Index: webrtc/video/rtc_event_log.proto |
diff --git a/webrtc/video/rtc_event_log.proto b/webrtc/video/rtc_event_log.proto |
index bdb6bb12750a0250683369fa2b2f0c6800838ddc..ae5cfd24b478f8a979d3f1bc6d428682e776c5e2 100644 |
--- a/webrtc/video/rtc_event_log.proto |
+++ b/webrtc/video/rtc_event_log.proto |
@@ -29,13 +29,15 @@ message Event { |
// receive the new events as UNKNOWN_EVENT. |
enum EventType { |
UNKNOWN_EVENT = 0; |
- RTP_EVENT = 1; |
- RTCP_EVENT = 2; |
- DEBUG_EVENT = 3; |
- VIDEO_RECEIVER_CONFIG_EVENT = 4; |
- VIDEO_SENDER_CONFIG_EVENT = 5; |
- AUDIO_RECEIVER_CONFIG_EVENT = 6; |
- AUDIO_SENDER_CONFIG_EVENT = 7; |
+ LOG_START = 1; |
+ LOG_END = 2; |
+ RTP_EVENT = 3; |
+ RTCP_EVENT = 4; |
+ AUDIO_PLAYOUT_EVENT = 5; |
+ VIDEO_RECEIVER_CONFIG_EVENT = 6; |
+ VIDEO_SENDER_CONFIG_EVENT = 7; |
+ AUDIO_RECEIVER_CONFIG_EVENT = 8; |
+ AUDIO_SENDER_CONFIG_EVENT = 9; |
} |
// required - Indicates the type of this event |
@@ -47,8 +49,8 @@ message Event { |
// optional - but required if type == RTCP_EVENT |
optional RtcpPacket rtcp_packet = 4; |
- // optional - but required if type == DEBUG_EVENT |
- optional DebugEvent debug_event = 5; |
+ // optional - but required if type == AUDIO_PLAYOUT_EVENT |
+ optional AudioPlayoutEvent audio_playout_event = 5; |
// optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT |
optional VideoReceiveConfig video_receiver_config = 6; |
@@ -92,22 +94,8 @@ message RtcpPacket { |
optional bytes packet_data = 3; |
} |
- |
-message DebugEvent { |
- // Indicates the type of the debug event. |
- // LOG_START and LOG_END indicate the start and end of the log respectively. |
- // AUDIO_PLAYOUT indicates a call to the PlayoutData10Ms() function in ACM. |
- enum EventType { |
- UNKNOWN_EVENT = 0; |
- LOG_START = 1; |
- LOG_END = 2; |
- AUDIO_PLAYOUT = 3; |
- } |
- |
- // required |
- optional EventType type = 1; |
- |
- // required if type == AUDIO_PLAYOUT |
+message AudioPlayoutEvent { |
+ // required - The SSRC of the audio stream associated with the playout event. |
optional uint32 local_ssrc = 2; |
} |
@@ -223,9 +211,21 @@ message EncoderConfig { |
message AudioReceiveConfig { |
// TODO(terelius): Add audio-receive config. |
terelius
2015/09/21 15:18:26
As far as I am concerned you can remove these TODO
ivoc
2015/09/25 08:02:06
Done.
|
+ |
+ // RTP header extensions used for the received audio stream. |
+ repeated RtpHeaderExtension header_extensions = 1; |
terelius
2015/09/21 15:18:25
We definitely need ssrc to be able to distinguish
ivoc
2015/09/25 08:02:06
I added some SSRC fields.
|
+ |
+ // List of decoders associated with the stream. |
+ repeated DecoderConfig decoders = 2; |
terelius
2015/09/21 15:18:26
The encoder/decoder configs are not important to m
ivoc
2015/09/25 08:02:06
We do want to eventually log that information, but
|
} |
message AudioSendConfig { |
// TODO(terelius): Add audio-receive config. |
+ |
+ // RTP header extensions used for the outgoing audio stream. |
+ repeated RtpHeaderExtension header_extensions = 1; |
+ |
+ // required - Encoder associated with the stream. |
+ optional EncoderConfig encoder = 2; |
} |