| /prebuilts/gcc/linux-x86/host/x86_64-linux-glibc2.7-4.6/sysroot/usr/include/sound/ |
| pcm_oss.h | 44 int rate; /* requested rate */ member in struct:snd_pcm_oss_runtime
|
| /system/extras/sound/ |
| playwav.c | 36 int pcm_play(unsigned rate, unsigned channels,
58 config.sample_rate = rate;
135 void play_file(unsigned rate, unsigned channels,
148 pcm_play(rate, channels, fill_buffer, 0);
154 unsigned rate, channels; local
191 int wav_rec(const char *fn, unsigned channels, unsigned rate)
208 hdr.sample_rate = rate;
334 unsigned rate = 44100; local
347 } else if (!strcmp(argv[0],"-rate")) {
351 fprintf(stderr,"playwav: -rate requires a parameter\n")
[all...] |
| /cts/tests/tests/media/src/android/media/cts/ |
| SoundPoolTest.java | 147 float rate = 1f; local
148 int streamID = mSoundPool.play(sampleID, leftVolume, rightVolume, priority, loop, rate);
151 rate = 1.4f;
152 mSoundPool.setRate(streamID, rate);
162 streamID = mSoundPool.play(sampleID, leftVolume, rightVolume, priority, loop, rate);
|
| /external/iproute2/misc/ |
| ifstat.c | 59 double rate[MAXS]; member in struct:ifstat_ent
109 memset(&n->rate, 0, sizeof(n->rate));
183 unsigned rate; local
194 if (sscanf(p, "%u", &rate) != 1)
196 n->rate[i] = rate;
220 double *rates = n->rate;
228 rates = h1->rate;
291 fprintf(fp, "%8s/%-6s ", "RX Pkts", "Rate");
[all...] |
| nstat.c | 79 double rate; member in struct:nstat_ent
123 double rate; local
134 nr = sscanf(buf, "%s%llu%lg", idbuf, &val, &rate);
138 rate = 0;
146 n->rate = rate;
194 n->rate = 0;
265 if (!dump_zeros && !val && !n->rate)
279 fprintf(fp, "%-32s%-16llu%6.1f\n", n->id, val, n->rate);
303 if (!dump_zeros && !val && !n->rate)
[all...] |
| rtacct.c | 70 double rate[256*4]; member in struct:rtacct_data
133 void format_rate(FILE *fp, double rate)
137 if (rate > 1024*1024) {
138 sprintf(temp, "%uM", (unsigned)rint(rate/(1024*1024)));
140 } else if (rate > 1024) {
141 sprintf(temp, "%uK", (unsigned)rint(rate/1024));
144 fprintf(fp, " %-10u", (unsigned)rate);
186 double *rate; local
192 rate = &kern_db->rate[realm*4]
248 double *rate; local
[all...] |
| /external/libvpx/libvpx/vp8/encoder/ |
| encodemb.c | 204 int rate; member in struct:vp8_token_state
283 tokens[eob][0].rate = 0;
305 rate0 = tokens[next][0].rate;
306 rate1 = tokens[next][1].rate;
330 tokens[i][0].rate = base_bits + (best ? rate1 : rate0);
337 rate0 = tokens[next][0].rate;
338 rate1 = tokens[next][1].rate;
400 tokens[i][1].rate = base_bits + (best ? rate1 : rate0);
420 tokens[next][0].rate += mb->token_costs[type][band][0][t0];
425 tokens[next][1].rate += mb->token_costs[type][band][0][t1]
[all...] |
| encodeframe.c | 1158 int rate; local
1206 int rate; local
[all...] |
| /external/speex/libspeex/ |
| sb_celp.c | 1170 spx_int32_t i=10, rate, target; local
1221 spx_int32_t rate, target; local
[all...] |
| nb_celp.c | 494 spx_int32_t rate; local
495 speex_encoder_ctl(state, SPEEX_GET_BITRATE, &rate);
496 if (rate > st->vbr_max)
498 rate = st->vbr_max;
499 speex_encoder_ctl(state, SPEEX_SET_BITRATE, &rate);
583 /*If we use low bit-rate pitch mode, transmit open-loop pitch*/
766 /* Low bit-rate pitch handling */
1618 spx_int32_t rate, target; local
1672 spx_int32_t rate, target; local
[all...] |
