zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .

Browse Source 
These are mostly trivial changes and are separated out just to reduce the
diff on that change to the minimum possible.

Note explanatory comments on patch set 1.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1235643003

Cr-Commit-Position: refs/heads/master@{#9617}
This commit is contained in:
pkasting
2015-07-22 15:17:22 -07:00
committed by Commit bot
parent 7c5304c791
commit b297c5a01f
29 changed files with 105 additions and 83 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
View File

@ -28,7 +28,7 @@ void WebRtcIlbcfix_CbMemEnergy(
int16_t *energyW16, /* (o) Energy in the CB vectors */
int16_t *energyShifts, /* (o) Shift value of the energy */
int scale, /* (i) The scaling of all energy values */
int16_t base_size /* (i) Index to where the energy values should be stored */
int16_t base_size /* (i) Index to where energy values should be stored */
);
#endif

3
webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
View File

@ -70,7 +70,8 @@ void WebRtcIlbcfix_DoThePlc(
/* Maximum 60 samples are correlated, preserve as high accuracy
as possible without getting overflow */
max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual, (int16_t)iLBCdec_inst->blockl);
max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual,
(int16_t)iLBCdec_inst->blockl);
scale3 = (WebRtcSpl_GetSizeInBits(max)<<1) - 25;
if (scale3 < 0) {
scale3 = 0;

6
webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
View File

@ -27,9 +27,9 @@
int WebRtcIsacfix_DecodeImpl(int16_t *signal_out16,
IsacFixDecoderInstance *ISACdec_obj,
int16_t *current_framesamples)
int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
IsacFixDecoderInstance* ISACdec_obj,
int16_t* current_framesamples)
{
int k;
int err;

35
webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
View File

@ -777,7 +777,7 @@ int WebRtcIsacfix_Decode(ISACFIX_MainStruct* ISAC_main_inst,
ISACFIX_SubStruct *ISAC_inst;
/* number of samples (480 or 960), output from decoder */
/* that were actually used in the encoder/decoder (determined on the fly) */
int16_t number_of_samples;
int16_t number_of_samples;
int declen = 0;
/* typecast pointer to real structure */
@ -807,8 +807,8 @@ int WebRtcIsacfix_Decode(ISACFIX_MainStruct* ISAC_main_inst,
/* added for NetEq purposes (VAD/DTX related) */
*speechType=1;
declen = WebRtcIsacfix_DecodeImpl(decoded,&ISAC_inst->ISACdec_obj, &number_of_samples);
declen = WebRtcIsacfix_DecodeImpl(decoded, &ISAC_inst->ISACdec_obj,
&number_of_samples);
if (declen < 0) {
/* Some error inside the decoder */
ISAC_inst->errorcode = -(int16_t)declen;
@ -818,14 +818,18 @@ int WebRtcIsacfix_Decode(ISACFIX_MainStruct* ISAC_main_inst,
/* error check */
if (declen & 0x0001) {
if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) & 0x00FF) ) {
if (declen & 1) {
if (len != declen &&
len != declen +
((ISAC_inst->ISACdec_obj.bitstr_obj.stream[declen >> 1]) & 0xFF)) {
ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
memset(decoded, 0, sizeof(int16_t) * number_of_samples);
return -1;
}
} else {
if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) >> 8) ) {
if (len != declen &&
len != declen +
((ISAC_inst->ISACdec_obj.bitstr_obj.stream[declen >> 1]) >> 8)) {
ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
memset(decoded, 0, sizeof(int16_t) * number_of_samples);
return -1;
@ -870,7 +874,7 @@ int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
ISACFIX_SubStruct *ISAC_inst;
/* twice the number of samples (480 or 960), output from decoder */
/* that were actually used in the encoder/decoder (determined on the fly) */
int16_t number_of_samples;
int16_t number_of_samples;
int declen = 0;
int16_t dummy[FRAMESAMPLES/2];
@ -901,8 +905,8 @@ int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
/* added for NetEq purposes (VAD/DTX related) */
*speechType=1;
declen = WebRtcIsacfix_DecodeImpl(decoded,&ISAC_inst->ISACdec_obj, &number_of_samples);
declen = WebRtcIsacfix_DecodeImpl(decoded, &ISAC_inst->ISACdec_obj,
&number_of_samples);
if (declen < 0) {
/* Some error inside the decoder */
ISAC_inst->errorcode = -(int16_t)declen;
@ -912,14 +916,18 @@ int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
/* error check */
if (declen & 0x0001) {
if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) & 0x00FF) ) {
if (declen & 1) {
if (len != declen &&
len != declen +
((ISAC_inst->ISACdec_obj.bitstr_obj.stream[declen >> 1]) & 0xFF)) {
ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
memset(decoded, 0, sizeof(int16_t) * number_of_samples);
return -1;
}
} else {
if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) >> 8) ) {
if (len != declen &&
len != declen +
((ISAC_inst->ISACdec_obj.bitstr_obj.stream[declen >>1]) >> 8)) {
ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
memset(decoded, 0, sizeof(int16_t) * number_of_samples);
return -1;
@ -1319,7 +1327,8 @@ int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded,
read_be16(encoded, kRequiredEncodedLenBytes, streamdata.stream);
/* decode frame length, needed to get to the rateIndex in the bitstream */
err = WebRtcIsacfix_DecodeFrameLen(&streamdata, rateIndex);
int16_t frameLength;
err = WebRtcIsacfix_DecodeFrameLen(&streamdata, &frameLength);
if (err<0) // error check
return err;

