Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .
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}
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
index 68dd7da..34ff8aa 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
@@ -28,7 +28,7 @@
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
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c b/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
index ecdd68a..b313b58 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
@@ -70,7 +70,8 @@
/* 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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
index f0ae07e..d0c59d6 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
@@ -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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index bdb807e..43a9e52 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -777,7 +777,7 @@
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 @@
/* 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 @@
/* 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 @@
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 @@
/* 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 @@
/* 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 @@
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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
index 7fcb9e3..13858d7 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
@@ -218,7 +218,7 @@
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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S b/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
index 4a0d99f..945d6ee 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
@@ -38,7 +38,7 @@
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 @@
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]
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
index ab7c640..2628f1f 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -624,8 +624,8 @@
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 @@
/* 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 @@
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);
}
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
index 6dbdb7e..71bd272 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
@@ -556,12 +556,13 @@
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 @@
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);
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
index 7a51a1e..ac211e9 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -750,7 +750,8 @@
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);
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index 4eeeed0..8584c76 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -662,8 +662,9 @@
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 */
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index e0d0f41..214dccd 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -350,6 +350,11 @@
}
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);
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 88e084f..9bf1ae3 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -115,7 +115,7 @@
// 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;
diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.cc b/webrtc/modules/audio_coding/main/test/PCMFile.cc
index 4f46098..4b08f75 100644
--- a/webrtc/modules/audio_coding/main/test/PCMFile.cc
+++ b/webrtc/modules/audio_coding/main/test/PCMFile.cc
@@ -150,8 +150,7 @@
}
} 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];
}
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 99ff95a..6a9b953 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -256,16 +256,14 @@
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);
diff --git a/webrtc/modules/audio_coding/neteq/dsp_helper.cc b/webrtc/modules/audio_coding/neteq/dsp_helper.cc
index 289e66d..3e5c61d 100644
--- a/webrtc/modules/audio_coding/neteq/dsp_helper.cc
+++ b/webrtc/modules/audio_coding/neteq/dsp_helper.cc
@@ -117,7 +117,7 @@
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 @@
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;
diff --git a/webrtc/modules/audio_coding/neteq/expand.cc b/webrtc/modules/audio_coding/neteq/expand.cc
index 10f6a9f..ae12e50 100644
--- a/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/webrtc/modules/audio_coding/neteq/expand.cc
@@ -441,8 +441,8 @@
&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|.
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.cc b/webrtc/modules/audio_coding/neteq/packet_buffer.cc
index 5792b22..07fa339 100644
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.cc
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.cc
@@ -255,7 +255,7 @@
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.
}
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index 1aacb40..7e778b8 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -454,7 +454,7 @@
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 @@
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;
}
}