/system/bt/audio_a2dp_hw/ |
audio_a2dp_hw.h | 47 // sample rate rather than being constant. 49 // FIXME: The BT HAL should consume data at a constant rate. 50 // AudioFlinger assumes that the HAL draws data at a constant rate, which is true 51 // for most audio devices; however, the BT engine reads data at a variable rate 53 // which deliver data at a (generally) fixed rate.
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/cts/tests/tests/media/src/android/media/cts/ |
IvfWriter.java | 41 * Timebase fraction is in format scale/rate, e.g. 1/1000 49 * @param rate timebase rate (or denominator of the timebase fraction) 53 int scale, int rate) throws IOException { 58 mRate = rate; 102 * Timebase fraction is in format scale/rate, e.g. 1/1000 108 * @param rate timebase rate (or denominator of the timebase fraction) 110 private static byte[] makeIvfHeader(int frameCount, int width, int height, int scale, int rate){ 124 lay32Bits(ivfHeader, 16, rate); // scale/rat [all...] |
/external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/boxes/mp4/samplegrouping/ |
RateShareEntry.java | 31 * which defines a record of rate-share information. Typically the same rate-share information applies to many 35 * The grouping type 'rash' (short for rate share) is defined as the grouping criterion for rate share information. 39 * Target rate share may be specified for several operation points that are defined in terms of the total available 40 * bitrate, i.e., the bitrate that should be shared. If only one operation point is defined, the target rate share 42 * target rate share. Target rate share values specified for the first and the last operation points also specify the 43 * target rate share values at lower and higher available bitrates, respectively. The target rate share between tw [all...] |
/external/opencv3/3rdparty/include/ffmpeg_/libavutil/ |
timecode.h | 44 AVRational rate; ///< frame rate in rational form member in struct:__anon20637 45 unsigned fps; ///< frame per second; must be consistent with the rate field 113 * @param rate frame rate in rational form 119 int av_timecode_init(AVTimecode *tc, AVRational rate, int flags, int frame_start, void *log_ctx); 127 * @param rate frame rate in rational form 131 int av_timecode_init_from_string(AVTimecode *tc, AVRational rate, const char *str, void *log_ctx); 134 * Check if the timecode feature is available for the given frame rate [all...] |
/frameworks/base/docs/html/ndk/guides/audio/ |
input-latency.jd | 77 even when both sides use the same sample rate. Your application should buffer the data with 80 <li>Don't assume that the actual sample rate exactly matches the nominal sample rate. For 81 example, if the nominal sample rate is 48,000 Hz, it is normal for the audio clock to advance 82 at a slightly different rate than the operating system {@code CLOCK_MONOTONIC}. This is because 85 <li>Don't assume that the actual playback sample rate exactly matches the actual capture sample 86 rate, especially if the endpoints are on separate paths. For example, if you are capturing from 87 the on-device microphone at 48,000 Hz nominal sample rate, and playing on USB audio 88 at 48,000 Hz nominal sample rate, the actual sample rates are likely to be slightly different 92 <p>A consequence of potentially independent audio clocks is the need for asynchronous sample rate [all...] |
/frameworks/base/media/java/android/media/ |
