Chromium Code Reviews| Index: webrtc/voice_engine/channel.h |
| diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h |
| index c86820f2a5d933ab2d039274ff4f91cd7e64f8cd..ddb2d8e4ad870b074c6181cfed3246e8c56c309d 100644 |
| --- a/webrtc/voice_engine/channel.h |
| +++ b/webrtc/voice_engine/channel.h |
| @@ -17,6 +17,7 @@ |
| #include "webrtc/api/call/audio_sink.h" |
| #include "webrtc/base/criticalsection.h" |
| #include "webrtc/base/optional.h" |
| +#include "webrtc/base/smoothing_filter.h" |
| #include "webrtc/common_audio/resampler/include/push_resampler.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/audio_coding/acm2/codec_manager.h" |
| @@ -443,6 +444,8 @@ class Channel |
| int GetRtpTimestampRateHz() const; |
| int64_t GetRTT(bool allow_associate_channel) const; |
| + void OnUplinkBandwidthUpdated(int bitrate_bps); |
| + |
| rtc::CriticalSection _fileCritSect; |
| rtc::CriticalSection _callbackCritSect; |
| rtc::CriticalSection volume_settings_critsect_; |
| @@ -546,6 +549,8 @@ class Channel |
| // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
| rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
| + |
| + SmoothingFilterImpl bitrate_bps_smoothed_; |
|
minyue-webrtc
2016/11/15 14:36:03
I hope to call this
uplink_bandwidth_estimator an
stefan-webrtc
2016/11/15 14:41:58
I would prefer not using uplink bandwidth, as it's
minyue-webrtc
2016/11/16 08:56:45
ok. Then please add "we give smoothed bitrate allo
michaelt
2016/11/16 09:50:24
Done.
|
| }; |
| } // namespace voe |