Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
- Functionality now implemented in AudioReceiveStream and Call.
- Added some missing function to MockChannelProxy.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2461523002
Cr-Commit-Position: refs/heads/master@{#15072}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index cc30939..e2f8f14 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -14,6 +14,7 @@
#include <utility>
#include "webrtc/api/call/audio_sink.h"
+#include "webrtc/audio/audio_send_stream.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/base/checks.h"
@@ -143,6 +144,7 @@
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
Stop();
+ channel_proxy_->DisassociateSendChannel();
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->ResetCongestionControlObjects();
channel_proxy_->SetRtcEventLog(nullptr);
@@ -230,6 +232,18 @@
return config_;
}
+void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (send_stream) {
+ VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
+ std::unique_ptr<voe::ChannelProxy> send_channel_proxy =
+ voe_impl->GetChannelProxy(send_stream->config().voe_channel_id);
+ channel_proxy_->AssociateSendChannel(*send_channel_proxy.get());
+ } else {
+ channel_proxy_->DisassociateSendChannel();
+ }
+}
+
void AudioReceiveStream::SignalNetworkState(NetworkState state) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 906ddb6..aeaadc6 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -30,6 +30,7 @@
} // namespace voe
namespace internal {
+class AudioSendStream;
class AudioReceiveStream final : public webrtc::AudioReceiveStream,
public AudioMixer::Source {
@@ -47,6 +48,7 @@
void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
void SetGain(float gain) override;
+ void AssociateSendStream(AudioSendStream* send_stream);
void SignalNetworkState(NetworkState state);
bool DeliverRtcp(const uint8_t* packet, size_t length);
bool DeliverRtp(const uint8_t* packet,
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index bde68ed..0331d9b 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -113,6 +113,7 @@
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
.Times(1)
.After(expect_set);
+ EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1);
return channel_proxy_;
}));
EXPECT_CALL(voice_engine_, StopPlayout(kChannelId)).WillOnce(Return(0));
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index ebc1aeb..3ed7947 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -407,6 +407,14 @@
audio_send_ssrcs_.end());
audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
}
+ {
+ ReadLockScoped read_lock(*receive_crit_);
+ for (const auto& kv : audio_receive_ssrcs_) {
+ if (kv.second->config().rtp.local_ssrc == config.rtp.ssrc) {
+ kv.second->AssociateSendStream(send_stream);
+ }
+ }
+ }
send_stream->SignalNetworkState(audio_network_state_);
UpdateAggregateNetworkState();
return send_stream;
@@ -421,11 +429,19 @@
webrtc::internal::AudioSendStream* audio_send_stream =
static_cast<webrtc::internal::AudioSendStream*>(send_stream);
+ uint32_t ssrc = audio_send_stream->config().rtp.ssrc;
{
WriteLockScoped write_lock(*send_crit_);
- size_t num_deleted = audio_send_ssrcs_.erase(
- audio_send_stream->config().rtp.ssrc);
- RTC_DCHECK(num_deleted == 1);
+ size_t num_deleted = audio_send_ssrcs_.erase(ssrc);
+ RTC_DCHECK_EQ(1, num_deleted);
+ }
+ {
+ ReadLockScoped read_lock(*receive_crit_);
+ for (const auto& kv : audio_receive_ssrcs_) {
+ if (kv.second->config().rtp.local_ssrc == ssrc) {
+ kv.second->AssociateSendStream(nullptr);
+ }
+ }
}
UpdateAggregateNetworkState();
delete audio_send_stream;
@@ -445,6 +461,13 @@
audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
ConfigureSync(config.sync_group);
}
+ {
+ ReadLockScoped read_lock(*send_crit_);
+ auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc);
+ if (it != audio_send_ssrcs_.end()) {
+ receive_stream->AssociateSendStream(it->second);
+ }
+ }
receive_stream->SignalNetworkState(audio_network_state_);
UpdateAggregateNetworkState();
return receive_stream;
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index 1cdd48a..2b93deb 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -32,6 +32,7 @@
}
webrtc::Call* operator->() { return call_.get(); }
+ webrtc::test::MockVoiceEngine* voice_engine() { return &voice_engine_; }
private:
testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
@@ -118,6 +119,102 @@
}
}
+TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
+ rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
+ new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
+ CallHelper call(decoder_factory);
+
+ constexpr int kRecvChannelId = 101;
+
+ // Set up the mock to create a channel proxy which we know of, so that we can
+ // add our expectations to it.
