| /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()
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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()
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| /external/libwebm/webm_parser/tests/ |
| D | audio_parser_test.cc | 27 const Audio audio = parser_.value(); in TEST_F() local
29 EXPECT_FALSE(audio.sampling_frequency.is_present()); in TEST_F()
30 EXPECT_EQ(8000, audio.sampling_frequency.value()); in TEST_F()
32 EXPECT_FALSE(audio.output_frequency.is_present()); in TEST_F()
33 EXPECT_EQ(8000, audio.output_frequency.value()); in TEST_F()
35 EXPECT_FALSE(audio.channels.is_present()); in TEST_F()
36 EXPECT_EQ(static_cast<std::uint64_t>(1), audio.channels.value()); in TEST_F()
38 EXPECT_FALSE(audio.bit_depth.is_present()); in TEST_F()
39 EXPECT_EQ(static_cast<std::uint64_t>(0), audio.bit_depth.value()); in TEST_F()
59 const Audio audio = parser_.value(); in TEST_F() local
[all …]
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| /external/autotest/metadata/tests/ |
| D | audio.star | 13 'audio/Aconnect',
14 suites = ['audio'],
18 'audio/ActiveStreamStress',
23 'audio/Aplay',
28 'audio/AudioBasicAssistant',
33 'audio/AudioBasicBluetoothPlayback',
38 'audio/AudioBasicBluetoothPlaybackRecord',
43 'audio/AudioBasicBluetoothRecord',
48 'audio/AudioBasicExternalMicrophone',
53 'audio/AudioBasicHDMI',
[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/webrtc/test/pc/e2e/ |
| D | peer_connection_e2e_smoke_test.cc | 158 AudioConfig audio; in TEST_F() local
159 audio.stream_label = "alice-audio"; in TEST_F()
160 audio.mode = AudioConfig::Mode::kFile; in TEST_F()
161 audio.input_file_name = in TEST_F()
163 audio.sampling_frequency_in_hz = 48000; in TEST_F()
164 audio.sync_group = "alice-media"; in TEST_F()
165 alice->SetAudioConfig(std::move(audio)); in TEST_F()
174 AudioConfig audio; in TEST_F() local
175 audio.stream_label = "charlie-audio"; in TEST_F()
176 audio.mode = AudioConfig::Mode::kFile; in TEST_F()
[all …]
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| /external/webrtc/modules/audio_processing/ |
| D | echo_control_mobile_impl.cc | 128 const AudioBuffer* audio, in PackRenderAudioBuffer() argument
133 audio->num_frames_per_band()); in PackRenderAudioBuffer()
134 RTC_DCHECK_EQ(num_channels, audio->num_channels()); in PackRenderAudioBuffer()
140 for (size_t j = 0; j < audio->num_channels(); j++) { in PackRenderAudioBuffer()
142 FloatS16ToS16(audio->split_bands_const(render_channel)[kBand0To8kHz], in PackRenderAudioBuffer()
143 audio->num_frames_per_band(), data_to_buffer.data()); in PackRenderAudioBuffer()
148 data_to_buffer.data() + audio->num_frames_per_band()); in PackRenderAudioBuffer()
149 render_channel = (render_channel + 1) % audio->num_channels(); in PackRenderAudioBuffer()
160 int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio, in ProcessCaptureAudio() argument
163 RTC_DCHECK_GE(160, audio->num_frames_per_band()); in ProcessCaptureAudio()
[all …]
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| D | gain_control_impl.cc | 122 const AudioBuffer& audio, in PackRenderAudioBuffer() argument
124 RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, audio.num_frames_per_band()); in PackRenderAudioBuffer()
128 mixed_16_kHz_render_data.data(), audio.num_frames_per_band()); in PackRenderAudioBuffer()
129 if (audio.num_channels() == 1) { in PackRenderAudioBuffer()
130 FloatS16ToS16(audio.split_bands_const(0)[kBand0To8kHz], in PackRenderAudioBuffer()
131 audio.num_frames_per_band(), mixed_16_kHz_render_data.data()); in PackRenderAudioBuffer()
133 const int num_channels = static_cast<int>(audio.num_channels()); in PackRenderAudioBuffer()
134 for (size_t i = 0; i < audio.num_frames_per_band(); ++i) { in PackRenderAudioBuffer()
137 sum += FloatS16ToS16(audio.split_channels_const(kBand0To8kHz)[ch][i]); in PackRenderAudioBuffer()
146 (mixed_16_kHz_render.data() + audio.num_frames_per_band())); in PackRenderAudioBuffer()
[all …]
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| D | high_pass_filter.cc | 67 void HighPassFilter::Process(AudioBuffer* audio, bool use_split_band_data) { in Process() argument