2
webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
View File

@ -218,7 +218,7 @@ void WebRtcIsacfix_NormLatticeFilterAr(int16_t orderCoef,
int16_t lo_hi,
int16_t *lat_outQ0)
{
int ii,n,k,i,u;
int ii, n, k, i, u;
int16_t sthQ15[MAX_AR_MODEL_ORDER];
int16_t cthQ15[MAX_AR_MODEL_ORDER];
int32_t tmp32;

4
webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
View File

@ -38,7 +38,7 @@ DEFINE_FUNCTION WebRtcIsacfix_FilterArLoop
mov r4, #HALF_SUBFRAMELEN
sub r4, #1 @ Outer loop counter = HALF_SUBFRAMELEN - 1
HALF_SUBFRAME_LOOP: @ for(n = 0; n < HALF_SUBFRAMELEN - 1; n++)
HALF_SUBFRAME_LOOP: @ for (n = 0; n < HALF_SUBFRAMELEN - 1; n++)
ldr r9, [sp, #32] @ Restore the inner loop counter to order_coef
ldrh r5, [r1] @ tmpAR = ar_f_Q0[n+1]
@ -46,7 +46,7 @@ HALF_SUBFRAME_LOOP: @ for(n = 0; n < HALF_SUBFRAMELEN - 1; n++)
add r2, r9, asl #1 @ Restore r2 to &cth_Q15[order_coef]
add r3, r9, asl #1 @ Restore r3 to &sth_Q15[order_coef]
ORDER_COEF_LOOP: @ for(k = order_coef ; k > 0; k--)
ORDER_COEF_LOOP: @ for (k = order_coef; k > 0; k--)
ldrh r7, [r3, #-2]! @ sth_Q15[k - 1]
ldrh r6, [r2, #-2]! @ cth_Q15[k - 1]

14
webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
View File

@ -624,8 +624,8 @@ int main(int argc, char* argv[])
errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
printf("\nError in encoder: %d.\n", errtype);
} else {
if (fwrite(streamdata, sizeof(char),
stream_len, outbits) != (size_t)stream_len) {
if (fwrite(streamdata, sizeof(char), stream_len, outbits) !=
(size_t)stream_len) {
return -1;
}
}
@ -731,12 +731,12 @@ int main(int argc, char* argv[])
/* iSAC decoding */
if( lostFrame && framecnt > 0) {
if (nbTest !=2) {
declen = WebRtcIsacfix_DecodePlc(ISAC_main_inst,
decoded, prevFrameSize );
declen =
WebRtcIsacfix_DecodePlc(ISAC_main_inst, decoded, prevFrameSize);
} else {
#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
declen = WebRtcIsacfix_DecodePlcNb(ISAC_main_inst, decoded,
prevFrameSize );
declen = WebRtcIsacfix_DecodePlcNb(
ISAC_main_inst, decoded, prevFrameSize);
#else
declen = -1;
#endif
@ -755,7 +755,7 @@ int main(int argc, char* argv[])
decoded,
speechType);
/* Error check */
if (err<0 || declen<0 || FL!=declen) {
if (err < 0 || declen < 0 || FL != declen) {
errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
printf("\nError in decode_B/or getFrameLen: %d.\n", errtype);
}

