Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(477)

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

Issue 1505253004: Support for remote audio into tracks (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Address comments Created 5 years 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
Index: talk/app/webrtc/rtpsenderreceiver_unittest.cc
diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
index c9871864bd5618548c89efbdbb354c455868355f..3f61504cae36509b4c20fe3803e4c6f1354dec18 100644
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
@@ -26,10 +26,12 @@
*/
#include <string>
+#include <utility>
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/mediastream.h"
#include "talk/app/webrtc/remoteaudiosource.h"
+#include "talk/app/webrtc/remoteaudiotrack.h"
#include "talk/app/webrtc/rtpreceiver.h"
#include "talk/app/webrtc/rtpsender.h"
#include "talk/app/webrtc/streamcollection.h"
@@ -57,7 +59,8 @@ namespace webrtc {
// Helper class to test RtpSender/RtpReceiver.
class MockAudioProvider : public AudioProviderInterface {
public:
- virtual ~MockAudioProvider() {}
+ ~MockAudioProvider() override {}
+
MOCK_METHOD2(SetAudioPlayout,
void(uint32_t ssrc,
bool enable));
@@ -67,6 +70,14 @@ class MockAudioProvider : public AudioProviderInterface {
const cricket::AudioOptions& options,
cricket::AudioRenderer* renderer));
MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
+
+ void SetRawAudioSink(uint32_t,
+ rtc::scoped_ptr<AudioSinkInterface> sink) override {
+ sink_ = std::move(sink);
+ }
+
+ private:
+ rtc::scoped_ptr<AudioSinkInterface> sink_;
};
// Helper class to test RtpSender/RtpReceiver.
@@ -151,8 +162,8 @@ class RtpSenderReceiverTest : public testing::Test {
}
void CreateAudioRtpReceiver() {
- audio_track_ =
- AudioTrack::Create(kAudioTrackId, RemoteAudioSource::Create().get());
+ audio_track_ = RemoteAudioTrack::Create(
+ kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL));
EXPECT_TRUE(stream_->AddTrack(audio_track_));
EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));
audio_rtp_receiver_ = new AudioRtpReceiver(stream_->GetAudioTracks()[0],

Powered by Google App Engine
This is Rietveld 408576698