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Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.

Browse Source 
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2721003002
Cr-Commit-Position: refs/heads/master@{#16956}
This commit is contained in:
solenberg
2017-03-01 17:02:23 -08:00
committed by Commit bot
parent d48f488bed
commit 796b8f9d71
19 changed files with 65 additions and 82 deletions
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5
webrtc/audio/audio_receive_stream.cc
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@ -207,6 +207,11 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
return stats;
}
int AudioReceiveStream::GetOutputLevel() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return channel_proxy_->GetSpeechOutputLevel();
}
void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_proxy_->SetSink(std::move(sink));

1
webrtc/audio/audio_receive_stream.h
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@ -46,6 +46,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
void Start() override;
void Stop() override;
webrtc::AudioReceiveStream::Stats GetStats() const override;
int GetOutputLevel() const override;
void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
void SetGain(float gain) override;

15
webrtc/audio/audio_send_stream.cc
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@ -25,7 +25,7 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
#include "webrtc/voice_engine/transmit_mixer.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
@ -193,16 +193,11 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
}
}
// Local speech level.
{
ScopedVoEInterface<VoEVolumeControl> volume(voice_engine());
unsigned int level = 0;
int error = volume->GetSpeechInputLevelFullRange(level);
RTC_DCHECK_EQ(0, error);
stats.audio_level = static_cast<int32_t>(level);
}
ScopedVoEInterface<VoEBase> base(voice_engine());
RTC_DCHECK(base->transmit_mixer());
stats.audio_level = base->transmit_mixer()->AudioLevelFullRange();
RTC_DCHECK_LE(0, stats.audio_level);
RTC_DCHECK(base->audio_processing());
auto audio_processing_stats = base->audio_processing()->GetStatistics();
stats.echo_delay_median_ms = audio_processing_stats.delay_median;

16
webrtc/audio/audio_send_stream_unittest.cc
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@ -26,6 +26,7 @@
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
#include "webrtc/test/mock_voice_engine.h"
#include "webrtc/voice_engine/transmit_mixer.h"
namespace webrtc {
namespace test {
@ -46,7 +47,7 @@ const int kEchoDelayStdDev = -3;
const int kEchoReturnLoss = -65;
const int kEchoReturnLossEnhancement = 101;
const float kResidualEchoLikelihood = -1.0f;
const unsigned int kSpeechInputLevel = 96;
const int32_t kSpeechInputLevel = 96;
const CallStatistics kCallStats = {
1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123};
const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
@ -63,6 +64,11 @@ class MockLimitObserver : public BitrateAllocator::LimitObserver {
uint32_t max_padding_bitrate_bps));
};
class MockTransmitMixer : public voe::TransmitMixer {
public:
MOCK_CONST_METHOD0(AudioLevelFullRange, int16_t());
};
struct ConfigHelper {
explicit ConfigHelper(bool audio_bwe_enabled)
: simulated_clock_(123456),
@ -213,11 +219,14 @@ struct ConfigHelper {
.WillRepeatedly(Return(report_blocks));
EXPECT_CALL(*channel_proxy_, GetSendCodec(_))
.WillRepeatedly(DoAll(SetArgPointee<0>(kIsacCodec), Return(true)));
EXPECT_CALL(voice_engine_, GetSpeechInputLevelFullRange(_))
.WillRepeatedly(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0)));
EXPECT_CALL(voice_engine_, transmit_mixer())
.WillRepeatedly(Return(&transmit_mixer_));
EXPECT_CALL(voice_engine_, audio_processing())
.WillRepeatedly(Return(&audio_processing_));
EXPECT_CALL(transmit_mixer_, AudioLevelFullRange())
.WillRepeatedly(Return(kSpeechInputLevel));
// We have to set the instantaneous value, the average, min and max. We only
// care about the instantaneous value, so we set all to the same value.
audio_processing_stats_.echo_return_loss.Set(
@ -241,6 +250,7 @@ struct ConfigHelper {
testing::NiceMock<MockCongestionObserver> bitrate_observer_;
testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
MockAudioProcessing audio_processing_;
MockTransmitMixer transmit_mixer_;
AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
PacketRouter packet_router_;
CongestionController congestion_controller_;

2
webrtc/call/audio_receive_stream.h
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@ -116,6 +116,8 @@ class AudioReceiveStream {
virtual void Stop() = 0;
virtual Stats GetStats() const = 0;
// TODO(solenberg): Remove, once AudioMonitor is gone.
virtual int GetOutputLevel() const = 0;
// Sets an audio sink that receives unmixed audio from the receive stream.
// Ownership of the sink is passed to the stream and can be used by the