7
webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
View File

@ -556,12 +556,13 @@ int main(int argc, char* argv[]) {
else
declen = WebRtcIsacfix_DecodePlcNb(ISACFIX_main_inst, decoded, 1);
} else {
if (nbTest != 2)
if (nbTest != 2) {
declen = WebRtcIsacfix_Decode(ISACFIX_main_inst, streamdata,
stream_len, decoded, speechType);
else
} else {
declen = WebRtcIsacfix_DecodeNb(ISACFIX_main_inst, streamdata,
stream_len, decoded, speechType);
}
}
if (declen <= 0) {
/* exit if returned with error */
@ -582,7 +583,7 @@ int main(int argc, char* argv[]) {
totalsmpls += declen;
totalbits += 8 * stream_len;
kbps = ((double)FS) / ((double)cur_framesmpls) * 8.0 * stream_len / 1000.0;
kbps = (double)FS / (double)cur_framesmpls * 8.0 * stream_len / 1000.0;
fy = fopen("bit_rate.dat", "a");
fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
fclose(fy);

3
webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
View File

@ -750,7 +750,8 @@ int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
streamLenUB + garbageLen, &crc);
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
encoded[streamLen - LEN_CHECK_SUM_WORD8 + k] = crc >> (24 - k * 8);
encoded[streamLen - LEN_CHECK_SUM_WORD8 + k] =
(uint8_t)(crc >> (24 - k * 8));
}
#else
memcpy(&encoded[streamLenLB + streamLenUB + 1], &crc, LEN_CHECK_SUM_WORD8);

3
webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@ -662,8 +662,9 @@ int main(int argc, char* argv[]) {
if (stream_len < 0) {
/* exit if returned with error */
errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
printf("\n\nError in encoder: %d.\n\n", errtype);
fprintf(stderr, "Error in encoder: %d.\n", errtype);
cout << flush;
exit(0);
}
cur_framesmpls += samplesIn10Ms;
/* exit encoder loop if the encoder returned a bitstream */

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

@ -350,6 +350,11 @@ int main(int argc, char* argv[]) {
}
rcuStreamLen = WebRtcIsac_GetRedPayload(ISAC_main_inst, payloadRCU);
if (rcuStreamLen < 0) {
fprintf(stderr, "\nError getting RED payload\n");
getc(stdin);
exit(EXIT_FAILURE);
}
get_arrival_time(cur_framesmpls, stream_len, bottleneck, &packetData,
sampFreqKHz * 1000, sampFreqKHz * 1000);

2
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -115,7 +115,7 @@ size_t AudioEncoderOpus::MaxEncodedBytes() const {
// Calculate the number of bytes we expect the encoder to produce,
// then multiply by two to give a wide margin for error.
size_t bytes_per_millisecond =
static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
size_t approx_encoded_bytes =
num_10ms_frames_per_packet_ * 10 * bytes_per_millisecond;
return 2 * approx_encoded_bytes;

3
webrtc/modules/audio_coding/main/test/PCMFile.cc
View File

@ -150,8 +150,7 @@ void PCMFile::Write10MsData(AudioFrame& audio_frame) {
}
} else {
int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_];
int k;
for (k = 0; k < audio_frame.samples_per_channel_; k++) {
for (int k = 0; k < audio_frame.samples_per_channel_; k++) {
stereo_audio[k << 1] = audio_frame.data_[k];
stereo_audio[(k << 1) + 1] = audio_frame.data_[k];
}

12
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
View File

@ -256,16 +256,14 @@ int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
static_cast<int16_t>(encoded_len / 2),
&decoded[decoded_len], &temp_type);
if (ret == decoded_len) {
decoded_len += ret;
ret += decoded_len; // Return total number of samples.
// Interleave output.
for (int k = decoded_len / 2; k < decoded_len; k++) {
for (int k = ret / 2; k < ret; k++) {
int16_t temp = decoded[k];
memmove(&decoded[2 * k - decoded_len + 2],
&decoded[2 * k - decoded_len + 1],
(decoded_len - k - 1) * sizeof(int16_t));
decoded[2 * k - decoded_len + 1] = temp;
memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
(ret - k - 1) * sizeof(int16_t));
decoded[2 * k - ret + 1] = temp;
}
ret = decoded_len; // Return total number of samples.
}
}
*speech_type = ConvertSpeechType(temp_type);

