generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

48 kHz extension to iSAC.

Browse Source 
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org 2012-12-03 17:43:52 +00:00
parent 0bacb635cb
commit b0dff12d2b
27 changed files with 340 additions and 132 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

31
webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
View File

@ -18,9 +18,6 @@
typedef struct WebRtcISACStruct ISACStruct;
enum IsacSamplingRate {kIsacWideband = 16, kIsacSuperWideband = 32};
#if defined(__cplusplus)
extern "C" {
#endif
@ -478,14 +475,11 @@ extern "C" {
* Input:
* - ISAC_main_inst : iSAC instance
*
* Return value : enumerator representing sampling frequency
* associated with the decoder, i.e. the
* sampling rate of the decoded audio.
* Return value : sampling frequency in Hertz.
*
*/
enum IsacSamplingRate WebRtcIsac_DecSampRate(
ISACStruct* ISAC_main_inst);
WebRtc_UWord16 WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst);
/******************************************************************************
@ -494,14 +488,11 @@ extern "C" {
* Input:
* - ISAC_main_inst : iSAC instance
*
* Return value : enumerator representing sampling frequency
* associated with the encoder, the input audio
* is expected to be sampled at this rate.
* Return value : sampling rate in Hertz.
*
*/
enum IsacSamplingRate WebRtcIsac_EncSampRate(
ISACStruct* ISAC_main_inst);
WebRtc_UWord16 WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst);
/******************************************************************************
@ -512,15 +503,14 @@ extern "C" {
*
* Input:
* - ISAC_main_inst : iSAC instance
* - sampRate : enumerator specifying the sampling rate.
* - sampRate : sampling rate in Hertz.
*
* Return value : 0 if successful
* -1 if failed.
*/
WebRtc_Word16 WebRtcIsac_SetDecSampRate(
ISACStruct* ISAC_main_inst,
enum IsacSamplingRate sampRate);
WebRtc_Word16 WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
WebRtc_UWord16 samp_rate_hz);
/******************************************************************************
@ -533,15 +523,14 @@ extern "C" {
*
* Input:
* - ISAC_main_inst : iSAC instance
* - sampRate : enumerator specifying the sampling rate.
* - sampRate : sampling rate in Hertz.
*
* Return value : 0 if successful
* -1 if failed.
*/
WebRtc_Word16 WebRtcIsac_SetEncSampRate(
ISACStruct* ISAC_main_inst,
enum IsacSamplingRate sampRate);
WebRtc_Word16 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
WebRtc_UWord16 sample_rate_hz);

1
webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
View File

@ -68,6 +68,7 @@ WebRtcIsac_EstimateBandwidth(
diffArrivalTime = (WebRtc_UWord32)diffArrivalTime >> 1;
diffSendTime = (WebRtc_UWord32)diffSendTime >> 1;
}
// arrival timestamp in 16 kHz
arrivalTimestampIn16kHz = (WebRtc_UWord32)((WebRtc_UWord32)
bwest_str->prev_rec_arr_ts + (WebRtc_UWord32)diffArrivalTime);

