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/external/libgdx/backends/gdx-backend-lwjgl/src/com/badlogic/gdx/backends/lwjgl/audio/
DOpenALSound.java17 package com.badlogic.gdx.backends.lwjgl.audio;
22 import com.badlogic.gdx.audio.Sound;
29 private final OpenALAudio audio; field in OpenALSound
32 public OpenALSound (OpenALAudio audio) { in OpenALSound() argument
33 this.audio = audio; in OpenALSound()
57 if (audio.noDevice) return 0; in play()
58 int sourceID = audio.obtainSource(false); in play()
61 audio.retain(this, true); in play()
62 sourceID = audio.obtainSource(false); in play()
63 } else audio.retain(this, false); in play()
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DOpenALMusic.java17 package com.badlogic.gdx.backends.lwjgl.audio;
25 import com.badlogic.gdx.audio.Music;
40 private final OpenALAudio audio; field in OpenALMusic
54 public OpenALMusic (OpenALAudio audio, FileHandle file) { in OpenALMusic() argument
55 this.audio = audio; in OpenALMusic()
67 if (audio.noDevice) return; in play()
69 sourceID = audio.obtainSource(true); in play()
72 audio.music.add(this); in play()
103 if (audio.noDevice) return; in stop()
105 audio.music.removeValue(this, true); in stop()
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DOgg.java17 package com.badlogic.gdx.backends.lwjgl.audio;
30 public Music (OpenALAudio audio, FileHandle file) { in Music() argument
31 super(audio, file); in Music()
32 if (audio.noDevice) return; in Music()
61 public Sound (OpenALAudio audio, FileHandle file) { in Sound() argument
62 super(audio); in Sound()
63 if (audio.noDevice) return; in Sound()
/external/libgdx/backends/gdx-backend-lwjgl3/src/com/badlogic/gdx/backends/lwjgl3/audio/
DOpenALSound.java17 package com.badlogic.gdx.backends.lwjgl3.audio;
22 import com.badlogic.gdx.audio.Sound;
29 private final OpenALAudio audio; field in OpenALSound
32 public OpenALSound (OpenALAudio audio) { in OpenALSound() argument
33 this.audio = audio; in OpenALSound()
57 if (audio.noDevice) return 0; in play()
58 int sourceID = audio.obtainSource(false); in play()
61 audio.retain(this, true); in play()
62 sourceID = audio.obtainSource(false); in play()
63 } else audio.retain(this, false); in play()
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DOpenALMusic.java17 package com.badlogic.gdx.backends.lwjgl3.audio;
25 import com.badlogic.gdx.audio.Music;
40 private final OpenALAudio audio; field in OpenALMusic
54 public OpenALMusic (OpenALAudio audio, FileHandle file) { in OpenALMusic() argument
55 this.audio = audio; in OpenALMusic()
67 if (audio.noDevice) return; in play()
69 sourceID = audio.obtainSource(true); in play()
72 audio.music.add(this); in play()
103 if (audio.noDevice) return; in stop()
105 audio.music.removeValue(this, true); in stop()
[all …]
DOgg.java17 package com.badlogic.gdx.backends.lwjgl3.audio;
30 public Music (OpenALAudio audio, FileHandle file) { in Music() argument
31 super(audio, file); in Music()
32 if (audio.noDevice) return; in Music()
61 public Sound (OpenALAudio audio, FileHandle file) { in Sound() argument
62 super(audio); in Sound()
63 if (audio.noDevice) return; in Sound()
/external/vboot_reference/firmware/lib/
Dvboot_audio.c62 static void VbGetDevMusicNotes(VbAudioContext *audio, int use_short) in VbGetDevMusicNotes() argument
85 if (!audio->background_beep) in VbGetDevMusicNotes()
192 audio->music_notes = notebuf; in VbGetDevMusicNotes()
193 audio->note_count = count; in VbGetDevMusicNotes()
194 audio->free_notes_when_done = 1; in VbGetDevMusicNotes()
200 audio->music_notes = builtin; in VbGetDevMusicNotes()
201 audio->note_count = count; in VbGetDevMusicNotes()
202 audio->free_notes_when_done = 0; in VbGetDevMusicNotes()
212 VbAudioContext *audio = &au; in VbAudioOpen() local
227 Memset(audio, 0, sizeof(*audio)); in VbAudioOpen()
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/external/v8/tools/gyp/tools/emacs/testdata/
Dmedia.gyp10 # Override to dynamically link the cras (ChromeOS audio) library.
