
process_thread_checker_.DetachFromThread();
RTC_DCHECK(config_.audio_mixer);
-
- auto* const device = voe_base_->audio_device_module();
- RTC_DCHECK(device);
-
- // This is needed for the Chrome implementation of RegisterAudioCallback.
- device->RegisterAudioCallback(nullptr);
- device->RegisterAudioCallback(&audio_transport_proxy_);
}
AudioState::~AudioState() {
diff --git a/audio/audio_state_unittest.cc b/audio/audio_state_unittest.cc
index 86be245..28b0a71 100644
--- a/audio/audio_state_unittest.cc
+++ b/audio/audio_state_unittest.cc
@@ -26,22 +26,9 @@
struct ConfigHelper {
ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) {
- EXPECT_CALL(mock_voice_engine, audio_device_module())
- .Times(testing::AtLeast(1));
EXPECT_CALL(mock_voice_engine, audio_transport())
.WillRepeatedly(testing::Return(&audio_transport));
- auto device = static_cast<MockAudioDeviceModule*>(
- voice_engine().audio_device_module());
-
- // Populate the audio transport proxy pointer to the most recent
- // transport connected to the Audio Device.
- ON_CALL(*device, RegisterAudioCallback(testing::_))
- .WillByDefault(testing::Invoke([this](AudioTransport* transport) {
- registered_audio_transport = transport;
- return 0;
- }));
-
audio_state_config.voice_engine = &mock_voice_engine;
audio_state_config.audio_mixer = audio_mixer;
audio_state_config.audio_processing =
@@ -51,14 +38,12 @@
MockVoiceEngine& voice_engine() { return mock_voice_engine; }
rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
MockAudioTransport& original_audio_transport() { return audio_transport; }
- AudioTransport* audio_transport_proxy() { return registered_audio_transport; }
private:
testing::StrictMock<MockVoiceEngine> mock_voice_engine;
AudioState::Config audio_state_config;
rtc::scoped_refptr<AudioMixer> audio_mixer;
MockAudioTransport audio_transport;
- AudioTransport* registered_audio_transport = nullptr;
};
class FakeAudioSource : public AudioMixer::Source {
@@ -112,7 +97,7 @@
kNumberOfChannels, kSampleRate, 0, 0, 0, false,
testing::Ref(new_mic_level)));
- helper.audio_transport_proxy()->RecordedDataIsAvailable(
+ audio_state->audio_transport()->RecordedDataIsAvailable(
nullptr, kSampleRate / 100, kBytesPerSample, kNumberOfChannels,
kSampleRate, 0, 0, 0, false, new_mic_level);
}
@@ -141,7 +126,7 @@
size_t n_samples_out;
int64_t elapsed_time_ms;
int64_t ntp_time_ms;
- helper.audio_transport_proxy()->NeedMorePlayData(
+ audio_state->audio_transport()->NeedMorePlayData(
kSampleRate / 100, kBytesPerSample, kNumberOfChannels, kSampleRate,
audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
}
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 98ebe8b..a6fe0b2 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -176,6 +176,7 @@
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 @@
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_),
diff --git a/call/call_unittest.cc b/call/call_unittest.cc
index aeba5a8..7b16271 100644
--- a/call/call_unittest.cc
+++ b/call/call_unittest.cc
@@ -41,7 +41,6 @@
audio_state_config.voice_engine = &voice_engine_;
audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
audio_state_config.audio_processing = webrtc::AudioProcessing::Create();
- EXPECT_CALL(voice_engine_, audio_device_module());
EXPECT_CALL(voice_engine_, audio_transport());
webrtc::Call::Config config(&event_log_);
config.audio_state = webrtc::AudioState::Create(audio_state_config);
diff --git a/media/engine/adm_helpers.cc b/media/engine/adm_helpers.cc
index b973ce6..119cc64 100644
--- a/media/engine/adm_helpers.cc
+++ b/media/engine/adm_helpers.cc
@@ -32,94 +32,50 @@
#define AUDIO_DEVICE_ID (0u)
#endif // defined(WEBRTC_WIN)
-void SetRecordingDevice(AudioDeviceModule* adm) {
+void Init(AudioDeviceModule* adm) {
RTC_DCHECK(adm);
- // Save recording status and stop recording.