68 RTC_DCHECK(audio); in Process()
69 RTC_DCHECK_EQ(filters_.size(), audio->num_channels()); in Process()
71 for (size_t k = 0; k < audio->num_channels(); ++k) { in Process()
73 audio->split_bands(k)[0], audio->num_frames_per_band()); in Process()
77 for (size_t k = 0; k < audio->num_channels(); ++k) { in Process()
79 rtc::ArrayView<float>(&audio->channels()[k][0], audio->num_frames()); in Process()
85 void HighPassFilter::Process(std::vector<std::vector<float>>* audio) { in Process() argument
86 RTC_DCHECK_EQ(filters_.size(), audio->size()); in Process()
87 for (size_t k = 0; k < audio->size(); ++k) { in Process()
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| D | voice_detection.cc | 66 bool VoiceDetection::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio() argument
68 audio->num_frames_per_band()); in ProcessCaptureAudio()
71 audio->num_frames_per_band()); in ProcessCaptureAudio()
72 if (audio->num_channels() == 1) { in ProcessCaptureAudio()
73 FloatS16ToS16(audio->split_bands_const(0)[kBand0To8kHz], in ProcessCaptureAudio()
74 audio->num_frames_per_band(), mixed_low_pass_data.data()); in ProcessCaptureAudio()
76 const int num_channels = static_cast<int>(audio->num_channels()); in ProcessCaptureAudio()
77 for (size_t i = 0; i < audio->num_frames_per_band(); ++i) { in ProcessCaptureAudio()
79 FloatS16ToS16(audio->split_channels_const(kBand0To8kHz)[0][i]); in ProcessCaptureAudio()
81 value += FloatS16ToS16(audio->split_channels_const(kBand0To8kHz)[j][i]); in ProcessCaptureAudio()
|
| /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()
|
| /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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| /external/webrtc/modules/audio_processing/ns/ |
| D | noise_suppressor_unittest.cc | 38 AudioBuffer* audio) { in PopulateInputFrameWithIdenticalChannels() argument
43 audio->split_bands(ch)[b][i] = (value > 0 ? 5000 * b + value : 0); in PopulateInputFrameWithIdenticalChannels()
52 const AudioBuffer& audio) { in VerifyIdenticalChannels() argument
57 EXPECT_EQ(audio.split_bands_const(ch)[b][i], in VerifyIdenticalChannels()
58 audio.split_bands_const(0)[b][i]); in VerifyIdenticalChannels()
78 AudioBuffer audio(rate, num_channels, rate, num_channels, rate, in TEST() local
84 audio.SplitIntoFrequencyBands(); in TEST()
88 frame_index, &audio); in TEST()
90 ns.Analyze(audio); in TEST()
91 ns.Process(&audio); in TEST()
[all …]
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| /external/oboe/samples/hello-oboe/ |
| D | README.md | 7 **Audio API:** Leave as "Unspecified" to select the best audio API for the current API level, or se…
9 …audio device, or leave as "Automatic" to use the default device. The list of audio devices is auto…
11 **Channel count:** Choose the number of audio channels to output. A different pitched sine wave wil…
13 …:** Choose the buffer size in bursts. A burst is an array of audio frames read by the audio device…
15 …audio stream between data entering the stream and it being presented to the audio device. This lat…
|
| /external/webrtc/audio/test/ |
| D | pc_low_bandwidth_audio_test.cc | 137 AudioConfig audio; in TEST() local
138 audio.stream_label = "alice-audio"; in TEST()
139 audio.mode = AudioConfig::Mode::kFile; in TEST()
140 audio.input_file_name = AudioInputFile(); in TEST()
141 audio.output_dump_file_name = AudioOutputFile(); in TEST()
142 audio.sampling_frequency_in_hz = absl::GetFlag(FLAGS_sample_rate_hz); in TEST()
143 alice->SetAudioConfig(std::move(audio)); in TEST()
163 AudioConfig audio; in TEST() local
164 audio.stream_label = "alice-audio"; in TEST()
165 audio.mode = AudioConfig::Mode::kFile; in TEST()
[all …]
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| /external/oboe/docs/ |
| D | OpenSLESMigration.md | 6 …ode from [OpenSL ES for Android](https://developer.android.com/ndk/guides/audio/opensl/opensl-for-…
13 …th an audio device capable of playing or recording audio samples. They also use a callback mechani…
17 …to reduce the amount of boilerplate code and guesswork associated with recording and playing audio.
