Index: talk/media/webrtc/fakewebrtccall.h |
diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h |
index fb271f22154a09f2ed6e66b2f94ff73feebc0a4d..de56a033bceaa3c53b71e37e256f558c16342008 100644 |
--- a/talk/media/webrtc/fakewebrtccall.h |
+++ b/talk/media/webrtc/fakewebrtccall.h |
@@ -25,6 +25,15 @@ |
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
*/ |
+// This file contains fake implementations, for use in unit tests, of the |
+// following classes: |
+// |
+// webrtc::Call |
+// webrtc::AudioSendStream |
+// webrtc::AudioReceiveStream |
+// webrtc::VideoSendStream |
+// webrtc::VideoReceiveStream |
+ |
#ifndef TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_ |
#define TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_ |
@@ -32,11 +41,35 @@ |
#include "webrtc/call.h" |
#include "webrtc/audio_receive_stream.h" |
+#include "webrtc/audio_send_stream.h" |
#include "webrtc/video_frame.h" |
#include "webrtc/video_receive_stream.h" |
#include "webrtc/video_send_stream.h" |
namespace cricket { |
+ |
+class FakeAudioSendStream : public webrtc::AudioSendStream { |
+ public: |
+ explicit FakeAudioSendStream( |
+ const webrtc::AudioSendStream::Config& config); |
+ |
+ // webrtc::AudioSendStream implementation. |
+ webrtc::AudioSendStream::Stats GetStats() const override; |
+ |
+ const webrtc::AudioSendStream::Config& GetConfig() const; |
+ |
+ private: |
+ // webrtc::SendStream 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; |
+ } |
+ |
+ webrtc::AudioSendStream::Config config_; |
+}; |
+ |
class FakeAudioReceiveStream : public webrtc::AudioReceiveStream { |
public: |
explicit FakeAudioReceiveStream( |
@@ -161,6 +194,8 @@ class FakeCall : public webrtc::Call, public webrtc::PacketReceiver { |
const std::vector<FakeVideoSendStream*>& GetVideoSendStreams(); |
const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams(); |
+ const std::vector<FakeAudioSendStream*>& GetAudioSendStreams(); |
+ const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc); |
const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams(); |
const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc); |
@@ -208,6 +243,7 @@ class FakeCall : public webrtc::Call, public webrtc::PacketReceiver { |
rtc::SentPacket last_sent_packet_; |
webrtc::Call::Stats stats_; |
std::vector<FakeVideoSendStream*> video_send_streams_; |
+ std::vector<FakeAudioSendStream*> audio_send_streams_; |
std::vector<FakeVideoReceiveStream*> video_receive_streams_; |
std::vector<FakeAudioReceiveStream*> audio_receive_streams_; |