AnotherFoxGuy/webrtc-audio-processing
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Update code to current Chromium master

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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
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6
webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h → webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
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@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
#include <stddef.h>
@@ -721,4 +721,4 @@ extern "C" {
#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ */
#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_ */

2
webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
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@@ -440,7 +440,7 @@ WebRtcIsac_CorrelateInterVec(
int16_t rowCntr;
int16_t colCntr;
int16_t interVecDim;
double myVec[UB16_LPC_VEC_PER_FRAME];
double myVec[UB16_LPC_VEC_PER_FRAME] = {0.0};
const double* interVecDecorrMat;
switch(bandwidth)

2
webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
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@@ -17,10 +17,10 @@
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "entropy_coding.h"
#include "settings.h"
#include "arith_routines.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "spectrum_ar_model_tables.h"
#include "lpc_tables.h"
#include "pitch_gain_tables.h"

2
webrtc/modules/audio_coding/codecs/isac/main/source/settings.h
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@@ -168,8 +168,6 @@ enum IsacSamplingRate {kIsacWideband = 16, kIsacSuperWideband = 32};
#define RCU_TRANSCODING_SCALE_UB 0.50f
#define RCU_TRANSCODING_SCALE_UB_INVERSE 2.0f
#define SIZE_RESAMPLER_STATE 6
/* Define Error codes */
/* 6000 General */
#define ISAC_MEMORY_ALLOCATION_FAILED 6010

9
webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
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@@ -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;