+ test::MockVoEChannelProxy* recv_channel_proxy = nullptr;
+ EXPECT_CALL(*call.voice_engine(), ChannelProxyFactory(testing::_))
+ .WillRepeatedly(testing::Invoke([&](int channel_id) {
+ test::MockVoEChannelProxy* channel_proxy =
+ new testing::NiceMock<test::MockVoEChannelProxy>();
+ EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
+ .WillRepeatedly(testing::ReturnRef(decoder_factory));
+ // If being called for the send channel, save a pointer to the channel
+ // proxy for later.
+ if (channel_id == kRecvChannelId) {
+ EXPECT_FALSE(recv_channel_proxy);
+ recv_channel_proxy = channel_proxy;
+ }
+ return channel_proxy;
+ }));
+
+ AudioReceiveStream::Config recv_config;
+ recv_config.rtp.remote_ssrc = 42;
+ recv_config.rtp.local_ssrc = 777;
+ recv_config.voe_channel_id = kRecvChannelId;
+ recv_config.decoder_factory = decoder_factory;
+ AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config);
+ EXPECT_NE(recv_stream, nullptr);
+
+ EXPECT_CALL(*recv_channel_proxy, AssociateSendChannel(testing::_)).Times(1);
+ AudioSendStream::Config send_config(nullptr);
+ send_config.rtp.ssrc = 777;
+ send_config.voe_channel_id = 123;
+ AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
+ EXPECT_NE(send_stream, nullptr);
+
+ EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1);
+ call->DestroyAudioSendStream(send_stream);
+
+ EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1);
+ call->DestroyAudioReceiveStream(recv_stream);
+}
+
+TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
+ rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
+ new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
+ CallHelper call(decoder_factory);
+
+ constexpr int kRecvChannelId = 101;
+
+ // Set up the mock to create a channel proxy which we know of, so that we can
+ // add our expectations to it.
+ test::MockVoEChannelProxy* recv_channel_proxy = nullptr;
+ EXPECT_CALL(*call.voice_engine(), ChannelProxyFactory(testing::_))
+ .WillRepeatedly(testing::Invoke([&](int channel_id) {
+ test::MockVoEChannelProxy* channel_proxy =
+ new testing::NiceMock<test::MockVoEChannelProxy>();
+ EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
+ .WillRepeatedly(testing::ReturnRef(decoder_factory));
+ // If being called for the send channel, save a pointer to the channel
+ // proxy for later.
+ if (channel_id == kRecvChannelId) {
+ EXPECT_FALSE(recv_channel_proxy);
+ recv_channel_proxy = channel_proxy;
+ // We need to set this expectation here since the channel proxy is
+ // created as a side effect of CreateAudioReceiveStream().
