Removes Set/GetRecordingChannel() from the ADM
These two unused APIs are removed: SetRecordingChannel(const ChannelType channel) RecordingChannel(ChannelType* channel) const Bug: webrtc:7306 Change-Id: I3289c4b9a5eebb64cc0aa3a1c1144e9c4d6a661d Reviewed-on: https://webrtc-review.googlesource.com/22681 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20667}
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@ -42,10 +42,7 @@ void SetRecordingDevice(AudioDeviceModule* adm) {
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return;
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}
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// Set device and stereo mode.
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if (adm->SetRecordingChannel(AudioDeviceModule::kChannelBoth) != 0) {
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RTC_LOG(LS_ERROR) << "Unable to set recording channel to kChannelBoth.";
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}
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// Set device to default.
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if (adm->SetRecordingDevice(AUDIO_DEVICE_ID) != 0) {
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RTC_LOG(LS_ERROR) << "Unable to set recording device.";
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return;
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@ -90,8 +90,6 @@ void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
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.WillOnce(Return(rtc::RefCountReleaseStatus::kDroppedLastRef));
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#if !defined(WEBRTC_IOS)
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EXPECT_CALL(*adm, Recording()).WillOnce(Return(false));
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EXPECT_CALL(*adm, SetRecordingChannel(webrtc::AudioDeviceModule::
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ChannelType::kChannelBoth)).WillOnce(Return(0));
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#if defined(WEBRTC_WIN)
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EXPECT_CALL(*adm, SetRecordingDevice(
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testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
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@ -221,22 +221,6 @@ int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
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return 0;
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}
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int32_t AudioDeviceBuffer::SetRecordingChannel(
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const AudioDeviceModule::ChannelType channel) {
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RTC_LOG(INFO) << "SetRecordingChannel(" << channel << ")";
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RTC_LOG(LS_WARNING) << "Not implemented";
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// Add DCHECK to ensure that user does not try to use this API with a non-
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// default parameter.
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RTC_DCHECK_EQ(channel, AudioDeviceModule::kChannelBoth);
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return -1;
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}
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int32_t AudioDeviceBuffer::RecordingChannel(
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AudioDeviceModule::ChannelType& channel) const {
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RTC_LOG(LS_WARNING) << "Not implemented";
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return -1;
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}
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size_t AudioDeviceBuffer::RecordingChannels() const {
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RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
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return rec_channels_;
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@ -92,8 +92,6 @@ class AudioDeviceBuffer {
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int32_t SetPlayoutChannels(size_t channels);
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size_t RecordingChannels() const;
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size_t PlayoutChannels() const;
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int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
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int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
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virtual int32_t SetRecordedBuffer(const void* audio_buffer,
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size_t samples_per_channel);
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@ -241,12 +241,6 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport {
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int32_t StereoRecording(bool* enabled) const override {
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return impl_->StereoRecording(enabled);
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}
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int32_t SetRecordingChannel(const ChannelType channel) override {
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return impl_->SetRecordingChannel(channel);
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}
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int32_t RecordingChannel(ChannelType* channel) const override {
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return impl_->RecordingChannel(channel);
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}
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int32_t PlayoutDelay(uint16_t* delay_ms) const override {
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return impl_->PlayoutDelay(delay_ms);
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}
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@ -524,41 +524,6 @@ int32_t AudioDeviceModuleImpl::StereoRecording(bool* enabled) const {
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return 0;
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}
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int32_t AudioDeviceModuleImpl::SetRecordingChannel(const ChannelType channel) {
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if (channel == kChannelBoth) {
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RTC_LOG(INFO) << __FUNCTION__ << "(both)";
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} else if (channel == kChannelLeft) {
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RTC_LOG(INFO) << __FUNCTION__ << "(left)";
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} else {
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RTC_LOG(INFO) << __FUNCTION__ << "(right)";
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}
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CHECKinitialized_();
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bool stereo = false;
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if (audio_device_->StereoRecording(stereo) == -1) {
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RTC_LOG(WARNING) << "recording in stereo is not supported";
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return -1;
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}
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return audio_device_buffer_.SetRecordingChannel(channel);
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}
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int32_t AudioDeviceModuleImpl::RecordingChannel(ChannelType* channel) const {
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RTC_LOG(INFO) << __FUNCTION__;
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CHECKinitialized_();
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ChannelType chType;
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if (audio_device_buffer_.RecordingChannel(chType) == -1) {
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return -1;
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}
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*channel = chType;
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if (*channel == kChannelBoth) {
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RTC_LOG(INFO) << "output: both";
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} else if (*channel == kChannelLeft) {
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RTC_LOG(INFO) << "output: left";
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} else {
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RTC_LOG(INFO) << "output: right";
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}
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return 0;
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}
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int32_t AudioDeviceModuleImpl::StereoPlayoutIsAvailable(bool* available) const {
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RTC_LOG(INFO) << __FUNCTION__;
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CHECKinitialized_();
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@ -128,8 +128,6 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
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int32_t StereoRecordingIsAvailable(bool* available) const override;
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int32_t SetStereoRecording(bool enable) override;
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int32_t StereoRecording(bool* enabled) const override;
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int32_t SetRecordingChannel(const ChannelType channel) override;
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int32_t RecordingChannel(ChannelType* channel) const override;
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// Delay information and control
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int32_t PlayoutDelay(uint16_t* delayMS) const override;
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@ -42,6 +42,7 @@ class AudioDeviceModule : public rtc::RefCountInterface {
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kDefaultDevice = -2
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};
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// TODO(bugs.webrtc.org/7306): deprecated.
