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Removes usage of AGC APIs in the ADM.

Browse Source 
Bug: webrtc:8598
Change-Id: I5ebc2e3549eba039797e40d2f8aea48341f3fe46
Reviewed-on: https://webrtc-review.googlesource.com/31520
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21254}
This commit is contained in:
henrika 2017-12-13 14:08:48 +01:00 committed by Commit Bot
parent d235d52d1d
commit e26456a4ed
29 changed files with 11 additions and 431 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

7
media/engine/apm_helpers.cc
View File

@ -64,10 +64,8 @@ void SetAgcConfig(AudioProcessing* apm,
}
void SetAgcStatus(AudioProcessing* apm,
AudioDeviceModule* adm,
bool enable) {
RTC_DCHECK(apm);
RTC_DCHECK(adm);
#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
GainControl::Mode agc_mode = GainControl::kFixedDigital;
#else
@ -82,11 +80,6 @@ void SetAgcStatus(AudioProcessing* apm,
RTC_LOG(LS_ERROR) << "Failed to enable/disable AGC: " << enable;
return;
}
// Set AGC state in the ADM when adaptive AGC mode has been selected.
if (adm->SetAGC(enable && agc_mode == GainControl::kAdaptiveAnalog) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set AGC mode in ADM: " << enable;
return;
}
RTC_LOG(LS_INFO) << "AGC set to " << enable << " with mode " << agc_mode;
}

2
media/engine/apm_helpers.h
View File

@ -16,7 +16,6 @@
namespace webrtc {
class AudioProcessing;
class AudioDeviceModule;
enum EcModes {
kEcConference, // Conferencing default (aggressive AEC).
@ -36,7 +35,6 @@ AgcConfig GetAgcConfig(AudioProcessing* apm);
void SetAgcConfig(AudioProcessing* apm,
const AgcConfig& config);
void SetAgcStatus(AudioProcessing* apm,
AudioDeviceModule* adm,
bool enable);
void SetEcStatus(AudioProcessing* apm,
bool enable,

15
media/engine/apm_helpers_unittest.cc
View File

@ -39,10 +39,6 @@ struct TestHelper {
const AudioProcessing* apm() const { return apm_.get(); }
test::MockAudioDeviceModule* adm() {
return &mock_audio_device_;
}
voe::TransmitMixer* transmit_mixer() {
return voe_wrapper_.base()->transmit_mixer();
}
@ -117,21 +113,18 @@ TEST(ApmHelpersTest, AgcStatus_EnableDisable) {
TestHelper helper;
GainControl* gc = helper.apm()->gain_control();
#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), false);
apm_helpers::SetAgcStatus(helper.apm(), false);
EXPECT_FALSE(gc->is_enabled());
EXPECT_EQ(GainControl::kFixedDigital, gc->mode());
apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), true);
apm_helpers::SetAgcStatus(helper.apm(), true);
EXPECT_TRUE(gc->is_enabled());
EXPECT_EQ(GainControl::kFixedDigital, gc->mode());
#else
EXPECT_CALL(*helper.adm(), SetAGC(false)).WillOnce(testing::Return(0));
apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), false);
apm_helpers::SetAgcStatus(helper.apm(), false);
EXPECT_FALSE(gc->is_enabled());
EXPECT_EQ(GainControl::kAdaptiveAnalog, gc->mode());
EXPECT_CALL(*helper.adm(), SetAGC(true)).WillOnce(testing::Return(0));
apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), true);
apm_helpers::SetAgcStatus(helper.apm(), true);
EXPECT_TRUE(gc->is_enabled());
EXPECT_EQ(GainControl::kAdaptiveAnalog, gc->mode());
#endif

2
media/engine/webrtcvoiceengine.cc
View File

@ -485,7 +485,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
<< "Disabling AGC since built-in AGC will be used instead";
}
}
webrtc::apm_helpers::SetAgcStatus(apm(), adm(), *options.auto_gain_control);
webrtc::apm_helpers::SetAgcStatus(apm(), *options.auto_gain_control);
}
if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain ||

