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Replace a few calls to VoEHardware with direct calls on the ADM, in WVoMC.

Browse Source 
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1875483002

Cr-Commit-Position: refs/heads/master@{#12293}
This commit is contained in:
solenberg 2016-04-08 05:35:48 -07:00 committed by Commit bot
parent f37ba84e82
commit 5b5129a2ad
4 changed files with 42 additions and 37 deletions
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22
webrtc/media/engine/fakewebrtcvoiceengine.h
View File

@ -456,22 +456,10 @@ class FakeWebRtcVoiceEngine
WEBRTC_STUB(SetPlayoutDevice, (int));
WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) {
recording_sample_rate_ = samples_per_sec;
return 0;
}
WEBRTC_FUNC_CONST(RecordingSampleRate, (unsigned int* samples_per_sec)) {
*samples_per_sec = recording_sample_rate_;
return 0;
}
WEBRTC_FUNC(SetPlayoutSampleRate, (unsigned int samples_per_sec)) {
playout_sample_rate_ = samples_per_sec;
return 0;
}
WEBRTC_FUNC_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)) {
*samples_per_sec = playout_sample_rate_;
return 0;
}
WEBRTC_STUB(SetRecordingSampleRate, (unsigned int samples_per_sec));
WEBRTC_STUB_CONST(RecordingSampleRate, (unsigned int* samples_per_sec));
WEBRTC_STUB(SetPlayoutSampleRate, (unsigned int samples_per_sec));
WEBRTC_STUB_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec));
WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
virtual bool BuiltInAECIsAvailable() const { return false; }
WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
@ -729,8 +717,6 @@ class FakeWebRtcVoiceEngine
webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
webrtc::AgcConfig agc_config_;
int playout_fail_channel_ = -1;
int recording_sample_rate_ = -1;
int playout_sample_rate_ = -1;
FakeAudioProcessing audio_processing_;
};

26
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -655,14 +655,14 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
// Android and in combination with Java based audio layer.
// TODO(henrika): investigate possibility to support built-in EC also
// in combination with Open SL ES audio.
const bool built_in_aec = voe_wrapper_->hw()->BuiltInAECIsAvailable();
const bool built_in_aec = adm()->BuiltInAECIsAvailable();
if (built_in_aec) {
// Built-in EC exists on this device and use_delay_agnostic_aec is not
// overriding it. Enable/Disable it according to the echo_cancellation
// audio option.
const bool enable_built_in_aec =
*options.echo_cancellation && !use_delay_agnostic_aec;
if (voe_wrapper_->hw()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
if (adm()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
enable_built_in_aec) {
// Disable internal software EC if built-in EC is enabled,
// i.e., replace the software EC with the built-in EC.
@ -694,10 +694,9 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
}
if (options.auto_gain_control) {
const bool built_in_agc = voe_wrapper_->hw()->BuiltInAGCIsAvailable();
const bool built_in_agc = adm()->BuiltInAGCIsAvailable();
if (built_in_agc) {
if (voe_wrapper_->hw()->EnableBuiltInAGC(*options.auto_gain_control) ==
0 &&
if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 &&
*options.auto_gain_control) {
// Disable internal software AGC if built-in AGC is enabled,
// i.e., replace the software AGC with the built-in AGC.
@ -741,10 +740,9 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
}
if (options.noise_suppression) {
const bool built_in_ns = voe_wrapper_->hw()->BuiltInNSIsAvailable();
const bool built_in_ns = adm()->BuiltInNSIsAvailable();
if (built_in_ns) {
if (voe_wrapper_->hw()->EnableBuiltInNS(*options.noise_suppression) ==
0 &&
if (adm()->EnableBuiltInNS(*options.noise_suppression) == 0 &&
*options.noise_suppression) {
// Disable internal software NS if built-in NS is enabled,
// i.e., replace the software NS with the built-in NS.
@ -848,16 +846,14 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
if (options.recording_sample_rate) {
LOG(LS_INFO) << "Recording sample rate is "
<< *options.recording_sample_rate;
if (voe_wrapper_->hw()->SetRecordingSampleRate(
*options.recording_sample_rate)) {
if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) {
LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
}
}
if (options.playout_sample_rate) {
LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
if (voe_wrapper_->hw()->SetPlayoutSampleRate(
*options.playout_sample_rate)) {
if (adm()->SetPlayoutSampleRate(*options.playout_sample_rate)) {
LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
}
}
@ -1078,6 +1074,12 @@ int WebRtcVoiceEngine::CreateVoEChannel() {
return voe_wrapper_->base()->CreateChannel(voe_config_);
}
webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(adm_);
return adm_;
}
class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
: public AudioSource::Sink {
public:

