| /external/ltp/testcases/kernel/device-drivers/v4l/user_space/ |
| D | test_VIDIOC_ENUMAUDIO.c | 41 struct v4l2_audio audio; in test_VIDIOC_ENUMAUDIO() local
47 memset(&audio, 0xff, sizeof(audio)); in test_VIDIOC_ENUMAUDIO()
48 audio.index = i; in test_VIDIOC_ENUMAUDIO()
49 ret_enum = ioctl(get_video_fd(), VIDIOC_ENUMAUDIO, &audio); in test_VIDIOC_ENUMAUDIO()
58 CU_ASSERT_EQUAL(audio.index, i); in test_VIDIOC_ENUMAUDIO()
60 CU_ASSERT(0 < strlen((char *)audio.name)); in test_VIDIOC_ENUMAUDIO()
62 ((char *)audio.name, sizeof(audio.name))); in test_VIDIOC_ENUMAUDIO()
66 CU_ASSERT_EQUAL(audio.reserved[0], 0); in test_VIDIOC_ENUMAUDIO()
67 CU_ASSERT_EQUAL(audio.reserved[1], 0); in test_VIDIOC_ENUMAUDIO()
75 audio2.index = audio.index; in test_VIDIOC_ENUMAUDIO()
[all …]
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| D | test_VIDIOC_AUDIO.c | 67 struct v4l2_audio audio; in test_VIDIOC_G_AUDIO() local
70 memset(&audio, 0xff, sizeof(audio)); in test_VIDIOC_G_AUDIO()
71 ret_get = ioctl(get_video_fd(), VIDIOC_G_AUDIO, &audio); in test_VIDIOC_G_AUDIO()
82 CU_ASSERT(0 < strlen((char *)audio.name)); in test_VIDIOC_G_AUDIO()
83 CU_ASSERT(valid_string((char *)audio.name, sizeof(audio.name))); in test_VIDIOC_G_AUDIO()
85 CU_ASSERT(valid_audio_capability(audio.capability)); in test_VIDIOC_G_AUDIO()
86 CU_ASSERT(valid_audio_mode(audio.mode)); in test_VIDIOC_G_AUDIO()
88 CU_ASSERT_EQUAL(audio.reserved[0], 0); in test_VIDIOC_G_AUDIO()
89 CU_ASSERT_EQUAL(audio.reserved[1], 0); in test_VIDIOC_G_AUDIO()
97 audio2.index = audio.index; in test_VIDIOC_G_AUDIO()
[all …]
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| /external/vboot_reference/firmware/lib/ |
| D | vboot_audio.c | 62 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()
[all …]
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| /external/tensorflow/tensorflow/core/kernels/ |
| D | encode_wav_op_test.cc | 60 {decode_wav_op.audio, decode_wav_op.sample_rate}, in TEST()
63 const Tensor& audio = outputs[0]; in TEST() local
66 EXPECT_EQ(2, audio.dims()); in TEST()
67 EXPECT_EQ(2, audio.dim_size(1)); in TEST()
68 EXPECT_EQ(4, audio.dim_size(0)); in TEST()
69 EXPECT_NEAR(0.0f, audio.flat<float>()(0), 1e-4f); in TEST()
70 EXPECT_NEAR(0.5f, audio.flat<float>()(1), 1e-4f); in TEST()
71 EXPECT_NEAR(1.0f, audio.flat<float>()(2), 1e-4f); in TEST()
72 EXPECT_NEAR(-1.0f, audio.flat<float>()(3), 1e-4f); in TEST()
73 EXPECT_NEAR(0.25f, audio.flat<float>()(4), 1e-4f); in TEST()
[all …]
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| D | decode_wav_op_test.cc | 71 {decode_wav_op.audio, decode_wav_op.sample_rate}, in TEST()
74 const Tensor& audio = outputs[0]; in TEST() local
77 EXPECT_EQ(2, audio.dims()); in TEST()
78 EXPECT_EQ(1, audio.dim_size(1)); in TEST()
79 EXPECT_EQ(4, audio.dim_size(0)); in TEST()
80 EXPECT_NEAR(0.0f, audio.flat<float>()(0), 1e-4f); in TEST()
81 EXPECT_NEAR(0.5f, audio.flat<float>()(1), 1e-4f); in TEST()
82 EXPECT_NEAR(1.0f, audio.flat<float>()(2), 1e-4f); in TEST()
83 EXPECT_NEAR(-1.0f, audio.flat<float>()(3), 1e-4f); in TEST()
|
| D | encode_wav_op.cc | 35 const Tensor& audio = context->input(0); in Compute() local
36 OP_REQUIRES(context, audio.dims() == 2, in Compute()
38 audio.shape().DebugString())); in Compute()
47 FastBoundsCheck(audio.NumElements(), std::numeric_limits<int32>::max()), in Compute()
51 const int32 channel_count = static_cast<int32>(audio.dim_size(1)); in Compute()
52 const int32 sample_count = static_cast<int32>(audio.dim_size(0)); in Compute()
60 audio.flat<float>().data(), sample_rate, channel_count, in Compute()
|
| D | summary_audio_op_test.cc | 65 ASSERT_FALSE(value->audio().encoded_audio_string().empty()) in CheckAndRemoveEncodedAudio()
71 value->audio().encoded_audio_string())); in CheckAndRemoveEncodedAudio()
101 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 2 in TEST_F()
104 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 2 in TEST_F()
107 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 2 in TEST_F()
135 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 1 in TEST_F()
138 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 1 in TEST_F()
141 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 1 in TEST_F()
|
| /external/python/cpython2/Doc/library/ |
| D | al.rst | 14 This module provides access to the audio facilities of the SGI Indy and Indigo
25 Symbolic constants from the C header file ``<audio.h>`` are defined in the
30 The current version of the audio library may dump core when bad argument values
44 :dfn:`audio port object`; methods of audio port objects are described below.
