Index: webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c |
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c |
index c4ceb590628783f488a26cfca01411a6c1d42c9c..ce8ceb217a7b1c981c85abfd8210602fe0ad248c 100644 |
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c |
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c |
@@ -132,12 +132,12 @@ int32_t WebRtcIsac_InitBandwidthEstimator( |
/* 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 |
/*, const uint16_t Index*/) |
{ |
float weight = 0.0f; |