25 OpenSL uses an audio engine object, created using `slCreateEngine`, to create other objects. Oboe's…
27 OpenSL uses audio player and audio recorder objects to communicate with audio devices. In Oboe an `…
29 …audio callback mechanism is a user-defined function which is called each time a buffer is enqueued…
75 This is a container array which you can read audio data from when recording, or write data into whe…
87 …audio stream by constructing an `AudioStreamDataCallback` object. [Here's an example.](https://git…
92 In OpenSL you cannot specify the size of the internal buffers of the audio player/recorder because …
94 …mation it has about the current audio device to configure its buffer size. It will determine the o…
[all …]
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| /external/webrtc/sdk/objc/native/src/audio/ |
| D | audio_device_ios.mm | 32 #import "components/audio/RTCAudioSession+Private.h"
33 #import "components/audio/RTCAudioSession.h"
34 #import "components/audio/RTCAudioSessionConfiguration.h"
35 #import "components/audio/RTCNativeAudioSessionDelegateAdapter.h"
152 // is not called until audio is about to start. However, it makes sense to
158 // Ensure that the audio device buffer (ADB) knows about the internal audio
159 // parameters. Note that, even if we are unable to get a mono audio session,
160 // we will always tell the I/O audio unit to do a channel format conversion
161 // to guarantee mono on the "input side" of the audio unit.
237 RTCLogError(@"StartPlayout failed to start audio unit.");
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| /external/oboe/samples/drumthumper/ |
| D | README.md | 6 **DrumThumper** is a "Drum Pad" app which demonstrates best-practices for low-latency audio playbac…
8 The audio samples are stored in application resources as WAV data. This is parsed and loaded (by ro…
9 The audio samples are mixed and played by routines in **iolib**.
15 * To demonstrate the most efficient means of playing audio with the lowest possible latency.
16 * To demonstrate how to play multiple sources of audio mixed into a single Oboe stream.
17 * To demonstrate the life-cycle of an Oboe audio stream and it's relationship to the application li…
21 * Using Android "assets" for audio data.
22 * The mechanism for calling native (C/C++) audio functionality from a Kotlin/Java app.
24 * A mechanism for parsing/loading one type (WAV) of audio data.
25 * How to control the relative levels (gain) of audio sources mixed into an output stream.
[all …]
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| /external/tensorflow/tensorflow/lite/experimental/microfrontend/python/kernel_tests/ |
| D | audio_microfrontend_op_test.py | 43 audio = tf.constant(
47 audio,
61 audio = tf.constant(
65 audio,
82 audio = tf.constant(
86 audio,
102 audio = tf.constant(
106 audio,
124 audio = tf.constant(
128 audio,
[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/oboe/samples/iolib/ |
| D | README.md | 3 Classes for playing audio data.
6 (Oboe) **iolib** contains classes implementing streaming audio playback and mixing from multiple so…
8 … chosen since it is presumed that this project will eventually implement audio capture capability.
14 Contains classes to support streaming playback from (potentially) multiple audio sources.
18 Declares the basic interface for audio data sources.
21 Extends the `DataSource` interface for audio data coming from SampleBuffer objects.
27 Loads and holds (in memory) audio sample data and provides read-only access to that data.
30 Implements an Oboe audio stream into which it mixes audio from some number of `SampleSource`s.
33 * Creation and lifetime management of an Oboe audio stream (`ManagedStream`)
|
| /external/autotest/test_suites/ |
| D | control.audio_essential | 5 AUTHOR = "The Chromium OS Authors,chromeos-audio-sw@google.com"
7 PURPOSE = "Suite for testing essential audio functionalities."
15 Audio tests that cover audio functionalities that are essential to the
16 Chrome OS audio stack.
18 Generally the tests require chameleon and audio boards connected.
19 Together with DUT and jack plugger bundled in audio-box environment for
21 go/chameleon-audio-conf and go/cras-test-green.
23 The audio boxes set up for this suites shouldn't be with USB audio peripherals
42 'chromeos-audio-bugs@google.com']
|