HomeSort by relevance Sort by last modified time
    Searched full:rate (Results 276 - 300 of 1951) sorted by null

<<11121314151617181920>>

  /external/opencore/codecs_v2/audio/gsm_amr/amr_nb/dec/src/
pvgsmamrdecoder.h 22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec
pvgsmamrdecoder_dpi.h 22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec
  /external/opencore/codecs_v2/audio/gsm_amr/amr_nb/enc/src/
convolve.h 22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec
prm2bits.h 22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec
  /external/opencore/codecs_v2/audio/gsm_amr/amr_wb/dec/include/
decoder_amr_wb.h 22 ANSI-C code for the Adaptive Multi-Rate - Wideband (AMR-WB) speech codec
  /external/opencore/codecs_v2/utilities/pv_config_parser/src/
pv_audio_config_parser.cpp 83 // This routine parses the wma config header and returns Sampling Rate, Number of Channels, and Bits Per Sample
204 // limit to M0-profile bitrate and sampling rate
241 // not a valid sample rate for WMA Std spec
287 // not a valid sample rate for WMA Std spec
  /external/opencore/fileformats/mp4/composer/include/
editlistatom.h 45 void addEditEntry(uint32 duration, int32 time, uint16 rate);
  /external/opencore/nodes/pvmp4ffcomposernode/src/
pvmp4ffcn_port.cpp 738 LOG_DEBUG((0, "PVMp4FFComposerPort::GetInputParametersFromPeer: Sampling rate info not available. Use default"));
749 // Get the sampling rate, number of channels and bits per sample for audio
750 // sampling rate
754 LOG_DEBUG((0, "PVMp4FFComposerPort::GetInputParametersFromPeer: Sampling rate info not available. Use default"));
853 // Get video frame rate from peer
857 LOG_DEBUG((0, "PVMp4FFComposerPort::GetInputParametersFromPeer: Frame rate not available. Use default"));
862 // Set input frame rate of container node
872 LOG_DEBUG((0, "PVMp4FFComposerPort::GetInputParametersFromPeer: Sampling rate info not available. Use default"));
    [all...]
  /external/opencore/protocols/rtp_payload_parser/rfc_3640/include/
rfc3640_payload_parser.h 28 // This implementation currently only supports AAC high-bit-rate (AAChbr).
  /external/opencore/protocols/sdp/common/include/
rfc3640_media_info.h 34 //For now, just default to AAC high bit-rate.
rfc3640_payload_info.h 40 //For now, just default to AAC high bit-rate.
  /external/qemu/distrib/sdl-1.2.12/src/audio/mint/
SDL_mintaudio_gsxb.h 45 #define SETRATE 7 /* Set sample rate */
  /external/quake/quake/src/QW/client/
snd_mem.c 60 // resample / decimate to the current source rate
134 stepscale = (float)info.rate / shm->speed;
145 sc->speed = info.rate;
292 info.rate = GetLittleLong();
  /external/quake/quake/src/QW/server/
newnet.txt 47 rate of extraction
  /external/quake/quake/src/WinQuake/
snd_mem.cpp 60 // resample / decimate to the current source rate
134 stepscale = (float)info.rate / shm->speed;
145 sc->speed = info.rate;
290 info.rate = GetLittleLong();
  /external/svox/pico/tts/
svox_ssml_parser.cpp 322 else if (strcmp(element, "prosody") == 0) /* only pitch, rate and volume attributes are supported */
369 else if (strcmp(attributes[i], "rate") == 0)
382 char* rate = new char[17 + strlen(svoxrate)]; local
383 if (!rate)
388 sprintf(rate, "<speed level='%s'>", svoxrate);
389 if (strlen(m_data) + strlen(rate) + 1 > (size_t)m_datasize)
397 strcat(m_data, rate);
410 delete [] rate;
662 Converts SSML rate labels to SVOX speed levels
  /external/webkit/WebCore/html/
HTMLMediaElement.cpp     [all...]
  /external/webkit/WebCore/platform/graphics/
MediaPlayer.cpp 414 float MediaPlayer::rate() const function in class:WebCore::MediaPlayer
419 void MediaPlayer::setRate(float rate)
421 m_rate = rate;
422 m_private->setRate(rate);
  /external/webkit/WebCore/platform/graphics/mac/
MediaPlayerProxy.h 95 - (void)_setRate:(float)rate;
  /external/webkit/WebCore/platform/graphics/wince/
PlatformPathWince.cpp 676 double rate = span / d01; local
678 startPoint.m_x = p1.m_x + v01.m_x * rate;
679 startPoint.m_y = p1.m_y + v01.m_y * rate;
684 rate = span / d21;
685 endPoint.m_x = p1.m_x + v21.m_x * rate;
686 endPoint.m_y = p1.m_y + v21.m_y * rate;
697 rate = d / dm1;
698 centerPoint.m_x = p1.m_x + vm1.m_x * rate;
699 centerPoint.m_y = p1.m_y + vm1.m_y * rate;
  /external/wpa_supplicant_6/wpa_supplicant/examples/
wpas-test.py 83 print " %s :: ssid='%s' wpa=%s wpa2=%s quality=%d%% rate=%d freq=%d" % (bssid, ssid, wpa, wpa2, qual, maxrate, freq)
  /frameworks/base/core/java/android/view/
WindowOrientationListener.java 56 * @param rate at which sensor events are processed (see also
63 * no way to get the period from SensorManager based on the rate constant.
65 private WindowOrientationListener(Context context, int rate) {
67 mRate = rate;
  /frameworks/base/docs/html/guide/topics/resources/
animation-resource.jd 270 <p>An interpolator is an animation modifier defined in XML that affects the rate of change in an
335 For example, you can adjust the rate of
373 <dd>The rate of change starts and ends slowly but accelerates through the
376 <dd>The rate of change starts out slowly, then accelerates.
380 <dd><em>Float</em>. The acceleration rate (default is 1).</dd>
406 <dd>Repeats the animation for a specified number of cycles. The rate of change follows a
415 <dd>The rate of change starts out quickly, then decelerates.
419 <dd><em>Float</em>. The deceleration rate (default is 1).</dd>
423 <dd>The rate of change is constant. <p>No attributes.</p></dd>
  /frameworks/base/media/java/android/media/
AudioTrack.java 44 * (high sampling rate, bits per sample ...)</li>
180 * The audio data sampling rate in Hz.
240 * @param sampleRateInHz the sample rate expressed in Hertz. Examples of rates are (but
315 // sample rate
318 + "Hz is not a supported sample rate."));
433 * Returns the configured audio data sample rate in Hz
440 * Returns the current playback rate in Hz.
533 * Returns the hardware output sample rate
544 * @param sampleRateInHz the sample rate expressed in Hertz.
579 loge("getMinBufferSize(): " + sampleRateInHz +"Hz is not a supported sample rate.")
    [all...]
  /frameworks/base/media/libstagefright/codecs/amrnb/common/include/
bits2prm.h 22 ANSI-C code for the Adaptive Multi-Rate (AMR) speech codec

Completed in 64 milliseconds

<<11121314151617181920>>