13
media/engine/webrtcvoiceengine_unittest.cc
View File

@ -124,7 +124,6 @@ void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
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));
@ -2221,7 +2220,6 @@ TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
EXPECT_CALL(adm_, BuiltInAGCIsAvailable())
.Times(1)
.WillRepeatedly(Return(false));
EXPECT_CALL(adm_, SetAGC(true)).Times(1).WillRepeatedly(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).Times(1).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, Enable(true)).Times(1).WillOnce(Return(0));
send_parameters_.options.tx_agc_target_dbov = 3;
@ -2768,7 +2766,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
SetSendParameters(send_parameters_);
// Turn off AGC
EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
@ -2777,7 +2774,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
SetSendParameters(send_parameters_);
// Turn AGC back on
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
@ -2786,7 +2782,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
SetSendParameters(send_parameters_);
// Turn off other options (and stereo swapping on).
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
@ -2803,7 +2798,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
EXPECT_FALSE(IsHighPassFilterEnabled());
// Set options again to ensure it has no impact.
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
@ -2856,7 +2850,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
parameters_options_all.options.echo_cancellation = true;
parameters_options_all.options.auto_gain_control = true;
parameters_options_all.options.noise_suppression = true;
EXPECT_CALL(adm_, SetAGC(true)).Times(2).WillRepeatedly(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).Times(2).WillRepeatedly(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).Times(2).WillRepeatedly(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).Times(2).WillOnce(Return(0));
@ -2871,7 +2864,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
// unset NS
cricket::AudioSendParameters parameters_options_no_ns = send_parameters_;
parameters_options_no_ns.options.noise_suppression = false;
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
@ -2888,7 +2880,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
// unset AGC
cricket::AudioSendParameters parameters_options_no_agc = send_parameters_;
parameters_options_no_agc.options.auto_gain_control = false;
EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
@ -2901,7 +2892,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
expected_options.noise_suppression = true;
EXPECT_EQ(expected_options, channel2->options());
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
@ -2910,7 +2900,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0));
EXPECT_TRUE(channel_->SetSendParameters(parameters_options_all));
EXPECT_CALL(adm_, SetAGC(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
@ -2919,7 +2908,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
channel1->SetSend(true);
EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));
@ -2933,7 +2921,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
send_parameters_;
parameters_options_no_agc_nor_ns.options.auto_gain_control = false;
parameters_options_no_agc_nor_ns.options.noise_suppression = false;
EXPECT_CALL(adm_, SetAGC(false)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));
EXPECT_CALL(apm_gc_, set_mode(kDefaultAgcMode)).WillOnce(Return(0));

12
modules/audio_device/android/audio_device_template.h
View File

@ -215,18 +215,6 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
bool Recording() const override { return input_.Recording(); }
int32_t SetAGC(bool enable) override {
if (enable) {
FATAL() << "Should never be called";
}
return -1;
}
bool AGC() const override {
RTC_LOG(INFO) << __FUNCTION__;
return false;
}
int32_t InitSpeaker() override {
RTC_LOG(INFO) << __FUNCTION__;
return 0;

2
modules/audio_device/audio_device_data_observer.cc
View File

@ -165,8 +165,6 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport {
int32_t StartRecording() override { return impl_->StartRecording(); }
int32_t StopRecording() override { return impl_->StopRecording(); }
bool Recording() const override { return impl_->Recording(); }
int32_t SetAGC(bool enable) override { return impl_->SetAGC(enable); }
bool AGC() const override { return impl_->AGC(); }
int32_t InitSpeaker() override { return impl_->InitSpeaker(); }
bool SpeakerIsInitialized() const override {
return impl_->SpeakerIsInitialized();

4
modules/audio_device/audio_device_generic.h
View File

@ -71,10 +71,6 @@ class AudioDeviceGeneric {
virtual int32_t StopRecording() = 0;
virtual bool Recording() const = 0;
// Microphone Automatic Gain Control (AGC)
virtual int32_t SetAGC(bool enable) = 0;
virtual bool AGC() const = 0;
// Audio mixer initialization
virtual int32_t InitSpeaker() = 0;
virtual bool SpeakerIsInitialized() const = 0;

12
modules/audio_device/audio_device_impl.cc
View File

@ -568,18 +568,6 @@ int32_t AudioDeviceModuleImpl::StereoPlayout(bool* enabled) const {
return 0;
}
int32_t AudioDeviceModuleImpl::SetAGC(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
return (audio_device_->SetAGC(enable));
}
bool AudioDeviceModuleImpl::AGC() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->AGC();
}
int32_t AudioDeviceModuleImpl::PlayoutIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();