- const bool was_recording = adm->Recording();
- if (was_recording && adm->StopRecording() != 0) {
- RTC_LOG(LS_ERROR) << "Unable to stop recording.";
- return;
- }
+ RTC_CHECK_EQ(0, adm->Init()) << "Failed to initialize the ADM.";
- // Set device to default.
- if (adm->SetRecordingDevice(AUDIO_DEVICE_ID) != 0) {
- RTC_LOG(LS_ERROR) << "Unable to set recording device.";
- return;
- }
-
- // Init microphone, so user can do volume settings etc.
- if (adm->InitMicrophone() != 0) {
- RTC_LOG(LS_ERROR) << "Unable to access microphone.";
- }
-
- // Set number of channels
- bool available = false;
- if (adm->StereoRecordingIsAvailable(&available) != 0) {
- RTC_LOG(LS_ERROR) << "Failed to query stereo recording.";
- }
- if (adm->SetStereoRecording(available) != 0) {
- RTC_LOG(LS_ERROR) << "Failed to set stereo recording mode.";
- }
-
- // Restore recording if it was enabled already when calling this function.
- if (was_recording) {
- if (adm->InitRecording() != 0) {
- RTC_LOG(LS_ERROR) << "Failed to initialize recording.";
+ // Playout device.
+ {
+ if (adm->SetPlayoutDevice(AUDIO_DEVICE_ID) != 0) {
+ RTC_LOG(LS_ERROR) << "Unable to set playout device.";
return;
}
- if (adm->StartRecording() != 0) {
- RTC_LOG(LS_ERROR) << "Failed to start recording.";
- return;
+ if (adm->InitSpeaker() != 0) {
+ RTC_LOG(LS_ERROR) << "Unable to access speaker.";
+ }
+
+ // Set number of channels
+ bool available = false;
+ if (adm->StereoPlayoutIsAvailable(&available) != 0) {
+ RTC_LOG(LS_ERROR) << "Failed to query stereo playout.";
+ }
+ if (adm->SetStereoPlayout(available) != 0) {
+ RTC_LOG(LS_ERROR) << "Failed to set stereo playout mode.";
}
}
- RTC_LOG(LS_INFO) << "Set recording device.";
+ // Recording device.
+ {
+ if (adm->SetRecordingDevice(AUDIO_DEVICE_ID) != 0) {
+ RTC_LOG(LS_ERROR) << "Unable to set recording device.";
+ return;
+ }
+ if (adm->InitMicrophone() != 0) {
+ RTC_LOG(LS_ERROR) << "Unable to access microphone.";
+ }
+
+ // Set number of channels
+ bool available = false;
+ if (adm->StereoRecordingIsAvailable(&available) != 0) {
+ RTC_LOG(LS_ERROR) << "Failed to query stereo recording.";
+ }
+ if (adm->SetStereoRecording(available) != 0) {
+ RTC_LOG(LS_ERROR) << "Failed to set stereo recording mode.";
+ }
+ }
}
-
-void SetPlayoutDevice(AudioDeviceModule* adm) {
- RTC_DCHECK(adm);
-
- // Save playing status and stop playout.
- const bool was_playing = adm->Playing();
- if (was_playing && adm->StopPlayout() != 0) {
- RTC_LOG(LS_ERROR) << "Unable to stop playout.";
- }
-
- // Set device.
- if (adm->SetPlayoutDevice(AUDIO_DEVICE_ID) != 0) {
- RTC_LOG(LS_ERROR) << "Unable to set playout device.";
- return;
- }
-
- // Init speaker, so user can do volume settings etc.
- if (adm->InitSpeaker() != 0) {
- RTC_LOG(LS_ERROR) << "Unable to access speaker.";
- }
-
- // Set number of channels
- bool available = false;
- if (adm->StereoPlayoutIsAvailable(&available) != 0) {
- RTC_LOG(LS_ERROR) << "Failed to query stereo playout.";
- }
- if (adm->SetStereoPlayout(available) != 0) {
- RTC_LOG(LS_ERROR) << "Failed to set stereo playout mode.";
- }
-
- // Restore recording if it was enabled already when calling this function.
