Restarting channel when swapping AudioReceiveStreams in WebrtcVoE.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2383143002
Cr-Commit-Position: refs/heads/master@{#14493}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 57e62f5..11a1c58 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -144,6 +144,7 @@
AudioReceiveStream::~AudioReceiveStream() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
+ Stop();
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->ResetCongestionControlObjects();
channel_proxy_->SetRtcEventLog(nullptr);
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 94bfabd..d30eb11 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -119,6 +119,7 @@
.After(expect_set);
return channel_proxy_;
}));
+ EXPECT_CALL(voice_engine_, StopPlayout(kChannelId)).WillOnce(Return(0));
stream_config_.voe_channel_id = kChannelId;
stream_config_.rtp.local_ssrc = kLocalSsrc;
stream_config_.rtp.remote_ssrc = kRemoteSsrc;
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 358c142..71803cb 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1424,6 +1424,7 @@
LOG(LS_INFO) << "Stopping playout for channel #" << channel();
stream_->Stop();
}
+ playout_ = playout;
}
private:
@@ -1444,6 +1445,7 @@
RTC_DCHECK(!stream_);
stream_ = call_->CreateAudioReceiveStream(config_);
RTC_CHECK(stream_);
+ SetPlayout(playout_);
}
rtc::ThreadChecker worker_thread_checker_;
@@ -1452,6 +1454,7 @@
// The stream is owned by WebRtcAudioReceiveStream and may be reallocated if
// configuration changes.
webrtc::AudioReceiveStream* stream_ = nullptr;
+ bool playout_ = false;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
};
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index e40bc77..8be3b9d 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -3340,6 +3340,21 @@
call_.GetNetworkState(webrtc::MediaType::VIDEO));
}
+// Test that playout is still started after changing parameters
+TEST_F(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) {
+ SetupRecvStream();
+ channel_->SetPlayout(true);
+ EXPECT_TRUE(GetRecvStream(kSsrc1).started());
+
+ // Changing RTP header extensions will recreate the AudioReceiveStream.
+ cricket::AudioRecvParameters parameters;
+ parameters.extensions.push_back(
+ webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12));
+ channel_->SetRecvParameters(parameters);
+
+ EXPECT_TRUE(GetRecvStream(kSsrc1).started());
+}
+
// Tests that the library initializes and shuts down properly.
TEST(WebRtcVoiceEngineTest, StartupShutdown) {
// If the VoiceEngine wants to gather available codecs early, that's fine but
