Chromium Code Reviews| Index: webrtc/voice_engine/channel.h |
| diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h |
| index 6448349a8be87348d2e2d731f66012549828ae24..a4d73a7fdc83ac9e646ed1a82ce5ee7dbfdd85cf 100644 |
| --- a/webrtc/voice_engine/channel.h |
| +++ b/webrtc/voice_engine/channel.h |
| @@ -17,6 +17,8 @@ |
| #include "webrtc/api/call/audio_sink.h" |
| #include "webrtc/base/criticalsection.h" |
| #include "webrtc/base/optional.h" |
| +#include "webrtc/base/random.h" |
| +#include "webrtc/base/timeutils.h" |
| #include "webrtc/common_audio/resampler/include/push_resampler.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/audio_coding/acm2/codec_manager.h" |
| @@ -518,6 +520,8 @@ class Channel |
| bool _includeAudioLevelIndication; |
| size_t transport_overhead_per_packet_; |
| size_t rtp_overhead_per_packet_; |
| + // For generation of random ssrc:s. |
| + webrtc::Random random_; |
|
the sun
2017/02/16 12:34:16
You're only using this member in the ctor, why do
nisse-webrtc
2017/02/16 12:58:16
Good catch. I think I originally put it in the Cha
|
| // VoENetwork |
| AudioFrame::SpeechType _outputSpeechType; |
| // VoEVideoSync |