33 'audio/android/audio_manager_android.cc',
34 'audio/android/audio_manager_android.h',
35 'audio/android/audio_track_output_android.cc',
36 'audio/android/audio_track_output_android.h',
37 'audio/android/opensles_input.cc',
38 'audio/android/opensles_input.h',
39 'audio/android/opensles_output.cc',
40 'audio/android/opensles_output.h',
41 'audio/async_socket_io_handler.h',
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/external/webrtc/webrtc/modules/audio_processing/
Dnoise_suppression_impl.cc70 void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { in AnalyzeCaptureAudio() argument
71 RTC_DCHECK(audio); in AnalyzeCaptureAudio()
78 RTC_DCHECK_GE(160u, audio->num_frames_per_band()); in AnalyzeCaptureAudio()
79 RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels()); in AnalyzeCaptureAudio()
82 audio->split_bands_const_f(i)[kBand0To8kHz]); in AnalyzeCaptureAudio()
87 void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio() argument
88 RTC_DCHECK(audio); in ProcessCaptureAudio()
94 RTC_DCHECK_GE(160u, audio->num_frames_per_band()); in ProcessCaptureAudio()
95 RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels()); in ProcessCaptureAudio()
99 audio->split_bands_const_f(i), in ProcessCaptureAudio()
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Dgain_control_impl.cc69 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) { in ProcessRenderAudio() argument
75 assert(audio->num_frames_per_band() <= 160); in ProcessRenderAudio()
81 WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band()); in ProcessRenderAudio()
88 render_queue_buffer_.end(), audio->mixed_low_pass_data(), in ProcessRenderAudio()
89 (audio->mixed_low_pass_data() + audio->num_frames_per_band())); in ProcessRenderAudio()
127 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { in AnalyzeCaptureAudio() argument
134 assert(audio->num_frames_per_band() <= 160); in AnalyzeCaptureAudio()
135 assert(audio->num_channels() == num_handles()); in AnalyzeCaptureAudio()
145 audio->split_bands(i), in AnalyzeCaptureAudio()
146 audio->num_bands(), in AnalyzeCaptureAudio()
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Decho_control_mobile_impl.cc93 int EchoControlMobileImpl::ProcessRenderAudio(const AudioBuffer* audio) { in ProcessRenderAudio() argument
100 assert(audio->num_frames_per_band() <= 160); in ProcessRenderAudio()
101 assert(audio->num_channels() == apm_->num_reverse_channels()); in ProcessRenderAudio()
108 for (size_t j = 0; j < audio->num_channels(); j++) { in ProcessRenderAudio()
111 my_handle, audio->split_bands_const(j)[kBand0To8kHz], in ProcessRenderAudio()
112 audio->num_frames_per_band()); in ProcessRenderAudio()
119 audio->split_bands_const(j)[kBand0To8kHz], in ProcessRenderAudio()
120 (audio->split_bands_const(j)[kBand0To8kHz] + in ProcessRenderAudio()
121 audio->num_frames_per_band())); in ProcessRenderAudio()
167 int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio() argument
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Decho_cancellation_impl.cc88 int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) { in ProcessRenderAudio() argument
94 assert(audio->num_frames_per_band() <= 160); in ProcessRenderAudio()
95 assert(audio->num_channels() == apm_->num_reverse_channels()); in ProcessRenderAudio()
