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Unified Diff: webrtc/modules/audio_coding/codecs/isac/main/source/structs.h

Issue 1392173004: Delete full-band mode from the iSAC codec (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove-isac-fb-neteq
Patch Set: Remove redundant DCHECK Created 5 years, 2 months ago
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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;
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