Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(507)

Unified Diff: talk/app/webrtc/webrtcsession_unittest.cc

Issue 1393343003: Revert of Remove AudioTrackRenderer (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « talk/app/webrtc/audiotrackrenderer.cc ('k') | talk/libjingle.gyp » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: talk/app/webrtc/webrtcsession_unittest.cc
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index ff41383c32abe0472342ab47f57f8b83be1d7d66..2853ca43a7301b5e28b202f60041f35784342bb2 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -330,16 +330,26 @@
class FakeAudioRenderer : public cricket::AudioRenderer {
public:
- FakeAudioRenderer() : sink_(NULL) {}
+ FakeAudioRenderer() : channel_id_(-1), sink_(NULL) {}
virtual ~FakeAudioRenderer() {
if (sink_)
sink_->OnClose();
}
+ void AddChannel(int channel_id) override {
+ ASSERT(channel_id_ == -1);
+ channel_id_ = channel_id;
+ }
+ void RemoveChannel(int channel_id) override {
+ ASSERT(channel_id == channel_id_);
+ channel_id_ = -1;
+ }
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_;
};
@@ -3107,10 +3117,12 @@
EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
EXPECT_EQ(0, left_vol);
EXPECT_EQ(0, right_vol);
+ EXPECT_EQ(0, renderer->channel_id());
session_->SetAudioPlayout(receive_ssrc, true, NULL);
EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
EXPECT_EQ(1, left_vol);
EXPECT_EQ(1, right_vol);
+ EXPECT_EQ(-1, renderer->channel_id());
}
TEST_F(WebRtcSessionTest, SetAudioSend) {
@@ -3130,6 +3142,7 @@
session_->SetAudioSend(send_ssrc, false, options, renderer.get());
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|.
@@ -3138,6 +3151,7 @@
bool value;
EXPECT_TRUE(channel->options().echo_cancellation.Get(&value));
EXPECT_TRUE(value);
+ EXPECT_EQ(-1, renderer->channel_id());
EXPECT_TRUE(renderer->sink() == NULL);
}
« no previous file with comments | « talk/app/webrtc/audiotrackrenderer.cc ('k') | talk/libjingle.gyp » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698