143
webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
View File

@ -249,6 +249,7 @@ WebRtc_Word16 WebRtcIsac_Assign(ISACStruct** ISAC_main_inst,
instISAC->encoderSamplingRateKHz = kIsacWideband;
instISAC->decoderSamplingRateKHz = kIsacWideband;
instISAC->bandwidthKHz = isac8kHz;
instISAC->in_sample_rate_hz = 16000;
return 0;
} else {
return -1;
@ -280,6 +281,7 @@ WebRtc_Word16 WebRtcIsac_Create(ISACStruct** ISAC_main_inst) {
instISAC->bandwidthKHz = isac8kHz;
instISAC->encoderSamplingRateKHz = kIsacWideband;
instISAC->decoderSamplingRateKHz = kIsacWideband;
instISAC->in_sample_rate_hz = 16000;
return 0;
} else {
return -1;
@ -458,6 +460,7 @@ WebRtc_Word16 WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst,
return -1;
}
}
memset(instISAC->state_in_resampler, 0, sizeof(instISAC->state_in_resampler));
/* Initialization is successful, set the flag. */
instISAC->initFlag |= BIT_MASK_ENC_INIT;
return 0;
@ -505,6 +508,8 @@ WebRtc_Word16 WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
ISACLBStruct* instLB = &(instISAC->instLB);
ISACUBStruct* instUB = &(instISAC->instUB);
const int16_t* speech_in_ptr = speechIn;
int16_t resampled_buff[FRAMESAMPLES_10ms * 2];
/* Check if encoder initiated. */
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
@ -513,8 +518,37 @@ WebRtc_Word16 WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
return -1;
}
if (instISAC->in_sample_rate_hz == 48000) {
/* Samples in 10 ms @ 48 kHz. */
const int kNumInputSamples = FRAMESAMPLES_10ms * 3;
/* Samples 10 ms @ 32 kHz. */
const int kNumOutputSamples = FRAMESAMPLES_10ms * 2;
/* Resampler divide the input into blocks of 3 samples, i.e.
* kNumInputSamples / 3. */
const int kNumResamplerBlocks = FRAMESAMPLES_10ms;
int32_t buffer32[FRAMESAMPLES_10ms * 3 + SIZE_RESAMPLER_STATE];
/* Restore last samples from the past to the beginning of the buffer
* and store the last samples of current frame for the next resampling. */
for (k = 0; k < SIZE_RESAMPLER_STATE; k++) {
buffer32[k] = instISAC->state_in_resampler[k];
instISAC->state_in_resampler[k] = speechIn[kNumInputSamples -
SIZE_RESAMPLER_STATE + k];
}
for (k = 0; k < kNumInputSamples; k++) {
buffer32[SIZE_RESAMPLER_STATE + k] = speechIn[k];
}
/* Resampling 3 samples to 2. Function divides the input in
* |kNumResamplerBlocks| number of 3-sample groups, and output is
* |kNumResamplerBlocks| number of 2-sample groups. */
WebRtcSpl_Resample48khzTo32khz(buffer32, buffer32, kNumResamplerBlocks);
WebRtcSpl_VectorBitShiftW32ToW16(resampled_buff, kNumOutputSamples,
buffer32, 15);
speech_in_ptr = resampled_buff;
}
if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) {
WebRtcSpl_AnalysisQMF(speechIn, speechInLB, speechInUB,
WebRtcSpl_AnalysisQMF(speech_in_ptr, speechInLB, speechInUB,
instISAC->analysisFBState1,
instISAC->analysisFBState2);
@ -728,7 +762,7 @@ WebRtc_Word16 WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
* even for 60 msec frames.
*
* NOTE 1! This function does not write in the ISACStruct, it is not allowed.
* NOTE 3! Rates larger than the bottleneck of the codec will be limited
* NOTE 2! Rates larger than the bottleneck of the codec will be limited
* to the current bottleneck.
*
* Input:
@ -946,6 +980,11 @@ WebRtc_Word16 WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
*
* This function updates the estimate of the bandwidth.
*
* NOTE:
* The estimates of bandwidth is not valid if the sample rate of the far-end
* encoder is set to 48 kHz and send timestamps are increamented according to
* 48 kHz sampling rate.
*
* Input:
* - ISAC_main_inst : ISAC instance.
* - encoded : encoded ISAC frame(s).
@ -1731,7 +1770,8 @@ WebRtc_Word16 WebRtcIsac_ReadBwIndex(const WebRtc_Word16* encoded,
/****************************************************************************
* WebRtcIsac_ReadFrameLen(...)
*
* This function returns the length of the frame represented in the packet.
* This function returns the number of samples the decoder will generate if
* the given payload is decoded.
*
* Input:
* - encoded : Encoded bitstream
@ -1798,11 +1838,12 @@ WebRtc_Word16 WebRtcIsac_GetNewFrameLen(ISACStruct* ISAC_main_inst) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
/* Return new frame length. */
if (instISAC->encoderSamplingRateKHz == kIsacWideband) {
if (instISAC->in_sample_rate_hz == 16000)
return (instISAC->instLB.ISACencLB_obj.new_framelength);
} else {
return ((instISAC->instLB.ISACencLB_obj.new_framelength) << 1);
}
else if (instISAC->in_sample_rate_hz == 32000)
return ((instISAC->instLB.ISACencLB_obj.new_framelength) * 2);
else
return ((instISAC->instLB.ISACencLB_obj.new_framelength) * 3);
}
@ -2015,7 +2056,7 @@ WebRtc_Word16 WebRtcIsac_SetMaxRate(ISACStruct* ISAC_main_inst,
} else {
if (maxRateInBytesPer30Ms < 120) {
/* 'maxRate' is out of valid range
* Sset to the acceptable value and return -1. */
* Set to the acceptable value and return -1. */
maxRateInBytesPer30Ms = 120;
status = -1;
}
@ -2153,31 +2194,45 @@ void WebRtcIsac_version(char* version) {
* and the bottleneck remain unchanged by this call, however, the maximum rate
* and maximum payload-size will be reset to their default values.
*
* NOTE:
* The maximum internal sampling rate is 32 kHz. If the encoder sample rate is
* set to 48 kHz the input is expected to be at 48 kHz but will be resampled to
* 32 kHz before any further processing.
* This mode is created for compatibility with full-band codecs if iSAC is used
* in dual-streaming. See SetDecSampleRate() for sampling rates at the decoder.
*
* Input:
* - ISAC_main_inst : iSAC instance
* - sampRate : enumerator specifying the sampling rate.
* - sample_rate_hz : sampling rate in Hertz, valid values are 16000,
* 32000 and 48000.
*
* Return value : 0 if successful
* -1 if failed.
*/
WebRtc_Word16 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
enum IsacSamplingRate sampRate) {
WebRtc_UWord16 sample_rate_hz) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
enum IsacSamplingRate encoder_operational_rate;
if ((sampRate != kIsacWideband) &&
(sampRate != kIsacSuperWideband)) {
if ((sample_rate_hz != 16000) && (sample_rate_hz != 32000) &&
(sample_rate_hz != 48000)) {
/* Sampling Frequency is not supported. */
instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY;
return -1;
} else if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
BIT_MASK_ENC_INIT) {
if (sampRate == kIsacWideband) {
}
if (sample_rate_hz == 16000) {
encoder_operational_rate = kIsacWideband;
} else {
encoder_operational_rate = kIsacSuperWideband;
}
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
BIT_MASK_ENC_INIT) {
if (encoder_operational_rate == kIsacWideband) {
instISAC->bandwidthKHz = isac8kHz;
} else {
instISAC->bandwidthKHz = isac16kHz;
}
instISAC->encoderSamplingRateKHz = sampRate;
return 0;
} else {
ISACUBStruct* instUB = &(instISAC->instUB);
ISACLBStruct* instLB = &(instISAC->instLB);
@ -2188,7 +2243,7 @@ WebRtc_Word16 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
WebRtc_Word16 frameSizeMs = instLB->ISACencLB_obj.new_framelength /
(FS / 1000);
if ((sampRate == kIsacWideband) &&
if ((encoder_operational_rate == kIsacWideband) &&
(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)) {
/* Changing from super-wideband to wideband.
* we don't need to re-initialize the encoder of the lower-band. */
@ -2199,7 +2254,7 @@ WebRtc_Word16 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
}
instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX_60;
instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX_30;
} else if ((sampRate == kIsacSuperWideband) &&
} else if ((encoder_operational_rate == kIsacSuperWideband) &&
(instISAC->encoderSamplingRateKHz == kIsacWideband)) {
if (codingMode == 1) {
WebRtcIsac_RateAllocation(bottleneck, &bottleneckLB, &bottleneckUB,
@ -2210,7 +2265,7 @@ WebRtc_Word16 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX;
instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX;
EncoderInitLb(instLB, codingMode, sampRate);
EncoderInitLb(instLB, codingMode, encoder_operational_rate);
EncoderInitUb(instUB, instISAC->bandwidthKHz);
memset(instISAC->analysisFBState1, 0,
@ -2230,9 +2285,10 @@ WebRtc_Word16 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
instLB->ISACencLB_obj.new_framelength = FRAMESAMPLES;
}
}
instISAC->encoderSamplingRateKHz = sampRate;
return 0;
}
instISAC->encoderSamplingRateKHz = encoder_operational_rate;
instISAC->in_sample_rate_hz = sample_rate_hz;
return 0;
}
@ -2244,23 +2300,29 @@ WebRtc_Word16 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
*
* Input:
* - ISAC_main_inst : iSAC instance
* - sampRate : enumerator specifying the sampling rate.
* - sample_rate_hz : sampling rate in Hertz, valid values are 16000
* and 32000.
*
* Return value : 0 if successful
* -1 if failed.
*/
WebRtc_Word16 WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
enum IsacSamplingRate sampRate) {
WebRtc_UWord16 sample_rate_hz) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
enum IsacSamplingRate decoder_operational_rate;
if ((sampRate != kIsacWideband) &&
(sampRate != kIsacSuperWideband)) {
if (sample_rate_hz == 16000) {
decoder_operational_rate = kIsacWideband;
} else if (sample_rate_hz == 32000) {
decoder_operational_rate = kIsacSuperWideband;
} else {
/* Sampling Frequency is not supported. */
instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY;
return -1;
} else {
if ((instISAC->decoderSamplingRateKHz == kIsacWideband) &&
(sampRate == kIsacSuperWideband)) {
}
if ((instISAC->decoderSamplingRateKHz == kIsacWideband) &&
(decoder_operational_rate == kIsacSuperWideband)) {
/* Switching from wideband to super-wideband at the decoder
* we need to reset the filter-bank and initialize upper-band decoder. */
memset(instISAC->synthesisFBState1, 0,
@ -2271,10 +2333,9 @@ WebRtc_Word16 WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
if (DecoderInitUb(&(instISAC->instUB)) < 0) {
return -1;
}
}
instISAC->decoderSamplingRateKHz = sampRate;
return 0;
}
instISAC->decoderSamplingRateKHz = decoder_operational_rate;
return 0;
}
@ -2284,14 +2345,13 @@ WebRtc_Word16 WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
* Input:
* - ISAC_main_inst : iSAC instance
*
* Return value : enumerator representing sampling frequency
* associated with the encoder, the input audio
* is expected to be sampled at this rate.
* Return value : sampling rate in Hertz. The input to encoder
* is expected to be sampled in this rate.
*
*/
enum IsacSamplingRate WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst) {
WebRtc_UWord16 WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
return instISAC->encoderSamplingRateKHz;
return instISAC->in_sample_rate_hz;
}
@ -2302,12 +2362,11 @@ enum IsacSamplingRate WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst) {
* Input:
* - ISAC_main_inst : iSAC instance
*
* Return value : enumerator representing sampling frequency
* associated with the decoder, i.e. the
* sampling rate of the decoded audio.
* Return value : sampling rate in Hertz. Decoder output is
* sampled at this rate.
*
*/
enum IsacSamplingRate WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst) {
WebRtc_UWord16 WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
return instISAC->decoderSamplingRateKHz;
return instISAC->decoderSamplingRateKHz == kIsacWideband ? 16000 : 32000;
}

