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Unified Diff: webrtc/media/engine/fakewebrtccall.h

Issue 1909333002: Switch voice transport to use Call and Stream instead of VoENetwork. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Remove VoENetwork from perf test. Created 4 years, 8 months ago
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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,

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