- if (was_playing) {
- if (adm->InitPlayout() != 0) {
- RTC_LOG(LS_ERROR) << "Failed to initialize playout.";
- return;
- }
- if (adm->StartPlayout() != 0) {
- RTC_LOG(LS_ERROR) << "Failed to start playout.";
- return;
- }
- }
-
- RTC_LOG(LS_INFO) << "Set playout device.";
-}
-
} // namespace adm_helpers
} // namespace webrtc
diff --git a/media/engine/adm_helpers.h b/media/engine/adm_helpers.h
index d04db17..c6ea3a2 100644
--- a/media/engine/adm_helpers.h
+++ b/media/engine/adm_helpers.h
@@ -19,8 +19,7 @@
namespace adm_helpers {
-void SetRecordingDevice(AudioDeviceModule* adm);
-void SetPlayoutDevice(AudioDeviceModule* adm);
+void Init(AudioDeviceModule* adm);
} // namespace adm_helpers
} // namespace webrtc
diff --git a/media/engine/fakewebrtcvoiceengine.h b/media/engine/fakewebrtcvoiceengine.h
index 55d3100..9c9a428 100644
--- a/media/engine/fakewebrtcvoiceengine.h
+++ b/media/engine/fakewebrtcvoiceengine.h
@@ -69,9 +69,6 @@
inited_ = false;
return 0;
}
- webrtc::AudioDeviceModule* audio_device_module() override {
- return nullptr;
- }
webrtc::voe::TransmitMixer* transmit_mixer() override {
return transmit_mixer_;
}
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index 19619a3..9630bc4 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -28,6 +28,7 @@
#include "media/engine/payload_type_mapper.h"
#include "media/engine/webrtcmediaengine.h"
#include "media/engine/webrtcvoe.h"
+#include "modules/audio_device/audio_device_impl.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/include/audio_processing.h"
@@ -251,6 +252,12 @@
if (initialized_) {
StopAecDump();
voe_wrapper_->base()->Terminate();
+
+ // Stop AudioDevice.
+ adm()->StopPlayout();
+ adm()->StopRecording();
+ adm()->RegisterAudioCallback(nullptr);
+ adm()->Terminate();
}
}
@@ -283,15 +290,17 @@
channel_config_.enable_voice_pacing = true;
- RTC_CHECK_EQ(0,
- voe_wrapper_->base()->Init(adm_.get(), apm(), decoder_factory_));
-
- // No ADM supplied? Get the default one from VoE.
+#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+ // No ADM supplied? Create a default one.
if (!adm_) {
- adm_ = voe_wrapper_->base()->audio_device_module();
+ adm_ = webrtc::AudioDeviceModule::Create(
+ webrtc::AudioDeviceModule::kPlatformDefaultAudio);
}
- RTC_DCHECK(adm_);
+#endif // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
+ RTC_CHECK(adm());
+ webrtc::adm_helpers::Init(adm());
+ RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm(), apm(), decoder_factory_));
transmit_mixer_ = voe_wrapper_->base()->transmit_mixer();
RTC_DCHECK(transmit_mixer_);
@@ -324,15 +333,16 @@
// Set default audio devices.
#if !defined(WEBRTC_IOS)
- webrtc::adm_helpers::SetRecordingDevice(adm_);
apm()->Initialize();
- webrtc::adm_helpers::SetPlayoutDevice(adm_);
#endif // !WEBRTC_IOS
// May be null for VoE injected for testing.
if (voe()->engine()) {
audio_state_ = webrtc::AudioState::Create(
MakeAudioStateConfig(voe(), audio_mixer_, apm_));
+
+ // Connect the ADM to our audio path.
+ adm()->RegisterAudioCallback(audio_state_->audio_transport());
}
initialized_ = true;
@@ -708,7 +718,7 @@
webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(adm_);
- return adm_;
+ return adm_.get();
}
webrtc::AudioProcessing* WebRtcVoiceEngine::apm() const {
diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc
index ef56473..f178262 100644
--- a/media/engine/webrtcvoiceengine_unittest.cc
+++ b/media/engine/webrtcvoiceengine_unittest.cc
@@ -85,23 +85,10 @@
void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
RTC_DCHECK(adm);
+
+ // Setup.
