| Index: webrtc/media/base/fakemediaengine.h
|
| diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h
|
| index af05144b529e0424a4a8b89f618470f69a26e01e..6feaf8e989f7e3e1569cfb060c816a6d651ba309 100644
|
| --- a/webrtc/media/base/fakemediaengine.h
|
| +++ b/webrtc/media/base/fakemediaengine.h
|
| @@ -20,6 +20,7 @@
|
|
|
| #include "webrtc/audio_sink.h"
|
| #include "webrtc/base/copyonwritebuffer.h"
|
| +#include "webrtc/base/networkroute.h"
|
| #include "webrtc/base/stringutils.h"
|
| #include "webrtc/media/base/audiosource.h"
|
| #include "webrtc/media/base/mediaengine.h"
|
| @@ -178,6 +179,12 @@ template <class Base> class RtpHelper : public Base {
|
| return ready_to_send_;
|
| }
|
|
|
| + NetworkRoute last_network_route() const { return last_network_route_; }
|
| + int num_network_route_changes() const { return num_network_route_changes_; }
|
| + void set_num_network_route_changes(int changes) {
|
| + num_network_route_changes_ = changes;
|
| + }
|
| +
|
| protected:
|
| bool MuteStream(uint32_t ssrc, bool mute) {
|
| if (!HasSendStream(ssrc) && ssrc != 0) {
|
| @@ -216,6 +223,11 @@ template <class Base> class RtpHelper : public Base {
|
| virtual void OnReadyToSend(bool ready) {
|
| ready_to_send_ = ready;
|
| }
|
| + virtual void OnNetworkRouteChanged(const std::string& transport_name,
|
| + const NetworkRoute& network_route) {
|
| + last_network_route_ = network_route;
|
| + ++num_network_route_changes_;
|
| + }
|
| bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
|
| bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
|
|
|
| @@ -235,6 +247,8 @@ template <class Base> class RtpHelper : public Base {
|
| uint32_t send_ssrc_;
|
| std::string rtcp_cname_;
|
| bool ready_to_send_;
|
| + NetworkRoute last_network_route_;
|
| + int num_network_route_changes_ = 0;
|
| };
|
|
|
| class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
|
|
|