| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g711/test/ |
| testG711.cc | 48 int16_t framelength = 80; local
72 printf("./testG711.exe framelength law infile outfile \n\n");
73 printf("framelength: Framelength in samples.\n");
88 framelength = atoi(argv[1]);
125 endfile = readframe(shortdata, inp, framelength);
130 stream_len = WebRtcG711_EncodeA(NULL, shortdata, framelength, streamdata);
142 stream_len = WebRtcG711_EncodeU(NULL, shortdata, framelength, streamdata);
161 if (fwrite(decoded, sizeof(short), framelength, outp) !=
162 static_cast<size_t>(framelength)) {
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g722/test/ |
| testG722.cc | 52 int16_t framelength = 160; local
74 printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
76 printf("framelength : Framelength in samples.\n\n");
85 framelength = atoi(argv[1]);
120 endfile = readframe(shortdata, inp, framelength);
126 stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
142 if (fwrite(decoded, sizeof(short), framelength,
143 outp) != static_cast<size_t>(framelength)) {
152 length_file = ((double)framecnt*(double)framelength/16000)
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| entropy_coding.h | 190 int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, int16_t* framelength);
191 int WebRtcIsac_EncodeFrameLen(int16_t framelength, Bitstr* streamdata);
|
| structs.h | 261 estimate. All values (except framelength and minBytes) is double size to
270 int16_t framelength; member in struct:__anon20397
|
| encode.c | 217 /* Set the framelength for the next packet. */
271 /* Save framelength for multiple packets memory. */
272 ISACencLB_obj->SaveEnc_obj.framelength =
[all...] |
| /external/aac/libAACenc/src/ |
| aacenc.cpp | 111 INT framelength,
119 INT frameLength,
131 while ( (frameLength & ~((1<<(shift+1))-1)) == frameLength
139 averageBitsPerFrame = (bitRate*(frameLength>>shift)) / (coreSamplingRate>>shift) / nSubFrames;
152 bitRate = FDKmax(bitRate, ((((40 * nChannels) + transportBits) * (coreSamplingRate)) / frameLength) );
155 bitRate = FDKmin(bitRate, ((nChannelsEff * MIN_BUFSIZE_PER_EFF_CHAN)*(coreSamplingRate>>shift)) / (frameLength>>shift)) ;
283 config->framelength = -1; /* Framesize not configured */
430 config->framelength,
451 switch (config->framelength)
[all...] |
| aacenc.h | 196 INT framelength; /* used frame size */ member in struct:AACENC_CONFIG
229 * \param frameLength the frameLength to be used for the AAC encoder
242 INT frameLength,
|
| aacenc_lib.cpp | 505 cc->samplesPerFrame = hAacConfig->framelength;
645 const INT frameLength,
682 frameLength,
800 hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 1024;
801 if (hAacConfig->framelength != 1024) {
812 hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 512;
813 if (hAacConfig->framelength != 512 && hAacConfig->framelength != 480) {
825 hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 512;
826 if (hAacConfig->framelength != 512 && hAacConfig->framelength != 480)
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| entropy_coding.h | 95 int16_t *framelength);
98 int WebRtcIsacfix_EncodeFrameLen(int16_t framelength,
|
| structs.h | 271 estimate. All values (except framelength and minBytes) is double size to
280 int16_t framelength; member in struct:__anon20379
|
| encode.c | 71 /* set the framelength for the next packet */
78 /* buffer speech samples (by 10ms packet) until the framelength is reached (30 or 60 ms) */
135 /* Save framelength for multiple packets memory */
137 (ISACenc_obj->SaveEnc_ptr)->framelength=ISACenc_obj->current_framesamples;
411 /* also update the framelength for next packet, in Adaptive mode only */
529 status = WebRtcIsacfix_EncodeFrameLen(SaveEnc_str->framelength, &ISACenc_obj->bitstr_obj);
|