103 for (size_t j = 0; j < audio->num_channels(); j++) { in ProcessRenderAudio()
108 my_handle, audio->split_bands_const_f(j)[kBand0To8kHz], in ProcessRenderAudio()
109 audio->num_frames_per_band()); in ProcessRenderAudio()
117 audio->split_bands_const_f(j)[kBand0To8kHz], in ProcessRenderAudio()
118 (audio->split_bands_const_f(j)[kBand0To8kHz] + in ProcessRenderAudio()
119 audio->num_frames_per_band())); in ProcessRenderAudio()
162 int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio() argument
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/external/webrtc/webrtc/audio/
Dwebrtc_audio.gypi17 'audio/audio_receive_stream.cc',
18 'audio/audio_receive_stream.h',
19 'audio/audio_send_stream.cc',
20 'audio/audio_send_stream.h',
21 'audio/audio_sink.h',
22 'audio/audio_state.cc',
23 'audio/audio_state.h',
24 'audio/conversion.h',
25 'audio/scoped_voe_interface.h',
/external/webrtc/webrtc/modules/audio_device/ios/
Daudio_device_ios.mm34 // audio session. This variable is used to ensure that we only activate an audio
60 // will be set to this value as well to avoid resampling the the audio unit's
67 // ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
74 // in the I/O audio unit. Initial tests have shown that it is possible to use
78 // audio unit. Hence, we will not hit a RTC_CHECK in
82 // Number of bytes per audio sample for 16-bit signed integer representation.
98 // Verifies that the current audio session supports input audio and that the
102 // Ensure that the device currently supports audio input.
104 LOG(LS_ERROR) << "No audio input path is available!";
121 // Activates an audio session suitable for full duplex VoIP sessions when
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/external/libgdx/backends/gdx-backend-moe/src/com/badlogic/gdx/backends/iosmoe/
DIOSAudio.java21 import com.badlogic.gdx.audio.AudioDevice;
22 import com.badlogic.gdx.audio.AudioRecorder;
23 import com.badlogic.gdx.audio.Music;
24 import com.badlogic.gdx.audio.Sound;
33 OALSimpleAudio audio = OALSimpleAudio.sharedInstance(); in IOSAudio() local
34 if (audio != null) { in IOSAudio()
35 audio.setAllowIpod(config.allowIpod); in IOSAudio()
36 audio.setHonorSilentSwitch(true); in IOSAudio()
/external/libgdx/backends/gdx-backend-robovm/src/com/badlogic/gdx/backends/iosrobovm/
DIOSAudio.java21 import com.badlogic.gdx.audio.AudioDevice;
22 import com.badlogic.gdx.audio.AudioRecorder;
23 import com.badlogic.gdx.audio.Music;
24 import com.badlogic.gdx.audio.Sound;
33 OALSimpleAudio audio = OALSimpleAudio.sharedInstance(); in IOSAudio() local
34 if (audio != null) { in IOSAudio()
35 audio.setAllowIpod(config.allowIpod); in IOSAudio()
36 audio.setHonorSilentSwitch(true); in IOSAudio()
/external/autotest/client/site_tests/audio_AudioCorruption/
Dcontrol7 PURPOSE = "Verify that Chrome can handle corrupted mp3 audio"
9 This test will fail if Chrome can't catch error for playing corrupted mp3 audio.
15 TEST_CLASS = "audio"
19 This test verifies Chrome can catch error for playing corrupted mp3 audio.
22 audio = 'http://commondatastorage.googleapis.com/chromiumos-test-assets-public/audio_AudioCorruptio…
23 job.run_test('audio_AudioCorruption', audio=audio)
/external/autotest/client/site_tests/audio_CrasLoopback/
Dcontrol5 AUTHOR = 'The Chromium OS Audiovideo Team, chromeos-audio@google.com'