1
webrtc/media/engine/fakewebrtccall.h
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@ -94,6 +94,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
void Stop() override { started_ = false; }
webrtc::AudioReceiveStream::Stats GetStats() const override;
int GetOutputLevel() const override { return 0; }
void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
void SetGain(float gain) override;

19
webrtc/media/engine/fakewebrtcvoiceengine.h
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@ -60,8 +60,7 @@ static const int kOpusBandwidthFb = 20000;
class FakeWebRtcVoiceEngine
: public webrtc::VoEBase, public webrtc::VoECodec,
public webrtc::VoEHardware,
public webrtc::VoEVolumeControl {
public webrtc::VoEHardware {
public:
struct Channel {
std::vector<webrtc::CodecInst> recv_codecs;
@ -224,22 +223,6 @@ class FakeWebRtcVoiceEngine
WEBRTC_STUB(EnableBuiltInNS, (bool enable));
bool BuiltInNSIsAvailable() const override { return false; }
// webrtc::VoEVolumeControl
WEBRTC_STUB(SetSpeakerVolume, (unsigned int));
WEBRTC_STUB(GetSpeakerVolume, (unsigned int&));
WEBRTC_STUB(SetMicVolume, (unsigned int));
WEBRTC_STUB(GetMicVolume, (unsigned int&));
WEBRTC_STUB(SetInputMute, (int, bool));
WEBRTC_STUB(GetInputMute, (int, bool&));
WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&));
WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&));
WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&));
WEBRTC_STUB(GetSpeechOutputLevelFullRange, (int, unsigned int&));
WEBRTC_STUB(SetChannelOutputVolumeScaling, (int channel, float scale));
WEBRTC_STUB(GetChannelOutputVolumeScaling, (int channel, float& scale));
WEBRTC_STUB(SetOutputVolumePan, (int channel, float left, float right));
WEBRTC_STUB(GetOutputVolumePan, (int channel, float& left, float& right));
size_t GetNetEqCapacity() const {
auto ch = channels_.find(last_channel_);
RTC_DCHECK(ch != channels_.end());

12
webrtc/media/engine/webrtcvoe.h
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@ -21,7 +21,6 @@
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_errors.h"
#include "webrtc/voice_engine/include/voe_hardware.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
namespace cricket {
// automatically handles lifetime of WebRtc VoiceEngine
@ -78,25 +77,21 @@ class VoEWrapper {
public:
VoEWrapper()
: engine_(webrtc::VoiceEngine::Create()),
base_(engine_), codec_(engine_), hw_(engine_),
volume_(engine_) {
base_(engine_), codec_(engine_), hw_(engine_) {
}
VoEWrapper(webrtc::VoEBase* base,
webrtc::VoECodec* codec,
webrtc::VoEHardware* hw,
webrtc::VoEVolumeControl* volume)
webrtc::VoEHardware* hw)
: engine_(NULL),
base_(base),
codec_(codec),
hw_(hw),
volume_(volume) {
hw_(hw) {
}
~VoEWrapper() {}
webrtc::VoiceEngine* engine() const { return engine_.get(); }
webrtc::VoEBase* base() const { return base_.get(); }
webrtc::VoECodec* codec() const { return codec_.get(); }
webrtc::VoEHardware* hw() const { return hw_.get(); }
webrtc::VoEVolumeControl* volume() const { return volume_.get(); }
int error() { return base_->LastError(); }
private:
@ -104,7 +99,6 @@ class VoEWrapper {
scoped_voe_ptr<webrtc::VoEBase> base_;
scoped_voe_ptr<webrtc::VoECodec> codec_;
scoped_voe_ptr<webrtc::VoEHardware> hw_;
scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
};
} // namespace cricket