+ EXPECT_CALL(*recv_channel_proxy,
+ AssociateSendChannel(testing::_)).Times(1);
+ }
+ return channel_proxy;
+ }));
+
+ AudioSendStream::Config send_config(nullptr);
+ send_config.rtp.ssrc = 777;
+ send_config.voe_channel_id = 123;
+ AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
+ EXPECT_NE(send_stream, nullptr);
+
+ AudioReceiveStream::Config recv_config;
+ recv_config.rtp.remote_ssrc = 42;
+ recv_config.rtp.local_ssrc = 777;
+ recv_config.voe_channel_id = kRecvChannelId;
+ recv_config.decoder_factory = decoder_factory;
+ AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config);
+ EXPECT_NE(recv_stream, nullptr);
+
+ EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1);
+ call->DestroyAudioReceiveStream(recv_stream);
+
+ call->DestroyAudioSendStream(send_stream);
+}
+
TEST(CallTest, CreateDestroy_FlexfecReceiveStream) {
CallHelper call;
FlexfecReceiveStream::Config config;
diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index 2641904..cd6c420 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -57,7 +57,6 @@
public webrtc::VoEVolumeControl {
public:
struct Channel {
- int associate_send_channel = -1;
std::vector<webrtc::CodecInst> recv_codecs;
size_t neteq_capacity = 0;
bool neteq_fast_accelerate = false;
@@ -79,10 +78,6 @@
fail_create_channel_ = fail_create_channel;
}
- int GetAssociateSendChannel(int channel) {
- return channels_[channel]->associate_send_channel;
- }
-
WEBRTC_STUB(Release, ());
// webrtc::VoEBase
@@ -125,11 +120,6 @@
}
WEBRTC_FUNC(DeleteChannel, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
- for (const auto& ch : channels_) {
- if (ch.second->associate_send_channel == channel) {
- ch.second->associate_send_channel = -1;
- }
- }
delete channels_[channel];
channels_.erase(channel);
return 0;
@@ -142,12 +132,8 @@
WEBRTC_STUB(StopSend, (int channel));
WEBRTC_STUB(GetVersion, (char version[1024]));
WEBRTC_STUB(LastError, ());
- WEBRTC_FUNC(AssociateSendChannel, (int channel,
- int accociate_send_channel)) {
- WEBRTC_CHECK_CHANNEL(channel);
- channels_[channel]->associate_send_channel = accociate_send_channel;
- return 0;
- }
+ WEBRTC_STUB(AssociateSendChannel, (int channel,
+ int accociate_send_channel));
// webrtc::VoECodec
WEBRTC_STUB(NumOfCodecs, ());
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index a4768db..35629da 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -2085,10 +2085,6 @@
// TODO(solenberg): Allow applications to set the RTCP SSRC of receive
// streams instead, so we can avoid recreating the streams here.
kv.second->RecreateAudioReceiveStream(ssrc);
- int recv_channel = kv.second->channel();
- engine()->voe()->base()->AssociateSendChannel(recv_channel, channel);
- LOG(LS_INFO) << "VoiceEngine channel #" << recv_channel
- << " is associated with channel #" << channel << ".";
}
}
@@ -2110,6 +2106,10 @@
it->second->SetSend(false);
+ // TODO(solenberg): If we're removing the receiver_reports_ssrc_ stream, find
+ // the first active send stream and use that instead, reassociating receive
+ // streams.
+
// Clean up and delete the send stream+channel.
int channel = it->second->channel();
LOG(LS_INFO) << "Removing audio send stream " << ssrc
@@ -2182,15 +2182,6 @@
}
}
- const int send_channel = GetSendChannelId(receiver_reports_ssrc_);
- if (send_channel != -1) {
- // Associate receive channel with first send channel (so the receive channel
- // can obtain RTT from the send channel)
- engine()->voe()->base()->AssociateSendChannel(channel, send_channel);
- LOG(LS_INFO) << "VoiceEngine channel #" << channel
- << " is associated with channel #" << send_channel << ".";
- }
-
recv_streams_.insert(std::make_pair(
ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
recv_transport_cc_enabled_,
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 93300e4..6f0ff63 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -50,6 +50,7 @@
const uint32_t kSsrc1 = 0x99;
const uint32_t kSsrc2 = 2;
const uint32_t kSsrc3 = 3;
+const uint32_t kSsrc4 = 0x42;
const uint32_t kSsrcs4[] = { 1, 2, 3, 4 };
constexpr int kRtpHistoryMs = 5000;
@@ -3239,42 +3240,29 @@
// All receive channels should be associated with the first send channel,
// since they do not send RTCP SR.
-TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel) {
+TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) {
EXPECT_TRUE(SetupSendStream());
- SetSendParameters(send_parameters_);
- int default_channel = voe_.GetLastChannel();
- EXPECT_TRUE(AddRecvStream(1));
- int recv_ch = voe_.GetLastChannel();
- EXPECT_NE(recv_ch, default_channel);
- EXPECT_EQ(voe_.GetAssociateSendChannel(recv_ch), default_channel);
- EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(2)));
- EXPECT_EQ(voe_.GetAssociateSendChannel(recv_ch), default_channel);
- EXPECT_TRUE(AddRecvStream(3));
- recv_ch = voe_.GetLastChannel();
- EXPECT_NE(recv_ch, default_channel);
- EXPECT_EQ(voe_.GetAssociateSendChannel(recv_ch), default_channel);
+ EXPECT_TRUE(AddRecvStream(kSsrc2));
+ EXPECT_EQ(kSsrc1, GetRecvStreamConfig(kSsrc2).rtp.local_ssrc);
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(kSsrc3)));
+ EXPECT_EQ(kSsrc1, GetRecvStreamConfig(kSsrc2).rtp.local_ssrc);
+ EXPECT_TRUE(AddRecvStream(kSsrc4));
+ EXPECT_EQ(kSsrc1, GetRecvStreamConfig(kSsrc4).rtp.local_ssrc);
}
-TEST_F(WebRtcVoiceEngineTestFake, AssociateChannelResetUponDeleteChannnel) {
- EXPECT_TRUE(SetupSendStream());
- SetSendParameters(send_parameters_);
-
- EXPECT_TRUE(AddRecvStream(1));
- int recv_ch = voe_.GetLastChannel();
-
- EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(2)));
- int send_ch = voe_.GetLastChannel();
-
- // Manually associate |recv_ch| to |send_ch|. This test is to verify a
- // deleting logic, i.e., deleting |send_ch| will reset the associate send
- // channel of |recv_ch|.This is not a common case, sinceļ¼ normally, only the
- // default channel can be associated. However, the default is not deletable.
- // So we force the |recv_ch| to associate with a non-default channel.
- EXPECT_EQ(0, voe_.AssociateSendChannel(recv_ch, send_ch));
- EXPECT_EQ(voe_.GetAssociateSendChannel(recv_ch), send_ch);
-
- EXPECT_TRUE(channel_->RemoveSendStream(2));
- EXPECT_EQ(voe_.GetAssociateSendChannel(recv_ch), -1);
+TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_RecvCreatedFirst) {
+ EXPECT_TRUE(SetupRecvStream());
+ EXPECT_EQ(0xFA17FA17u, GetRecvStreamConfig(kSsrc1).rtp.local_ssrc);
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(kSsrc2)));
+ EXPECT_EQ(kSsrc2, GetRecvStreamConfig(kSsrc1).rtp.local_ssrc);
+ EXPECT_TRUE(AddRecvStream(kSsrc3));
+ EXPECT_EQ(kSsrc2, GetRecvStreamConfig(kSsrc3).rtp.local_ssrc);
+ EXPECT_TRUE(channel_->AddSendStream(
+ cricket::StreamParams::CreateLegacy(kSsrc4)));
+ EXPECT_EQ(kSsrc2, GetRecvStreamConfig(kSsrc1).rtp.local_ssrc);
+ EXPECT_EQ(kSsrc2, GetRecvStreamConfig(kSsrc3).rtp.local_ssrc);
}
TEST_F(WebRtcVoiceEngineTestFake, SetRawAudioSink) {
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index 1f4b14f..aea30e8 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -44,9 +44,10 @@
MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type));
MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
- MOCK_METHOD1(SetInputMute, void(bool muted));
+ MOCK_METHOD1(SetBitrate, void(int bitrate_bps));
// TODO(solenberg): Talk the compiler into accepting this mock method:
// MOCK_METHOD1(SetSink, void(std::unique_ptr<AudioSinkInterface> sink));
+ MOCK_METHOD1(SetInputMute, void(bool muted));
MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport));
MOCK_METHOD0(DeRegisterExternalTransport, void());
MOCK_METHOD3(ReceivedRTPPacket, bool(const uint8_t* packet,
@@ -57,13 +58,19 @@
const rtc::scoped_refptr<AudioDecoderFactory>&());
MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
- MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
- MOCK_METHOD1(SetBitrate, void(int bitrate_bps));
MOCK_METHOD1(EnableAudioNetworkAdaptor,
void(const std::string& config_string));
MOCK_METHOD0(DisableAudioNetworkAdaptor, void());
MOCK_METHOD2(SetReceiverFrameLengthRange,
void(int min_frame_length_ms, int max_frame_length_ms));
+ MOCK_METHOD2(GetAudioFrameWithInfo,
+ AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
+ AudioFrame* audio_frame));
+ MOCK_CONST_METHOD0(NeededFrequency, int());
+ MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
+ MOCK_METHOD1(AssociateSendChannel,
+ void(const ChannelProxy& send_channel_proxy));
+ MOCK_METHOD0(DisassociateSendChannel, void());
};
} // namespace test
} // namespace webrtc
diff --git a/webrtc/test/mock_voice_engine.h b/webrtc/test/mock_voice_engine.h
index 85d27bd..84c1f5a 100644
--- a/webrtc/test/mock_voice_engine.h
+++ b/webrtc/test/mock_voice_engine.h
@@ -36,7 +36,7 @@
++_ref_count;
// We add this default behavior to make the mock easier to use in tests. It
// will create a NiceMock of a voe::ChannelProxy.