2
webrtc/modules/audio_coding/codecs/isac/main/source/settings.h
View File

@ -82,6 +82,7 @@
#define LB_TOTAL_DELAY_SAMPLES 48
enum ISACBandwidth {isac8kHz = 8, isac12kHz = 12, isac16kHz = 16};
enum ISACBand {kIsacLowerBand = 0, kIsacUpperBand12 = 1, kIsacUpperBand16 = 2};
enum IsacSamplingRate {kIsacWideband = 16, kIsacSuperWideband = 32};
#define UB_LPC_GAIN_DIM SUBFRAMES
#define FB_STATE_SIZE_WORD32 6
@ -167,6 +168,7 @@ enum ISACBand {kIsacLowerBand = 0, kIsacUpperBand12 = 1, kIsacUpperBand16 = 2};
#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 */

6
webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
View File

@ -472,6 +472,12 @@ typedef struct {
WebRtc_Word16 maxRateBytesPer30Ms;
// Maximum allowed payload-size, measured in Bytes.
WebRtc_Word16 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. */
WebRtc_UWord16 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];
} ISACMainStruct;
#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */

15
webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@ -477,8 +477,9 @@ int main(int argc, char* argv[])
if(!useAssign)
{
err =WebRtcIsac_Create(&ISAC_main_inst);
WebRtcIsac_SetEncSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
WebRtcIsac_SetDecSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000);
WebRtcIsac_SetDecSampRate(ISAC_main_inst, sampFreqKHz >= 32 ?
32000 : 16000);
}
else
{
@ -488,8 +489,9 @@ int main(int argc, char* argv[])
err = WebRtcIsac_AssignSize(&sss);
ppp = malloc(sss);
err = WebRtcIsac_Assign(&ISAC_main_inst, ppp);
WebRtcIsac_SetEncSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
WebRtcIsac_SetDecSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000);
WebRtcIsac_SetDecSampRate(ISAC_main_inst, sampFreqKHz >= 32 ?
32000 : 16000);
}
/* Error check */
if(err < 0)
@ -510,8 +512,9 @@ int main(int argc, char* argv[])
if(doTransCoding)
{
WebRtcIsac_Create(&decoderTransCoding);
WebRtcIsac_SetEncSampRate(decoderTransCoding, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
WebRtcIsac_SetDecSampRate(decoderTransCoding, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
WebRtcIsac_SetEncSampRate(decoderTransCoding, sampFreqKHz * 1000);
WebRtcIsac_SetDecSampRate(decoderTransCoding, sampFreqKHz >= 32 ?
32000 : 16000);
WebRtcIsac_DecoderInit(decoderTransCoding);
transCodingFile = fopen(transCodingFileName, "wb");
if(transCodingFile == NULL)

