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Remove voe::Channel::StopReceive() and associated logic.

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- The legacy API is not used in WVoE/MC.
- Removed use of the API (along with StartReceive()) from unit tests.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2453243003
Cr-Commit-Position: refs/heads/master@{#14858}
This commit is contained in:
solenberg 2016-10-31 12:52:33 -07:00 committed by Commit bot
parent 9c41e47b12
commit e566ac7341
15 changed files with 6 additions and 90 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/call/call_perf_tests.cc
View File

@ -268,14 +268,12 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
fake_audio_device.Start();
EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
EXPECT_EQ(0, voe_base->StartReceive(recv_channel_id));
EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
EXPECT_TRUE(observer.Wait())
<< "Timed out while waiting for audio and video to be synchronized.";
EXPECT_EQ(0, voe_base->StopSend(send_channel_id));
EXPECT_EQ(0, voe_base->StopReceive(recv_channel_id));
EXPECT_EQ(0, voe_base->StopPlayout(recv_channel_id));
fake_audio_device.Stop();

2
webrtc/test/call_test.cc
View File

@ -128,7 +128,6 @@ void CallTest::Start() {
if (!audio_receive_streams_.empty()) {
fake_recv_audio_device_->Start();
EXPECT_EQ(0, voe_recv_.base->StartPlayout(voe_recv_.channel_id));
EXPECT_EQ(0, voe_recv_.base->StartReceive(voe_recv_.channel_id));
}
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Start();
@ -139,7 +138,6 @@ void CallTest::Stop() {
frame_generator_capturer_->Stop();
if (!audio_receive_streams_.empty()) {
fake_recv_audio_device_->Stop();
EXPECT_EQ(0, voe_recv_.base->StopReceive(voe_recv_.channel_id));
EXPECT_EQ(0, voe_recv_.base->StopPlayout(voe_recv_.channel_id));
}
for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)

2
webrtc/video/video_quality_test.cc
View File

@ -1366,7 +1366,6 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
// Start receiving audio.
audio_receive_stream->Start();
EXPECT_EQ(0, voe.base->StartPlayout(voe.receive_channel_id));
EXPECT_EQ(0, voe.base->StartReceive(voe.receive_channel_id));
// Start sending audio.
audio_send_stream_->Start();
@ -1381,7 +1380,6 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
audio_send_stream_->Stop();
// Stop receiving audio.
EXPECT_EQ(0, voe.base->StopReceive(voe.receive_channel_id));
EXPECT_EQ(0, voe.base->StopPlayout(voe.receive_channel_id));
audio_receive_stream->Stop();
}

26
webrtc/voice_engine/channel.cc
View File

@ -1222,26 +1222,10 @@ int32_t Channel::StopSend() {
return 0;
}
int32_t Channel::StartReceiving() {
void Channel::ResetDiscardedPacketCount() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StartReceiving()");
if (channel_state_.Get().receiving) {
return 0;
}
channel_state_.SetReceiving(true);
"Channel::ResetDiscardedPacketCount()");
_numberOfDiscardedPackets = 0;
return 0;
}
int32_t Channel::StopReceiving() {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::StopReceiving()");
if (!channel_state_.Get().receiving) {
return 0;
}
channel_state_.SetReceiving(false);
return 0;
}
int32_t Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) {
@ -1370,12 +1354,6 @@ int32_t Channel::SetRecPayloadType(const CodecInst& codec) {
"SetRecPayloadType() unable to set PT while playing");
return -1;
}
if (channel_state_.Get().receiving) {
_engineStatisticsPtr->SetLastError(
VE_ALREADY_LISTENING, kTraceError,
"SetRecPayloadType() unable to set PT while listening");
return -1;
}
if (codec.pltype == -1) {
// De-register the selected codec (RTP/RTCP module and ACM)

11
webrtc/voice_engine/channel.h
View File

@ -88,7 +88,6 @@ class ChannelState {
bool input_file_playing = false;
bool playing = false;
bool sending = false;
bool receiving = false;
};
ChannelState() {}
@ -129,11 +128,6 @@ class ChannelState {
state_.sending = enable;
}
void SetReceiving(bool enable) {
rtc::CritScope lock(&lock_);
state_.receiving = enable;
}
private:
rtc::CriticalSection lock_;
State state_;
@ -189,9 +183,7 @@ class Channel
int32_t StopPlayout();
int32_t StartSend();
int32_t StopSend();
int32_t StartReceiving();
int32_t StopReceiving();
void ResetDiscardedPacketCount();
int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
int32_t DeRegisterVoiceEngineObserver();
@ -393,7 +385,6 @@ class Channel
int32_t ChannelId() const { return _channelId; }
bool Playing() const { return channel_state_.Get().playing; }
bool Sending() const { return channel_state_.Get().sending; }
bool Receiving() const { return channel_state_.Get().receiving; }
bool ExternalTransport() const {
rtc::CritScope cs(&_callbackCritSect);
return _externalTransport;