- // TODO(ossu): As long as AudioReceiveStream is implmented as a wrapper
+ // TODO(ossu): As long as AudioReceiveStream is implemented as a wrapper
// around Channel, we need to make sure ChannelProxy returns the same
// decoder factory as the one passed in when creating an AudioReceiveStream.
ON_CALL(*this, ChannelProxyFactory(testing::_))
@@ -48,7 +48,7 @@
return proxy;
}));
}
- ~MockVoiceEngine() /* override */ {
+ virtual ~MockVoiceEngine() /* override */ {
// Decrease ref count before base class d-tor is called; otherwise it will
// trigger an assertion.
--_ref_count;
@@ -57,7 +57,7 @@
MOCK_METHOD1(ChannelProxyFactory, voe::ChannelProxy*(int channel_id));
// VoiceEngineImpl
- std::unique_ptr<voe::ChannelProxy> GetChannelProxy(
+ virtual std::unique_ptr<voe::ChannelProxy> GetChannelProxy(
int channel_id) /* override */ {
return std::unique_ptr<voe::ChannelProxy>(ChannelProxyFactory(channel_id));
}
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 66f14bd..6a30fbc 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -2813,6 +2813,13 @@
return 0;
}
+void Channel::set_associate_send_channel(const ChannelOwner& channel) {
+ RTC_DCHECK(!channel.channel() ||
+ channel.channel()->ChannelId() != _channelId);
+ rtc::CritScope lock(&assoc_send_channel_lock_);
+ associate_send_channel_ = channel;
+}
+
void Channel::DisassociateSendChannel(int channel_id) {
rtc::CritScope lock(&assoc_send_channel_lock_);
Channel* channel = associate_send_channel_.channel();
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 4bfcd74..da020e6 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -405,12 +405,7 @@
// Associate to a send channel.
// Used for obtaining RTT for a receive-only channel.
- void set_associate_send_channel(const ChannelOwner& channel) {
- assert(_channelId != channel.channel()->ChannelId());
- rtc::CritScope lock(&assoc_send_channel_lock_);
- associate_send_channel_ = channel;
- }
-
+ void set_associate_send_channel(const ChannelOwner& channel);
// Disassociate a send channel if it was associated.
void DisassociateSendChannel(int channel_id);
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index 1f9d7f3..bfd5b17 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -236,6 +236,17 @@
channel()->SetTransportOverhead(transport_overhead_per_packet);
}
+void ChannelProxy::AssociateSendChannel(
+ const ChannelProxy& send_channel_proxy) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ channel()->set_associate_send_channel(send_channel_proxy.channel_owner_);
+}
+
+void ChannelProxy::DisassociateSendChannel() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ channel()->set_associate_send_channel(ChannelOwner(nullptr));
+}
+
Channel* ChannelProxy::channel() const {
RTC_DCHECK(channel_owner_.channel());
return channel_owner_.channel();
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 44b25fd..182e252 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -91,10 +91,10 @@
virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame);
-
virtual int NeededFrequency() const;
-
virtual void SetTransportOverhead(int transport_overhead_per_packet);
+ virtual void AssociateSendChannel(const ChannelProxy& send_channel_proxy);
+ virtual void DisassociateSendChannel();
private:
Channel* channel() const;