18
webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
View File

@ -38,7 +38,7 @@ int main(int argc, char* argv[])
ISACStruct* codecInstance[MAX_NUM_CLIENTS];
WebRtc_Word32 resamplerState[MAX_NUM_CLIENTS][8];
enum IsacSamplingRate encoderSampRate[MAX_NUM_CLIENTS];
int encoderSampRate[MAX_NUM_CLIENTS];
int minBn = 16000;
int maxBn = 56000;
@ -86,11 +86,11 @@ int main(int argc, char* argv[])
}
// THE FIRST CLIENT STARTS IN WIDEBAND
encoderSampRate[0] = kIsacWideband;
encoderSampRate[0] = 16000;
OPEN_FILE_RB(inFile[0], fileNameWB);
// THE SECOND CLIENT STARTS IN SUPER-WIDEBAND
encoderSampRate[1] = kIsacSuperWideband;
encoderSampRate[1] = 32000;
OPEN_FILE_RB(inFile[1], fileNameSWB);
strcpy(myFlag, "-I");
@ -104,7 +104,7 @@ int main(int argc, char* argv[])
printf("Client %d\n", clientCntr + 1);
printf("---------\n");
printf("Starting %s",
(encoderSampRate[clientCntr] == kIsacWideband)
(encoderSampRate[clientCntr] == 16000)
? "wideband":"super-wideband");
// Open output File Name
@ -242,17 +242,17 @@ int main(int argc, char* argv[])
printf("Changing Encoder Sampling frequency in client %d to ", senderIdx+1);
fclose(inFile[senderIdx]);
numSampFreqChanged++;
if(encoderSampRate[senderIdx] == kIsacWideband)
if(encoderSampRate[senderIdx] == 16000)
{
printf("super-wideband.\n");
OPEN_FILE_RB(inFile[senderIdx], fileNameSWB);
encoderSampRate[senderIdx] = kIsacSuperWideband;
encoderSampRate[senderIdx] = 32000;
}
else
{
printf("wideband.\n");
OPEN_FILE_RB(inFile[senderIdx], fileNameWB);
encoderSampRate[senderIdx] = kIsacWideband;
encoderSampRate[senderIdx] = 16000;
}
WebRtcIsac_SetEncSampRate(codecInstance[senderIdx], encoderSampRate[senderIdx]);
WebRtcIsac_SetDecSampRate(codecInstance[receiverIdx], encoderSampRate[senderIdx]);
@ -264,7 +264,7 @@ int main(int argc, char* argv[])
if(numSamplesRead != samplesIn10ms[senderIdx])
{
printf(" File %s for client %d has not enough audio\n",
(encoderSampRate[senderIdx]==kIsacWideband)? "wideband":"super-wideband",
(encoderSampRate[senderIdx]==16000)? "wideband":"super-wideband",
senderIdx + 1);
return -1;
}
@ -428,7 +428,7 @@ int main(int argc, char* argv[])
}
if(encoderSampRate[senderIdx] == kIsacWideband)
if(encoderSampRate[senderIdx] == 16000)
{
WebRtcSpl_UpsampleBy2(audioBuff60ms, lenDecodedAudio, resampledAudio60ms,
resamplerState[receiverIdx]);

5
webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
View File

@ -254,8 +254,9 @@ valid values are 8 and 16.\n", sampFreqKHz);
/* Initialize the ISAC and BN structs */
err = WebRtcIsac_Create(&ISAC_main_inst);
WebRtcIsac_SetEncSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband: kIsacSuperWideband);
WebRtcIsac_SetDecSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband: kIsacSuperWideband);
WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000);
WebRtcIsac_SetDecSampRate(ISAC_main_inst, sampFreqKHz >= 32 ? 32000 :
16000);
/* Error check */
if (err < 0) {
fprintf(stderr,"\n\n Error in create.\n\n");

7
webrtc/modules/audio_coding/main/source/acm_codec_database.cc
View File

@ -107,10 +107,10 @@ namespace webrtc {
// codecs. Note! There are a limited number of payload types. If more codecs
// are defined they will receive reserved fixed payload types (values 69-95).
const int kDynamicPayloadtypes[ACMCodecDB::kMaxNumCodecs] = {
105, 107, 108, 109, 111, 112, 113, 114, 115, 116, 117, 92,
107, 108, 109, 111, 112, 113, 114, 115, 116, 117, 92,
91, 90, 89, 88, 87, 86, 85, 84, 83, 82, 81, 80,
79, 78, 77, 76, 75, 74, 73, 72, 71, 70, 69, 68,
67, 66
67, 66, 65
};
// Creates database with all supported codecs at compile time.
@ -129,6 +129,7 @@ const CodecInst ACMCodecDB::database_[] = {
{103, "ISAC", 16000, kIsacPacSize480, 1, kIsacWbDefaultRate},
# if (defined(WEBRTC_CODEC_ISAC))
{104, "ISAC", 32000, kIsacPacSize960, 1, kIsacSwbDefaultRate},
{105, "ISAC", 48000, kIsacPacSize1440, 1, kIsacSwbDefaultRate},
# endif
#endif
#ifdef WEBRTC_CODEC_PCM16
@ -221,6 +222,7 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
{2, {kIsacPacSize480, kIsacPacSize960}, 0, 1},
# if (defined(WEBRTC_CODEC_ISAC))
{1, {kIsacPacSize960}, 0, 1},
{1, {kIsacPacSize1440}, 0, 1},
# endif
#endif
#ifdef WEBRTC_CODEC_PCM16
@ -311,6 +313,7 @@ const WebRtcNetEQDecoder ACMCodecDB::neteq_decoders_[] = {
kDecoderISAC,
# if (defined(WEBRTC_CODEC_ISAC))
kDecoderISACswb,
kDecoderISACfb,
# endif
#endif
#ifdef WEBRTC_CODEC_PCM16

2
webrtc/modules/audio_coding/main/source/acm_codec_database.h
View File

@ -33,6 +33,7 @@ class ACMCodecDB {
, kISAC
# if (defined(WEBRTC_CODEC_ISAC))
, kISACSWB
, kISACFB
# endif
#endif
#ifdef WEBRTC_CODEC_PCM16
@ -115,6 +116,7 @@ class ACMCodecDB {
// Set unsupported codecs to -1
#ifndef WEBRTC_CODEC_ISAC
enum {kISACSWB = -1};
enum {kISACFB = -1};
# ifndef WEBRTC_CODEC_ISACFX
enum {kISAC = -1};
# endif

1
webrtc/modules/audio_coding/main/source/acm_common_defs.h
View File

@ -54,6 +54,7 @@ const int kIsacWbDefaultRate = 32000;
const int kIsacSwbDefaultRate = 56000;
const int kIsacPacSize480 = 480;
const int kIsacPacSize960 = 960;
const int kIsacPacSize1440 = 1440;
// An encoded bit-stream is labeled by one of the following enumerators.
//