7 PURPOSE = 'Test that audio played to line out can be heard at mic in.'
9 Check if the audio played to line out is heard by cras_test_client at mic in.
11 ATTRIBUTES = "suite:audio, suite:partners"
12 SUITE = 'audio, partners'
15 TEST_CLASS = "audio"
20 Test that audio playback and capture are working.
/external/autotest/client/site_tests/audio_AlsaLoopback/
Dcontrol5 AUTHOR = 'The Chromium OS Audiovideo Team, chromeos-audio@google.com'
7 PURPOSE = 'Test that audio played to line out can be heard at mic in.'
9 Check if the audio played to line out is heard by arecord at mic in.
11 ATTRIBUTES = "suite:audio"
12 SUITE = 'audio'
15 TEST_CLASS = "audio"
20 Test that audio playback and capture are working.
/external/libgdx/tests/gdx-tests/src/com/badlogic/gdx/tests/
DAudioRecorderTest.java20 import com.badlogic.gdx.audio.AudioDevice;
21 import com.badlogic.gdx.audio.AudioRecorder;
32 device = Gdx.audio.newAudioDevice(44100, true); in create()
33 recorder = Gdx.audio.newAudioRecorder(44100, true); in create()
62 device = Gdx.audio.newAudioDevice(44100, true); in resume()
63 recorder = Gdx.audio.newAudioRecorder(44100, true); in resume()
/external/libvorbis/doc/
Da1-encapsulation-ogg.tex9 streams to encapsulate Vorbis compressed audio packet data into file
13 of Vorbis audio packets.
36 The Ogg stream must be unmultiplexed (only one stream, a Vorbis audio stream, per link)
44 for low-bitrate movies consisting of DivX video and Vorbis audio.
45 However, a 'Vorbis I audio file' is taken to imply Vorbis audio
47 audio player' is not required to implement Ogg support beyond the
59 while visual media should use \literal{video/ogg}, and audio
60 \literal{audio/ogg}. Vorbis data encapsulated in Ogg may appear
62 \literal{audio/vorbis} + \literal{audio/vorbis-config}.
73 uniquely identifies a stream as Vorbis audio, is placed alone in the
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/external/libgdx/backends/gdx-backend-lwjgl3/src/com/badlogic/gdx/backends/lwjgl3/audio/mock/
DMockAudio.java17 package com.badlogic.gdx.backends.lwjgl3.audio.mock;
20 import com.badlogic.gdx.audio.AudioDevice;
21 import com.badlogic.gdx.audio.AudioRecorder;
22 import com.badlogic.gdx.audio.Music;
23 import com.badlogic.gdx.audio.Sound;
/external/libgdx/backends/gdx-backend-headless/src/com/badlogic/gdx/backends/headless/mock/audio/
DMockAudio.java17 package com.badlogic.gdx.backends.headless.mock.audio;
20 import com.badlogic.gdx.audio.AudioDevice;
21 import com.badlogic.gdx.audio.AudioRecorder;
22 import com.badlogic.gdx.audio.Music;
23 import com.badlogic.gdx.audio.Sound;
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/
Daudio_encoder_pcm.cc82 rtc::ArrayView<const int16_t> audio, in EncodeInternal() argument
88 speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end()); in EncodeInternal()
110 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, in EncodeCall() argument
113 return WebRtcG711_EncodeA(audio, input_len, encoded); in EncodeCall()
123 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, in EncodeCall() argument
126 return WebRtcG711_EncodeU(audio, input_len, encoded); in EncodeCall()
/external/autotest/client/site_tests/audio_LoopbackLatency/
Dcontrol7 PURPOSE = 'Test that audio loopback latency'
9 Check if the audio played to line out can be heard mic in, and assert
12 ATTRIBUTES = "suite:audio"
13 SUITE = 'audio'
16 TEST_CLASS = "audio"
21 Test that audio loopback latency is within certain limit.

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