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/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