| /external/tremolo/Tremolo/ |
| codec_internal.h | 97 long rate; member in struct:__anon15862
|
| /external/webkit/Source/WebCore/platform/graphics/openvg/ |
| PathOpenVG.cpp | 255 double rate = span / d01; local
256 FloatPoint startPoint = FloatPoint(point1.x() + v01.width() * rate,
257 point1.y() + v01.height() * rate);
258 rate = span / d21;
259 FloatPoint endPoint = FloatPoint(point1.x() + v21.width() * rate,
260 point1.y() + v21.height() * rate);
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| bandwidth_estimator.c | 45 /* Received header rate. First value is for 30 ms packets and second for 60 ms */
125 * This function updates bottle neck rate received from other side in payload
234 /* store far-side transmission rate */
370 /* don't allow it to be less than frame rate - 10 ms */
375 /* compute inverse receiving rate for last packet, in Q19 */
396 /* Limit inv rate. Note that minBwInv > maxBwInv! */
403 /* update bottle neck rate estimate */
436 /* Q9 (only shift arrTimeDiff by 5 to simulate divide by 16 (need to revisit if change sampling rate) DH */
489 /* Limit to minimum or maximum bottle neck rate (in Q30) */
500 /* store far-side transmission rate */
610 WebRtc_Word32 rate; local
[all...] |
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
| bandwidth_estimator.c | 158 // We have to adjust the header-rate if the first packet has a
167 /* compute far-side transmission rate */
183 /* store far-side transmission rate */
368 /* don't allow it to be less than frame rate - 10 ms */
372 /* compute inverse receiving rate for last packet */
380 // don't allow inv rate to be larger than MAX
385 /* update bottle neck rate estimate */
435 /* limit minimum bottle neck rate */
454 /* store far-side transmission rate */
503 /* This function updates the send bottle neck rate */
602 float rate; local
[all...] |
| /external/webrtc/src/modules/audio_processing/test/ |
| unit_test.cc | 484 int rate[] = {16000, 44100, 48000}; local
485 for (size_t i = 0; i < sizeof(rate)/sizeof(*rate); i++) {
487 apm_->echo_cancellation()->set_device_sample_rate_hz(rate[i]));
488 EXPECT_EQ(rate[i],
[all...] |
| /frameworks/av/media/libmedia/ |
| SoundPool.cpp | 242 int priority, int loop, float rate)
244 ALOGV("play sampleID=%d, leftVolume=%f, rightVolume=%f, priority=%d, loop=%d, rate=%f",
245 sampleID, leftVolume, rightVolume, priority, loop, rate);
276 channel->play(sample, channelID, leftVolume, rightVolume, priority, loop, rate);
403 void SoundPool::setRate(int channelID, float rate)
405 ALOGV("setRate(%d, %f)", channelID, rate);
409 channel->setRate(rate);
518 ALOGE("Sample rate (%u) out of range", sampleRate);
548 float rightVolume, int priority, int loop, float rate)
558 " priority=%d, loop=%d, rate=%f"
662 float rate; local
[all...] |
| /hardware/qcom/audio/legacy/libalsa-intf/ |
| aplay.c | 59 {"Rate", 1, 0, 'R'},
90 unsigned int requestedRate = pcm->rate;
115 param_set_int(params, SNDRV_PCM_HW_PARAM_RATE, pcm->rate);
164 static int play_file(unsigned rate, unsigned channels, int fd,
235 pcm->rate = rate;
472 int play_raw(const char *fg, int rate, int ch, const char *device, const char *fn)
495 return play_file(rate, ch, fd, flag, device, 0);