EncoderCapabilities.java | 38 * <li>Bit rate: the compressed output bit rate in bits per second; 39 * <li>Frame rate: the output number of frames per second. 46 public final int mMinBitRate, mMaxBitRate; // min and max bit rate (bps) 47 public final int mMinFrameRate, mMaxFrameRate; // min and max frame rate (fps) 74 * <li>Bit rate: the compressed output bit rate in bits per second; 75 * <li>Sample rate: the sampling rate used for recording the audio in samples per second; 84 public final int mMinSampleRate, mMaxSampleRate; // min and max sample rate (hz [all...] |
/external/libopus/silk/ |
control.h | 53 /* I: Input signal sampling rate in Hertz; 8000/12000/16000/24000/32000/44100/48000 */ 56 /* I: Maximum internal sampling rate in Hertz; 8000/12000/16000 */ 59 /* I: Minimum internal sampling rate in Hertz; 8000/12000/16000 */ 62 /* I: Soft request for internal sampling rate in Hertz; 8000/12000/16000 */ 98 /* O: Internal sampling rate used, in Hertz; 8000/12000/16000 */ 125 /* I: Output signal sampling rate in Hertz; 8000/12000/16000/24000/32000/44100/48000 */ 128 /* I: Internal sampling rate used, in Hertz; 8000/12000/16000 */
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/hardware/intel/img/hwcomposer/merrifield/common/observers/ |
SoftVsyncObserver.cpp | 84 void SoftVsyncObserver::setRefreshRate(int rate) 87 WTRACE("too late to set refresh rate"); 88 } else if (rate < 1 || rate > 120) { 89 WTRACE("invalid refresh rate %d", rate); 91 mRefreshRate = rate;
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/hardware/intel/img/hwcomposer/moorefield_hdmi/common/observers/ |
SoftVsyncObserver.cpp | 80 void SoftVsyncObserver::setRefreshRate(int rate) 83 WLOGTRACE("too late to set refresh rate"); 84 } else if (rate < 1 || rate > 120) { 85 WLOGTRACE("invalid refresh rate %d", rate); 87 mRefreshRate = rate;
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/system/media/alsa_utils/ |
alsa_device_profile.c | 52 The order here determines the default sample rate for the device. 56 TODO: remove 32000, 22050, 12000, 11025? Each sample rate check 108 * Returns the system defined minimum period size based on the supplied sample rate. 112 ALOGV("profile_calc_min_period_size(%p, rate:%d)", profile, sample_rate); 129 ALOGV("profile_get_period_size(rate:%d) = %d", sample_rate, period_size); 134 * Sample Rate 139 * TODO this won't be right in general. we should store a preferred rate as we are scanning. 140 * But right now it will return the highest rate, which may be correct. 145 bool profile_is_sample_rate_valid(alsa_device_profile* profile, unsigned rate) 150 if (profile->sample_rates[index] == rate) { [all...] |
/system/media/audio_utils/include/audio_utils/spdif/ |
FrameScanner.h | 27 * Parse the sample rate and the size of the encoded frame. 59 * @return sample rate of the encoded audio 65 * a sample rate that is a multiple of the encoded sample rate. 67 * @return sample rate multiplier for the SP/DIF PCM data bursts 113 uint32_t mSampleRate; // encoded sample rate 114 uint32_t mRateMultiplier; // SPDIF output data burst rate = msampleRate * mRateMultiplier
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/external/autotest/client/cros/audio/ |
audio_test_data.py | 26 file_type, sample_format, channel, and rate. 31 rate: sampling rate. 63 channel=2, rate=48000) 84 rate_src=self.data_format['rate'], 89 rate_dst=data_format['rate'], 126 each sample being a signed 16-bit integer in little-endian with sampling rate 134 rate=48000)) 140 16-bit integer in little-endian with sampling rate 48000 samples/sec. 147 rate=48000) [all...] |
