Chromium Code Reviews| Index: webrtc/media/engine/fakewebrtccall.h |
| diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h |
| index ee3e449b199445d4601e850c26f88e2cd114fdb5..e01d63c4af1570f3761421ec52c40b5038f4a21c 100644 |
| --- a/webrtc/media/engine/fakewebrtccall.h |
| +++ b/webrtc/media/engine/fakewebrtccall.h |
| @@ -74,22 +74,18 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { |
| const webrtc::AudioReceiveStream::Config& GetConfig() const; |
| void SetStats(const webrtc::AudioReceiveStream::Stats& stats); |
| int received_packets() const { return received_packets_; } |
| - void IncrementReceivedPackets(); |
| + bool VerifyLastPacket(const void* data, size_t length) const; |
|
the sun
2016/04/22 12:40:31
uint8_t*
mflodman
2016/04/27 13:42:17
Done.
|
| const webrtc::AudioSinkInterface* sink() const { return sink_.get(); } |
| + bool DeliverRtp(const uint8_t* packet, |
| + size_t length, |
| + const webrtc::PacketTime& packet_time) override; |
| + bool DeliverRtcp(const uint8_t* packet, size_t length) override; |
| private: |
| // webrtc::ReceiveStream implementation. |
| void Start() override {} |
| void Stop() override {} |
| void SignalNetworkState(webrtc::NetworkState state) override {} |
| - bool DeliverRtcp(const uint8_t* packet, size_t length) override { |
| - return true; |
| - } |
| - bool DeliverRtp(const uint8_t* packet, |
| - size_t length, |
| - const webrtc::PacketTime& packet_time) override { |
| - return true; |
| - } |
| // webrtc::AudioReceiveStream implementation. |
| webrtc::AudioReceiveStream::Stats GetStats() const override; |
| @@ -99,6 +95,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { |
| webrtc::AudioReceiveStream::Stats stats_; |
| int received_packets_; |
| std::unique_ptr<webrtc::AudioSinkInterface> sink_; |
| + std::string last_packet_; |
| }; |
| class FakeVideoSendStream final : public webrtc::VideoSendStream, |