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; |