Chromium Code Reviews| Index: webrtc/logging/rtc_event_log/rtc_event_log.proto |
| diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.proto b/webrtc/logging/rtc_event_log/rtc_event_log.proto |
| index a6d169579659e496379394de86cb9b4fac220b13..14b8266a66aea5b3fce6b3359f40696a046cdde5 100644 |
| --- a/webrtc/logging/rtc_event_log/rtc_event_log.proto |
| +++ b/webrtc/logging/rtc_event_log/rtc_event_log.proto |
| @@ -37,6 +37,7 @@ message Event { |
| VIDEO_SENDER_CONFIG_EVENT = 9; |
| AUDIO_RECEIVER_CONFIG_EVENT = 10; |
| AUDIO_SENDER_CONFIG_EVENT = 11; |
| + AUDIO_NETWORK_ADAPTOR_EVENT = 12; |
| } |
| // required - Indicates the type of this event |
| @@ -65,6 +66,9 @@ message Event { |
| // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT |
| optional AudioSendConfig audio_sender_config = 11; |
| + |
| + // optional - but required if type == AUDIO_NETWORK_ADAPTOR_EVENT |
| + optional AudioNetworkAdaptorDecition audio_network_adaptor_decition = 12; |
|
minyue-webrtc
2016/12/12 10:33:12
We call the output of AudioNetworkAdaptor, Runtime
michaelt
2016/12/12 10:50:08
Sure why not "AudioEncoderRuntimeConfig"
|
| } |
| message RtpPacket { |
| @@ -227,3 +231,12 @@ message AudioSendConfig { |
| // RTP header extensions used for the outgoing audio stream. |
| repeated RtpHeaderExtension header_extensions = 2; |
| } |
| + |
| +message AudioNetworkAdaptorDecition { |
|
minyue-webrtc
2016/12/13 10:52:02
is it possible to merge this and EncoderRuntimeCon
michaelt
2016/12/13 15:44:39
Done.
|
| + optional int32 bitrate_bps = 1; |
| + optional int32 frame_length_ms = 2; |
| + optional float uplink_packet_loss_fraction = 3; |
| + optional bool enable_fec = 4; |
| + optional bool enable_dtx = 5; |
| + optional uint32 num_channels = 6; |
| +} |