EXPECT_CALL(*adm, AddRef()).Times(1);
- EXPECT_CALL(*adm, Release())
- .WillOnce(Return(rtc::RefCountReleaseStatus::kDroppedLastRef));
-#if !defined(WEBRTC_IOS)
- EXPECT_CALL(*adm, Recording()).WillOnce(Return(false));
-#if defined(WEBRTC_WIN)
- EXPECT_CALL(*adm, SetRecordingDevice(
- testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
- webrtc::AudioDeviceModule::kDefaultCommunicationDevice)))
- .WillOnce(Return(0));
-#else
- EXPECT_CALL(*adm, SetRecordingDevice(0)).WillOnce(Return(0));
-#endif // #if defined(WEBRTC_WIN)
- EXPECT_CALL(*adm, InitMicrophone()).WillOnce(Return(0));
- EXPECT_CALL(*adm, StereoRecordingIsAvailable(testing::_)).WillOnce(Return(0));
- EXPECT_CALL(*adm, SetStereoRecording(false)).WillOnce(Return(0));
- EXPECT_CALL(*adm, Playing()).WillOnce(Return(false));
+ EXPECT_CALL(*adm, Init()).WillOnce(Return(0));
#if defined(WEBRTC_WIN)
EXPECT_CALL(*adm, SetPlayoutDevice(
testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
@@ -113,11 +100,29 @@
EXPECT_CALL(*adm, InitSpeaker()).WillOnce(Return(0));
EXPECT_CALL(*adm, StereoPlayoutIsAvailable(testing::_)).WillOnce(Return(0));
EXPECT_CALL(*adm, SetStereoPlayout(false)).WillOnce(Return(0));
-#endif // #if !defined(WEBRTC_IOS)
+#if defined(WEBRTC_WIN)
+ EXPECT_CALL(*adm, SetRecordingDevice(
+ testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
+ webrtc::AudioDeviceModule::kDefaultCommunicationDevice)))
+ .WillOnce(Return(0));
+#else
+ EXPECT_CALL(*adm, SetRecordingDevice(0)).WillOnce(Return(0));
+#endif // #if defined(WEBRTC_WIN)
+ EXPECT_CALL(*adm, InitMicrophone()).WillOnce(Return(0));
+ EXPECT_CALL(*adm, StereoRecordingIsAvailable(testing::_)).WillOnce(Return(0));
+ EXPECT_CALL(*adm, SetStereoRecording(false)).WillOnce(Return(0));
EXPECT_CALL(*adm, BuiltInAECIsAvailable()).WillOnce(Return(false));
EXPECT_CALL(*adm, BuiltInAGCIsAvailable()).WillOnce(Return(false));
EXPECT_CALL(*adm, BuiltInNSIsAvailable()).WillOnce(Return(false));
EXPECT_CALL(*adm, SetAGC(true)).WillOnce(Return(0));
+
+ // Teardown.
+ EXPECT_CALL(*adm, StopPlayout()).WillOnce(Return(0));
+ EXPECT_CALL(*adm, StopRecording()).WillOnce(Return(0));
+ EXPECT_CALL(*adm, RegisterAudioCallback(nullptr)).WillOnce(Return(0));
+ EXPECT_CALL(*adm, Terminate()).WillOnce(Return(0));
+ EXPECT_CALL(*adm, Release())
+ .WillOnce(Return(rtc::RefCountReleaseStatus::kDroppedLastRef));
}
} // namespace
diff --git a/test/call_test.cc b/test/call_test.cc
index f4b3877..3210e64 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -75,6 +75,8 @@
audio_state_config.audio_mixer = AudioMixerImpl::Create();
audio_state_config.audio_processing = apm_send_;
send_config.audio_state = AudioState::Create(audio_state_config);
+ fake_send_audio_device_->RegisterAudioCallback(
+ send_config.audio_state->audio_transport());
}
CreateSenderCall(send_config);
if (sender_call_transport_controller_ != nullptr) {
@@ -89,7 +91,8 @@
audio_state_config.audio_mixer = AudioMixerImpl::Create();
audio_state_config.audio_processing = apm_recv_;
recv_config.audio_state = AudioState::Create(audio_state_config);
- }
+ fake_recv_audio_device_->RegisterAudioCallback(
+ recv_config.audio_state->audio_transport()); }
CreateReceiverCall(recv_config);
}
test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
@@ -427,6 +430,7 @@
void CallTest::CreateVoiceEngines() {
voe_send_.voice_engine = VoiceEngine::Create();
voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
+ EXPECT_EQ(0, fake_send_audio_device_->Init());
EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(),
apm_send_.get(), decoder_factory_));
VoEBase::ChannelConfig config;
@@ -436,6 +440,7 @@
voe_recv_.voice_engine = VoiceEngine::Create();
voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
+ EXPECT_EQ(0, fake_recv_audio_device_->Init());
EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(),
apm_recv_.get(), decoder_factory_));
voe_recv_.channel_id = voe_recv_.base->CreateChannel();
diff --git a/test/fake_audio_device.cc b/test/fake_audio_device.cc
index fa3f1f4..eef8bfd 100644
--- a/test/fake_audio_device.cc
+++ b/test/fake_audio_device.cc
@@ -303,13 +303,9 @@
}
int32_t FakeAudioDevice::Init() {
- // TODO(solenberg): Temporarily allow multiple init calls.