/prebuilts/gdb/darwin-x86/lib/python2.7/ |
sndhdr.py | 8 - sampling rate (0 if unknown or hard to decode) 17 sampling rate (a frame contains a sample for each channel). 90 rate = f(h[16:20]) 103 return type, rate, nchannels, data_size//frame_size, sample_bits 121 rate = 0 124 rate = int(1000000.0 / (256 - ratecode)) 125 return 'voc', rate, 1, -1, 8 136 rate = get_long_le(h[24:28]) 138 return 'wav', rate, nchannels, -1, sample_bits 155 rate = get_short_le(h[20:22] [all...] |
/prebuilts/gdb/linux-x86/lib/python2.7/ |
sndhdr.py | 8 - sampling rate (0 if unknown or hard to decode) 17 sampling rate (a frame contains a sample for each channel). 90 rate = f(h[16:20]) 103 return type, rate, nchannels, data_size//frame_size, sample_bits 121 rate = 0 124 rate = int(1000000.0 / (256 - ratecode)) 125 return 'voc', rate, 1, -1, 8 136 rate = get_long_le(h[24:28]) 138 return 'wav', rate, nchannels, -1, sample_bits 155 rate = get_short_le(h[20:22] [all...] |
/prebuilts/python/darwin-x86/2.7.5/lib/python2.7/ |
sndhdr.py | 8 - sampling rate (0 if unknown or hard to decode) 17 sampling rate (a frame contains a sample for each channel). 90 rate = f(h[16:20]) 103 return type, rate, nchannels, data_size//frame_size, sample_bits 121 rate = 0 124 rate = int(1000000.0 / (256 - ratecode)) 125 return 'voc', rate, 1, -1, 8 136 rate = get_long_le(h[24:28]) 138 return 'wav', rate, nchannels, -1, sample_bits 155 rate = get_short_le(h[20:22] [all...] |
/prebuilts/python/linux-x86/2.7.5/lib/python2.7/ |
sndhdr.py | 8 - sampling rate (0 if unknown or hard to decode) 17 sampling rate (a frame contains a sample for each channel). 90 rate = f(h[16:20]) 103 return type, rate, nchannels, data_size//frame_size, sample_bits 121 rate = 0 124 rate = int(1000000.0 / (256 - ratecode)) 125 return 'voc', rate, 1, -1, 8 136 rate = get_long_le(h[24:28]) 138 return 'wav', rate, nchannels, -1, sample_bits 155 rate = get_short_le(h[20:22] [all...] |
/device/google/contexthub/util/nanoapp_cmd/ |
nanoapp_cmd.c | 54 uint32_t rate; member in struct:ConfigCmd 102 cmd->rate = SENSOR_RATE_ONCHANGE; 105 cmd->rate = SENSOR_RATE_ONCHANGE; 108 cmd->rate = SENSOR_RATE_ONCHANGE; 111 cmd->rate = SENSOR_RATE_ONCHANGE; 114 cmd->rate = SENSOR_RATE_ONCHANGE; 117 cmd->rate = SENSOR_RATE_ONCHANGE; 120 cmd->rate = SENSOR_RATE_ONCHANGE; 123 cmd->rate = SENSOR_RATE_ONCHANGE; 126 cmd->rate = SENSOR_RATE_ONESHOT [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...] |
/prebuilts/gcc/linux-x86/host/x86_64-linux-glibc2.11-4.8/sysroot/usr/include/ |
esd.h | 24 /* default sample rate for the EsounD server */ 67 ESD_PROTO_SERVER_INFO, /* get server info (ver, sample rate, format) */ 131 /* rate, format = (bits | channels | stream | func) */ 147 int esd_play_stream( esd_format_t format, int rate, 149 int esd_play_stream_fallback( esd_format_t format, int rate, 151 int esd_monitor_stream( esd_format_t format, int rate, 153 /* int esd_monitor_stream_fallback( esd_format_t format, int rate ); */ 154 int esd_record_stream( esd_format_t format, int rate, 156 int esd_record_stream_fallback( esd_format_t format, int rate, 158 int esd_filter_stream( esd_format_t format, int rate, 203 int rate; \/* sample rate *\/ member in struct:esd_server_info 214 int rate; \/* sample rate *\/ member in struct:esd_player_info 229 int rate; \/* sample rate *\/ member in struct:esd_sample_info [all...] |
