diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h index 275b2f5932..b101dc7ebe 100644 --- a/media/base/mediachannel.h +++ b/media/base/mediachannel.h @@ -169,8 +169,6 @@ struct AudioOptions { SetFrom(&tx_agc_digital_compression_gain, change.tx_agc_digital_compression_gain); SetFrom(&tx_agc_limiter, change.tx_agc_limiter); - SetFrom(&recording_sample_rate, change.recording_sample_rate); - SetFrom(&playout_sample_rate, change.playout_sample_rate); SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe); SetFrom(&audio_network_adaptor, change.audio_network_adaptor); SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config); @@ -202,8 +200,6 @@ struct AudioOptions { tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain && tx_agc_limiter == o.tx_agc_limiter && - recording_sample_rate == o.recording_sample_rate && - playout_sample_rate == o.playout_sample_rate && combined_audio_video_bwe == o.combined_audio_video_bwe && audio_network_adaptor == o.audio_network_adaptor && audio_network_adaptor_config == o.audio_network_adaptor_config && @@ -240,8 +236,6 @@ struct AudioOptions { ost << ToStringIfSet("tx_agc_digital_compression_gain", tx_agc_digital_compression_gain); ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter); - ost << ToStringIfSet("recording_sample_rate", recording_sample_rate); - ost << ToStringIfSet("playout_sample_rate", playout_sample_rate); ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe); ost << ToStringIfSet("audio_network_adaptor", audio_network_adaptor); // The adaptor config is a serialized proto buffer and therefore not human @@ -284,8 +278,6 @@ struct AudioOptions { rtc::Optional tx_agc_target_dbov; rtc::Optional tx_agc_digital_compression_gain; rtc::Optional tx_agc_limiter; - rtc::Optional recording_sample_rate; - rtc::Optional playout_sample_rate; // Enable combined audio+bandwidth BWE. // TODO(pthatcher): This flag is set from the // "googCombinedAudioVideoBwe", but not used anywhere. So delete it, diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc index 1505012009..cab9f82f00 100644 --- a/media/engine/webrtcvoiceengine.cc +++ b/media/engine/webrtcvoiceengine.cc @@ -625,24 +625,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { apm()->SetExtraOptions(config); apm()->ApplyConfig(apm_config); - - if (options.recording_sample_rate) { - RTC_LOG(LS_INFO) << "Recording sample rate is " - << *options.recording_sample_rate; - if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) { - RTC_LOG(LS_WARNING) << "SetRecordingSampleRate(" - << *options.recording_sample_rate << ") failed."; - } - } - - if (options.playout_sample_rate) { - RTC_LOG(LS_INFO) << "Playout sample rate is " - << *options.playout_sample_rate; - if (adm()->SetPlayoutSampleRate(*options.playout_sample_rate)) { - RTC_LOG(LS_WARNING) << "SetPlayoutSampleRate(" - << *options.playout_sample_rate << ") failed."; - } - } return true; } diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc index 793fc7d611..e370612cb3 100644 --- a/media/engine/webrtcvoiceengine_unittest.cc +++ b/media/engine/webrtcvoiceengine_unittest.cc @@ -2238,16 +2238,6 @@ TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) { SetSendParameters(send_parameters_); } -TEST_F(WebRtcVoiceEngineTestFake, SampleRatesViaOptions) { - EXPECT_TRUE(SetupSendStream()); - EXPECT_CALL(adm_, SetRecordingSampleRate(48000)).WillOnce(Return(0)); - EXPECT_CALL(adm_, SetPlayoutSampleRate(44100)).WillOnce(Return(0)); - send_parameters_.options.recording_sample_rate = - rtc::Optional(48000); - send_parameters_.options.playout_sample_rate = rtc::Optional(44100); - SetSendParameters(send_parameters_); -} - TEST_F(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) { EXPECT_TRUE(SetupSendStream()); send_parameters_.options.audio_network_adaptor = rtc::Optional(true); diff --git a/modules/audio_device/android/audio_device_template.h b/modules/audio_device/android/audio_device_template.h index bb577ad73c..0b883bd76a 100644 --- a/modules/audio_device/android/audio_device_template.h +++ b/modules/audio_device/android/audio_device_template.h @@ -385,12 +385,6 @@ class AudioDeviceTemplate : public AudioDeviceGeneric { input_.AttachAudioBuffer(audioBuffer); } - // TODO(henrika): remove - int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) override { - FATAL() << "Should never be called"; - return -1; - } - int32_t SetLoudspeakerStatus(bool enable) override { FATAL() << "Should never be called"; return -1; diff --git a/modules/audio_device/audio_device_data_observer.cc b/modules/audio_device/audio_device_data_observer.cc index f37d808e23..0dead0779f 100644 --- a/modules/audio_device/audio_device_data_observer.cc +++ b/modules/audio_device/audio_device_data_observer.cc @@ -244,18 +244,6 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport { int32_t