- if (!inited_) {
- RTC_CHECK(tick_->StartTimer(true, kFrameLengthMs / speed_));
- thread_.Start();
- thread_.SetPriority(rtc::kHighPriority);
- inited_ = true;
- }
+ RTC_CHECK(tick_->StartTimer(true, kFrameLengthMs / speed_));
+ thread_.Start();
+ thread_.SetPriority(rtc::kHighPriority);
return 0;
}
diff --git a/test/fake_audio_device.h b/test/fake_audio_device.h
index 4d90619..f71fd01 100644
--- a/test/fake_audio_device.h
+++ b/test/fake_audio_device.h
@@ -137,7 +137,6 @@
std::unique_ptr<EventTimerWrapper> tick_;
rtc::PlatformThread thread_;
- bool inited_ = false;
};
} // namespace test
} // namespace webrtc
diff --git a/test/mock_voice_engine.h b/test/mock_voice_engine.h
index 67bc993..7fec120 100644
--- a/test/mock_voice_engine.h
+++ b/test/mock_voice_engine.h
@@ -13,7 +13,6 @@
#include <memory>
-#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_device/include/mock_audio_transport.h"
#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "test/gmock.h"
@@ -63,8 +62,6 @@
return proxy;
}));
- ON_CALL(*this, audio_device_module())
- .WillByDefault(testing::Return(&mock_audio_device_));
ON_CALL(*this, audio_transport())
.WillByDefault(testing::Return(&mock_audio_transport_));
}
@@ -97,7 +94,6 @@
int(AudioDeviceModule* external_adm,
AudioProcessing* external_apm,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory));
- MOCK_METHOD0(audio_device_module, AudioDeviceModule*());
MOCK_METHOD0(transmit_mixer, voe::TransmitMixer*());
MOCK_METHOD0(Terminate, int());
MOCK_METHOD0(CreateChannel, int());
@@ -120,7 +116,6 @@
std::map<int, std::unique_ptr<MockRtpRtcp>> mock_rtp_rtcps_;
- MockAudioDeviceModule mock_audio_device_;
MockAudioTransport mock_audio_transport_;
};
} // namespace test
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index 512a9bb..e40d295 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -92,11 +92,13 @@
void CreateVoiceEngine(
VoiceEngineState* voe,
+ webrtc::AudioDeviceModule* adm,
webrtc::AudioProcessing* apm,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory) {
voe->voice_engine = webrtc::VoiceEngine::Create();
voe->base = webrtc::VoEBase::GetInterface(voe->voice_engine);
- EXPECT_EQ(0, voe->base->Init(nullptr, apm, decoder_factory));
+ EXPECT_EQ(0, adm->Init());
+ EXPECT_EQ(0, voe->base->Init(adm, apm, decoder_factory));
webrtc::VoEBase::ChannelConfig config;
config.enable_voice_pacing = true;
voe->send_channel_id = voe->base->CreateChannel(config);
@@ -1968,6 +1970,7 @@
void VideoQualityTest::RunWithRenderers(const Params& params) {
std::unique_ptr<test::LayerFilteringTransport> send_transport;
std::unique_ptr<test::DirectTransport> recv_transport;
+ std::unique_ptr<test::FakeAudioDevice> fake_audio_device;
::VoiceEngineState voe;
std::unique_ptr<test::VideoRenderer> local_preview;
std::vector<std::unique_ptr<test::VideoRenderer>> loopback_renderers;
@@ -1982,16 +1985,24 @@
Call::Config call_config(event_log_.get());
call_config.bitrate_config = params_.call.call_bitrate_config;
