Searched refs:audio (Results 1 - 25 of 212) sorted by relevance

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/external/ltp/testcases/kernel/device-drivers/v4l/user_space/
H A Dtest_VIDIOC_ENUMAUDIO.c41 struct v4l2_audio audio; local
47 memset(&audio, 0xff, sizeof(audio));
48 audio.index = i;
49 ret_enum = ioctl(get_video_fd(), VIDIOC_ENUMAUDIO, &audio);
58 CU_ASSERT_EQUAL(audio.index, i);
60 CU_ASSERT(0 < strlen((char *)audio.name));
62 ((char *)audio.name, sizeof(audio.name)));
64 //CU_ASSERT_EQUAL(audio
110 struct v4l2_audio audio; local
130 struct v4l2_audio audio; local
150 struct v4l2_audio audio; local
171 struct v4l2_audio audio; local
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H A Dtest_VIDIOC_AUDIO.c67 struct v4l2_audio audio; local
70 memset(&audio, 0xff, sizeof(audio));
71 ret_get = ioctl(get_video_fd(), VIDIOC_G_AUDIO, &audio);
80 //CU_ASSERT_EQUAL(audio.index, ?);
82 CU_ASSERT(0 < strlen((char *)audio.name));
83 CU_ASSERT(valid_string((char *)audio.name, sizeof(audio.name)));
85 CU_ASSERT(valid_audio_capability(audio.capability));
86 CU_ASSERT(valid_audio_mode(audio
129 struct v4l2_audio audio; local
166 struct v4l2_audio audio; local
296 struct v4l2_audio audio; local
353 struct v4l2_audio audio; local
410 struct v4l2_audio audio; local
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/external/vboot_reference/firmware/lib/include/
H A Dvboot_audio.h23 int VbAudioLooping(VbAudioContext *audio);
28 void VbAudioClose(VbAudioContext *audio);
/external/vboot_reference/firmware/lib/
H A Dvboot_audio.c62 static void VbGetDevMusicNotes(VbAudioContext *audio, int use_short) argument
85 if (!audio->background_beep)
192 audio->music_notes = notebuf;
193 audio->note_count = count;
194 audio->free_notes_when_done = 1;
200 audio->music_notes = builtin;
201 audio->note_count = count;
202 audio->free_notes_when_done = 0;
212 VbAudioContext *audio = &au; local
218 /* Calibrate audio dela
256 VbAudioLooping(VbAudioContext *audio) argument
295 VbAudioClose(VbAudioContext *audio) argument
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/external/webrtc/webrtc/modules/audio_processing/
H A Dlevel_estimator_impl.cc31 void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) { argument
32 RTC_DCHECK(audio);
38 for (size_t i = 0; i < audio->num_channels(); i++) {
39 rms_->Process(audio->channels_const()[i], audio->num_frames());
H A Dnoise_suppression_impl.cc70 void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { argument
71 RTC_DCHECK(audio);
78 RTC_DCHECK_GE(160u, audio->num_frames_per_band());
79 RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
82 audio->split_bands_const_f(i)[kBand0To8kHz]);
87 void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) { argument
88 RTC_DCHECK(audio);
94 RTC_DCHECK_GE(160u, audio->num_frames_per_band());
95 RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
99 audio
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H A Dgain_control_impl.cc69 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) { argument
75 assert(audio->num_frames_per_band() <= 160);
81 WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band());
88 render_queue_buffer_.end(), audio->mixed_low_pass_data(),
89 (audio->mixed_low_pass_data() + audio->num_frames_per_band()));
127 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { argument
134 assert(audio->num_frames_per_band() <= 160);
135 assert(audio->num_channels() == num_handles());
145 audio
179 ProcessCaptureAudio(AudioBuffer* audio) argument
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H A Dhigh_pass_filter_impl.cc104 void HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) { argument
105 RTC_DCHECK(audio);
111 RTC_DCHECK_GE(160u, audio->num_frames_per_band());
112 RTC_DCHECK_EQ(filters_.size(), audio->num_channels());
114 filters_[i]->Process(audio->split_bands(i)[kBand0To8kHz],
115 audio->num_frames_per_band());
H A Decho_control_mobile_impl.cc93 int EchoControlMobileImpl::ProcessRenderAudio(const AudioBuffer* audio) { argument
100 assert(audio->num_frames_per_band() <= 160);
101 assert(audio->num_channels() == apm_->num_reverse_channels());
108 for (size_t j = 0; j < audio->num_channels(); j++) {
111 my_handle, audio->split_bands_const(j)[kBand0To8kHz],
112 audio->num_frames_per_band());
119 audio->split_bands_const(j)[kBand0To8kHz],
120 (audio->split_bands_const(j)[kBand0To8kHz] +
121 audio->num_frames_per_band()));
167 int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) { argument
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/
H A Daudio_checksum.h29 bool WriteArray(const int16_t* audio, size_t num_samples) override {
36 checksum_.Update(audio, num_samples * sizeof(*audio));
H A Doutput_audio_file.h37 bool WriteArray(const int16_t* audio, size_t num_samples) override {
39 return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
H A Daudio_sink.h21 // Interface class for an object receiving raw output audio from test
28 // Writes |num_samples| from |audio| to the AudioSink. Returns true if
30 virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
44 // Forks the output audio to two AudioSink objects.