PlayoutDelay(uint16_t* delay_ms) const override { return impl_->PlayoutDelay(delay_ms); } - int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override { - return impl_->SetRecordingSampleRate(samples_per_sec); - } - int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override { - return impl_->RecordingSampleRate(samples_per_sec); - } - int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override { - return impl_->SetPlayoutSampleRate(samples_per_sec); - } - int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override { - return impl_->PlayoutSampleRate(samples_per_sec); - } int32_t SetLoudspeakerStatus(bool enable) override { return impl_->SetLoudspeakerStatus(enable); } diff --git a/modules/audio_device/audio_device_generic.cc b/modules/audio_device/audio_device_generic.cc index e7c1f0ec69..41ed2f1d84 100644 --- a/modules/audio_device/audio_device_generic.cc +++ b/modules/audio_device/audio_device_generic.cc @@ -13,17 +13,6 @@ namespace webrtc { -int32_t AudioDeviceGeneric::SetRecordingSampleRate( - const uint32_t samplesPerSec) { - RTC_LOG_F(LS_ERROR) << "Not supported on this platform"; - return -1; -} - -int32_t AudioDeviceGeneric::SetPlayoutSampleRate(const uint32_t samplesPerSec) { - RTC_LOG_F(LS_ERROR) << "Not supported on this platform"; - return -1; -} - int32_t AudioDeviceGeneric::SetLoudspeakerStatus(bool enable) { RTC_LOG_F(LS_ERROR) << "Not supported on this platform"; return -1; diff --git a/modules/audio_device/audio_device_generic.h b/modules/audio_device/audio_device_generic.h index a8c6270e72..61efd009e0 100644 --- a/modules/audio_device/audio_device_generic.h +++ b/modules/audio_device/audio_device_generic.h @@ -116,10 +116,6 @@ class AudioDeviceGeneric { // Delay information and control virtual int32_t PlayoutDelay(uint16_t& delayMS) const = 0; - // Native sample rate controls (samples/sec) - virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec); - virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec); - // Speaker audio routing (for mobile devices) virtual int32_t SetLoudspeakerStatus(bool enable); virtual int32_t GetLoudspeakerStatus(bool& enable) const; diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc index 46489dafb2..24c28f47bf 100644 --- a/modules/audio_device/audio_device_impl.cc +++ b/modules/audio_device/audio_device_impl.cc @@ -823,54 +823,6 @@ int32_t AudioDeviceModuleImpl::PlayoutDelay(uint16_t* delayMS) const { return 0; } -int32_t AudioDeviceModuleImpl::SetRecordingSampleRate( - const uint32_t samplesPerSec) { - RTC_LOG(INFO) << __FUNCTION__ << "(" << samplesPerSec << ")"; - CHECKinitialized_(); - if (audio_device_->SetRecordingSampleRate(samplesPerSec) != 0) { - return -1; - } - return 0; -} - -int32_t AudioDeviceModuleImpl::RecordingSampleRate( - uint32_t* samplesPerSec) const { - RTC_LOG(INFO) << __FUNCTION__; - CHECKinitialized_(); - int32_t sampleRate = audio_device_buffer_.RecordingSampleRate(); - if (sampleRate == -1) { - RTC_LOG(LERROR) << "failed to retrieve the sample rate"; - return -1; - } - *samplesPerSec = sampleRate; - RTC_LOG(INFO) << "output: " << *samplesPerSec; - return 0; -} - -int32_t AudioDeviceModuleImpl::SetPlayoutSampleRate( - const uint32_t samplesPerSec) { - RTC_LOG(INFO) << __FUNCTION__ << "(" << samplesPerSec << ")"; - CHECKinitialized_(); - if (audio_device_->SetPlayoutSampleRate(samplesPerSec) != 0) { - return -1; - } - return 0; -} - -int32_t AudioDeviceModuleImpl::PlayoutSampleRate( - uint32_t* samplesPerSec) const { - RTC_LOG(INFO) << __FUNCTION__; - CHECKinitialized_(); - int32_t sampleRate = audio_device_buffer_.PlayoutSampleRate(); - if (sampleRate == -1) { - RTC_LOG(LERROR) << "failed to retrieve the sample rate"; - return -1; - } - *samplesPerSec = sampleRate; - RTC_LOG(INFO) << "output: " << *samplesPerSec; - return 0; -} - int32_t AudioDeviceModuleImpl::SetLoudspeakerStatus(bool enable) { RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")"; CHECKinitialized_(); diff --git a/modules/audio_device/audio_device_impl.h b/modules/audio_device/audio_device_impl.h index cd4d0ccefd..54af2e64f6 100644 --- a/modules/audio_device/audio_device_impl.h +++ b/modules/audio_device/audio_device_impl.h @@ -132,12 +132,6 @@ class AudioDeviceModuleImpl : public AudioDeviceModule { // Delay information and control int32_t PlayoutDelay(uint16_t* delayMS) const override; - // Native sample rate controls (samples/sec) - int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) override; - int32_t RecordingSampleRate(uint32_t* samplesPerSec) const override; - int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) override; - int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const override; - // Mobile device specific functions int32_t SetLoudspeakerStatus(bool enable) override; int32_t GetLoudspeakerStatus(bool* enabled) const