+ fake_audio_device.reset(new test::FakeAudioDevice(
+ test::FakeAudioDevice::CreatePulsedNoiseCapturer(32000, 48000),
+ test::FakeAudioDevice::CreateDiscardRenderer(48000),
+ 1.f));
+
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing(
webrtc::AudioProcessing::Create());
if (params_.audio.enabled) {
- CreateVoiceEngine(&voe, audio_processing.get(), decoder_factory_);
+ CreateVoiceEngine(&voe, fake_audio_device.get(), audio_processing.get(),
+ decoder_factory_);
AudioState::Config audio_state_config;
audio_state_config.voice_engine = voe.voice_engine;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
audio_state_config.audio_processing = audio_processing;
call_config.audio_state = AudioState::Create(audio_state_config);
+ fake_audio_device->RegisterAudioCallback(
+ call_config.audio_state->audio_transport());
}
CreateCalls(call_config, call_config);
diff --git a/voice_engine/include/voe_base.h b/voice_engine/include/voe_base.h
index 06a8cf3..2f74fcc 100644
--- a/voice_engine/include/voe_base.h
+++ b/voice_engine/include/voe_base.h
@@ -92,15 +92,10 @@
// functionality in a separate (reference counted) module.
// - The AudioProcessing module handles capture-side processing.
// - An AudioDecoderFactory - used to create audio decoders.
- // If NULL is passed for ADM, VoiceEngine
- // will create its own. Returns -1 in case of an error, 0 otherwise.
virtual int Init(
- AudioDeviceModule* external_adm,
- AudioProcessing* external_apm,
+ AudioDeviceModule* audio_device,
+ AudioProcessing* audio_processing,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) = 0;
- // This method is WIP - DO NOT USE!
- // Returns NULL before Init() is called.
- virtual AudioDeviceModule* audio_device_module() = 0;
// This method is WIP - DO NOT USE!
// Returns NULL before Init() is called.
diff --git a/voice_engine/voe_base_impl.cc b/voice_engine/voe_base_impl.cc
index 9848557..695b06d 100644
--- a/voice_engine/voe_base_impl.cc
+++ b/voice_engine/voe_base_impl.cc
@@ -142,91 +142,17 @@
}
int VoEBaseImpl::Init(
- AudioDeviceModule* external_adm,
+ AudioDeviceModule* audio_device,
AudioProcessing* audio_processing,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+ RTC_DCHECK(audio_device);
RTC_DCHECK(audio_processing);
rtc::CritScope cs(shared_->crit_sec());
- WebRtcSpl_Init();
if (shared_->process_thread()) {
shared_->process_thread()->Start();
}
- // Create an internal ADM if the user has not added an external
- // ADM implementation as input to Init().
- if (external_adm == nullptr) {
-#if !defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
- return -1;
-#else
- // Create the internal ADM implementation.
- shared_->set_audio_device(AudioDeviceModule::Create(
- AudioDeviceModule::kPlatformDefaultAudio));
- if (shared_->audio_device() == nullptr) {
- RTC_LOG(LS_ERROR) << "Init() failed to create the ADM";
- return -1;
- }
-#endif // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
- } else {
- // Use the already existing external ADM implementation.
