AnotherFoxGuy/webrtc-audio-processing
1
0
Fork 0
Code Pull Requests Activity

Update code to current Chromium master

Browse Source 
This corresponds to:

Chromium: 6555f9456074c0c0e5f7713564b978588ac04a5d
webrtc: c8b569e0a7ad0b369e15f0197b3a558699ec8efa
This commit is contained in:
Arun Raghavan
2015-11-04 10:07:52 +05:30
parent 9bc60d3e10
commit 34abadd258
108 changed files with 893 additions and 384 deletions
Download Patch File Download Diff File Expand all files Collapse all files

9
webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
View File

@ -19,7 +19,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/source/settings.h"
#include "webrtc/typedefs.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;