| 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;
|
|
|