- shared_->set_audio_device(external_adm);
- RTC_LOG_F(LS_INFO)
- << "An external ADM implementation will be used in VoiceEngine";
- }
-
- bool available = false;
-
- // --------------------
- // Reinitialize the ADM
-
- // Register the AudioTransport implementation
- if (shared_->audio_device()->RegisterAudioCallback(this) != 0) {
- RTC_LOG(LS_ERROR) << "Init() failed to register audio callback for the ADM";
- }
-
- // ADM initialization
- if (shared_->audio_device()->Init() != 0) {
- RTC_LOG(LS_ERROR) << "Init() failed to initialize the ADM";
- return -1;
- }
-
- // Initialize the default speaker
- if (shared_->audio_device()->SetPlayoutDevice(
- WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0) {
- RTC_LOG(LS_ERROR) << "Init() failed to set the default output device";
- }
- if (shared_->audio_device()->InitSpeaker() != 0) {
- RTC_LOG(LS_ERROR) << "Init() failed to initialize the speaker";
- }
-
- // Initialize the default microphone
- if (shared_->audio_device()->SetRecordingDevice(
- WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0) {
- RTC_LOG(LS_ERROR) << "Init() failed to set the default input device";
- }
- if (shared_->audio_device()->InitMicrophone() != 0) {
- RTC_LOG(LS_ERROR) << "Init() failed to initialize the microphone";
- }
-
- // Set number of channels
- if (shared_->audio_device()->StereoPlayoutIsAvailable(&available) != 0) {
- RTC_LOG(LS_ERROR) << "Init() failed to query stereo playout mode";
- }
- if (shared_->audio_device()->SetStereoPlayout(available) != 0) {
- RTC_LOG(LS_ERROR) << "Init() failed to set mono/stereo playout mode";
- }
-
- // TODO(andrew): These functions don't tell us whether stereo recording
- // is truly available. We simply set the AudioProcessing input to stereo
- // here, because we have to wait until receiving the first frame to
- // determine the actual number of channels anyway.
- //
- // These functions may be changed; tracked here:
- // http://code.google.com/p/webrtc/issues/detail?id=204
- shared_->audio_device()->StereoRecordingIsAvailable(&available);
- if (shared_->audio_device()->SetStereoRecording(available) != 0) {
- RTC_LOG(LS_ERROR) << "Init() failed to set mono/stereo recording mode";
- }
-
+ shared_->set_audio_device(audio_device);
shared_->set_audio_processing(audio_processing);
// Configure AudioProcessing components.
@@ -518,23 +444,7 @@
shared_->process_thread()->Stop();
}
- if (shared_->audio_device()) {
- if (shared_->audio_device()->StopPlayout() != 0) {
- RTC_LOG(LS_ERROR) << "TerminateInternal() failed to stop playout";
- }
- if (shared_->audio_device()->StopRecording() != 0) {
- RTC_LOG(LS_ERROR) << "TerminateInternal() failed to stop recording";
- }
- if (shared_->audio_device()->RegisterAudioCallback(nullptr) != 0) {
- RTC_LOG(LS_ERROR) << "TerminateInternal() failed to de-register audio "
- "callback for the ADM";
- }
- if (shared_->audio_device()->Terminate() != 0) {
- RTC_LOG(LS_ERROR) << "TerminateInternal() failed to terminate the ADM";
- }
- shared_->set_audio_device(nullptr);
- }
-
+ shared_->set_audio_device(nullptr);
shared_->set_audio_processing(nullptr);
return 0;
diff --git a/voice_engine/voe_base_impl.h b/voice_engine/voe_base_impl.h
index e647124..4cd6c86 100644
--- a/voice_engine/voe_base_impl.h
+++ b/voice_engine/voe_base_impl.h
@@ -25,12 +25,9 @@
public AudioTransport {
public:
int Init(
- AudioDeviceModule* external_adm,
+ AudioDeviceModule* audio_device,
AudioProcessing* audio_processing,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) override;
- AudioDeviceModule* audio_device_module() override {
- return shared_->audio_device();
- }
voe::TransmitMixer* transmit_mixer() override {
return shared_->transmit_mixer();
}
diff --git a/voice_engine/voice_engine_defines.h b/voice_engine/voice_engine_defines.h
index 4397662..35f9492 100644
--- a/voice_engine/voice_engine_defines.h
+++ b/voice_engine/voice_engine_defines.h
@@ -47,23 +47,4 @@
} // namespace webrtc
-namespace webrtc {
-
-inline int VoEId(int veId, int chId) {
- if (chId == -1) {
- const int dummyChannel(99);
- return (int)((veId << 16) + dummyChannel);
- }
- return (int)((veId << 16) + chId);
-}
-
-} // namespace webrtc
-
-#if defined(_WIN32)
-#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE \
- AudioDeviceModule::kDefaultCommunicationDevice
-#else
-#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
-#endif // #if (defined(_WIN32)
-
#endif // VOICE_ENGINE_VOICE_ENGINE_DEFINES_H_
