/external/chromium_org/media/formats/mp4/ |
H A D | mp4_stream_parser.cc | 187 // TODO(strobe): Only the first audio and video track present in a file are 220 MEDIA_LOG(log_cb_) << "Unsupported audio format 0x" 228 MEDIA_LOG(log_cb_) << "audio object type 0x" << std::hex << audio_type 248 MEDIA_LOG(log_cb_) << "Unsupported audio object type 0x" << std::hex 393 // Append an ADTS header to every audio sample. 443 bool audio = has_audio_ && audio_track_id_ == runs_->track_id(); 447 if (!audio && !video) { 490 if (audio) { 509 } else if ((audio && is_audio_track_encrypted_) || 517 StreamParserBuffer::Type buffer_type = audio [all...] |
/external/chromium_org/content/browser/media/ |
H A D | webrtc_internals.h | 71 // security origin of the getUserMedia call, |audio| is true if audio stream 73 // |audio_constraints| is the constraints for the audio, |video_constraints| 78 bool audio, 165 // "audio" -- the serialized audio constraints if audio is requested.
|
/external/chromium_org/content/browser/renderer_host/media/ |
H A D | audio_input_sync_writer.cc | 33 // Create vector of audio buses by wrapping existing blocks of memory. 39 media::AudioBus::WrapMemory(params, buffer->audio); 63 oss << "AISW::Write: audio input data received for the first time: delay " 69 oss << "AISW::Write: audio input data delay unexpectedly long: delay = " 81 // Write audio parameters to shared memory. 90 // Copy data from the native audio layer into shared memory using pre- 91 // allocated audio buses.
|
/external/chromium_org/media/cast/test/ |
H A D | fake_media_source.h | 5 // A fake media source that generates video and audio frames to a cast 8 // generate an animation and audio of fixed frequency. 21 #include "media/audio/audio_parameters.h" 54 // Transcode this file as the source of video and audio frames. 87 ScopedAVPacket DemuxOnePacket(bool* audio); 111 int audio_frame_count_; // Each audio frame is exactly 10ms. 131 // These are used for audio resampling. 137 // Track the timestamp of audio sent to the receiver.
|
/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
H A D | mediamessages.h | 50 // A collection of audio and video and data streams. Most of the 66 const std::vector<StreamParams>& audio() const { return audio_; } function in struct:cricket::MediaStreams
|
H A D | mediasession.cc | 153 // For audio, HMAC 32 is prefered because of the low overhead. 179 // For video support only 80-bit SHA1 HMAC. For audio 32-bit HMAC is 185 bool audio = offer->type() == MEDIA_TYPE_AUDIO; local 191 (CS_AES_CM_128_HMAC_SHA1_32 == i->cipher_suite && audio && !bundle)) { 396 // Helper class used for finding duplicate RTP payload types among audio, video 411 // audio and video extensions. 1063 type_str = "audio"; 1320 const AudioContentDescription* audio = local 1322 if (audio) { 1323 *audio_codecs = audio 1361 const AudioContentDescription* audio = local [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtcp_format_remb_unittest.cc | 91 configuration.audio = false;
|
/external/libvorbis/ |
H A D | libvorbis.spec | 26 general-purpose compressed audio format for audio and music at fixed
|
/external/qemu/android/ |
H A D | cmdline-options.h | 110 OPT_FLAG ( no_audio, "disable audio support" ) 111 OPT_FLAG ( noaudio, "same as -no-audio" ) 112 OPT_PARAM( audio, "<backend>", "use specific audio backend" )
|
/external/srec/audio/AudioIn/UNIX/src/ |
H A D | audioinwrapper.cpp | 29 #include <system/audio.h> 83 // possibly lame attempt to get Sooner audio input working
|
/external/srec/srec_jni/ |
H A D | android_speech_srec_MicrophoneInputStream.cpp | 31 #include <system/audio.h>
|
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
H A D | peerconnection_unittest.cc | 150 virtual void Negotiate(bool audio, bool video) = 0; 157 void AddMediaStream(bool audio, bool video) { argument 164 if (audio && can_receive_audio()) { 166 // Disable highpass filter so that we can get all the test audio frames. 171 // TODO(perkj): Test audio source when it is implemented. Currently audio 243 // We can't create a DTMF sender with an invalid audio track or a non local 608 virtual void Negotiate(bool audio, bool video) { argument 612 if (offer->description()->GetContentByName("audio")) { 613 offer->description()->GetContentByName("audio") 663 SetReceiveAudioVideo(bool audio, bool video) argument 670 SetReceiveAudio(bool audio) argument [all...] |
H A D | peerconnectionendtoend_unittest.cc | 96 InjectAfter("a=mid:audio\r\n", kAudioSdes, sdp); 114 InjectAfter("a=mid:audio\r\n", ufrag_pwd, sdp); 161 void GetAndAddUserMedia(bool audio, FakeConstraints audio_constraints, argument 163 caller_->GetAndAddUserMedia(audio, audio_constraints, 165 callee_->GetAndAddUserMedia(audio, audio_constraints,
|
/external/chromium_org/chrome/browser/extensions/api/audio/ |
H A D | audio_service_chromeos.cc | 5 #include "chrome/browser/extensions/api/audio/audio_service.h" 10 #include "chromeos/audio/audio_device.h" 11 #include "chromeos/audio/cras_audio_handler.h" 18 using api::audio::OutputDeviceInfo; 19 using api::audio::InputDeviceInfo; 31 // Start to query audio device information.
