| 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(
|
|
|