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Removes unused sample-rate APIs from the ADM.

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The following four methods are removed:

SetRecordingSampleRate(const uint32_t samplesPerSec)
RecordingSampleRate(uint32_t* samplesPerSec) const
SetPlayoutSampleRate(const uint32_t samplesPerSec)
PlayoutSampleRate(uint32_t* samplesPerSec) const

Bug: webrtc:7306
Change-Id: I2c3c2e7bd3fb1264da197699fd5de15ab6c35c1b
Reviewed-on: https://webrtc-review.googlesource.com/22001
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20703}
This commit is contained in:
henrika
2017-11-15 14:28:59 +01:00
committed by Commit Bot
parent 19d77c1bbb
commit c97cf03ede
14 changed files with 14 additions and 173 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
media/base/mediachannel.h
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@ -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<uint16_t> tx_agc_target_dbov;
rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
rtc::Optional<bool> tx_agc_limiter;
rtc::Optional<uint32_t> recording_sample_rate;
rtc::Optional<uint32_t> playout_sample_rate;
// Enable combined audio+bandwidth BWE.
// TODO(pthatcher): This flag is set from the
// "googCombinedAudioVideoBwe", but not used anywhere. So delete it,

18
media/engine/webrtcvoiceengine.cc
View File

@ -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;
}

10
media/engine/webrtcvoiceengine_unittest.cc
View File

@ -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<uint32_t>(48000);
send_parameters_.options.playout_sample_rate = rtc::Optional<uint32_t>(44100);
SetSendParameters(send_parameters_);
}
TEST_F(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) {
EXPECT_TRUE(SetupSendStream());
send_parameters_.options.audio_network_adaptor = rtc::Optional<bool>(true);

6
modules/audio_device/android/audio_device_template.h
View File

@ -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;

12
modules/audio_device/audio_device_data_observer.cc
View File

@ -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);
}

11
modules/audio_device/audio_device_generic.cc
View File

@ -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;

4
modules/audio_device/audio_device_generic.h
View File

@ -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;

48
modules/audio_device/audio_device_impl.cc
View File

@ -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_();

6
modules/audio_device/audio_device_impl.h
View File

@ -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;

19
modules/audio_device/include/audio_device.h
View File

@ -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;

12
modules/audio_device/include/fake_audio_device.h
View File

@ -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; }

4
modules/audio_device/include/mock_audio_device.h
View File

@ -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());

24
pc/test/fakeaudiocapturemodule.cc
View File

@ -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;

5
pc/test/fakeaudiocapturemodule.h
View File

@ -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; }