61
webrtc/modules/audio_coding/main/source/acm_isac.cc
View File

@ -465,7 +465,8 @@ WebRtc_Word16 ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codecParams) {
}
// set decoder sampling frequency.
if (codecParams->codecInstant.plfreq == 32000) {
if (codecParams->codecInstant.plfreq == 32000 ||
codecParams->codecInstant.plfreq == 48000) {
UpdateDecoderSampFreq(ACMCodecDB::kISACSWB);
} else {
UpdateDecoderSampFreq(ACMCodecDB::kISAC);
@ -556,9 +557,17 @@ WebRtc_Word32 ACMISAC::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
#endif
} else {
#ifdef WEBRTC_CODEC_ISAC
SET_CODEC_PAR((codecDef), kDecoderISACswb, codecInst.pltype,
_codecInstPtr->inst, 32000);
SET_ISACSWB_FUNCTIONS((codecDef));
// Decoder is either @ 16 kHz or 32 kHz. Even if encoder is set @ 48 kHz
// decoding is @ 32 kHz.
if (codecInst.plfreq == 32000) {
SET_CODEC_PAR((codecDef), kDecoderISACswb, codecInst.pltype,
_codecInstPtr->inst, 32000);
SET_ISACSWB_FUNCTIONS((codecDef));
} else {
SET_CODEC_PAR((codecDef), kDecoderISACfb, codecInst.pltype,
_codecInstPtr->inst, 32000);
SET_ISACFB_FUNCTIONS((codecDef));
}
#else
return -1;
#endif
@ -659,7 +668,7 @@ WebRtc_Word16 ACMISAC::SetBitRateSafe(WebRtc_Word32 bitRate) {
WebRtc_Word32 ACMISAC::GetEstimatedBandwidthSafe() {
WebRtc_Word16 bandwidthIndex = 0;
WebRtc_Word16 delayIndex = 0;
IsacSamplingRate sampRate;
int sampRate;
// Get bandwidth information
ACM_ISAC_GETSENDBWE(_codecInstPtr->inst, &bandwidthIndex, &delayIndex);
@ -671,7 +680,7 @@ WebRtc_Word32 ACMISAC::GetEstimatedBandwidthSafe() {
// Check sample frequency
sampRate = ACM_ISAC_GETDECSAMPRATE(_codecInstPtr->inst);
if (sampRate == kIsacWideband) {
if (sampRate == 16000) {
return isacRatesWB[bandwidthIndex];
} else {
return isacRatesSWB[bandwidthIndex];
@ -680,13 +689,13 @@ WebRtc_Word32 ACMISAC::GetEstimatedBandwidthSafe() {
WebRtc_Word32 ACMISAC::SetEstimatedBandwidthSafe(
WebRtc_Word32 estimatedBandwidth) {
IsacSamplingRate sampRate;
int sampRate;
WebRtc_Word16 bandwidthIndex;
// Check sample frequency and choose appropriate table
sampRate = ACM_ISAC_GETENCSAMPRATE(_codecInstPtr->inst);
if (sampRate == kIsacWideband) {
if (sampRate == 16000) {
// Search through the WB rate table to find the index
bandwidthIndex = NR_ISAC_BANDWIDTHS / 2 - 1;
for (int i = 0; i < (NR_ISAC_BANDWIDTHS / 2); i++) {
@ -738,10 +747,12 @@ WebRtc_Word32 ACMISAC::GetRedPayloadSafe(
WebRtc_Word16 ACMISAC::UpdateDecoderSampFreq(
#ifdef WEBRTC_CODEC_ISAC
WebRtc_Word16 codecId) {
// The decoder supports only wideband and super-wideband.
if (ACMCodecDB::kISAC == codecId) {
return WebRtcIsac_SetDecSampRate(_codecInstPtr->inst, kIsacWideband);
} else if (ACMCodecDB::kISACSWB == codecId) {
return WebRtcIsac_SetDecSampRate(_codecInstPtr->inst, kIsacSuperWideband);
return WebRtcIsac_SetDecSampRate(_codecInstPtr->inst, 16000);
} else if (ACMCodecDB::kISACSWB == codecId ||
ACMCodecDB::kISACFB == codecId) {
return WebRtcIsac_SetDecSampRate(_codecInstPtr->inst, 32000);
} else {
return -1;
}
@ -758,24 +769,18 @@ WebRtc_Word16 ACMISAC::UpdateEncoderSampFreq(
EncoderSampFreq(currentSampRateHz);
if (currentSampRateHz != encoderSampFreqHz) {
if ((encoderSampFreqHz != 16000) && (encoderSampFreqHz != 32000)) {
if ((encoderSampFreqHz != 16000) && (encoderSampFreqHz != 32000) &&
(encoderSampFreqHz != 48000)) {
return -1;
} else {
_inAudioIxRead = 0;
_inAudioIxWrite = 0;
_inTimestampIxWrite = 0;
if (encoderSampFreqHz == 16000) {
if (WebRtcIsac_SetEncSampRate(_codecInstPtr->inst, kIsacWideband) < 0) {
return -1;
}
_samplesIn10MsAudio = 160;
} else {
if (WebRtcIsac_SetEncSampRate(_codecInstPtr->inst, kIsacSuperWideband)
< 0) {
return -1;
}
_samplesIn10MsAudio = 320;
if (WebRtcIsac_SetEncSampRate(_codecInstPtr->inst,
encoderSampFreqHz) < 0) {
return -1;
}
_samplesIn10MsAudio = encoderSampFreqHz / 100;
_frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
_encoderParams.codecInstant.pacsize = _frameLenSmpl;
_encoderParams.codecInstant.plfreq = encoderSampFreqHz;
@ -789,13 +794,7 @@ WebRtc_Word16 ACMISAC::UpdateEncoderSampFreq(
}
WebRtc_Word16 ACMISAC::EncoderSampFreq(WebRtc_UWord16& sampFreqHz) {
IsacSamplingRate sampRate;
sampRate = ACM_ISAC_GETENCSAMPRATE(_codecInstPtr->inst);
if (sampRate == kIsacSuperWideband) {
sampFreqHz = 32000;
} else {
sampFreqHz = 16000;
}
sampFreqHz = ACM_ISAC_GETENCSAMPRATE(_codecInstPtr->inst);
return 0;
}
@ -809,7 +808,7 @@ WebRtc_Word32 ACMISAC::ConfigISACBandwidthEstimator(
// TODO(turajs): at 32kHz we hardcode calling with 30ms and enforce
// the frame-size otherwise we might get error. Revise if
// control-bwe is changed.
if (sampFreqHz == 32000) {
if (sampFreqHz == 32000 || sampFreqHz == 48000) {
status = ACM_ISAC_CONTROL_BWE(_codecInstPtr->inst, initRateBitPerSec, 30,
1);
} else {