2
webrtc/voice_engine/include/voe_base.h
View File

@ -165,7 +165,7 @@ class WEBRTC_DLLEXPORT VoEBase {
virtual int StartReceive(int channel) = 0;
// Stops receiving incoming RTP/RTCP packets on the specified |channel|.
virtual int StopReceive(int channel) = 0;
virtual int StopReceive(int channel) { return 0; }
// Starts forwarding the packets to the mixer/soundcard for a
// specified |channel|.

2
webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
View File

@ -47,11 +47,9 @@ void BeforeStreamingFixture::RestartFakeMicrophone() {
void BeforeStreamingFixture::PausePlaying() {
EXPECT_EQ(0, voe_base_->StopSend(channel_));
EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
EXPECT_EQ(0, voe_base_->StopReceive(channel_));
}
void BeforeStreamingFixture::ResumePlaying() {
EXPECT_EQ(0, voe_base_->StartReceive(channel_));
EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
EXPECT_EQ(0, voe_base_->StartSend(channel_));
}

2
webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc
View File

@ -48,9 +48,7 @@ TEST_F(DtmfTest, ManualCanChangeDtmfPayloadType) {
codec_instance.pltype = 88; // Use 88 instead of default 106.
EXPECT_EQ(0, voe_base_->StopSend(channel_));
EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
EXPECT_EQ(0, voe_base_->StopReceive(channel_));
EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance));
EXPECT_EQ(0, voe_base_->StartReceive(channel_));
EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
EXPECT_EQ(0, voe_base_->StartSend(channel_));
break;

2
webrtc/voice_engine/test/auto_test/standard/mixing_test.cc
View File

@ -185,7 +185,6 @@ class MixingTest : public AfterInitializationFixture {
EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(
stream, static_cast<unsigned int>(stream)));
transport_->AddChannel(stream, stream);
EXPECT_EQ(0, voe_base_->StartReceive(stream));
EXPECT_EQ(0, voe_base_->StartPlayout(stream));
EXPECT_EQ(0, voe_codec_->SetSendCodec(stream, codec_inst));
EXPECT_EQ(0, voe_base_->StartSend(stream));
@ -197,7 +196,6 @@ class MixingTest : public AfterInitializationFixture {
for (size_t i = 0; i < streams.size(); ++i) {
EXPECT_EQ(0, voe_base_->StopSend(streams[i]));
EXPECT_EQ(0, voe_base_->StopPlayout(streams[i]));
EXPECT_EQ(0, voe_base_->StopReceive(streams[i]));
EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(streams[i]));
EXPECT_EQ(0, voe_base_->DeleteChannel(streams[i]));
}

1
webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
View File

@ -67,7 +67,6 @@ class RtpRtcpTest : public AfterStreamingFixture {
EXPECT_EQ(0, voe_network_->RegisterExternalTransport(second_channel_,
*transport_));
EXPECT_EQ(0, voe_base_->StartReceive(second_channel_));
EXPECT_EQ(0, voe_base_->StartPlayout(second_channel_));
EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(second_channel_, 5678));
EXPECT_EQ(0, voe_base_->StartSend(second_channel_));

2
webrtc/voice_engine/test/auto_test/voe_cpu_test.cc
View File

@ -74,7 +74,6 @@ int VoECpuTest::DoTest() {
CHECK(codec->SetRecPayloadType(channel, isac));
CHECK(codec->SetSendCodec(channel, isac));
CHECK(base->StartReceive(channel));
CHECK(base->StartPlayout(channel));
CHECK(base->StartSend(channel));
CHECK(file->StartPlayingFileAsMicrophone(channel, _mgr.AudioFilename(),
@ -92,7 +91,6 @@ int VoECpuTest::DoTest() {
CHECK(base->StopSend(channel));
CHECK(base->StopPlayout(channel));
CHECK(base->StopReceive(channel));
base->DeleteChannel(channel);
CHECK(base->Terminate());

