| Index: webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
|
| index 1fe5d312b8c7ae296eb876c7b7c1ad3df637b86d..27998923f03b6c3a04004e6f946af07d40e7ee6c 100644
|
| --- a/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
|
| +++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
|
| @@ -91,6 +91,15 @@ struct IsacFloat {
|
| uint16_t sample_rate_hz) {
|
| return WebRtcIsac_SetEncSampRate(inst, sample_rate_hz);
|
| }
|
| + static inline void SetEncSampRateInDecoder(instance_type* inst,
|
| + uint16_t sample_rate_hz) {
|
| + WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz);
|
| + }
|
| + static inline void SetInitialBweBottleneck(
|
| + instance_type* inst,
|
| + int bottleneck_bits_per_second) {
|
| + WebRtcIsac_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
|
| + }
|
| static inline int16_t UpdateBwEstimate(instance_type* inst,
|
| const uint8_t* encoded,
|
| int32_t packet_size,
|
|
|