4
modules/audio_device/audio_device_impl.h
View File

@ -87,10 +87,6 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
int32_t StopRecording() override;
bool Recording() const override;
// Microphone Automatic Gain Control (AGC)
int32_t SetAGC(bool enable) override;
bool AGC() const override;
// Audio mixer initialization
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;

2
modules/audio_device/audio_device_unittest.cc
View File

@ -496,8 +496,6 @@ class AudioDeviceTest : public ::testing::Test {
// See https://bugs.chromium.org/p/webrtc/issues/detail?id=7397 for
// details.
EXPECT_EQ(0, audio_device_->SetStereoRecording(false));
EXPECT_EQ(0, audio_device_->SetAGC(false));
EXPECT_FALSE(audio_device_->AGC());
}
}

4
modules/audio_device/dummy/audio_device_dummy.cc
View File

@ -79,10 +79,6 @@ int32_t AudioDeviceDummy::StopRecording() { return 0; }
bool AudioDeviceDummy::Recording() const { return false; }
int32_t AudioDeviceDummy::SetAGC(bool enable) { return -1; }
bool AudioDeviceDummy::AGC() const { return false; }
int32_t AudioDeviceDummy::InitSpeaker() { return -1; }
bool AudioDeviceDummy::SpeakerIsInitialized() const { return false; }

4
modules/audio_device/dummy/audio_device_dummy.h
View File

@ -65,10 +65,6 @@ class AudioDeviceDummy : public AudioDeviceGeneric {
int32_t StopRecording() override;
bool Recording() const override;
// Microphone Automatic Gain Control (AGC)
int32_t SetAGC(bool enable) override;
bool AGC() const override;
// Audio mixer initialization
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;

8
modules/audio_device/dummy/file_audio_device.cc
View File

@ -310,14 +310,6 @@ bool FileAudioDevice::Recording() const {
return _recording;
}
int32_t FileAudioDevice::SetAGC(bool enable) {
return -1;
}
bool FileAudioDevice::AGC() const {
return false;
}
int32_t FileAudioDevice::InitSpeaker() {
return -1;
}

4
modules/audio_device/dummy/file_audio_device.h
View File

@ -85,10 +85,6 @@ class FileAudioDevice : public AudioDeviceGeneric {
int32_t StopRecording() override;
bool Recording() const override;
// Microphone Automatic Gain Control (AGC)
int32_t SetAGC(bool enable) override;
bool AGC() const override;
// Audio mixer initialization
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;

6
modules/audio_device/include/audio_device.h
View File

@ -101,9 +101,9 @@ class AudioDeviceModule : public rtc::RefCountInterface {
virtual int32_t StopRecording() = 0;
virtual bool Recording() const = 0;
// Microphone Automatic Gain Control (AGC)
virtual int32_t SetAGC(bool enable) = 0;
virtual bool AGC() const = 0;
// TODO(bugs.webrtc.org/7306): deprecated (to be removed).
virtual int32_t SetAGC(bool enable) { return -1; }
virtual bool AGC() const { return false; }
// Audio mixer initialization
virtual int32_t InitSpeaker() = 0;

1
modules/audio_device/include/fake_audio_device.h
View File

@ -41,7 +41,6 @@ class FakeAudioDeviceModule : public AudioDeviceModule {
int32_t SetRecordingDevice(uint16_t index) override { return 0; }
int32_t SetRecordingDevice(WindowsDeviceType device) override { return 0; }
int32_t SetStereoRecording(bool enable) override { return 0; }
int32_t SetAGC(bool enable) override { return 0; }
int32_t StopRecording() override { return 0; }
int32_t Terminate() override { return 0; }

2
modules/audio_device/ios/audio_device_ios.h
View File

@ -93,8 +93,6 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
AudioDeviceModule::AudioLayer& audioLayer) const override;
int32_t PlayoutIsAvailable(bool& available) override;
int32_t RecordingIsAvailable(bool& available) override;
int32_t SetAGC(bool enable) override;
bool AGC() const override;
int16_t PlayoutDevices() override;
int16_t RecordingDevices() override;
int32_t PlayoutDeviceName(uint16_t index,