4
webrtc/voice_engine/test/auto_test/voe_stress_test.cc
View File

@ -149,7 +149,6 @@ int VoEStressTest::StartStopTest() {
for (i = 0; i < numberOfLoops; ++i) {
voice_channel_transport->SetSendDestination("127.0.0.1", 4800);
voice_channel_transport->SetLocalReceiver(4800);
VALIDATE_STRESS(base->StartReceive(0));
VALIDATE_STRESS(base->StartPlayout(0));
VALIDATE_STRESS(base->StartSend(0));
if (!(i % markInterval))
@ -157,21 +156,18 @@ int VoEStressTest::StartStopTest() {
SleepMs(loopSleep);
VALIDATE_STRESS(base->StopSend(0));
VALIDATE_STRESS(base->StopPlayout(0));
VALIDATE_STRESS(base->StopReceive(0));
}
ANL();
VALIDATE_STRESS(voice_channel_transport->SetSendDestination("127.0.0.1",
4800));
VALIDATE_STRESS(voice_channel_transport->SetLocalReceiver(4800));
VALIDATE_STRESS(base->StartReceive(0));
VALIDATE_STRESS(base->StartPlayout(0));
VALIDATE_STRESS(base->StartSend(0));
printf("Verify that audio is good. \n");
PAUSE_OR_SLEEP(20000);
VALIDATE_STRESS(base->StopSend(0));
VALIDATE_STRESS(base->StopPlayout(0));
VALIDATE_STRESS(base->StopReceive(0));
///////////// End test /////////////

18
webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
View File

@ -378,12 +378,6 @@ void RunTest(std::string out_path) {
const bool receive = !(call_selection == 2);
if (receive) {
#ifndef EXTERNAL_TRANSPORT
printf("Start Listen \n");
res = base1->StartReceive(chan);
VALIDATE;
#endif
printf("Start Playout \n");
res = base1->StartPlayout(chan);
VALIDATE;
@ -700,8 +694,6 @@ void RunTest(std::string out_path) {
VALIDATE;
} else if (option_selection == option_index++) {
if (channel_index < kMaxNumChannels) {
res = base1->StartReceive(channels[channel_index]);
VALIDATE;
res = base1->StartPlayout(channels[channel_index]);
VALIDATE;
res = base1->StartSend(channels[channel_index]);
@ -725,8 +717,6 @@ void RunTest(std::string out_path) {
VALIDATE;
res = base1->StopPlayout(channels[channel_index]);
VALIDATE;
res = base1->StopReceive(channels[channel_index]);
VALIDATE;
printf("Using %d additional channels\n", channel_index);
} else {
printf("All additional channels stopped\n");
@ -787,12 +777,6 @@ void RunTest(std::string out_path) {
printf("Stop Playout \n");
res = base1->StopPlayout(chan);
VALIDATE;
#ifndef EXTERNAL_TRANSPORT
printf("Stop Listen \n");
res = base1->StopReceive(chan);
VALIDATE;
#endif
}
while (channel_index > 0) {
@ -803,8 +787,6 @@ void RunTest(std::string out_path) {
VALIDATE;
res = base1->StopPlayout(channels[channel_index]);
VALIDATE;
res = base1->StopReceive(channels[channel_index]);
VALIDATE;
}
printf("\n1. New call \n");

19
webrtc/voice_engine/voe_base_impl.cc
View File

@ -436,23 +436,8 @@ int VoEBaseImpl::StartReceive(int channel) {
"StartReceive() failed to locate channel");
return -1;
}
return channelPtr->StartReceiving();
}
int VoEBaseImpl::StopReceive(int channel) {
rtc::CritScope cs(shared_->crit_sec());
if (!shared_->statistics().Initialized()) {
shared_->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
voe::Channel* channelPtr = ch.channel();
if (channelPtr == nullptr) {
shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
"SetLocalReceiver() failed to locate channel");
return -1;
}
return channelPtr->StopReceiving();
channelPtr->ResetDiscardedPacketCount();
return 0;
}
int VoEBaseImpl::StartPlayout(int channel) {

1
webrtc/voice_engine/voe_base_impl.h
View File

@ -47,7 +47,6 @@ class VoEBaseImpl : public VoEBase,
int StartReceive(int channel) override;
int StartPlayout(int channel) override;
int StartSend(int channel) override;
int StopReceive(int channel) override;
int StopPlayout(int channel) override;
int StopSend(int channel) override;