Replace VoEBase::[Start/Stop]Playout().
The functionality is moved into AudioState.
TBR: henrika@webrtc.org
Bug: webrtc:4690
Change-Id: I015482ad18a39609634f6ead9e991d5210107f0f
Reviewed-on: https://webrtc-review.googlesource.com/34502
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21338}diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index d24bed5..4d4af2e 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -48,7 +48,8 @@
return audio_decode_stats;
}
-const int kChannelId = 2;
+const int kChannelId1 = 2;
+const int kChannelId2 = 29;
const uint32_t kRemoteSsrc = 1234;
const uint32_t kLocalSsrc = 5678;
const size_t kOneByteExtensionHeaderLength = 4;
@@ -84,7 +85,7 @@
new rtc::RefCountedObject<MockAudioDeviceModule>();
audio_state_ = AudioState::Create(config);
- EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
+ EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId1))
.WillOnce(Invoke([this](int channel_id) {
EXPECT_FALSE(channel_proxy_);
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
@@ -118,7 +119,15 @@
}));
return channel_proxy_;
}));
- stream_config_.voe_channel_id = kChannelId;
+ EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId2))
+ .WillRepeatedly(Invoke([this](int channel_id) {
+ testing::NiceMock<MockVoEChannelProxy>* proxy =
+ new testing::NiceMock<MockVoEChannelProxy>();
+ EXPECT_CALL(*proxy, GetAudioDecoderFactory())
+ .WillOnce(ReturnRef(decoder_factory_));
+ return proxy;
+ }));
+ stream_config_.voe_channel_id = kChannelId1;
stream_config_.rtp.local_ssrc = kLocalSsrc;
stream_config_.rtp.remote_ssrc = kRemoteSsrc;
stream_config_.rtp.nack.rtp_history_ms = 300;
@@ -231,7 +240,7 @@
AudioReceiveStream::Config config;
config.rtp.remote_ssrc = kRemoteSsrc;
config.rtp.local_ssrc = kLocalSsrc;
- config.voe_channel_id = kChannelId;
+ config.voe_channel_id = kChannelId1;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
EXPECT_EQ(
@@ -354,32 +363,36 @@
recv_stream.SetGain(0.765f);
}
-TEST(AudioReceiveStreamTest, StreamShouldNotBeAddedToMixerWhenVoEReturnsError) {
+TEST(AudioReceiveStreamTest, StreamsShouldBeAddedToMixerOnceOnStart) {
ConfigHelper helper;
- internal::AudioReceiveStream recv_stream(
+ internal::AudioReceiveStream recv_stream1(
helper.rtp_stream_receiver_controller(),
helper.packet_router(),
helper.config(), helper.audio_state(), helper.event_log());
-
- EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(-1));
- EXPECT_CALL(*helper.audio_mixer(), AddSource(_)).Times(0);
-
- recv_stream.Start();
-}
-
-TEST(AudioReceiveStreamTest, StreamShouldBeAddedToMixerOnStart) {
- ConfigHelper helper;
- internal::AudioReceiveStream recv_stream(
+ AudioReceiveStream::Config config2 = helper.config();
+ config2.voe_channel_id = kChannelId2;
+ internal::AudioReceiveStream recv_stream2(
helper.rtp_stream_receiver_controller(),
helper.packet_router(),
- helper.config(), helper.audio_state(), helper.event_log());
+ config2, helper.audio_state(), helper.event_log());
- EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0));
- EXPECT_CALL(helper.voice_engine(), StopPlayout(_));
- EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream))
+ EXPECT_CALL(*helper.channel_proxy(), StartPlayout()).Times(1);
+ EXPECT_CALL(*helper.channel_proxy(), StopPlayout()).Times(1);
+ EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream1))
.WillOnce(Return(true));
+ EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream2))
+ .WillOnce(Return(true));
+ EXPECT_CALL(*helper.audio_mixer(), RemoveSource(&recv_stream1)).Times(1);
+ EXPECT_CALL(*helper.audio_mixer(), RemoveSource(&recv_stream2)).Times(1);
- recv_stream.Start();
+ recv_stream1.Start();
+ recv_stream2.Start();
+
+ // One more should not result in any more mixer sources added.
+ recv_stream1.Start();
+
+ // Stop stream before it is being destructed.
+ recv_stream2.Stop();
}
} // namespace test
} // namespace webrtc