/external/webrtc/webrtc/modules/audio_coding/acm2/ |
audio_coding_module_impl.cc | 111 bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"), 153 input_data.audio, input_data.audio_channel * 273 // Add 10MS of raw (PCM) audio data to the encoder. 286 "Cannot Add 10 ms audio, payload length is zero"); 293 "Cannot Add 10 ms audio, input frequency not valid"); 301 "Cannot Add 10 ms audio, input frequency and length doesn't" 308 "Cannot Add 10 ms audio, invalid number of channels."); 342 // When adding data to encoders this pointer is pointing to an audio buffer 351 input_data->audio = ptr_audio; 361 // |*ptr_out| will point to the pre-processed audio-frame. If no pre-processin 398 int16_t audio[WEBRTC_10MS_PCM_AUDIO]; local [all...] |
audio_coding_module_unittest_oldapi.cc | 77 rtp_header->type.Audio.channel = 1; 78 rtp_header->type.Audio.isCNG = false; 174 "audio frame too small"); 257 // Insert some packets and pull audio. Check statistics are valid. Then, 598 // This is a multi-threaded ACM test using iSAC. The test encodes audio 617 // Set up input audio source to read from specified file, loop after 5 736 // Set up input audio source to read from specified file, loop after 5 810 // Pull audio. [all...] |
/external/webrtc/webrtc/modules/include/ |
module_common_types.h | 251 RTPAudioHeader Audio; 469 /* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
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/packages/apps/Settings/src/com/android/settings/notification/ |
SoundSettings.java | 414 new String[] { MediaStore.Audio.Media.TITLE }, null, null, null);
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/frameworks/base/docs/html/training/ |
training_toc.cs | 358 <a href="<?cs var:toroot ?>training/managing-audio/index.html" 360 "How to respond to hardware audio key presses, request audio focus 361 when playing audio, and respond appropriately to changes in audio focus." 362 >Managing Audio Playback</a> 365 <li><a href="<?cs var:toroot ?>training/managing-audio/volume-playback.html"> 369 <li><a href="<?cs var:toroot ?>training/managing-audio/audio-focus.html"> 370 Managing Audio Focu [all...] |
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/frameworks/base/ |
compiled-classes-phone | [all...] |
/external/libvorbis/doc/ |
04-codec.tex | 39 is an audio packet). The packets must occur in the order of 48 relevant pieces of information about the audio stream. The 180 multichannel audio with varying channel mapping applications. Vorbis I 277 \subsection{Audio packet decode and synthesis} 280 are audio. The first step of audio packet decode is to read and 281 verify the packet type. \emph{A non-audio packet when audio is expected 283 must ignore the packet and not attempt decoding it to audio}. 289 \item read 1 bit \varname{[packet_type]}; check that packet type is 0 (audio) [all...] |
/frameworks/base/docs/html/guide/ |
guide_toc.cs | 439 <li><a href="<?cs var:toroot ?>guide/topics/media/audio-capture.html"> 440 <span class="en">Audio Capture</span></a>
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/external/webrtc/webrtc/voice_engine/ |
channel.cc | 263 // Store current audio level in the RTP/RTCP module. 269 // Push data from ACM to RTP/RTCP-module to deliver audio frame for 465 rtpHeader->type.Audio.channel); 537 // In all likelihood, the audio in this frame is garbage. We return an 538 // error so that the audio mixer module doesn't add it to the mix. As 565 // Pass the audio buffers to an optional sink callback, before applying 567 // External recipients of the audio (e.g. via AudioTrack), will do their 609 // Do the panning operation (the audio frame contains stereo at this 644 // Measure audio level (0-9) [all...] |
/prebuilts/gcc/linux-x86/host/x86_64-w64-mingw32-4.8/x86_64-w64-mingw32/include/ddk/ |
scsi.h | 718 /* CDROM audio control */ [all...] |
/external/v8/test/mjsunit/asm/embenchen/ |
copy.js | [all...] |