6
webrtc/modules/audio_coding/neteq/codec_db.c
View File

@ -115,6 +115,9 @@ int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
#ifdef NETEQ_ISAC_SWB_CODEC
case kDecoderISACswb :
#endif
#ifdef NETEQ_ISAC_FB_CODEC
case kDecoderISACfb :
#endif
#ifdef NETEQ_OPUS_CODEC
case kDecoderOpus :
#endif
@ -463,6 +466,9 @@ int WebRtcNetEQ_DbGetSplitInfo(SplitInfo_t *inst, enum WebRtcNetEQDecoder codecI
#ifdef NETEQ_ISAC_SWB_CODEC
case kDecoderISACswb:
#endif
#ifdef NETEQ_ISAC_FB_CODEC
case kDecoderISACfb:
#endif
#ifdef NETEQ_OPUS_CODEC
case kDecoderOpus:
#endif

1
webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h
View File

@ -37,6 +37,7 @@ enum WebRtcNetEQDecoder
kDecoderILBC,
kDecoderISAC,
kDecoderISACswb,
kDecoderISACfb,
kDecoderPCM16B,
kDecoderPCM16Bwb,
kDecoderPCM16Bswb32kHz,

12
webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h
View File

@ -102,6 +102,18 @@
inst.funcDurationEst=NULL; \
inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsac_GetErrorCode;
#define SET_ISACFB_FUNCTIONS(inst) \
inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsac_Decode; \
inst.funcDecodeRCU=(WebRtcNetEQ_FuncDecode)WebRtcIsac_DecodeRcu; \
inst.funcDecodePLC=NULL; \
inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsac_DecoderInit; \
inst.funcAddLatePkt=NULL; \
inst.funcGetMDinfo=NULL; \
inst.funcGetPitch=NULL; \
inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsac_UpdateBwEstimate; \
inst.funcDurationEst=NULL; \
inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsac_GetErrorCode;
#define SET_G729_FUNCTIONS(inst) \
inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG729_Decode; \
inst.funcDecodeRCU=NULL; \

3
webrtc/modules/audio_coding/neteq/neteq.gypi
View File

@ -122,6 +122,7 @@
'CODEC_ISAC',
'CODEC_PCM16B_WB',
'CODEC_ISAC_SWB',
'CODEC_ISAC_FB',
'CODEC_PCM16B_32KHZ',
'CODEC_CNGCODEC8',
'CODEC_CNGCODEC16',
@ -159,6 +160,7 @@
'CODEC_ISAC',
'CODEC_PCM16B_WB',
'CODEC_ISAC_SWB',
'CODEC_ISAC_FB',
'CODEC_PCM16B_32KHZ',
'CODEC_CNGCODEC8',
'CODEC_CNGCODEC16',
@ -271,6 +273,7 @@
'CODEC_ISAC',
'CODEC_PCM16B_WB',
'CODEC_ISAC_SWB',
'CODEC_ISAC_FB',
'CODEC_PCM16B_32KHZ',
'CODEC_CNGCODEC8',
'CODEC_CNGCODEC16',

6
webrtc/modules/audio_coding/neteq/neteq_defines.h
View File

@ -65,6 +65,10 @@
*
* NETEQ_ISAC_SWB_CODEC Enable iSAC-SWB
*
* Note that the decoder of iSAC full-band operates at 32 kHz, that is the
* decoded signal is at 32 kHz.
* NETEQ_ISAC_FB_CODEC Enable iSAC-FB
*
* NETEQ_G722_CODEC Enable G.722
*
* NETEQ_G729_CODEC Enable G.729
@ -302,6 +306,7 @@
/* Fullband 48 kHz codecs */
#define NETEQ_OPUS_CODEC
#define NETEQ_ISAC_FB_CODEC
#endif
#if (defined(NETEQ_ALL_CODECS))
@ -339,6 +344,7 @@
/* Super wideband 48kHz codecs */
#define NETEQ_48KHZ_WIDEBAND
#define NETEQ_OPUS_CODEC
#define NETEQ_ISAC_FB
#endif
/* Max output size from decoding one frame */

3
webrtc/modules/audio_coding/neteq/packet_buffer.c
View File

@ -615,7 +615,8 @@ int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID,
codecBytes = 960; /* 240ms @ 32kbps (60ms frames) */
codecBuffers = 8;
}
else if (codecID[i] == kDecoderISACswb)
else if ((codecID[i] == kDecoderISACswb) ||
(codecID[i] == kDecoderISACfb))
{
codecBytes = 1560; /* 240ms @ 52kbps (30ms frames) */
codecBuffers = 8;

7
webrtc/modules/audio_coding/neteq/recin.c
View File

@ -380,11 +380,12 @@ int WebRtcNetEQ_GetTimestampScaling(MCUInst_t *MCU_inst, int rtpPayloadType)
MCU_inst->scalingFactor = kTSscalingTwo;
break;
}
case kDecoderISACfb:
case kDecoderOpus:
{
/* We resample Opus internally to 32 kHz, but timestamps
* are counted at 48 kHz. So there are two output samples
* per three RTP timestamp ticks. */
/* We resample Opus internally to 32 kHz, and isac-fb decodes at
* 32 kHz, but timestamps are counted at 48 kHz. So there are two
* output samples per three RTP timestamp ticks. */
MCU_inst->scalingFactor = kTSscalingTwoThirds;
break;
}