override; diff --git a/modules/audio_device/include/audio_device.h b/modules/audio_device/include/audio_device.h index 52d367adbd..928e6332b3 100644 --- a/modules/audio_device/include/audio_device.h +++ b/modules/audio_device/include/audio_device.h @@ -149,11 +149,20 @@ class AudioDeviceModule : public rtc::RefCountInterface { // Playout delay virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0; - // Native sample rate controls (samples/sec) - virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) = 0; - virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const = 0; - virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) = 0; - virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const = 0; + // Deprecated. Don't use. + // TODO(henrika): remove these methods. + virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) { + return -1; + } + virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const { + return -1; + } + virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) { + return -1; + } + virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const { + return -1; + } // Mobile device specific functions virtual int32_t SetLoudspeakerStatus(bool enable) = 0; diff --git a/modules/audio_device/include/fake_audio_device.h b/modules/audio_device/include/fake_audio_device.h index 74822d5d56..c271d6e3ba 100644 --- a/modules/audio_device/include/fake_audio_device.h +++ b/modules/audio_device/include/fake_audio_device.h @@ -103,18 +103,6 @@ class FakeAudioDeviceModule : public AudioDeviceModule { *delayMS = 0; return 0; } - int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) override { - return 0; - } - int32_t RecordingSampleRate(uint32_t* samplesPerSec) const override { - return 0; - } - int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) override { - return 0; - } - int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const override { - return 0; - } int32_t SetLoudspeakerStatus(bool enable) override { return 0; } int32_t GetLoudspeakerStatus(bool* enabled) const override { return 0; } bool BuiltInAECIsAvailable() const override { return false; } diff --git a/modules/audio_device/include/mock_audio_device.h b/modules/audio_device/include/mock_audio_device.h index bfc76601a1..268b358346 100644 --- a/modules/audio_device/include/mock_audio_device.h +++ b/modules/audio_device/include/mock_audio_device.h @@ -84,10 +84,6 @@ class MockAudioDeviceModule : public AudioDeviceModule { MOCK_METHOD1(SetStereoRecording, int32_t(bool enable)); MOCK_CONST_METHOD1(StereoRecording, int32_t(bool* enabled)); MOCK_CONST_METHOD1(PlayoutDelay, int32_t(uint16_t* delayMS)); - MOCK_METHOD1(SetRecordingSampleRate, int32_t(const uint32_t samplesPerSec)); - MOCK_CONST_METHOD1(RecordingSampleRate, int32_t(uint32_t* samplesPerSec)); - MOCK_METHOD1(SetPlayoutSampleRate, int32_t(const uint32_t samplesPerSec)); - MOCK_CONST_METHOD1(PlayoutSampleRate, int32_t(uint32_t* samplesPerSec)); MOCK_METHOD1(SetLoudspeakerStatus, int32_t(bool enable)); MOCK_CONST_METHOD1(GetLoudspeakerStatus, int32_t(bool* enabled)); MOCK_CONST_METHOD0(BuiltInAECIsAvailable, bool()); diff --git a/pc/test/fakeaudiocapturemodule.cc b/pc/test/fakeaudiocapturemodule.cc index 901137623e..26557c0adc 100644 --- a/pc/test/fakeaudiocapturemodule.cc +++ b/pc/test/fakeaudiocapturemodule.cc @@ -394,30 +394,6 @@ int32_t FakeAudioCaptureModule::PlayoutDelay(uint16_t* delay_ms) const { return 0; } -int32_t FakeAudioCaptureModule::SetRecordingSampleRate( - const uint32_t /*samples_per_sec*/) { - RTC_NOTREACHED(); - return 0; -} - -int32_t FakeAudioCaptureModule::RecordingSampleRate( - uint32_t* /*samples_per_sec*/) const { - RTC_NOTREACHED(); - return 0; -} - -int32_t FakeAudioCaptureModule::SetPlayoutSampleRate( - const uint32_t /*samples_per_sec*/) { - RTC_NOTREACHED(); - return 0; -} - -int32_t FakeAudioCaptureModule::PlayoutSampleRate( - uint32_t* /*samples_per_sec*/) const { - RTC_NOTREACHED(); - return 0; -} - int32_t FakeAudioCaptureModule::SetLoudspeakerStatus(bool /*enable*/) { RTC_NOTREACHED(); return 0; diff --git a/pc/test/fakeaudiocapturemodule.h b/pc/test/fakeaudiocapturemodule.h index 9851bc3529..a2988089b7 100644 --- a/pc/test/fakeaudiocapturemodule.h +++ b/pc/test/fakeaudiocapturemodule.h @@ -128,11 +128,6 @@ class FakeAudioCaptureModule int32_t PlayoutDelay(uint16_t* delay_ms) const override; - int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override; - int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override; - int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override; - int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override; - int32_t SetLoudspeakerStatus(bool enable) override; int32_t GetLoudspeakerStatus(bool* enabled) const override; bool BuiltInAECIsAvailable() const override { return false; }