25
webrtc/media/engine/webrtcvoiceengine.cc
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@ -962,11 +962,12 @@ void WebRtcVoiceEngine::SetDefaultDevices() {
#endif // !WEBRTC_IOS
}
// TODO(solenberg): Remove, once AudioMonitor is gone.
int WebRtcVoiceEngine::GetInputLevel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
unsigned int ulevel;
return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
static_cast<int>(ulevel) : -1;
int8_t level = transmit_mixer()->AudioLevel();
RTC_DCHECK_LE(0, level);
return level;
}
const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const {
@ -1577,6 +1578,12 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
return stream_->GetStats();
}
int GetOutputLevel() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->GetOutputLevel();
}
int channel() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return config_.voe_channel_id;
@ -2341,12 +2348,13 @@ bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc,
return true;
}
// TODO(solenberg): Remove, once AudioMonitor is gone.
bool WebRtcVoiceMediaChannel::GetActiveStreams(
AudioInfo::StreamList* actives) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
actives->clear();
for (const auto& ch : recv_streams_) {
int level = GetOutputLevel(ch.second->channel());
int level = ch.second->GetOutputLevel();
if (level > 0) {
actives->push_back(std::make_pair(ch.first, level));
}
@ -2354,11 +2362,12 @@ bool WebRtcVoiceMediaChannel::GetActiveStreams(
return true;
}
// TODO(solenberg): Remove, once AudioMonitor is gone.
int WebRtcVoiceMediaChannel::GetOutputLevel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int highest = 0;
for (const auto& ch : recv_streams_) {
highest = std::max(GetOutputLevel(ch.second->channel()), highest);
highest = std::max(ch.second->GetOutputLevel(), highest);
}
return highest;
}
@ -2657,12 +2666,6 @@ void WebRtcVoiceMediaChannel::SetRawAudioSink(
it->second->SetRawAudioSink(std::move(sink));
}
int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
unsigned int ulevel = 0;
int ret = engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
return (ret == 0) ? static_cast<int>(ulevel) : -1;
}
int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
const auto it = recv_streams_.find(ssrc);

1
webrtc/media/engine/webrtcvoiceengine.h
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@ -237,7 +237,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
WebRtcVoiceEngine* engine() { return engine_; }
int GetLastEngineError() { return engine()->GetLastEngineError(); }
int GetOutputLevel(int channel);
void ChangePlayout(bool playout);
int CreateVoEChannel();
bool DeleteVoEChannel(int channel);

3
webrtc/media/engine/webrtcvoiceengine_unittest.cc
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@ -63,8 +63,7 @@ class FakeVoEWrapper : public cricket::VoEWrapper {
explicit FakeVoEWrapper(cricket::FakeWebRtcVoiceEngine* engine)
: cricket::VoEWrapper(engine, // base
engine, // codec
engine, // hw
engine) { // volume
engine) { // hw
}
};

1
webrtc/test/mock_voe_channel_proxy.h
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@ -42,6 +42,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
MOCK_CONST_METHOD0(GetSpeechOutputLevel, int32_t());
MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t());
MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
MOCK_METHOD2(SetSendTelephoneEventPayloadType, bool(int payload_type,

22
webrtc/test/mock_voice_engine.h
View File

@ -21,6 +21,10 @@
#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
namespace voe {
class TransmitMixer;
} // namespace voe
namespace test {
// NOTE: This class inherits from VoiceEngineImpl so that its clients will be
@ -122,6 +126,7 @@ class MockVoiceEngine : public VoiceEngineImpl {
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory));
MOCK_METHOD0(audio_processing, AudioProcessing*());
MOCK_METHOD0(audio_device_module, AudioDeviceModule*());
MOCK_METHOD0(transmit_mixer, voe::TransmitMixer*());
MOCK_METHOD0(Terminate, int());
MOCK_METHOD0(CreateChannel, int());
MOCK_METHOD1(CreateChannel, int(const ChannelConfig& config));
@ -289,23 +294,6 @@ class MockVoiceEngine : public VoiceEngineImpl {
int(int channel, bool& enable, int& redPayloadtype));
MOCK_METHOD3(SetNACKStatus, int(int channel, bool enable, int maxNoPackets));
// VoEVolumeControl
MOCK_METHOD1(SetSpeakerVolume, int(unsigned int volume));
MOCK_METHOD1(GetSpeakerVolume, int(unsigned int& volume));
MOCK_METHOD1(SetMicVolume, int(unsigned int volume));
MOCK_METHOD1(GetMicVolume, int(unsigned int& volume));
MOCK_METHOD2(SetInputMute, int(int channel, bool enable));
MOCK_METHOD2(GetInputMute, int(int channel, bool& enabled));
MOCK_METHOD1(GetSpeechInputLevel, int(unsigned int& level));
MOCK_METHOD2(GetSpeechOutputLevel, int(int channel, unsigned int& level));
MOCK_METHOD1(GetSpeechInputLevelFullRange, int(unsigned int& level));
MOCK_METHOD2(GetSpeechOutputLevelFullRange,
int(int channel, unsigned& level));
MOCK_METHOD2(SetChannelOutputVolumeScaling, int(int channel, float scaling));
MOCK_METHOD2(GetChannelOutputVolumeScaling, int(int channel, float& scaling));
MOCK_METHOD3(SetOutputVolumePan, int(int channel, float left, float right));
MOCK_METHOD3(GetOutputVolumePan, int(int channel, float& left, float& right));
private:
// TODO(ossu): I'm not particularly happy about keeping the decoder factory
// here, but due to how gmock is implemented, I cannot just keep it in the