1
webrtc/media/engine/webrtcvoiceengine.h
View File

@ -103,6 +103,7 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
void StartAecDump(const std::string& filename);
int CreateVoEChannel();
webrtc::AudioDeviceModule* adm();
rtc::ThreadChecker signal_thread_checker_;
rtc::ThreadChecker worker_thread_checker_;

30
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -67,6 +67,9 @@ TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
StrictMock<webrtc::test::MockAudioDeviceModule> adm;
EXPECT_CALL(adm, AddRef()).WillOnce(Return(0));
EXPECT_CALL(adm, Release()).WillOnce(Return(0));
EXPECT_CALL(adm, BuiltInAECIsAvailable()).WillOnce(Return(false));
EXPECT_CALL(adm, BuiltInAGCIsAvailable()).WillOnce(Return(false));
EXPECT_CALL(adm, BuiltInNSIsAvailable()).WillOnce(Return(false));
cricket::FakeWebRtcVoiceEngine voe;
EXPECT_FALSE(voe.IsInited());
{
@ -93,6 +96,9 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
: call_(webrtc::Call::Config()), override_field_trials_(field_trials) {
EXPECT_CALL(adm_, AddRef()).WillOnce(Return(0));
EXPECT_CALL(adm_, Release()).WillOnce(Return(0));
EXPECT_CALL(adm_, BuiltInAECIsAvailable()).WillOnce(Return(false));
EXPECT_CALL(adm_, BuiltInAGCIsAvailable()).WillOnce(Return(false));
EXPECT_CALL(adm_, BuiltInNSIsAvailable()).WillOnce(Return(false));
engine_.reset(new cricket::WebRtcVoiceEngine(&adm_,
new FakeVoEWrapper(&voe_)));
send_parameters_.codecs.push_back(kPcmuCodec);
@ -2402,6 +2408,8 @@ TEST_F(WebRtcVoiceEngineTestFake, CodianSend) {
TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
EXPECT_TRUE(SetupSendStream());
EXPECT_CALL(adm_,
BuiltInAGCIsAvailable()).Times(2).WillRepeatedly(Return(false));
webrtc::AgcConfig agc_config;
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
EXPECT_EQ(0, agc_config.targetLeveldBOv);
@ -2426,16 +2434,12 @@ TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
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);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
unsigned int recording_sample_rate, playout_sample_rate;
EXPECT_EQ(0, voe_.RecordingSampleRate(&recording_sample_rate));
EXPECT_EQ(0, voe_.PlayoutSampleRate(&playout_sample_rate));
EXPECT_EQ(48000u, recording_sample_rate);
EXPECT_EQ(44100u, playout_sample_rate);
}
// Test that we can set the outgoing SSRC properly.
@ -2739,7 +2743,12 @@ TEST_F(WebRtcVoiceEngineTestFake, TestSetPlayoutError) {
TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
EXPECT_TRUE(SetupSendStream());
EXPECT_CALL(adm_,
BuiltInAECIsAvailable()).Times(9).WillRepeatedly(Return(false));
EXPECT_CALL(adm_,
BuiltInAGCIsAvailable()).Times(4).WillRepeatedly(Return(false));
EXPECT_CALL(adm_,
BuiltInNSIsAvailable()).Times(2).WillRepeatedly(Return(false));
bool ec_enabled;
webrtc::EcModes ec_mode;
webrtc::AecmModes aecm_mode;
@ -2937,6 +2946,13 @@ TEST_F(WebRtcVoiceEngineTestFake, InitDoesNotOverwriteDefaultAgcConfig) {
TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
EXPECT_TRUE(SetupSendStream());
EXPECT_CALL(adm_,
BuiltInAECIsAvailable()).Times(9).WillRepeatedly(Return(false));
EXPECT_CALL(adm_,
BuiltInAGCIsAvailable()).Times(9).WillRepeatedly(Return(false));
EXPECT_CALL(adm_,
BuiltInNSIsAvailable()).Times(9).WillRepeatedly(Return(false));
std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel1(
static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel(
&call_, cricket::MediaConfig(), cricket::AudioOptions())));