11
modules/audio_device/ios/audio_device_not_implemented_ios.mm
View File

@ -143,17 +143,6 @@ int32_t AudioDeviceIOS::StereoPlayout(bool& enabled) const {
return 0;
}
int32_t AudioDeviceIOS::SetAGC(bool enable) {
if (enable) {
RTC_NOTREACHED() << "Should never be called";
}
return -1;
}
bool AudioDeviceIOS::AGC() const {
return false;
}
int32_t AudioDeviceIOS::MicrophoneVolumeIsAvailable(bool& available) {
available = false;
return 0;

36
modules/audio_device/linux/audio_device_alsa_linux.cc
View File

@ -90,7 +90,6 @@ AudioDeviceLinuxALSA::AudioDeviceLinuxALSA()
_playing(false),
_recIsInitialized(false),
_playIsInitialized(false),
_AGC(false),
_recordingDelay(0),
_playoutDelay(0) {
memset(_oldKeyState, 0, sizeof(_oldKeyState));
@ -519,16 +518,6 @@ int32_t AudioDeviceLinuxALSA::StereoPlayout(bool& enabled) const {
return 0;
}
int32_t AudioDeviceLinuxALSA::SetAGC(bool enable) {
_AGC = enable;
return 0;
}
bool AudioDeviceLinuxALSA::AGC() const {
return _AGC;
}
int32_t AudioDeviceLinuxALSA::MicrophoneVolumeIsAvailable(bool& available) {
bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
@ -1593,19 +1582,6 @@ bool AudioDeviceLinuxALSA::RecThreadProcess() {
_ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
_recordingFramesIn10MS);
uint32_t currentMicLevel = 0;
uint32_t newMicLevel = 0;
if (AGC()) {
// store current mic level in the audio buffer if AGC is enabled
if (MicrophoneVolume(currentMicLevel) == 0) {
if (currentMicLevel == 0xffffffff)
currentMicLevel = 100;
// this call does not affect the actual microphone volume
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
}
}
// calculate delay
_playoutDelay = 0;
_recordingDelay = 0;
@ -1640,18 +1616,6 @@ bool AudioDeviceLinuxALSA::RecThreadProcess() {
UnLock();
_ptrAudioBuffer->DeliverRecordedData();
Lock();
if (AGC()) {
newMicLevel = _ptrAudioBuffer->NewMicLevel();
if (newMicLevel != 0) {
// The VQE will only deliver non-zero microphone levels when a
// change is needed. Set this new mic level (received from the
// observer as return value in the callback).
if (SetMicrophoneVolume(newMicLevel) == -1)
RTC_LOG(LS_WARNING)
<< "the required modification of the microphone volume failed";
}
}
}
}

5
modules/audio_device/linux/audio_device_alsa_linux.h
View File

@ -79,10 +79,6 @@ public:
int32_t StopRecording() override;
bool Recording() const override;
// Microphone Automatic Gain Control (AGC)
int32_t SetAGC(bool enable) override;
bool AGC() const override;
// Audio mixer initialization
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;
@ -191,7 +187,6 @@ private:
bool _playing;
bool _recIsInitialized;
bool _playIsInitialized;
bool _AGC;
snd_pcm_sframes_t _recordingDelay;
snd_pcm_sframes_t _playoutDelay;