50 bool WriteArray(const int16_t* audio, size_t num_samples) override {
51 return left_sink_->WriteArray(audio, num_samples) &&
52 right_sink_->WriteArray(audio, num_samples);
H A Doutput_wav_file.h30 bool WriteArray(const int16_t* audio, size_t num_samples) override {
31 wav_writer_.WriteSamples(audio, num_samples);
/external/webrtc/webrtc/modules/audio_processing/agc/
H A Dmock_agc.h23 MOCK_METHOD2(AnalyzePreproc, float(const int16_t* audio, size_t length));
24 MOCK_METHOD3(Process, int(const int16_t* audio, size_t length,
H A Dagc.cc42 float Agc::AnalyzePreproc(const int16_t* audio, size_t length) { argument
46 if (audio[i] == 32767 || audio[i] == -32768)
52 int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) { argument
53 vad_.ProcessChunk(audio, length, sample_rate_hz);
H A Dagc.h30 virtual float AnalyzePreproc(const int16_t* audio, size_t length);
31 // |audio| must be mono; in a multi-channel stream, provide the first (usually
33 virtual int Process(const int16_t* audio, size_t length, int sample_rate_hz);
/external/autotest/client/site_tests/audio_AlsaLoopback/
H A Daudio_AlsaLoopback.py8 from autotest_lib.client.cros.audio import audio_helper
9 from autotest_lib.client.cros.audio import alsa_utils
10 from autotest_lib.client.cros.audio import cmd_utils
16 """Verifies audio playback and capture function."""
37 # Make sure the audio is still playing.
/external/webrtc/webrtc/modules/audio_coding/codecs/
H A Daudio_encoder.cc28 rtc::ArrayView<const int16_t> audio,
32 RTC_CHECK_EQ(audio.size(),
35 EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
26 Encode( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
/external/autotest/client/site_tests/security_Minijail0/src/
H A Dtest-usergroups7 needgroup=audio # a group chronos is in and root isn't
/external/webrtc/talk/media/base/
H A Daudioframe.h41 AudioFrame(int16_t* audio, size_t audio_length, int sample_freq, bool stereo) argument
42 : audio10ms_(audio),
/external/webrtc/webrtc/common_audio/vad/mock/
H A Dmock_vad.h26 enum Activity(const int16_t* audio,
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/
H A Daudio_encoder_pcm.cc82 rtc::ArrayView<const int16_t> audio,
88 speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
110 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, argument
113 return WebRtcG711_EncodeA(audio, input_len, encoded);
123 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, argument
126 return WebRtcG711_EncodeU(audio, input_len, encoded);
80 EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
/external/webrtc/webrtc/tools/e2e_quality/audio/
H A Daudio_e2e_harness.cc11 // Sets up a simple VoiceEngine loopback call with the default audio devices
36 VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe); local
37 ASSERT_TRUE(audio != NULL);
87 // Disable all audio processing.
88 ASSERT_EQ(0, audio->SetAgcStatus(false));
89 ASSERT_EQ(0, audio->SetEcStatus(false));
90 ASSERT_EQ(0, audio->EnableHighPassFilter(false));
91 ASSERT_EQ(0, audio->SetNsStatus(false));
/external/autotest/client/site_tests/audio_AudioCorruption/
H A Daudio_AudioCorruption.py14 """This test verifies playing corrupted audio in Chrome."""
17 def run_once(self, audio):
18 """Tests whether Chrome handles corrupted audio gracefully.
20 @param audio: Sample corrupted audio file to be played in Chrome.
26 os.path.join(self.bindir, 'audio.html')))
30 'loadSourceAndRunCorruptionTest("%s")' % audio)
32 # Expect corruption being detected after playing corrupted audio.
/external/webrtc/webrtc/common_audio/vad/
H A Dvad.cc28 Activity VoiceActivity(const int16_t* audio,
31 int ret = WebRtcVad_Process(handle_, sample_rate_hz, audio, num_samples);

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