49 The return value is a new :dfn:`audio configuration object`; methods of audio
79 .. method:: audio configuration.getqueuesize()
84 .. method:: audio configuration.setqueuesize(size)
89 .. method:: audio configuration.getwidth()
94 .. method:: audio configuration.setwidth(width)
99 .. method:: audio configuration.getchannels()
[all …]
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| D | sunaudio.rst | 2 :mod:`sunaudiodev` --- Access to Sun audio hardware
7 :synopsis: Access to Sun audio hardware.
17 This module allows you to access the Sun audio interface. The Sun audio hardware
18 is capable of recording and playing back audio data in u-LAW format with a
20 :manpage:`audio(7I)` manual page.
38 This function opens the audio device and returns a Sun audio device object. This
43 to open the device only for the activity needed. See :manpage:`audio(7I)` for
47 ``AUDIODEV`` for the base audio device filename. If not found, it falls back to
48 :file:`/dev/audio`. The control device is calculated by appending "ctl" to the
49 base audio device.
[all …]
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| D | ossaudiodev.rst | 2 :mod:`ossaudiodev` --- Access to OSS-compatible audio devices
7 :synopsis: Access to OSS-compatible audio devices.
12 This module allows you to access the OSS (Open Sound System) audio interface.
14 the standard audio interface for Linux and recent versions of FreeBSD.
20 majority of Linux audio apps anyway.
32 > * This is an OSS (Linux) audio emulator.
38 audio interface. That's the great thing about standards, there are so
72 Open an audio device and return an OSS audio device object. This object
75 read/write semantics and those of OSS audio devices). It also supports a number
76 of audio-specific methods; see below for the complete list of methods.
[all …]
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| D | aifc.rst | 5 :synopsis: Read and write audio files in AIFF or AIFC format.
18 AIFF is Audio Interchange File Format, a format for storing digital audio
20 ability to compress the audio data.