41
modules/audio_device/linux/audio_device_pulse_linux.cc
View File

@ -49,7 +49,6 @@ AudioDeviceLinuxPulse::AudioDeviceLinuxPulse()
_stopRec(false),
_startPlay(false),
_stopPlay(false),
_AGC(false),
update_speaker_volume_at_startup_(false),
_sndCardPlayDelay(0),
_sndCardRecDelay(0),
@ -579,18 +578,6 @@ int32_t AudioDeviceLinuxPulse::StereoPlayout(bool& enabled) const {
return 0;
}
int32_t AudioDeviceLinuxPulse::SetAGC(bool enable) {
rtc::CritScope lock(&_critSect);
_AGC = enable;
return 0;
}
bool AudioDeviceLinuxPulse::AGC() const {
rtc::CritScope lock(&_critSect);
return _AGC;
}
int32_t AudioDeviceLinuxPulse::MicrophoneVolumeIsAvailable(bool& available) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
@ -1964,20 +1951,10 @@ int32_t AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData,
int32_t AudioDeviceLinuxPulse::ProcessRecordedData(int8_t* bufferData,
uint32_t bufferSizeInSamples,
uint32_t recDelay)
RTC_EXCLUSIVE_LOCKS_REQUIRED(_critSect) {
uint32_t currentMicLevel(0);
uint32_t newMicLevel(0);
RTC_EXCLUSIVE_LOCKS_REQUIRED(_critSect) {
_ptrAudioBuffer->SetRecordedBuffer(bufferData, bufferSizeInSamples);
if (AGC()) {
// Store current mic level in the audio buffer if AGC is enabled
if (MicrophoneVolume(currentMicLevel) == 0) {
// This call does not affect the actual microphone volume
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
}
}
const uint32_t clockDrift(0);
// TODO(andrew): this is a temporary hack, to avoid non-causal far- and
// near-end signals at the AEC for PulseAudio. I think the system delay is
@ -2001,22 +1978,6 @@ int32_t AudioDeviceLinuxPulse::ProcessRecordedData(int8_t* bufferData,
return -1;
}
if (AGC()) {
newMicLevel = _ptrAudioBuffer->NewMicLevel();
if (newMicLevel != 0) {
// The VQE will only deliver non-zero microphone levels when a
// change is needed.
// Set this new mic level (received from the observer as return
// value in the callback).
RTC_LOG(LS_VERBOSE) << "AGC change of volume: old=" << currentMicLevel
<< " => new=" << newMicLevel;
if (SetMicrophoneVolume(newMicLevel) == -1) {
RTC_LOG(LS_WARNING)
<< "the required modification of the microphone volume failed";
}
}
}
return 0;
}

5
modules/audio_device/linux/audio_device_pulse_linux.h
View File

@ -141,10 +141,6 @@ public:
int32_t StopRecording() override;
bool Recording() const override;
// Microphone Automatic Gain Control (AGC)
int32_t SetAGC(bool enable) override;
bool AGC() const override;
// Audio mixer initialization
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;
@ -289,7 +285,6 @@ private:
bool _stopRec;
bool _startPlay;
bool _stopPlay;
bool _AGC;
bool update_speaker_volume_at_startup_;
uint32_t _sndCardPlayDelay;

49
modules/audio_device/mac/audio_device_mac.cc
View File

@ -132,7 +132,6 @@ AudioDeviceMac::AudioDeviceMac()
_playing(false),
_recIsInitialized(false),
_playIsInitialized(false),
_AGC(false),
_renderDeviceIsAlive(1),
_captureDeviceIsAlive(1),
_twoDevices(true),
@ -149,8 +148,7 @@ AudioDeviceMac::AudioDeviceMac()
_paRenderBuffer(NULL),
_captureBufSizeSamples(0),
_renderBufSizeSamples(0),
prev_key_state_(),
get_mic_volume_counter_ms_(0) {
prev_key_state_() {
RTC_LOG(LS_INFO) << __FUNCTION__ << " created";
RTC_DCHECK(&_stopEvent != NULL);
@ -337,8 +335,6 @@ AudioDeviceGeneric::InitStatus AudioDeviceMac::Init() {
}
}
get_mic_volume_counter_ms_ = 0;
_initialized = true;
return InitStatus::OK;
@ -733,16 +729,6 @@ int32_t AudioDeviceMac::StereoPlayout(bool& enabled) const {
return 0;
}
int32_t AudioDeviceMac::SetAGC(bool enable) {
_AGC = enable;
return 0;
}
bool AudioDeviceMac::AGC() const {
return _AGC;
}
int32_t AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available) {
bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
@ -2494,8 +2480,6 @@ bool AudioDeviceMac::CaptureWorkerThread() {
// TODO(xians): what if the returned size is incorrect?
if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES) {
uint32_t currentMicLevel(0);
uint32_t newMicLevel(0);
int32_t msecOnPlaySide;
int32_t msecOnRecordSide;
@ -2515,43 +2499,12 @@ bool AudioDeviceMac::CaptureWorkerThread() {
// store the recorded buffer (no action will be taken if the
// #recorded samples is not a full buffer)
_ptrAudioBuffer->SetRecordedBuffer((int8_t*)&recordBuffer, (uint32_t)size);
if (AGC()) {
// Use mod to ensure we check the volume on the first pass.
if (get_mic_volume_counter_ms_ % kGetMicVolumeIntervalMs == 0) {
get_mic_volume_counter_ms_ = 0;
// store current mic level in the audio buffer if AGC is enabled
if (MicrophoneVolume(currentMicLevel) == 0) {
// this call does not affect the actual microphone volume
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
}
}
get_mic_volume_counter_ms_ += kBufferSizeMs;
}
_ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
_ptrAudioBuffer->SetTypingStatus(KeyPressed());
// deliver recorded samples at specified sample rate, mic level etc.
// to the observer using callback
_ptrAudioBuffer->DeliverRecordedData();
if (AGC()) {
newMicLevel = _ptrAudioBuffer->NewMicLevel();
if (newMicLevel != 0) {
// The VQE will only deliver non-zero microphone levels when
// a change is needed.
// Set this new mic level (received from the observer as return
// value in the callback).
RTC_LOG(LS_VERBOSE) << "AGC change of volume: old=" << currentMicLevel
<< " => new=" << newMicLevel;
if (SetMicrophoneVolume(newMicLevel) == -1) {
RTC_LOG(LS_WARNING)
<< "the required modification of the microphone volume failed";
}
}
}
}
return true;