4
webrtc/modules/audio_coding/neteq/dsp_helper.cc
View File

@ -117,7 +117,7 @@ void DspHelper::PeakDetection(int16_t* data, int data_length,
peak_index[i] = WebRtcSpl_MaxIndexW16(data, data_length - 1);
if (i != num_peaks - 1) {
min_index = std::max(0, peak_index[i] - 2);
min_index = (peak_index[i] > 2) ? (peak_index[i] - 2) : 0;
max_index = std::min(data_length - 1, peak_index[i] + 2);
}
@ -238,7 +238,7 @@ void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
int DspHelper::MinDistortion(const int16_t* signal, int min_lag,
int max_lag, int length,
int32_t* distortion_value) {
int best_index = -1;
int best_index = 0;
int32_t min_distortion = WEBRTC_SPL_WORD32_MAX;
for (int i = min_lag; i <= max_lag; i++) {
int32_t sum_diff = 0;

4
webrtc/modules/audio_coding/neteq/expand.cc
View File

@ -441,8 +441,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
&audio_history[signal_length - correlation_length - start_index
- correlation_lags],
correlation_length + start_index + correlation_lags - 1);
correlation_scale = ((31 - WebRtcSpl_NormW32(signal_max * signal_max))
+ (31 - WebRtcSpl_NormW32(correlation_length))) - 31;
correlation_scale = (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
(31 - WebRtcSpl_NormW32(correlation_length)) - 31;
correlation_scale = std::max(0, correlation_scale);
// Calculate the correlation, store in |correlation_vector2|.

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

@ -255,7 +255,7 @@ int PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database,
continue;
}
int duration =
decoder->PacketDuration(packet->payload, packet->payload_length);
decoder->PacketDuration(packet->payload, packet->payload_length);
if (duration >= 0) {
last_duration = duration; // Save the most up-to-date (valid) duration.
}

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

@ -454,7 +454,7 @@ int main(int argc, char* argv[]) {
printf("Packet size %d must be positive", packet_size);
return -1;
}
printf("Packet size: %i\n", packet_size);
printf("Packet size: %d\n", packet_size);
// check for stereo
if (argv[4][strlen(argv[4]) - 1] == '*') {
@ -1572,29 +1572,31 @@ int NetEQTest_encode(int coder,
if (useVAD) {
*vad = 0;
int sampleRate_10 = 10 * sampleRate / 1000;
int sampleRate_20 = 20 * sampleRate / 1000;
int sampleRate_30 = 30 * sampleRate / 1000;
for (int k = 0; k < numChannels; k++) {
tempLen = frameLen;
tempdata = &indata[k * frameLen];
int localVad = 0;
/* Partition the signal and test each chunk for VAD.
All chunks must be VAD=0 to produce a total VAD=0. */
while (tempLen >= 10 * sampleRate / 1000) {
if ((tempLen % 30 * sampleRate / 1000) ==
0) { // tempLen is multiple of 30ms
while (tempLen >= sampleRate_10) {
if ((tempLen % sampleRate_30) == 0) { // tempLen is multiple of 30ms
localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
30 * sampleRate / 1000);
tempdata += 30 * sampleRate / 1000;
tempLen -= 30 * sampleRate / 1000;
} else if (tempLen >= 20 * sampleRate / 1000) { // tempLen >= 20ms
sampleRate_30);
tempdata += sampleRate_30;
tempLen -= sampleRate_30;
} else if (tempLen >= sampleRate_20) { // tempLen >= 20ms
localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
20 * sampleRate / 1000);
tempdata += 20 * sampleRate / 1000;
tempLen -= 20 * sampleRate / 1000;
sampleRate_20);
tempdata += sampleRate_20;
tempLen -= sampleRate_20;
} else { // use 10ms
localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
10 * sampleRate / 1000);
tempdata += 10 * sampleRate / 1000;
tempLen -= 10 * sampleRate / 1000;
sampleRate_10);
tempdata += sampleRate_10;
tempLen -= sampleRate_10;
}
}