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Revert 5590 "description"

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> description

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8949006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5593 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
xians@webrtc.org
2014-02-21 10:31:29 +00:00
parent 0f2809a5ac
commit ef2215110c
20 changed files with 106 additions and 686 deletions
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6
talk/app/webrtc/mediastreamhandler.cc
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@@ -58,11 +58,7 @@ void TrackHandler::OnChanged() {
LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(NULL) {}
LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
talk_base::CritScope lock(&lock_);
if (sink_)
sink_->OnClose();
}
LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {}
void LocalAudioSinkAdapter::OnData(const void* audio_data,
int bits_per_sample,

1
talk/app/webrtc/test/fakeperiodicvideocapturer.h
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@@ -56,7 +56,6 @@ class FakePeriodicVideoCapturer : public cricket::FakeVideoCapturer {
virtual cricket::CaptureState Start(const cricket::VideoFormat& format) {
cricket::CaptureState state = FakeVideoCapturer::Start(format);
if (state != cricket::CS_FAILED) {
set_enable_video_adapter(false); // Simplify testing.
talk_base::Thread::Current()->Post(this, MSG_CREATEFRAME);
}
return state;

37
talk/app/webrtc/webrtcsession_unittest.cc
View File

@@ -249,11 +249,7 @@ class WebRtcSessionCreateSDPObserverForTest
class FakeAudioRenderer : public cricket::AudioRenderer {
public:
FakeAudioRenderer() : channel_id_(-1), sink_(NULL) {}
virtual ~FakeAudioRenderer() {
if (sink_)
sink_->OnClose();
}
FakeAudioRenderer() : channel_id_(-1) {}
virtual void AddChannel(int channel_id) OVERRIDE {
ASSERT(channel_id_ == -1);
@@ -263,15 +259,10 @@ class FakeAudioRenderer : public cricket::AudioRenderer {
ASSERT(channel_id == channel_id_);
channel_id_ = -1;
}
virtual void SetSink(Sink* sink) OVERRIDE {
sink_ = sink;
}
int channel_id() const { return channel_id_; }
cricket::AudioRenderer::Sink* sink() const { return sink_; }
private:
int channel_id_;
cricket::AudioRenderer::Sink* sink_;
};
class WebRtcSessionTest : public testing::Test {
@@ -2196,39 +2187,13 @@ TEST_F(WebRtcSessionTest, SetAudioSend) {
EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
EXPECT_FALSE(channel->options().echo_cancellation.IsSet());
EXPECT_EQ(0, renderer->channel_id());
EXPECT_TRUE(renderer->sink() != NULL);
// This will trigger SetSink(NULL) to the |renderer|.
session_->SetAudioSend(send_ssrc, true, options, NULL);
EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
bool value;
EXPECT_TRUE(channel->options().echo_cancellation.Get(&value));
EXPECT_TRUE(value);
EXPECT_EQ(-1, renderer->channel_id());
EXPECT_TRUE(renderer->sink() == NULL);
}
TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
ASSERT_TRUE(channel != NULL);
ASSERT_EQ(1u, channel->send_streams().size());
uint32 send_ssrc = channel->send_streams()[0].first_ssrc();
talk_base::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
cricket::AudioOptions options;
session_->SetAudioSend(send_ssrc, true, options, renderer.get());
EXPECT_TRUE(renderer->sink() != NULL);
// Delete the |renderer| and it will trigger OnClose() to the sink, and this
// will invalidate the |renderer_| pointer in the sink and prevent getting a
// SetSink(NULL) callback afterwards.
renderer.reset();
// This will trigger SetSink(NULL) if no OnClose() callback.
session_->SetAudioSend(send_ssrc, true, options, NULL);
}
TEST_F(WebRtcSessionTest, SetVideoPlayout) {