|
/external/chromium_org/chrome/browser/resources/cryptotoken/ |
H A D | webrequest.js | 93 // But video and audio is. 94 var audio = new Audio(); 95 audio.src = opt_logMsgUrl + logMsg;
|
/external/chromium_org/chrome/browser/resources/hotword/ |
H A D | state_manager.js | 47 * Hotword trigger audio notification... a.k.a The Chime (tm). 50 this.chime_ = document.createElement('audio'); 57 hotword.constants.SHARED_MODULE_ROOT + '/audio/chime.wav'); 147 // googDucking set to false so that audio output level from other tabs 151 ({audio: {optional: [{googDucking: false}]}});
|
/external/chromium_org/content/shell/renderer/test_runner/ |
H A D | mock_web_user_media_client.cc | 168 WebVector<WebMediaStreamTrack> audio_tracks(request.audio() ? one : zero); 171 if (request.audio()) { 175 "Mock audio device");
|
/external/chromium_org/google_apis/gaia/ |
H A D | google_service_auth_error.cc | 20 const std::string& token, const GURL& audio, const GURL& img, 22 : token(token), audio_url(audio), image_url(img), unlock_url(unlock), 19 Captcha( const std::string& token, const GURL& audio, const GURL& img, const GURL& unlock, int width, int height) argument
|
/external/chromium_org/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ |
H A D | APPRTCConnectionManager.m | 185 RTCPair* audio = 189 NSArray* mandatory = @[ audio, video ]; 268 @"Expected audio or video track"); 270 @"Expected at most 1 audio stream"); 381 RTCPair* audio = [[RTCPair alloc] initWithKey:@"OfferToReceiveAudio" 385 NSArray* mandatory = @[ audio, video ]; 430 // Mangle |origSDP| to prefer the ISAC/16k audio codec. 443 if ([line hasPrefix:@"m=audio "]) { 450 NSLog(@"No m=audio line, so can't prefer iSAC");
|
/external/chromium_org/third_party/webrtc/test/channel_transport/ |
H A D | udp_transport_impl.h | 82 const bool audio = false) OVERRIDE; 182 int32_t EnableQoS(int32_t serviceType, bool audio,
|
/external/libvorbis/doc/ |
H A D | footer.tex | 17 Ogg Vorbis is the first Ogg audio CODEC. Anyone may freely use and
|
H A D | 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 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) 538 vectors are the length \varname{[n]}/2 audio spectru [all...] |
/external/chromium_org/third_party/WebKit/Tools/Scripts/webkitpy/layout_tests/port/ |
H A D | driver.py | 61 def __init__(self, text, image, image_hash, audio, crash=False, 69 self.audio = audio # Binary format is port-dependent. 163 text, audio = self._read_first_block(deadline) # First block is either text or audio 205 return DriverOutput(text, image, actual_image_hash, audio, 414 if block.content_type == 'audio/wav':
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_generic_codec.cc | 119 // This is not 10 ms of audio, given the sampling frequency of the codec. 204 // There is not enough audio 234 // Process the audio through VAD. The function will set |_vad_labels|. 249 // Setting the following makes sure that the move of audio data and 280 // We should encode the audio frame. Either VAD and/or DTX is off, or the 281 // audio was considered "active". 297 // audio with the steps of the basic-coding-block. 352 // Remove encoded audio and move next audio to be encoded to the beginning 503 // Fresh start of audio buffe [all...] |
/external/chromium_org/media/cast/net/pacing/ |
H A D | paced_sender_unittest.cc | 88 bool audio) { 98 audio ? kAudioSsrc : kVideoSsrc, // ssrc 107 bool success = writer.WriteU32(audio ? kAudioSsrc : kVideoSsrc); 389 // Send normal audio packet. This is queued and will be sent 86 CreateSendPacketVector(size_t packet_size, int num_of_packets_in_frame, bool audio) argument
|