| Index: webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h b/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
|
| index 84428788d77d20609b580a9003afd8075754906f..7e2cf85f31bd16bab76d0e8cf40ae5adddc84430 100644
|
| --- a/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
|
| +++ b/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
|
| @@ -484,12 +484,9 @@ typedef struct {
|
| int16_t maxRateBytesPer30Ms;
|
| // Maximum allowed payload-size, measured in Bytes.
|
| int16_t maxPayloadSizeBytes;
|
| - /* The expected sampling rate of the input signal. Valid values are 16000,
|
| - * 32000 and 48000. This is not the operation sampling rate of the codec.
|
| - * Input signals at 48 kHz are resampled to 32 kHz, then encoded. */
|
| + /* The expected sampling rate of the input signal. Valid values are 16000
|
| + * and 32000. This is not the operation sampling rate of the codec. */
|
| uint16_t in_sample_rate_hz;
|
| - /* State for the input-resampler. It is only used for 48 kHz input signals. */
|
| - int16_t state_in_resampler[SIZE_RESAMPLER_STATE];
|
|
|
| // Trig tables for WebRtcIsac_Time2Spec and WebRtcIsac_Spec2time.
|
| TransformTables transform_tables;
|
|
|