498 int play_wav(const char *fg, int rate, int ch, const char *device, const char *fn)
516 hdr.sample_rate = rate;
564 int rate = 44100; local
[all...] |
| arec.c | 63 {"Rate", 1, 0, 'R'},
94 unsigned int requestedRate = pcm->rate;
118 param_set_int(params, SNDRV_PCM_HW_PARAM_RATE, pcm->rate);
178 int record_file(unsigned rate, unsigned channels, int fd, unsigned count, unsigned flags, const char *device)
208 pcm->rate = rate;
387 int rec_raw(const char *fg, const char *device, int rate, int ch,
408 count = rate * ch * 2;
414 ch, rate, 16, format);
421 return record_file(rate, ch, fd, count, flag, device)
513 int rate = 48000; local
[all...] |
| /external/esd/include/ |
| esd.h | 22 /* default sample rate for the EsounD server */
65 ESD_PROTO_SERVER_INFO, /* get server info (ver, sample rate, format) */
129 /* rate, format = (bits | channels | stream | func) */
145 int esd_play_stream( esd_format_t format, int rate,
147 int esd_play_stream_fallback( esd_format_t format, int rate,
149 int esd_monitor_stream( esd_format_t format, int rate,
151 /* int esd_monitor_stream_fallback( esd_format_t format, int rate ); */
152 int esd_record_stream( esd_format_t format, int rate,
154 int esd_record_stream_fallback( esd_format_t format, int rate,
156 int esd_filter_stream( esd_format_t format, int rate,
201 int rate; \/* sample rate *\/ member in struct:esd_server_info
212 int rate; \/* sample rate *\/ member in struct:esd_player_info
227 int rate; \/* sample rate *\/ member in struct:esd_sample_info
[all...] |
| /external/qemu/audio/ |
| audio_int.h | 116 void *rate; member in struct:SWVoiceOut
132 void *rate; member in struct:SWVoiceIn
|
| /external/quake/quake/src/QW/server/ |
| sv_user.c | 1007 int rate; local
1011 SV_ClientPrintf (host_client, PRINT_HIGH, "Current rate is %i\n",
1012 (int)(1.0/host_client->netchan.rate + 0.5));
1016 rate = atoi(Cmd_Argv(1));
1017 if (rate < 500)
1018 rate = 500;
1019 if (rate > 10000)
1020 rate = 10000;
1022 SV_ClientPrintf (host_client, PRINT_HIGH, "Net rate set to %i\n", rate);
[all...] |
| /external/webkit/Source/WebCore/platform/graphics/wince/ |
| SharedBitmap.cpp | 426 double rate = static_cast<double>(bmpWidth) / origSourceSize.width(); local
427 double temp = tileRect.width() * rate;
428 tileRect.setX(tileRect.x() * rate);
430 temp = tileRect.height() * rate;
431 tileRect.setY(tileRect.y() * rate);
|
| /external/wpa_supplicant_8/wpa_supplicant/ |
| bss.c | 1090 * wpa_bss_get_max_rate - Get maximum legacy TX rate supported in a BSS
1092 * Returns: Maximum legacy rate in units of 500 kbps
1096 int rate = 0; local
1102 if ((ie[i + 2] & 0x7f) > rate)
1103 rate = ie[i + 2] & 0x7f;
1108 if ((ie[i + 2] & 0x7f) > rate)
1109 rate = ie[i + 2] & 0x7f;
1112 return rate;
|
| scan.c | 1214 int rate = 0; local
1220 if ((ie[i + 2] & 0x7f) > rate)
1221 rate = ie[i + 2] & 0x7f;
1226 if ((ie[i + 2] & 0x7f) > rate)
1227 rate = ie[i + 2] & 0x7f;
1230 return rate;
[all...] |
| /frameworks/av/include/media/ |
| SoundPool.h | 97 float rightVolume, int priority, int loop, float rate);
104 float rate() { return mRate; } function in class:android::SoundEvent
126 int priority, int loop, float rate);
135 void setRate(float rate);
173 int loop, float rate);
182 void setRate(int channelID, float rate);
|
