/external/qemu/audio/ |
H A D | paaudio.c | 3 #include "audio.h" 18 # define D(...) VERBOSE_PRINT(audio,__VA_ARGS__) 19 # define D_ACTIVE VERBOSE_CHECK(audio) 295 dolog ("Internal logic error: Bad audio format %d\n", afmt); 528 D("%s: error opening open pulse audio library: %s",
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/external/chromium_org/content/renderer/pepper/ |
H A D | pepper_media_stream_audio_track_host.cc | 143 // Clear |buffers_|, so the audio thread will drop all incoming audio data. 162 // we just cleared |buffers_| , so the audio thread will drop all incoming 163 // audio data, and not use buffers in |host_|. 172 // Fill the |buffers_|, so the audio thread can continue receiving audio data. 188 // If |InitBuffers()| is called after this task being posted from the audio 205 // the incomming audio buffer. However, this doesn't necessarily equal 236 ->audio);
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
H A D | rtp_rtcp.h | 33 * audio - True for a audio version of the RTP/RTCP module 58 bool audio; member in struct:webrtc::RtpRtcp::Configuration 315 * Used by the codec module to deliver a video or audio frame for 622 * set audio packet size, used to determine when it's time to send a DTMF 667 * Store the audio level in dBov for header-extension-for-audio-level-
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | nack_rtx_unittest.cc | 181 configuration.audio = false;
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H A D | rtcp_sender.h | 72 RTCPSender(const int32_t id, const bool audio,
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H A D | rtp_rtcp_impl.cc | 29 audio(false), 61 configuration.audio, 70 configuration.audio, 76 audio_(configuration.audio), 1013 // Set audio packet size, used to determine when it's time to send a DTMF
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H A D | rtp_sender_audio.cc | 76 // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG) 124 payload->audio = true; 136 // for audio true for first packet in a speech burst 271 // A source MAY send events and coded audio packets for the same time 338 // we don't send empty audio RTP packets 438 // Update audio level extension, if included.
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H A D | rtp_sender_video.cc | 98 payload->audio = false;
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H A D | rtp_sender.h | 70 RTPSender(const int32_t id, const bool audio, Clock *clock, 235 // Set audio packet size, used to determine when it's time to send a DTMF 239 // Store the audio level in d_bov for 240 // header-extension-for-audio-level-indication.
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | mt_rx_tx_test.cc | 156 configuration.audio = false;
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H A D | media_opt_test.cc | 205 configuration.audio = false;
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H A D | rtp_player.cc | 221 configuration.audio = false;
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/external/libvorbis/doc/ |
H A D | 05-comment.tex | 114 The artist generally considered responsible for the work. In popular music this is usually the performing band or singer. For classical music it would be the composer. For an audio book it would be the author of the original text. 117 The artist(s) who performed the work. In classical music this would be the conductor, orchestra, soloists. In an audio book it would be the actor who did the reading. In popular music this is typically the same as the ARTIST and is omitted.
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/external/chromium_org/chrome/browser/media/ |
H A D | media_capture_devices_dispatcher.cc | 47 #include "media/audio/audio_manager_base.h" 170 // Use the special loopback device ID for system audio capture. 260 // AVFoundation is used for video/audio device monitoring and video capture in 520 // Currently loopback audio capture is supported only on Windows and ChromeOS. 550 // Currently loopback audio capture is supported only on Windows and ChromeOS. 733 // Set an initial error result. If neither audio or video is allowed, we'll 739 // result + a dcheck to ensure at least one of audio or video types is 745 // Get the exact audio or video device if an id is specified. 765 // If either or both audio and video devices were requested but not 872 bool audio, 870 GetDefaultDevicesForProfile( Profile* profile, bool audio, bool video, content::MediaStreamDevices* devices) argument [all...] |
/external/chromium_org/content/browser/media/ |
H A D | webrtc_internals_browsertest.cc | 194 << request.origin << "', audio:'" << request.audio_constraints 250 std::string origin, audio, video; local 254 ASSERT_TRUE(dict->GetString("audio", &audio)); 259 EXPECT_EQ(requests[i].audio_constraints, audio);
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/external/chromium_org/content/renderer/media/ |
H A D | peer_connection_tracker.cc | 104 result += ", audio: ["; 524 user_media_request.audio(),
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api_audio.cc | 131 configuration.audio = true;
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H A D | test_api_rtcp.cc | 111 configuration.audio = true;
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/external/sepolicy/ |
H A D | file_contexts | 40 /dev/audio.* u:object_r:audio_device:s0 200 /data/misc/audio(/.*)? u:object_r:audio_data_file:s0
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/external/wpa_supplicant_8/hostapd/src/utils/ |
H A D | http_curl.c | 329 STACK_OF(LogotypeAudio) *audio; 402 ASN1_IMP_SEQUENCE_OF_OPT(LogotypeData, audio, LogotypeAudio, 1) 660 num = data->audio ? sk_LogotypeAudio_num(data->audio) : 0;
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/external/wpa_supplicant_8/src/utils/ |
H A D | http_curl.c | 329 STACK_OF(LogotypeAudio) *audio; 402 ASN1_IMP_SEQUENCE_OF_OPT(LogotypeData, audio, LogotypeAudio, 1) 660 num = data->audio ? sk_LogotypeAudio_num(data->audio) : 0;
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/external/wpa_supplicant_8/wpa_supplicant/src/utils/ |
H A D | http_curl.c | 329 STACK_OF(LogotypeAudio) *audio; 402 ASN1_IMP_SEQUENCE_OF_OPT(LogotypeData, audio, LogotypeAudio, 1) 660 num = data->audio ? sk_LogotypeAudio_num(data->audio) : 0;
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/external/bluetooth/bluedroid/audio_a2dp_hw/ |
H A D | audio_a2dp_hw.c | 23 * Description: Implements hal for bluedroid a2dp audio device 42 #include <system/audio.h> 43 #include <hardware/audio.h> 503 /* disconnect audio path */ 528 /* disconnect audio path */ 539 ** audio output callbacks
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/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
H A D | channel_unittest.cc | 470 // kPcmuCodec is used as audio codec and kH264Codec is used as video codec. 1850 cricket::AudioContentDescription* audio) { 1851 audio->AddCodec(audio_codec); 1852 audio->set_rtcp_mux((flags & RTCP_MUX) != 0); 1854 audio->AddCrypto(cricket::CryptoParams( 1863 cricket::AudioContentDescription* audio) { 1864 *audio = source; 1876 uint32 ssrc, int flags, cricket::AudioContentDescription* audio) { 1877 audio->AddLegacyStream(ssrc); 1846 CreateContent( int flags, const cricket::AudioCodec& audio_codec, const cricket::VideoCodec& video_codec, cricket::AudioContentDescription* audio) argument 1861 CopyContent( const cricket::AudioContentDescription& source, cricket::AudioContentDescription* audio) argument 1875 AddLegacyStreamInContent( uint32 ssrc, int flags, cricket::AudioContentDescription* audio) argument
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | transmit_mixer.cc | 335 // --- Resample input audio and create/store the initial audio frame 352 // --- Near-end audio processing. 409 // --- Measure audio level of speech after all processing. 1137 void TransmitMixer::GenerateAudioFrame(const int16_t* audio, argument 1160 DownConvertToCodecFormat(audio, 1235 // Replace ACM audio with file.
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