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enum ChannelType {
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kChannelLeft = 0,
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kChannelRight = 1,
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@ -141,8 +142,9 @@ class AudioDeviceModule : public rtc::RefCountInterface {
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virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
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virtual int32_t SetStereoRecording(bool enable) = 0;
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virtual int32_t StereoRecording(bool* enabled) const = 0;
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virtual int32_t SetRecordingChannel(const ChannelType channel) = 0;
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virtual int32_t RecordingChannel(ChannelType* channel) const = 0;
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// TODO(bugs.webrtc.org/7306): deprecated.
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virtual int32_t SetRecordingChannel(const ChannelType channel) { return -1; }
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virtual int32_t RecordingChannel(ChannelType* channel) const { return -1; }
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// Playout delay
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virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
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@ -99,8 +99,6 @@ class FakeAudioDeviceModule : public AudioDeviceModule {
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return 0;
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}
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int32_t StereoRecording(bool* enabled) const override { return 0; }
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int32_t SetRecordingChannel(const ChannelType channel) override { return 0; }
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int32_t RecordingChannel(ChannelType* channel) const override { return 0; }
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int32_t PlayoutDelay(uint16_t* delayMS) const override {
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*delayMS = 0;
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return 0;
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@ -83,8 +83,6 @@ class MockAudioDeviceModule : public AudioDeviceModule {
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MOCK_CONST_METHOD1(StereoRecordingIsAvailable, int32_t(bool* available));
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MOCK_METHOD1(SetStereoRecording, int32_t(bool enable));
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MOCK_CONST_METHOD1(StereoRecording, int32_t(bool* enabled));
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MOCK_METHOD1(SetRecordingChannel, int32_t(const ChannelType channel));
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MOCK_CONST_METHOD1(RecordingChannel, int32_t(ChannelType* channel));
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MOCK_CONST_METHOD1(PlayoutDelay, int32_t(uint16_t* delayMS));
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MOCK_METHOD1(SetRecordingSampleRate, int32_t(const uint32_t samplesPerSec));
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MOCK_CONST_METHOD1(RecordingSampleRate, int32_t(uint32_t* samplesPerSec));
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@ -388,24 +388,6 @@ int32_t FakeAudioCaptureModule::StereoRecording(bool* /*enabled*/) const {
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return 0;
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}
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int32_t FakeAudioCaptureModule::SetRecordingChannel(
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const ChannelType channel) {
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if (channel != AudioDeviceModule::kChannelBoth) {
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// There is no right or left in mono. I.e. kChannelBoth should be used for
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// mono.
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RTC_NOTREACHED();
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return -1;
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}
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return 0;
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}
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int32_t FakeAudioCaptureModule::RecordingChannel(ChannelType* channel) const {
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// Stereo recording not supported. However, WebRTC ADM returns kChannelBoth
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// in that case. Do the same here.
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*channel = AudioDeviceModule::kChannelBoth;
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return 0;
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}
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int32_t FakeAudioCaptureModule::PlayoutDelay(uint16_t* delay_ms) const {
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// No delay since audio frames are dropped.
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*delay_ms = 0;
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@ -125,8 +125,6 @@ class FakeAudioCaptureModule
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int32_t StereoRecordingIsAvailable(bool* available) const override;
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int32_t SetStereoRecording(bool enable) override;
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int32_t StereoRecording(bool* enabled) const override;
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int32_t SetRecordingChannel(const ChannelType channel) override;
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int32_t RecordingChannel(ChannelType* channel) const override;
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int32_t PlayoutDelay(uint16_t* delay_ms) const override;
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