30
webrtc/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc
View File

@ -53,7 +53,7 @@ NETEQTEST_Decoder(kDecoderISAC, 16000, "iSAC", pt)
}
WebRtcIsac_EncoderInit((ISACStruct *) _decoder, 0);
WebRtcIsac_SetDecSampRate((ISACStruct *) _decoder, kIsacWideband);
WebRtcIsac_SetDecSampRate((ISACStruct *) _decoder, 16000);
}
@ -90,7 +90,7 @@ NETEQTEST_Decoder(kDecoderISACswb, 32000, "iSAC swb", pt)
}
WebRtcIsac_EncoderInit((ISACStruct *) _decoder, 0);
WebRtcIsac_SetDecSampRate((ISACStruct *) _decoder, kIsacSuperWideband);
WebRtcIsac_SetDecSampRate((ISACStruct *) _decoder, 32000);
}
decoder_iSACSWB::~decoder_iSACSWB()
@ -113,6 +113,32 @@ int decoder_iSACSWB::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
}
#endif
#ifdef CODEC_ISAC_FB
decoder_iSACFB::decoder_iSACFB(WebRtc_UWord8 pt)
: NETEQTEST_Decoder(kDecoderISACfb, 32000, "iSAC fb", pt) {
WebRtc_Word16 err = WebRtcIsac_Create((ISACStruct **) &_decoder);
if (err) {
exit(EXIT_FAILURE);
}
WebRtcIsac_EncoderInit((ISACStruct *) _decoder, 0);
WebRtcIsac_SetDecSampRate((ISACStruct *) _decoder, 32000);
}
decoder_iSACFB::~decoder_iSACFB() {
if (_decoder) {
WebRtcIsac_Free((ISACStruct *) _decoder);
_decoder = NULL;
}
}
int decoder_iSACFB::loadToNetEQ(NETEQTEST_NetEQClass & neteq){
WebRtcNetEQ_CodecDef codecInst;
SET_ISACFB_FUNCTIONS(codecInst);
return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
}
#endif
// PCM u/A
#ifdef CODEC_G711
#include "g711_interface.h"

8
webrtc/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h
View File

@ -64,6 +64,14 @@ public:
};
class decoder_iSACFB : public NETEQTEST_Decoder {
public:
decoder_iSACFB(WebRtc_UWord8 pt = 0);
virtual ~decoder_iSACFB();
int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
};
class decoder_PCMU : public NETEQTEST_Decoder
{
public:

11
webrtc/modules/audio_coding/neteq/test/NetEqRTPplay.cc
View File

@ -955,6 +955,12 @@ void parsePtypeFile(FILE *ptypeFile, std::map<WebRtc_UWord8, decoderStruct>* dec
tempDecoder.fs = 32000;
}
#endif
#ifdef CODEC_ISAC_FB
else if(strcmp(codec, "isacfb") == 0) {
tempDecoder.codec = kDecoderISACfb;
tempDecoder.fs = 32000;
}
#endif
#ifdef CODEC_IPCMWB
else if(strcmp(codec, "ipcmwb") == 0) {
tempDecoder.codec = kDecoderIPCMwb;
@ -1358,6 +1364,11 @@ void createAndInsertDecoders (NETEQTEST_NetEQClass *neteq, std::map<WebRtc_UWord
*dec = new decoder_iSACSWB( pt );
break;
#endif
#ifdef CODEC_ISAC_FB
case kDecoderISACfb:
*dec = new decoder_iSACFB(pt);
break;
#endif
#ifdef CODEC_G729
case kDecoderG729:
*dec = new decoder_G729( pt );

3
webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
View File

@ -35,6 +35,7 @@
#define NETEQ_CODEC_ISAC_PT 103
#define NETEQ_CODEC_ISACLC_PT 119
#define NETEQ_CODEC_ISACSWB_PT 104
#define NETEQ_CODEC_ISACFB_PT 124
#define NETEQ_CODEC_AVT_PT 106
#define NETEQ_CODEC_G722_1_16_PT 108
#define NETEQ_CODEC_G722_1_24_PT 109
@ -53,7 +54,7 @@
#define NETEQ_CODEC_CN_SWB_PT 126
#define NETEQ_CODEC_G729_1_PT 107
#define NETEQ_CODEC_G729D_PT 123
#define NETEQ_CODEC_MELPE_PT 124
//#define NETEQ_CODEC_MELPE_PT 124
#define NETEQ_CODEC_CELT32_PT 114
/* Extra dynamic codepoints */