/prebuilts/gcc/linux-x86/host/x86_64-linux-glibc2.15-4.8/sysroot/usr/include/ |
esd.h | 24 /* default sample rate for the EsounD server */ 67 ESD_PROTO_SERVER_INFO, /* get server info (ver, sample rate, format) */ 131 /* rate, format = (bits | channels | stream | func) */ 147 int esd_play_stream( esd_format_t format, int rate, 149 int esd_play_stream_fallback( esd_format_t format, int rate, 151 int esd_monitor_stream( esd_format_t format, int rate, 153 /* int esd_monitor_stream_fallback( esd_format_t format, int rate ); */ 154 int esd_record_stream( esd_format_t format, int rate, 156 int esd_record_stream_fallback( esd_format_t format, int rate, 158 int esd_filter_stream( esd_format_t format, int rate, 203 int rate; \/* sample rate *\/ member in struct:esd_server_info 214 int rate; \/* sample rate *\/ member in struct:esd_player_info 229 int rate; \/* sample rate *\/ member in struct:esd_sample_info [all...] |
/frameworks/wilhelm/tests/mimeUri/ |
slesTestSlowDownUri.cpp | 71 SLpermille minRate, maxRate, stepSize, rate = 1000; local 75 SLresult res = (*pRateItf)->GetRate(pRateItf, &rate); CheckErr(res); 78 fprintf(stdout, "old rate = %d, minRate=%d, maxRate=%d\n", rate, minRate, maxRate); 79 rate /= 2; 80 if (rate < minRate) { 81 rate = minRate; 83 fprintf(stdout, "new rate = %d\n", rate); 84 res = (*pRateItf)->SetRate(pRateItf, rate); CheckErr(res) 290 SLpermille rate = 1234; local [all...] |
/external/llvm/lib/Target/PowerPC/ |
PPCScheduleE500mc.td | 84 [7, 1, 1], // Latency = 4, Repeat rate = 1 89 [7, 1, 1], // Latency = 4, Repeat rate = 1 94 [7, 1, 1], // Latency = 4, Repeat rate = 1 109 [5, 1], // Latency = 2, Repeat rate = 2 130 [6, 1], // Latency = 3, Repeat rate = 1 221 [6, 1, 1], // Latency = 3, Repeat rate = 3 236 [5, 1], // Latency = 2, Repeat rate = 4 254 [7, 1], // Latency = 4, Repeat rate = 4 258 [4, 1], // Latency = 1, Repeat rate = 1 262 [7, 1], // Latency = 4, Repeat rate = [all...] |
/external/webrtc/webrtc/modules/audio_coding/acm2/ |
acm_codec_database.cc | 33 bool IsISACRateValid(int rate) { 34 return (rate == -1) || ((rate <= 56000) && (rate >= 10000)); 38 bool IsILBCRateValid(int rate, int frame_size_samples) { 40 (rate == 13300)) { 43 (rate == 15200)) { 51 bool IsOpusRateValid(int rate) { 52 return (rate >= 6000) && (rate <= 510000) [all...] |
/external/iproute2/examples/diffserv/ |
Edge32-cb-chains | 26 meter1="police rate $CIR1 burst $CBS1 " 27 meter1a="police rate $CIR2 burst $CBS1 " 28 meter2="police rate $CIR1 burst $CBS2 " 29 meter2a="police rate $CIR2 burst $CBS2 " 30 meter3="police rate $CIR2 burst $CBS1 " 31 meter3a="police rate $CIR2 burst $CBS1 " 32 meter4="police rate $CIR2 burst $CBS2 " 33 meter5="police rate $CIR1 burst $CBS2 " 53 #if it doesnt exceed its allocated rate (CIR/CBS) 62 # if it exceeds the above but not the extra rate/burst below, it gets a [all...] |
Edge32-cb-u32 | 21 # going to the PIR --> average rate is CIR+PIR) 32 meter1=" police rate $CIR1 burst $CBS1 " 33 meter1a=" police rate $PIR1 burst $EBS1 " 34 meter2=" police rate $CIR2 burst $CBS1 " 35 meter2a="police rate $PIR2 burst $CBS1 " 36 meter3=" police rate $CIR2 burst $CBS2 " 37 meter3a=" police rate $PIR2 burst $EBS2 " 38 meter4=" police rate $CIR1 burst $CBS2 " 39 meter5=" police rate $CIR1 burst $CBS2 " 63 # if it exceeds the above but not the extra rate/burst below, it gets a [all...] |