| /external/ipsec-tools/src/libipsec/ |
| H A D | pfkey.c | 312 * set the rate for SOFT lifetime against HARD one.
313 * If rate is more than 100 or equal to zero, then set to 100.
321 pfkey_set_softrate(type, rate)
322 u_int type, rate;
326 if (rate > 100 || rate == 0)
327 rate = 100;
331 soft_lifetime_allocations_rate = rate;
334 soft_lifetime_bytes_rate = rate;
337 soft_lifetime_addtime_rate = rate;
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_sender.cc | 300 uint32_t rate) {
317 (payload->typeSpecific.Audio.rate == rate ||
318 payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
319 payload->typeSpecific.Audio.rate = rate;
320 // Ensure that we update the rate if new or old is zero.
334 frequency, channels, rate, &payload);
336 payload = video_->CreateVideoPayload(payload_name, payload_number, rate);
295 RegisterPayload( const char payload_name[RTP_PAYLOAD_NAME_SIZE], int8_t payload_number, uint32_t frequency, size_t channels, uint32_t rate) argument
[all...] |
| H A D | rtp_receiver_audio.cc | 125 // - The correct rate is 4 bits/sample.
128 // - encoding sample/frame bits/sample rate ms/frame ms/packet
270 specific_payload.Audio.channels, specific_payload.Audio.rate)) {
|
| /external/webrtc/webrtc/modules/video_coding/ |
| H A D | jitter_buffer.cc | 416 float rate = 0.5f + ((incoming_frame_count_ * 1000.0f) / diff); local
417 if (rate < 1.0f) {
418 rate = 1.0f;
421 // Calculate frame rate
422 // Let r be rate.
427 *framerate = (incoming_frame_rate_ + static_cast<unsigned int>(rate)) / 2;
428 incoming_frame_rate_ = static_cast<unsigned int>(rate);
430 // Calculate bit rate
|
| /external/wpa_supplicant_8/hostapd/src/common/ |
| H A D | ieee802_11_common.c | 1037 static int is_11b(u8 rate) argument
1039 return rate == 0x02 || rate == 0x04 || rate == 0x0b || rate == 0x16;
|
| /external/wpa_supplicant_8/src/common/ |
| H A D | ieee802_11_common.c | 1037 static int is_11b(u8 rate) argument
1039 return rate == 0x02 || rate == 0x04 || rate == 0x0b || rate == 0x16;
|
| /external/wpa_supplicant_8/wpa_supplicant/src/common/ |
| H A D | ieee802_11_common.c | 1037 static int is_11b(u8 rate) argument
1039 return rate == 0x02 || rate == 0x04 || rate == 0x0b || rate == 0x16;
|
| /external/webrtc/webrtc/ |
| H A D | common_types.h | 247 uint32_t packet_rate; // Packet rate in packets per second.
248 uint64_t timestamp_ms; // Ntp timestamp in ms at time of rate estimation.
295 int rate; // bits/sec unlike {start,min,max}Bitrate elsewhere in this file! member in struct:webrtc::CodecInst
303 rate == other.rate;
337 // Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
339 // Late loss rate; fraction between 0 and 1, scaled to Q14.
|
| /external/esd/include/ |
| H A D | audiofile.h | 136 AF_INST_SAMP_RATE = 315, /* sample rate of this inst's sample */
259 AF_BAD_RATE = 14, /* invalid sample rate */
304 AF_WARNING_CODEC_RATE = 60, /* using 8k instead of codec rate 8012 */
305 AF_WARNING_RATECVT = 61, /* warning about rate conversion used */
339 AF_ERR_BAD_RATE = 14+AF_ERR_BASE, /* invalid sample rate */
487 /* track data: sampling rate */
488 void afInitRate (AFfilesetup, int track, double rate);
492 int afSetVirtualRate (AFfilehandle, int track, double rate);
|
| /external/fio/ |
| H A D | thread_options.h | 221 unsigned int rate[DDIR_RWDIR_CNT]; member in struct:thread_options
445 uint32_t rate[DDIR_RWDIR_CNT]; member in struct:thread_options_pack
|
| /external/libopus/ |
| H A D | Android.mk | 24 celt/rate.c \
|
| /external/webrtc/webrtc/voice_engine/ |
| H A D | voe_codec_impl.cc | 68 "channels=%" PRIuS ", rate=%d",
70 codec.channels, codec.rate);
166 "pacsize=%d, rate=%d",
168 codec.pacsize, codec.rate);
|
| /external/speex/libspeex/ |
| H A D | 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 */
1256 /* Get open-loop pitch estimation for low bit-rate pitch coding */
1618 spx_int32_t rate, target; local
1624 speex_encoder_ctl(st, SPEEX_GET_BITRATE, &rate);
1672 spx_int32_t rate, target; local
[all...] |
| /external/autotest/client/cros/chameleon/ |
| H A D | audio_test_utils.py | 288 normalized_signal, data_format['rate'])
310 rate=data_format['rate'],
|
| /external/iproute2/include/linux/ |
| H A D | pkt_cls.h | 61 struct tc_ratespec rate; member in struct:tc_police
|
| /external/kernel-headers/original/uapi/linux/ |
| H A D | pkt_cls.h | 115 struct tc_ratespec rate; member in struct:tc_police
|
| /external/kernel-headers/original/uapi/sound/ |
| H A D | asoc.h | 204 __le32 rate_min; /* min rate */
205 __le32 rate_max; /* max rate */
223 __le32 rate; /* SNDRV_PCM_RATE_* */ member in struct:snd_soc_tplg_stream
|
| /external/libnl/include/linux/ |
| H A D | pkt_cls.h | 134 struct tc_ratespec rate; member in struct:tc_police
|
| /external/pdfium/third_party/libopenjpeg20/ |
| H A D | tcd.h | 70 OPJ_UINT32 rate; member in struct:opj_tcd_pass
|
| /external/sonivox/arm-wt-22k/host_src/ |
| H A D | eas.h | 221 * Set the playback rate.
226 * rate - rate (28-bit fractional amount)
234 EAS_PUBLIC EAS_RESULT EAS_SetPlaybackRate (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_U32 rate);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| H A D | isacfix.c | 276 * - CodingMode : 0 -> Bit rate and frame length are automatically
280 * rate which is taken as the maximum short-term
281 * average bit rate.
1069 * This function sets the limit on the short-term average bit rate and the
1074 * - rate : limit on the short-term average bit rate,
1083 int16_t rate,
1098 if (rate >= 10000 && rate <= 32000)
1099 ISAC_inst->ISACenc_obj.BottleNeck = rate;
1082 WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst, int16_t rate, int framesize) argument
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| H A D | audio_encoder_opus.cc | 30 config.bitrate_bps = codec_inst.rate;
37 // Optimize the loss rate to configure Opus. Basically, optimized loss rate is
38 // the input loss rate rounded down to various levels, because a robustly good
41 // a loss rate from below, a higher threshold is used than jumping to the same
|
| /external/webrtc/webrtc/modules/audio_coding/test/ |
| H A D | utility.cc | 135 codecInst.plfreq, codecInst.rate);
|
| /external/webrtc/webrtc/modules/audio_processing/include/ |
| H A D | mock_audio_processing.h | 194 int(int rate));
|
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| H A D | apmtest.m | 284 % Assume the last init gives the sample rate of interest.
285 str_idx = strfind(result, 'Sample rate:');
|