Index: webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h |
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h |
index edabdff5cf37757ad11f861aefc0a2f3410858c0..8482a8c70e9070f26bf231dc0bea1456e99efe17 100644 |
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h |
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h |
@@ -90,12 +90,12 @@ extern "C" { |
/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */ |
/* returns 0 if everything went fine, -1 otherwise */ |
int16_t WebRtcIsac_UpdateBandwidthEstimator( |
- BwEstimatorstr* bwest_str, |
+ BwEstimatorstr* bwest_str, |
const uint16_t rtp_number, |
- const int32_t frame_length, |
+ const int32_t frame_length, |
const uint32_t send_ts, |
const uint32_t arr_ts, |
- const int32_t pksize); |
+ const int32_t pksize); |
/* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */ |
int16_t WebRtcIsac_UpdateUplinkBwImpl( |