admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Move ADM initialization into WebRtcVoiceEngine

Browse Source 
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
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
audio/audio_receive_stream_unittest.cc
View File

@ -75,7 +75,6 @@ struct ConfigHelper {
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;

1
audio/audio_send_stream_unittest.cc
View File

@ -148,7 +148,6 @@ struct ConfigHelper {
audio_encoder_(nullptr) {
using testing::Invoke;
EXPECT_CALL(voice_engine_, audio_device_module());
EXPECT_CALL(voice_engine_, audio_transport());
AudioState::Config config;

7
audio/audio_state.cc
View File

@ -29,13 +29,6 @@ AudioState::AudioState(const AudioState::Config& config)
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() {

19
audio/audio_state_unittest.cc
View File

@ -26,22 +26,9 @@ const int kBytesPerSample = 2;
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 @@ struct ConfigHelper {
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 @@ TEST(AudioStateAudioPathTest, RecordedAudioArrivesAtOriginalTransport) {
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 @@ TEST(AudioStateAudioPathTest,
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);
}

7
call/call_perf_tests.cc
View File

@ -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_),

1
call/call_unittest.cc
View File

@ -41,7 +41,6 @@ struct CallHelper {
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);

84
media/engine/adm_helpers.cc
View File

@ -32,67 +32,17 @@ namespace adm_helpers {
#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.";
return;
}
if (adm->StartRecording() != 0) {
RTC_LOG(LS_ERROR) << "Failed to start recording.";
return;
}
}
RTC_LOG(LS_INFO) << "Set recording device.";
}
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.
// Playout 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.";
}
@ -105,21 +55,27 @@ void SetPlayoutDevice(AudioDeviceModule* adm) {
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.";
// 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.";
}
RTC_LOG(LS_INFO) << "Set playout device.";
// 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.";
}
}
}
} // namespace adm_helpers
} // namespace webrtc

3
media/engine/adm_helpers.h
View File

@ -19,8 +19,7 @@ class AudioDeviceModule;
namespace adm_helpers {
void SetRecordingDevice(AudioDeviceModule* adm);
void SetPlayoutDevice(AudioDeviceModule* adm);
void Init(AudioDeviceModule* adm);
} // namespace adm_helpers
} // namespace webrtc

3
media/engine/fakewebrtcvoiceengine.h
View File

@ -69,9 +69,6 @@ class FakeWebRtcVoiceEngine : public webrtc::VoEBase {
inited_ = false;
return 0;
}
webrtc::AudioDeviceModule* audio_device_module() override {
return nullptr;
}
webrtc::voe::TransmitMixer* transmit_mixer() override {
return transmit_mixer_;
}

28
media/engine/webrtcvoiceengine.cc
View File

@ -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 @@ WebRtcVoiceEngine::~WebRtcVoiceEngine() {
if (initialized_) {
StopAecDump();
voe_wrapper_->base()->Terminate();
// Stop AudioDevice.
adm()->StopPlayout();
adm()->StopRecording();
adm()->RegisterAudioCallback(nullptr);
adm()->Terminate();
}
}
@ -283,15 +290,17 @@ void WebRtcVoiceEngine::Init() {
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 @@ void WebRtcVoiceEngine::Init() {
// 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 @@ int WebRtcVoiceEngine::CreateVoEChannel() {
webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(adm_);
return adm_;
return adm_.get();
}
webrtc::AudioProcessing* WebRtcVoiceEngine::apm() const {

39
media/engine/webrtcvoiceengine_unittest.cc
View File

@ -85,23 +85,10 @@ class MockTransmitMixer : public webrtc::voe::TransmitMixer {
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 @@ void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
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

7
test/call_test.cc
View File

@ -75,6 +75,8 @@ void CallTest::RunBaseTest(BaseTest* test) {
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 @@ void CallTest::RunBaseTest(BaseTest* test) {
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::SetFakeVideoCaptureRotation(VideoRotation rotation) {
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 @@ void CallTest::CreateVoiceEngines() {
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();

4
test/fake_audio_device.cc
View File

@ -303,13 +303,9 @@ int32_t FakeAudioDevice::StopRecording() {
}
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;
}
return 0;
}

1
test/fake_audio_device.h
View File

@ -137,7 +137,6 @@ class FakeAudioDevice : public FakeAudioDeviceModule {
std::unique_ptr<EventTimerWrapper> tick_;
rtc::PlatformThread thread_;
bool inited_ = false;
};
} // namespace test
} // namespace webrtc

5
test/mock_voice_engine.h
View File

@ -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 @@ class MockVoiceEngine : public VoiceEngineImpl {
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 @@ class MockVoiceEngine : public VoiceEngineImpl {
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 @@ class MockVoiceEngine : public VoiceEngineImpl {
std::map<int, std::unique_ptr<MockRtpRtcp>> mock_rtp_rtcps_;
MockAudioDeviceModule mock_audio_device_;
MockAudioTransport mock_audio_transport_;
};
} // namespace test

15
video/video_quality_test.cc
View File

@ -92,11 +92,13 @@ struct VoiceEngineState {
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::SetupAudio(int send_channel_id,
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 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
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);

9
voice_engine/include/voe_base.h
View File

@ -92,15 +92,10 @@ class WEBRTC_DLLEXPORT VoEBase {
// 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.

96
voice_engine/voe_base_impl.cc
View File

@ -142,91 +142,17 @@ void VoEBaseImpl::PullRenderData(int bits_per_sample,
}
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 @@ int32_t VoEBaseImpl::TerminateInternal() {
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_processing(nullptr);
return 0;

5
voice_engine/voe_base_impl.h
View File

@ -25,12 +25,9 @@ class VoEBaseImpl : public VoEBase,
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();
}

19
voice_engine/voice_engine_defines.h
View File

@ -47,23 +47,4 @@ enum { kVoiceEngineMaxMinPlayoutDelayMs = 10000 };
} // 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_