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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74
webrtc/modules/audio_coding/BUILD.gn
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@ -9,17 +9,48 @@
import("//build/config/arm.gni")
import("../../build/webrtc.gni")
source_set("rent_a_codec") {
sources = [
"main/acm2/acm_codec_database.cc",
"main/acm2/acm_codec_database.h",
"main/acm2/rent_a_codec.cc",
"main/acm2/rent_a_codec.h",
]
configs += [ "../..:common_config" ]
public_configs = [ "../..:common_inherited_config" ]
deps = [
"../..:webrtc_common",
]
defines = []
if (rtc_include_opus) {
defines += [ "WEBRTC_CODEC_OPUS" ]
}
if (!build_with_mozilla) {
if (current_cpu == "arm") {
defines += [ "WEBRTC_CODEC_ISACFX" ]
} else {
defines += [ "WEBRTC_CODEC_ISAC" ]
}
defines += [ "WEBRTC_CODEC_G722" ]
}
if (!build_with_mozilla && !build_with_chromium) {
defines += [
"WEBRTC_CODEC_ILBC",
"WEBRTC_CODEC_RED",
]
}
}
config("audio_coding_config") {
include_dirs = [
"main/interface",
"main/include",
"../interface",
]
}
source_set("audio_coding") {
sources = [
"main/acm2/acm_codec_database.cc",
"main/acm2/acm_codec_database.h",
"main/acm2/acm_common_defs.h",
"main/acm2/acm_receiver.cc",
"main/acm2/acm_receiver.h",
@ -36,10 +67,8 @@ source_set("audio_coding") {
"main/acm2/codec_owner.h",
"main/acm2/initial_delay_manager.cc",
"main/acm2/initial_delay_manager.h",
"main/acm2/nack.cc",
"main/acm2/nack.h",
"main/interface/audio_coding_module.h",
"main/interface/audio_coding_module_typedefs.h",
"main/include/audio_coding_module.h",
"main/include/audio_coding_module_typedefs.h",
]
defines = []
@ -69,6 +98,7 @@ source_set("audio_coding") {
":g711",
":neteq",
":pcm16b",
":rent_a_codec",
"../..:rtc_event_log",
"../..:webrtc_common",
"../../common_audio",
@ -245,7 +275,7 @@ source_set("g722") {
config("ilbc_config") {
include_dirs = [
"../../..",
"codecs/ilbc/interface",
"codecs/ilbc/include",
]
}
@ -323,6 +353,7 @@ source_set("ilbc") {
"codecs/ilbc/ilbc.c",
"codecs/ilbc/include/audio_decoder_ilbc.h",
"codecs/ilbc/include/audio_encoder_ilbc.h",
"codecs/ilbc/include/ilbc.h",
"codecs/ilbc/index_conv_dec.c",
"codecs/ilbc/index_conv_dec.h",
"codecs/ilbc/index_conv_enc.c",
@ -331,7 +362,6 @@ source_set("ilbc") {
"codecs/ilbc/init_decode.h",
"codecs/ilbc/init_encode.c",
"codecs/ilbc/init_encode.h",
"codecs/ilbc/interface/ilbc.h",
"codecs/ilbc/interpolate.c",
"codecs/ilbc/interpolate.h",
"codecs/ilbc/interpolate_samples.c",
@ -422,15 +452,15 @@ source_set("isac_common") {
config("isac_config") {
include_dirs = [
"../../..",
"codecs/isac/main/interface",
"codecs/isac/main/include",
]
}
source_set("isac") {
sources = [
"codecs/isac/main/interface/audio_decoder_isac.h",
"codecs/isac/main/interface/audio_encoder_isac.h",
"codecs/isac/main/interface/isac.h",
"codecs/isac/main/include/audio_decoder_isac.h",
"codecs/isac/main/include/audio_encoder_isac.h",
"codecs/isac/main/include/isac.h",
"codecs/isac/main/source/arith_routines.c",
"codecs/isac/main/source/arith_routines.h",
"codecs/isac/main/source/arith_routines_hist.c",
@ -506,15 +536,15 @@ source_set("isac") {
config("isac_fix_config") {
include_dirs = [
"../../..",
"codecs/isac/fix/interface",
"codecs/isac/fix/include",
]
}
source_set("isac_fix") {
sources = [
"codecs/isac/fix/interface/audio_decoder_isacfix.h",
"codecs/isac/fix/interface/audio_encoder_isacfix.h",
"codecs/isac/fix/interface/isacfix.h",
"codecs/isac/fix/include/audio_decoder_isacfix.h",
"codecs/isac/fix/include/audio_encoder_isacfix.h",
"codecs/isac/fix/include/isacfix.h",
"codecs/isac/fix/source/arith_routines.c",
"codecs/isac/fix/source/arith_routines_hist.c",
"codecs/isac/fix/source/arith_routines_logist.c",
@ -694,9 +724,9 @@ source_set("webrtc_opus") {
sources = [
"codecs/opus/audio_decoder_opus.cc",
"codecs/opus/audio_encoder_opus.cc",
"codecs/opus/interface/audio_decoder_opus.h",
"codecs/opus/interface/audio_encoder_opus.h",
"codecs/opus/interface/opus_interface.h",
"codecs/opus/include/audio_decoder_opus.h",
"codecs/opus/include/audio_encoder_opus.h",
"codecs/opus/include/opus_interface.h",
"codecs/opus/opus_inst.h",
"codecs/opus/opus_interface.c",
]
@ -764,9 +794,11 @@ source_set("neteq") {
"neteq/dtmf_tone_generator.h",
"neteq/expand.cc",
"neteq/expand.h",
"neteq/interface/neteq.h",
"neteq/include/neteq.h",
"neteq/merge.cc",
"neteq/merge.h",
"neteq/nack.cc",
"neteq/nack.h",
"neteq/neteq.cc",
"neteq/neteq_impl.cc",
"neteq/neteq_impl.h",

2
webrtc/modules/audio_coding/Makefile.am
View File

@ -1,6 +1,6 @@
noinst_LTLIBRARIES = libaudio_coding.la
libaudio_coding_la_SOURCES = codecs/isac/main/interface/isac.h \
libaudio_coding_la_SOURCES = codecs/isac/main/include/isac.h \
codecs/isac/main/source/arith_routines.c \
codecs/isac/main/source/arith_routines.h \
codecs/isac/main/source/arith_routines_hist.c \

6
webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h → webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
View File

@ -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
View File

@ -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
View File

@ -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
View File

@ -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
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;