| /external/aac/libSACenc/src/ |
| sacenc_dmx_tdom_enh.h | 119 HANDLE_ENHANCED_TIME_DOMAIN_DMX *hEnhancedTimeDmx, const INT framelength);
124 const FIXP_DBL outputGain_m, const INT outputGain_e, const INT framelength);
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| sacenc_dmx_tdom_enh.cpp | 126 int framelength; member in struct:T_ENHANCED_TIME_DOMAIN_DMX
244 HANDLE_ENHANCED_TIME_DOMAIN_DMX *phEnhancedTimeDmx, const INT framelength) {
252 FDK_ALLOCATE_MEMORY_1D(hEnhancedTimeDmx->sinusWindow_m, 1 + framelength,
254 hEnhancedTimeDmx->maxFramelength = framelength;
268 const INT framelength) {
275 if (framelength > hEnhancedTimeDmx->maxFramelength) {
280 hEnhancedTimeDmx->framelength = framelength;
286 PI_M, (FIXP_DBL)(2 * hEnhancedTimeDmx->framelength), &deltax_e);
290 for (smp = 0; smp < hEnhancedTimeDmx->framelength + 1; smp++)
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/test/ |
| testG711.cc | 43 size_t framelength = 80; local
65 printf("./testG711.exe framelength law infile outfile \n\n");
66 printf("framelength: Framelength in samples.\n");
83 printf(" G.722: Invalid framelength %d.\n", framelength_int);
86 framelength = static_cast<size_t>(framelength_int);
123 endfile = readframe(shortdata, inp, framelength);
128 stream_len = WebRtcG711_EncodeA(shortdata, framelength, streamdata);
139 stream_len = WebRtcG711_EncodeU(shortdata, framelength, streamdata);
153 if (fwrite(decoded, sizeof(short), framelength, outp) != framelength)
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/test/ |
| testG722.cc | 48 size_t framelength = 160; local
69 printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
71 printf("framelength : Framelength in samples.\n\n");
82 printf(" G.722: Invalid framelength %d.\n", framelength_int);
85 framelength = static_cast<size_t>(framelength_int);
120 endfile = readframe(shortdata, inp, framelength);
126 stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
139 if (fwrite(decoded, sizeof(short), framelength, outp) !=
140 framelength) {
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| entropy_coding.h | 192 int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, int16_t* framelength);
193 int WebRtcIsac_EncodeFrameLen(int16_t framelength, Bitstr* streamdata);
|
| structs.h | 264 estimate. All values (except framelength and minBytes) is double size to
273 int16_t framelength; member in struct:__anon44071
|
| encode.c | 218 /* Set the framelength for the next packet. */
272 /* Save framelength for multiple packets memory. */
273 ISACencLB_obj->SaveEnc_obj.framelength =
[all...] |
| /external/aac/libAACenc/src/ |
| mps_main.h | 166 * \param framelength Number of samples to be processes within one
178 const UINT sbrRatio, const UINT framelength,
|
| aacenc.cpp | 122 INT FDKaacEnc_CalcBitsPerFrame(const INT bitRate, const INT frameLength,
125 while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength &&
130 return (bitRate * (frameLength >> shift)) / (samplingRate >> shift);
133 INT FDKaacEnc_CalcBitrate(const INT bitsPerFrame, const INT frameLength,
136 while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength &&
141 return (bitsPerFrame * (samplingRate >> shift)) / (frameLength >> shift);
145 INT bitRate, INT framelength, INT ancillaryRate, INT *ancillaryBitsPerFrame,
149 INT coreSamplingRate, INT frameLength, INT nChannels
[all...] |
| aacenc_lib.cpp | 468 cc->samplesPerFrame = hAacConfig->framelength;
711 const INT frameLength, const INT nChannels,
731 bitRate = FDKaacEnc_LimitBitrate(hTpEnc, aot, coreSamplingRate, frameLength,
915 hAacConfig->framelength = (config->userFramelength != (UINT)-1)
918 if (hAacConfig->framelength != 1024 && hAacConfig->framelength != 960) {
[all...] |
| mps_main.cpp | 218 const UINT sbrRatio, const UINT framelength,
255 nTimeSlots = framelength >> nQmfBandsLd;
257 /* check if number of qmf bands is usable for given framelength */
258 if (framelength != (nTimeSlots << nQmfBandsLd)) {
|
| aacenc.h | 232 INT framelength; /* used frame size */ member in struct:AACENC_CONFIG
272 * \param frameLength Number of audio samples in one frame.
277 INT FDKaacEnc_CalcBitsPerFrame(const INT bitRate, const INT frameLength,
285 * \param frameLength Number of audio samples in one frame.
290 INT FDKaacEnc_CalcBitrate(const INT bitsPerFrame, const INT frameLength,
298 * \param frameLength the frameLength to be used for the AAC encoder
312 INT coreSamplingRate, INT frameLength, INT nChannels,
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| entropy_coding.h | 95 size_t *framelength);
98 int WebRtcIsacfix_EncodeFrameLen(int16_t framelength,
|
| structs.h | 270 estimate. All values (except framelength and minBytes) is double size to
279 int16_t framelength; member in struct:__anon44052
|
| 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) */
136 /* Save framelength for multiple packets memory */
138 (ISACenc_obj->SaveEnc_ptr)->framelength=ISACenc_obj->current_framesamples;
413 /* also update the framelength for next packet, in Adaptive mode only */
532 status = WebRtcIsacfix_EncodeFrameLen(SaveEnc_str->framelength, &ISACenc_obj->bitstr_obj);
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| /external/syslinux/gpxe/src/arch/i386/drivers/net/ |
| undiisr.S | 83 framelength: .word 0 label
|