7
modules/audio_device/mac/audio_device_mac.h
View File

@ -107,10 +107,6 @@ class AudioDeviceMac : public AudioDeviceGeneric {
virtual int32_t StopRecording();
virtual bool Recording() const;
// Microphone Automatic Gain Control (AGC)
virtual int32_t SetAGC(bool enable);
virtual bool AGC() const;
// Audio mixer initialization
virtual int32_t InitSpeaker();
virtual bool SpeakerIsInitialized() const;
@ -294,7 +290,6 @@ class AudioDeviceMac : public AudioDeviceGeneric {
bool _playing;
bool _recIsInitialized;
bool _playIsInitialized;
bool _AGC;
// Atomically set varaibles
int32_t _renderDeviceIsAlive;
@ -335,8 +330,6 @@ class AudioDeviceMac : public AudioDeviceGeneric {
// Typing detection
// 0x5c is key "9", after that comes function keys.
bool prev_key_state_[0x5d];
int get_mic_volume_counter_ms_;
};
} // namespace webrtc

153
modules/audio_device/win/audio_device_core_win.cc
View File

@ -417,9 +417,6 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore()
_hShutdownCaptureEvent(NULL),
_hRenderStartedEvent(NULL),
_hCaptureStartedEvent(NULL),
_hGetCaptureVolumeThread(NULL),
_hSetCaptureVolumeThread(NULL),
_hSetCaptureVolumeEvent(NULL),
_hMmTask(NULL),
_initialized(false),
_recording(false),
@ -428,15 +425,13 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore()
_playIsInitialized(false),
_speakerIsInitialized(false),
_microphoneIsInitialized(false),
_AGC(false),
_playBufDelay(80),
_usingInputDeviceIndex(false),
_usingOutputDeviceIndex(false),
_inputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
_outputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
_inputDeviceIndex(0),
_outputDeviceIndex(0),
_newMicLevel(0) {
_outputDeviceIndex(0) {
RTC_LOG(LS_INFO) << __FUNCTION__ << " created";
assert(_comInit.succeeded());
@ -487,7 +482,6 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore()
_hShutdownCaptureEvent = CreateEvent(NULL, TRUE, FALSE, NULL);
_hRenderStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hCaptureStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hSetCaptureVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_perfCounterFreq.QuadPart = 1;
_perfCounterFactor = 0.0;
@ -575,11 +569,6 @@ AudioDeviceWindowsCore::~AudioDeviceWindowsCore() {
_hShutdownCaptureEvent = NULL;
}
if (NULL != _hSetCaptureVolumeEvent) {
CloseHandle(_hSetCaptureVolumeEvent);
_hSetCaptureVolumeEvent = NULL;
}
if (_avrtLibrary) {
BOOL freeOK = FreeLibrary(_avrtLibrary);
if (!freeOK) {
@ -1270,25 +1259,6 @@ int32_t AudioDeviceWindowsCore::StereoPlayout(bool& enabled) const {
return 0;
}
// ----------------------------------------------------------------------------
// SetAGC
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetAGC(bool enable) {
rtc::CritScope lock(&_critSect);
_AGC = enable;
return 0;
}
// ----------------------------------------------------------------------------
// AGC
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::AGC() const {
rtc::CritScope lock(&_critSect);
return _AGC;
}
// ----------------------------------------------------------------------------
// MicrophoneVolumeIsAvailable
// ----------------------------------------------------------------------------
@ -2423,21 +2393,6 @@ int32_t AudioDeviceWindowsCore::StartRecording() {
// Set thread priority to highest possible
SetThreadPriority(_hRecThread, THREAD_PRIORITY_TIME_CRITICAL);
assert(_hGetCaptureVolumeThread == NULL);
_hGetCaptureVolumeThread =
CreateThread(NULL, 0, GetCaptureVolumeThread, this, 0, NULL);
if (_hGetCaptureVolumeThread == NULL) {
RTC_LOG(LS_ERROR) << "failed to create the volume getter thread";
return -1;
}
assert(_hSetCaptureVolumeThread == NULL);
_hSetCaptureVolumeThread =
CreateThread(NULL, 0, SetCaptureVolumeThread, this, 0, NULL);
if (_hSetCaptureVolumeThread == NULL) {
RTC_LOG(LS_ERROR) << "failed to create the volume setter thread";
return -1;
}
} // critScoped
DWORD ret = WaitForSingleObject(_hCaptureStartedEvent, 1000);
@ -2490,24 +2445,6 @@ int32_t AudioDeviceWindowsCore::StopRecording() {
} else {
RTC_LOG(LS_VERBOSE) << "webrtc_core_audio_capture_thread is now closed";
