diff options
Diffstat (limited to 'webrtc/audio_send_stream.h')
| -rw-r--r-- | webrtc/audio_send_stream.h | 13 |
1 files changed, 12 insertions, 1 deletions
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h index 89b73e6e3e..d1af9e0103 100644 --- a/webrtc/audio_send_stream.h +++ b/webrtc/audio_send_stream.h @@ -23,6 +23,11 @@ namespace webrtc { +// WORK IN PROGRESS +// This class is under development and is not yet intended for for use outside +// of WebRtc/Libjingle. Please use the VoiceEngine API instead. +// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 + class AudioSendStream : public SendStream { public: struct Stats { @@ -59,8 +64,11 @@ class AudioSendStream : public SendStream { // Sender SSRC. uint32_t ssrc = 0; - // RTP header extensions used for the received stream. + // RTP header extensions used for the sent stream. std::vector<RtpExtension> extensions; + + // RTCP CNAME, see RFC 3550. + std::string c_name; } rtp; // Transport for outgoing packets. The transport is expected to exist for @@ -81,6 +89,9 @@ class AudioSendStream : public SendStream { int red_payload_type = -1; // pt, or -1 to disable REDundant coding. }; + // TODO(solenberg): Make payload_type a config property instead. + virtual bool SendTelephoneEvent(int payload_type, uint8_t event, + uint32_t duration_ms) = 0; virtual Stats GetStats() const = 0; }; } // namespace webrtc |