Index: webrtc/pc/mediasession.h |
diff --git a/webrtc/pc/mediasession.h b/webrtc/pc/mediasession.h |
index 6ac74f2d335dadca819b3c4acfca7209de7a6bb1..39ac26bd8dad104c5b10b0d450caf7cf67e85233 100644 |
--- a/webrtc/pc/mediasession.h |
+++ b/webrtc/pc/mediasession.h |
@@ -76,18 +76,21 @@ extern const char kMediaProtocolTcpDtlsSctp[]; |
const int kAutoBandwidth = -1; |
const int kBufferedModeDisabled = 0; |
+// Default RTCP CNAME for unit tests. |
+const char kDefaultRtcpCname[] = "DefaultRtcpCname"; |
+ |
struct MediaSessionOptions { |
- MediaSessionOptions() : |
- recv_audio(true), |
- recv_video(false), |
- data_channel_type(DCT_NONE), |
- is_muc(false), |
- vad_enabled(true), // When disabled, removes all CN codecs from SDP. |
- rtcp_mux_enabled(true), |
- bundle_enabled(false), |
- video_bandwidth(kAutoBandwidth), |
- data_bandwidth(kDataMaxBandwidth) { |
- } |
+ MediaSessionOptions() |
+ : recv_audio(true), |
+ recv_video(false), |
+ data_channel_type(DCT_NONE), |
+ is_muc(false), |
+ vad_enabled(true), // When disabled, removes all CN codecs from SDP. |
+ rtcp_mux_enabled(true), |
+ bundle_enabled(false), |
+ video_bandwidth(kAutoBandwidth), |
+ data_bandwidth(kDataMaxBandwidth), |
+ rtcp_cname(kDefaultRtcpCname) {} |
bool has_audio() const { |
return recv_audio || HasSendMediaStream(MEDIA_TYPE_AUDIO); |
@@ -133,6 +136,7 @@ struct MediaSessionOptions { |
int data_bandwidth; |
// content name ("mid") => options. |
std::map<std::string, TransportOptions> transport_options; |
+ std::string rtcp_cname; |
struct Stream { |
Stream(MediaType type, |