}
ret = WaitForSingleObject(_hGetCaptureVolumeThread, 2000);
if (ret != WAIT_OBJECT_0) {
// the thread did not stop as it should
RTC_LOG(LS_ERROR) << "failed to close down volume getter thread";
err = -1;
} else {
RTC_LOG(LS_VERBOSE) << "volume getter thread is now closed";
}
ret = WaitForSingleObject(_hSetCaptureVolumeThread, 2000);
if (ret != WAIT_OBJECT_0) {
// the thread did not stop as it should
RTC_LOG(LS_ERROR) << "failed to close down volume setter thread";
err = -1;
} else {
RTC_LOG(LS_VERBOSE) << "volume setter thread is now closed";
}
_Lock();
ResetEvent(_hShutdownCaptureEvent); // Must be manually reset.
@ -2523,12 +2460,6 @@ int32_t AudioDeviceWindowsCore::StopRecording() {
CloseHandle(_hRecThread);
_hRecThread = NULL;
CloseHandle(_hGetCaptureVolumeThread);
_hGetCaptureVolumeThread = NULL;
CloseHandle(_hSetCaptureVolumeThread);
_hSetCaptureVolumeThread = NULL;
if (_builtInAecEnabled) {
assert(_dmo != NULL);
// This is necessary. Otherwise the DMO can generate garbage render
@ -2736,72 +2667,6 @@ DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThreadPollDMO(LPVOID context) {
->DoCaptureThreadPollDMO();
}
DWORD WINAPI AudioDeviceWindowsCore::GetCaptureVolumeThread(LPVOID context) {
return reinterpret_cast<AudioDeviceWindowsCore*>(context)
->DoGetCaptureVolumeThread();
}
DWORD WINAPI AudioDeviceWindowsCore::SetCaptureVolumeThread(LPVOID context) {
return reinterpret_cast<AudioDeviceWindowsCore*>(context)
->DoSetCaptureVolumeThread();
}
DWORD AudioDeviceWindowsCore::DoGetCaptureVolumeThread() {
HANDLE waitObject = _hShutdownCaptureEvent;
while (1) {
if (AGC()) {
uint32_t currentMicLevel = 0;
if (MicrophoneVolume(currentMicLevel) == 0) {
// This doesn't set the system volume, just stores it.
_Lock();
if (_ptrAudioBuffer) {
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
}
_UnLock();
}
}
DWORD waitResult =
WaitForSingleObject(waitObject, GET_MIC_VOLUME_INTERVAL_MS);
switch (waitResult) {
case WAIT_OBJECT_0: // _hShutdownCaptureEvent
return 0;
case WAIT_TIMEOUT: // timeout notification
break;
default: // unexpected error
RTC_LOG(LS_WARNING) << "unknown wait termination on get volume thread";
return 1;
}
}
}
DWORD AudioDeviceWindowsCore::DoSetCaptureVolumeThread() {
HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hSetCaptureVolumeEvent};
while (1) {
DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
switch (waitResult) {
case WAIT_OBJECT_0: // _hShutdownCaptureEvent
return 0;
case WAIT_OBJECT_0 + 1: // _hSetCaptureVolumeEvent
break;
default: // unexpected error
RTC_LOG(LS_WARNING) << "unknown wait termination on set volume thread";
return 1;
}
_Lock();
uint32_t newMicLevel = _newMicLevel;
_UnLock();
if (SetMicrophoneVolume(newMicLevel) == -1) {
RTC_LOG(LS_WARNING)
<< "the required modification of the microphone volume failed";
}
}
}
// ----------------------------------------------------------------------------
// DoRenderThread
// ----------------------------------------------------------------------------
@ -3188,8 +3053,6 @@ DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO() {
break;
}
// TODO(andrew): handle AGC.
if (bytesProduced > 0) {
const int kSamplesProduced = bytesProduced / _recAudioFrameSize;
// TODO(andrew): verify that this is always satisfied. It might
@ -3451,20 +3314,6 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread() {
syncBufIndex -= _recBlockSize;
sndCardRecDelay -= 10;
}
if (_AGC) {
uint32_t newMicLevel = _ptrAudioBuffer->NewMicLevel();
if (newMicLevel != 0) {
// The VQE will only deliver non-zero microphone levels when a
// change is needed. Set this new mic level (received from the
// observer as return value in the callback).
RTC_LOG(LS_VERBOSE) << "AGC change of volume: new=" << newMicLevel;
// We store this outside of the audio buffer to avoid
// having it overwritten by the getter thread.
_newMicLevel = newMicLevel;
SetEvent(_hSetCaptureVolumeEvent);
}
}
} else {
// If GetBuffer returns AUDCLNT_E_BUFFER_ERROR, the thread consuming the
// audio samples must wait for the next processing pass. The client

