| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g711/test/ |
| testG711.cc | 57 int16_t stream_len = 0; local
130 stream_len = WebRtcG711_EncodeA(NULL, shortdata, framelength, streamdata);
133 if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
134 static_cast<size_t>(stream_len)) {
138 err = WebRtcG711_DecodeA(NULL, streamdata, stream_len, decoded,
142 stream_len = WebRtcG711_EncodeU(NULL, shortdata, framelength, streamdata);
145 if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
146 static_cast<size_t>(stream_len)) {
150 err = WebRtcG711_DecodeU(NULL, streamdata, stream_len, decoded,
156 if (stream_len < 0 || err < 0)
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g722/test/ |
| testG722.cc | 62 int16_t stream_len = 0; local
126 stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
127 err = WebRtcG722_Decode((G722DecInst *)G722dec_inst, streamdata, stream_len, decoded, speechType);
132 if (stream_len < 0 || err < 0) {
137 if (fwrite(streamdata, sizeof(short), stream_len/2,
138 outbitp) != static_cast<size_t>(stream_len/2)) {
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| isacfix.c | 362 int16_t stream_len, stream_len_even; local
377 stream_len = WebRtcIsacfix_EncodeImpl((int16_t*)speechIn,
381 if (stream_len<0) {
382 ISAC_inst->errorcode = - stream_len;
389 stream_len_even = stream_len % 2 == 0 ? stream_len : stream_len + 1;
408 return stream_len;
445 int16_t stream_len; local
475 stream_len = WebRtcIsacfix_EncodeImpl((int16_t*)speechInWB
528 int16_t stream_len; local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
| test_iSACfixfloat.c | 112 int16_t stream_len = 0; local
442 stream_len = WebRtcIsac_Encode(ISAC_main_inst,
445 if (stream_len < 0) {
454 stream_len = WebRtcIsacfix_Encode(
459 stream_len = WebRtcIsacfix_EncodeNb(ISACFIX_main_inst, shortdata, streamdata);
461 if (stream_len < 0) {
487 if (stream_len != 0) break;
490 fwrite(streamdata, 1, stream_len, bitsp); /* NOTE! Writes bytes to file */
493 get_arrival_time(cur_framesmpls, stream_len, bottleneck,
500 stream_len,
[all...] |
| kenny.cc | 110 int16_t stream_len = 0; local
566 stream_len = WebRtcIsacfix_Encode(ISAC_main_inst,
572 if (stream_len>0) {
575 stream_len = WebRtcIsacfix_GetNewBitStream(
604 stream_len = WebRtcIsacfix_EncodeNb(ISAC_main_inst,
608 stream_len = -1;
617 if (stream_len < 0 || err < 0) {
623 stream_len, outbits) != (size_t)stream_len) {
647 if (stream_len != 0) break
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ |
| ReleaseTest-API.cc | 59 int16_t stream_len = 0; local
664 stream_len = WebRtcIsac_Encode(ISAC_main_inst,
667 if((payloadSize != 0) && (stream_len > payloadSize))
674 printf("\nError: Streamsize out of range %d\n", stream_len - payloadSize);
680 if(stream_len>0)
732 if(stream_len < 0)
741 if(stream_len != 0) break;
764 maxStreamLen30 = (stream_len > maxStreamLen30)? stream_len:maxStreamLen30;
768 maxStreamLen60 = (stream_len > maxStreamLen60)? stream_len:maxStreamLen60
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/test/ |
| simpleKenny.c | 76 int16_t stream_len = 0; local
339 stream_len = 0;
350 stream_len = ((uint8_t)auxUW8) << 8;
355 stream_len |= (uint16_t)auxUW8;
356 auxSizet = (size_t)stream_len;
365 while(stream_len == 0)
376 stream_len = WebRtcIsac_Encode(
381 if(stream_len < 0)
400 get_arrival_time(cur_framesmpls, stream_len, bottleneck, &packetData,
403 payload, stream_len, packetData.rtp_number
[all...] |