4
webrtc/voice_engine/channel.cc
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@ -39,7 +39,6 @@
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/output_mixer.h"
#include "webrtc/voice_engine/statistics.h"
#include "webrtc/voice_engine/transmit_mixer.h"
#include "webrtc/voice_engine/utility.h"
namespace webrtc {
@ -908,7 +907,6 @@ Channel::Channel(int32_t channelId,
capture_start_ntp_time_ms_(-1),
_engineStatisticsPtr(NULL),
_outputMixerPtr(NULL),
_transmitMixerPtr(NULL),
_moduleProcessThreadPtr(NULL),
_audioDeviceModulePtr(NULL),
_voiceEngineObserverPtr(NULL),
@ -1119,7 +1117,6 @@ int32_t Channel::Init() {
int32_t Channel::SetEngineInformation(Statistics& engineStatistics,
OutputMixer& outputMixer,
voe::TransmitMixer& transmitMixer,
ProcessThread& moduleProcessThread,
AudioDeviceModule& audioDeviceModule,
VoiceEngineObserver* voiceEngineObserver,
@ -1128,7 +1125,6 @@ int32_t Channel::SetEngineInformation(Statistics& engineStatistics,
"Channel::SetEngineInformation()");
_engineStatisticsPtr = &engineStatistics;
_outputMixerPtr = &outputMixer;
_transmitMixerPtr = &transmitMixer,
_moduleProcessThreadPtr = &moduleProcessThread;
_audioDeviceModulePtr = &audioDeviceModule;
_voiceEngineObserverPtr = voiceEngineObserver;

3
webrtc/voice_engine/channel.h
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@ -71,7 +71,6 @@ class RtcpRttStatsProxy;
class RtpPacketSenderProxy;
class Statistics;
class TransportFeedbackProxy;
class TransmitMixer;
class TransportSequenceNumberProxy;
class VoERtcpObserver;
@ -155,7 +154,6 @@ class Channel
int32_t Init();
int32_t SetEngineInformation(Statistics& engineStatistics,
OutputMixer& outputMixer,
TransmitMixer& transmitMixer,
ProcessThread& moduleProcessThread,
AudioDeviceModule& audioDeviceModule,
VoiceEngineObserver* voiceEngineObserver,
@ -478,7 +476,6 @@ class Channel
// uses
Statistics* _engineStatisticsPtr;
OutputMixer* _outputMixerPtr;
TransmitMixer* _transmitMixerPtr;
ProcessThread* _moduleProcessThreadPtr;
AudioDeviceModule* _audioDeviceModulePtr;
VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base

8
webrtc/voice_engine/channel_proxy.cc
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@ -128,6 +128,14 @@ AudioDecodingCallStats ChannelProxy::GetDecodingCallStatistics() const {
return stats;
}
int32_t ChannelProxy::GetSpeechOutputLevel() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
uint32_t level = 0;
int error = channel()->GetSpeechOutputLevel(level);
RTC_DCHECK_EQ(0, error);
return static_cast<int32_t>(level);
}
int32_t ChannelProxy::GetSpeechOutputLevelFullRange() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
uint32_t level = 0;

1
webrtc/voice_engine/channel_proxy.h
View File

@ -73,6 +73,7 @@ class ChannelProxy {
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
virtual NetworkStatistics GetNetworkStatistics() const;
virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
virtual int32_t GetSpeechOutputLevel() const;
virtual int32_t GetSpeechOutputLevelFullRange() const;
virtual uint32_t GetDelayEstimate() const;
virtual bool SetSendTelephoneEventPayloadType(int payload_type,

3
webrtc/voice_engine/transmit_mixer.h
View File

@ -87,7 +87,8 @@ public:
int8_t AudioLevel() const;
int16_t AudioLevelFullRange() const;
// 'virtual' to allow mocking.
virtual int16_t AudioLevelFullRange() const;
bool IsRecordingCall();

5
webrtc/voice_engine/voe_base_impl.cc
View File

@ -374,9 +374,8 @@ int VoEBaseImpl::CreateChannel(const ChannelConfig& config) {
int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) {
if (channel_owner->channel()->SetEngineInformation(
shared_->statistics(), *shared_->output_mixer(),
*shared_->transmit_mixer(), *shared_->process_thread(),
*shared_->audio_device(), voiceEngineObserverPtr_,
&callbackCritSect_) != 0) {
*shared_->process_thread(), *shared_->audio_device(),
voiceEngineObserverPtr_, &callbackCritSect_) != 0) {
shared_->SetLastError(
VE_CHANNEL_NOT_CREATED, kTraceError,
"CreateChannel() failed to associate engine and channel."