18
modules/audio_device/win/audio_device_core_win.h
View File

@ -129,10 +129,6 @@ public:
virtual int32_t StopRecording();
virtual bool Recording() const;
// Microphone Automatic Gain Control (AGC)
virtual int32_t SetAGC(bool enable);
virtual bool AGC() const;
// Audio mixer initialization
virtual int32_t InitSpeaker();
virtual bool SpeakerIsInitialized() const;
@ -201,12 +197,6 @@ private: // thread functions
static DWORD WINAPI WSAPIRenderThread(LPVOID context);
DWORD DoRenderThread();
static DWORD WINAPI GetCaptureVolumeThread(LPVOID context);
DWORD DoGetCaptureVolumeThread();
static DWORD WINAPI SetCaptureVolumeThread(LPVOID context);
DWORD DoSetCaptureVolumeThread();
void _Lock() { _critSect.Enter(); };
void _UnLock() { _critSect.Leave(); };
@ -274,10 +264,6 @@ private: // thread functions
HANDLE _hCaptureStartedEvent;
HANDLE _hShutdownCaptureEvent;
HANDLE _hGetCaptureVolumeThread;
HANDLE _hSetCaptureVolumeThread;
HANDLE _hSetCaptureVolumeEvent;
HANDLE _hMmTask;
UINT _playAudioFrameSize;
@ -318,12 +304,8 @@ private:
uint16_t _inputDeviceIndex;
uint16_t _outputDeviceIndex;
bool _AGC;
uint16_t _playBufDelay;
uint16_t _newMicLevel;
mutable char _str[512];
};

10
pc/test/fakeaudiocapturemodule.cc
View File

@ -233,16 +233,6 @@ bool FakeAudioCaptureModule::Recording() const {
return recording_;
}
int32_t FakeAudioCaptureModule::SetAGC(bool /*enable*/) {
// No AGC but not needed since audio is pregenerated. Return success.
return 0;
}
bool FakeAudioCaptureModule::AGC() const {
RTC_NOTREACHED();
return 0;
}
int32_t FakeAudioCaptureModule::InitSpeaker() {
// No speaker, just playing from file. Return success.
return 0;

3
pc/test/fakeaudiocapturemodule.h
View File

@ -90,9 +90,6 @@ class FakeAudioCaptureModule
int32_t StopRecording() override;
bool Recording() const override;
int32_t SetAGC(bool enable) override;
bool AGC() const override;
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;
int32_t InitMicrophone() override;