81
webrtc/modules/audio_coding/neteq/test/RTPencode.cc
View File

@ -110,7 +110,7 @@ void stereoInterleave(unsigned char* data, int dataLen, int stride);
#ifdef CODEC_ILBC
#include "ilbc.h"
#endif
#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB || defined CODEC_ISAC_FB)
#include "isac.h"
#endif
#ifdef NETEQ_ISACFIX_CODEC
@ -217,6 +217,9 @@ WebRtcVadInst *VAD_inst[2];
#ifdef CODEC_ISAC_SWB
ISACStruct *ISACSWB_inst[2];
#endif
#ifdef CODEC_ISAC_FB
ISACStruct *ISACFB_inst[2];
#endif
#ifdef CODEC_GSMFR
GSMFR_encinst_t *GSMFRenc_inst[2];
#endif
@ -362,6 +365,9 @@ int main(int argc, char* argv[])
#ifdef CODEC_ISAC_SWB
printf(" : isacswb iSAC SWB (32kHz and 32.0-52.0 kbps). To set rate specify a rate parameter as last parameter\n");
#endif
#ifdef CODEC_ISAC_FB
printf(" : isacfb iSAC FB (48kHz encoder 32kHz decoder and 32.0-52.0 kbps). To set rate specify a rate parameter as last parameter\n");
#endif
#ifdef CODEC_GSMFR
printf(" : gsmfr GSM FR codec (8kHz and 13kbps)\n");
#endif
@ -482,7 +488,8 @@ int main(int argc, char* argv[])
}
}
if ((usedCodec == kDecoderISAC) || (usedCodec == kDecoderISACswb))
if ((usedCodec == kDecoderISAC) || (usedCodec == kDecoderISACswb) ||
(usedCodec == kDecoderISACfb))
{
if (argc != 7)
{
@ -492,7 +499,7 @@ int main(int argc, char* argv[])
printf(
"Running iSAC at default bitrate of 32000 bps (to specify explicitly add the bps as last parameter)\n");
}
else // (usedCodec==kDecoderISACswb)
else // usedCodec == kDecoderISACswb || usedCodec == kDecoderISACfb
{
bitrate = 56000;
printf(
@ -513,12 +520,12 @@ int main(int argc, char* argv[])
}
printf("Running iSAC at bitrate of %i bps\n", bitrate);
}
else // (usedCodec==kDecoderISACswb)
else // usedCodec == kDecoderISACswb || usedCodec == kDecoderISACfb
{
if ((bitrate < 32000) || (bitrate > 56000))
{
printf(
"Error: iSAC SWB bitrate must be between 32000 and 56000 bps (%i is invalid)\n",
"Error: iSAC SWB/FB bitrate must be between 32000 and 56000 bps (%i is invalid)\n",
bitrate);
exit(0);
}
@ -970,11 +977,16 @@ void NetEQTest_GetCodec_and_PT(char * name, enum WebRtcNetEQDecoder *codec, int
*codec=kDecoderISAC;
*PT=NETEQ_CODEC_ISAC_PT;
}
else if(!strcmp(name,"isacswb")){
*fs=32000;
*codec=kDecoderISACswb;
*PT=NETEQ_CODEC_ISACSWB_PT;
else if(!strcmp(name,"isacswb")){
*fs=32000;
*codec=kDecoderISACswb;
*PT=NETEQ_CODEC_ISACSWB_PT;
}
else if(!strcmp(name,"isacfb")){
*fs=48000;
*codec=kDecoderISACfb;
*PT=NETEQ_CODEC_ISACFB_PT;
}
else if(!strcmp(name,"g729")){
*fs=8000;
*codec=kDecoderG729;
@ -1481,7 +1493,7 @@ int NetEQTest_init_coders(enum WebRtcNetEQDecoder coder, int enc_frameSize, int
printf("\nError - iSAC SWB only supports frameSize 30 ms\n");
exit(0);
}
ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], kIsacSuperWideband);
ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], 32000);
if (ok!=0) {
printf("Error: Couldn't set sample rate for iSAC SWB instance\n");
exit(0);
@ -1498,6 +1510,38 @@ int NetEQTest_init_coders(enum WebRtcNetEQDecoder coder, int enc_frameSize, int
}
break;
#endif
#ifdef CODEC_ISAC_FB
case kDecoderISACfb:
if (sampfreq == 48000) {
ok = WebRtcIsac_Create(&ISACFB_inst[k]);
if (ok != 0) {
printf("Error: Couldn't allocate memory for iSAC FB "
"instance\n");
exit(0);
}
if (enc_frameSize != 1440) {
printf("\nError - iSAC FB only supports frameSize 30 ms\n");
exit(0);
}
ok = WebRtcIsac_SetEncSampRate(ISACFB_inst[k], 48000);
if (ok != 0) {
printf("Error: Couldn't set sample rate for iSAC FB "
"instance\n");
exit(0);
}
WebRtcIsac_EncoderInit(ISACFB_inst[k], 1);
if ((bitrate < 32000) || (bitrate > 56000)) {
printf("\nError - iSAC FB bitrate has to be between 32000 and"
"56000 bps (not %i)\n", bitrate);
exit(0);
}
WebRtcIsac_Control(ISACFB_inst[k], bitrate, 30);
} else {
printf("\nError - iSAC FB only support 48 kHz sampling rate.\n");
exit(0);
}
break;
#endif
#ifdef CODEC_GSMFR
case kDecoderGSMFR:
if (sampfreq==8000) {
@ -1657,6 +1701,11 @@ int NetEQTest_free_coders(enum WebRtcNetEQDecoder coder, int numChannels) {
WebRtcIsac_Free(ISACSWB_inst[k]);
break;
#endif
#ifdef CODEC_ISAC_FB
case kDecoderISACfb:
WebRtcIsac_Free(ISACFB_inst[k]);
break;
#endif
#ifdef CODEC_GSMFR
case kDecoderGSMFR:
WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
@ -1859,6 +1908,18 @@ int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned ch
}
}
#endif
#ifdef CODEC_ISAC_FB
else if (coder == kDecoderISACfb) { /* iSAC FB */
int noOfCalls = 0;
cdlen = 0;
while (cdlen <= 0) {
cdlen = WebRtcIsac_Encode(ISACFB_inst[k],
&indata[noOfCalls * 480],
(WebRtc_Word16*)encoded);
noOfCalls++;
}
}
#endif
#ifdef CODEC_GSMFR
else if (coder==kDecoderGSMFR) { /* GSM FR */
cdlen=WebRtcGSMFR_Encode(GSMFRenc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);

1
webrtc/modules/audio_coding/neteq/test/ptypes.txt
View File

@ -7,6 +7,7 @@ cn 13
ilbc 102
isac 103
isacswb 104
isacfb 124
avt 106
red 117
cn_wb 98

2
webrtc/modules/audio_coding/neteq/webrtc_neteq_unittest.cc
View File

@ -239,6 +239,8 @@ void NetEqDecodingTest::SelectDecoders(WebRtcNetEQDecoder* used_codec) {
dec_.push_back(new decoder_iSAC(103));
*used_codec++ = kDecoderISACswb;
dec_.push_back(new decoder_iSACSWB(104));
*used_codec++ = kDecoderISACfb;
dec_.push_back(new decoder_iSACFB(105));
*used_codec++ = kDecoderPCM16B;
dec_.push_back(new decoder_PCM16B_NB(93));
*used_codec++ = kDecoderPCM16Bwb;

6
webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
View File

@ -340,8 +340,10 @@ void RunTest(std::string out_path) {
if (strncmp(cinst.plname, "ISAC", 4) == 0 && cinst.plfreq == 32000) {
printf("%i. ISAC-swb pltype:%i plfreq:%i channels:%i\n", i, cinst.pltype,
cinst.plfreq, cinst.channels);
}
else {
} else if (strncmp(cinst.plname, "ISAC", 4) == 0 && cinst.plfreq == 48000) {
printf("%i. ISAC-fb pltype:%i plfreq:%i channels:%i\n", i, cinst.pltype,
cinst.plfreq, cinst.channels);
} else {
printf("%i. %s pltype:%i plfreq:%i channels:%i\n", i, cinst.plname,
cinst.pltype, cinst.plfreq, cinst.channels);
}