28 Audio files have a number of parameters that describe the audio data. The
30 sampled. The number of channels indicate if the audio is mono, stereo, or
33 *nchannels*\*\ *samplesize* bytes, and a second's worth of audio consists of
36 For example, CD quality audio has a sample size of two bytes (16 bits), uses two
61 Return the number of audio channels (1 for mono, 2 for stereo).
71 Return the sampling rate (number of audio frames per second).
76 Return the number of audio frames in the file.
[all …]
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| /external/webrtc/webrtc/modules/audio_processing/ |
| D | noise_suppression_impl.cc | 70 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()
[all …]
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| D | gain_control_impl.cc | 69 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()
[all …]
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| D | echo_control_mobile_impl.cc | 93 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
[all …]
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| D | echo_cancellation_impl.cc | 88 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
[all …]
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| /external/u-boot/drivers/sound/ |
| D | Kconfig | 6 Support making sounds through an audio codec. This is normally a
12 audio codecs are called from the sound-i2s code. This could be
19 I2S is a serial bus often used to transmit audio data from the
20 SoC to the audio codec. This option enables sound support using
28 Samsung Exynos SoCs support an I2S interface for sending audio
29 data to an audio codec. This option enables support for this,
30 using one of the available audio codec drivers. Enabling this
35 bool "Support Maxim max98095 audio codec"
38 Enable the max98095 audio codec. This is connected via I2S for
39 audio data and I2C for codec control. At present it only works
[all …]
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| /external/python/cpython3/Lib/test/ |
| D | mime.types | 266 # atx: audio/ATRAC-X
828 # stm: audio/x-stm
936 application/vnd.yamaha.smaf-audio saf
966 # mod: audio/x-mod
977 audio/1d-interleaved-parityfec
978 audio/32kadpcm 726
980 audio/3gpp
982 audio/3gpp2
983 audio/ac3 ac3
984 audio/AMR amr
[all …]
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| /external/webrtc/webrtc/audio/ |
| D | webrtc_audio.gypi | 17 '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/ |
| D | audio_device_ios.mm | 34 // 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
[all …]
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| /external/tensorflow/tensorflow/lite/experimental/microfrontend/python/kernel_tests/ |
| D | audio_microfrontend_op_test.py | 40 audio = tf.constant(
44 audio,
59 audio = tf.constant(
63 audio,
81 audio = tf.constant(
85 audio,
101 audio = tf.constant(
105 audio,
124 audio = tf.constant(
128 audio,
[all …]
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| /external/tensorflow/tensorflow/core/lib/wav/ |
| D | wav_io_test.cc | 36 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f}; in TEST() local
42 EncodeAudioAsS16LEWav(audio, 44100, 2, 3, nullptr).code()); in TEST()
49 EncodeAudioAsS16LEWav(audio, 0, 2, 3, &result).code()); in TEST()
51 EncodeAudioAsS16LEWav(audio, 44100, 0, 3, &result).code()); in TEST()
53 EncodeAudioAsS16LEWav(audio, 44100, 2, 0, &result).code()); in TEST()
58 EncodeAudioAsS16LEWav(audio, kuint32max_plus_one, 2, 3, &result).code()); in TEST()
62 EncodeAudioAsS16LEWav(audio, 44100, kuint16max_plus_one, 3, &result) in TEST()
67 EncodeAudioAsS16LEWav(audio, 44100, 2, 1073741813, &result).code()); in TEST()
71 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f}; in TEST() local
73 TF_EXPECT_OK(EncodeAudioAsS16LEWav(audio, 44100, 2, 3, &result)); in TEST()
[all …]
|
| /external/python/cpython3/Doc/library/ |
| D | ossaudiodev.rst | 1 :mod:`ossaudiodev` --- Access to OSS-compatible audio devices
6 :synopsis: Access to OSS-compatible audio devices.
10 This module allows you to access the OSS (Open Sound System) audio interface.
12 the standard audio interface for Linux and recent versions of FreeBSD.
18 majority of Linux audio apps anyway.
30 > * This is an OSS (Linux) audio emulator.
36 audio interface. That's the great thing about standards, there are so
74 Open an audio device and return an OSS audio device object. This object
77 read/write semantics and those of OSS audio devices). It also supports a number
78 of audio-specific methods; see below for the complete list of methods.
[all …]
|
| D | aifc.rst | 5 :synopsis: Read and write audio files in AIFF or AIFC format.
17 AIFF is Audio Interchange File Format, a format for storing digital audio
19 ability to compress the audio data.
21 Audio files have a number of parameters that describe the audio data. The
23 sampled. The number of channels indicate if the audio is mono, stereo, or
26 ``nchannels * samplesize`` bytes, and a second's worth of audio consists of
29 For example, CD quality audio has a sample size of two bytes (16 bits), uses two
59 Return the number of audio channels (1 for mono, 2 for stereo).
69 Return the sampling rate (number of audio frames per second).
74 Return the number of audio frames in the file.
[all …]
|
| /external/autotest/client/site_tests/audio_AudioCorruption/ |
| D | control | 7 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.
14 TEST_CLASS = "audio"
18 This test verifies Chrome can catch error for playing corrupted mp3 audio.
21 audio = 'http://commondatastorage.googleapis.com/chromiumos-test-assets-public/audio_AudioCorruptio…
22 job.run_test('audio_AudioCorruption', audio=audio)
|
| /external/tensorflow/tensorflow/core/api_def/base_api/ |
| D | api_def_AudioSummaryV2.pbtxt | 33 Max number of batch elements to generate audio for.
36 summary: "Outputs a `Summary` protocol buffer with audio."
38 The summary has up to `max_outputs` summary values containing audio. The
39 audio is built from `tensor` which must be 3-D with shape `[batch_size,
46 * If `max_outputs` is 1, the summary value tag is '*tag*/audio'.
48 generated sequentially as '*tag*/audio/0', '*tag*/audio/1', etc.
|