Chromium Code Reviews| Index: webrtc/audio_send_stream.h |
| diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h |
| index 695d28f21e02ddfed09391a4019266d4b861a4a6..457aa5cb0df27a26bce9c5dd80ea7907665baf87 100644 |
| --- a/webrtc/audio_send_stream.h |
| +++ b/webrtc/audio_send_stream.h |
| @@ -48,6 +48,11 @@ class AudioSendStream : public SendStream { |
| // Transport for outgoing packets. |
| Transport* send_transport = nullptr; |
| + // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level |
| + // components. Temporarily used while VoiceEngine channels are created |
| + // outside of Call. |
|
kwiberg-webrtc
2015/09/30 12:04:30
It might not be immediately clear if "temporarily"
|
| + int voe_channel_id = -1; |
| + |
| rtc::scoped_ptr<AudioEncoder> encoder; |
| int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator. |
| int red_payload_type = -1; // pt, or -1 to disable REDundant coding. |