/external/ltp/testcases/kernel/device-drivers/v4l/user_space/ |
H A D | test_VIDIOC_ENUMAUDIO.c | 41 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 [all...] |
H A D | test_VIDIOC_AUDIO.c | 67 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 [all...] |
/external/vboot_reference/firmware/lib/include/ |
H A D | vboot_audio.h | 23 int VbAudioLooping(VbAudioContext *audio); 28 void VbAudioClose(VbAudioContext *audio);
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/external/vboot_reference/firmware/lib/ |
H A D | vboot_audio.c | 62 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 [all...] |
/external/webrtc/webrtc/modules/audio_processing/ |
H A D | level_estimator_impl.cc | 31 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());
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H A D | noise_suppression_impl.cc | 70 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 [all...] |
H A D | gain_control_impl.cc | 69 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 [all...] |
H A D | high_pass_filter_impl.cc | 104 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());
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H A D | echo_control_mobile_impl.cc | 93 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 [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
H A D | audio_checksum.h | 29 bool WriteArray(const int16_t* audio, size_t num_samples) override { 36 checksum_.Update(audio, num_samples * sizeof(*audio));
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H A D | output_audio_file.h | 37 bool WriteArray(const int16_t* audio, size_t num_samples) override { 39 return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
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H A D | audio_sink.h | 21 // 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);
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H A D | output_wav_file.h | 30 bool WriteArray(const int16_t* audio, size_t num_samples) override { 31 wav_writer_.WriteSamples(audio, num_samples);
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/external/webrtc/webrtc/modules/audio_processing/agc/ |
H A D | mock_agc.h | 23 MOCK_METHOD2(AnalyzePreproc, float(const int16_t* audio, size_t length)); 24 MOCK_METHOD3(Process, int(const int16_t* audio, size_t length,
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H A D | agc.cc | 42 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);
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H A D | agc.h | 30 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);
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/external/autotest/client/site_tests/audio_AlsaLoopback/ |
H A D | audio_AlsaLoopback.py | 8 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.
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/external/webrtc/webrtc/modules/audio_coding/codecs/ |
H A D | audio_encoder.cc | 28 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
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/external/autotest/client/site_tests/security_Minijail0/src/ |
H A D | test-usergroups | 7 needgroup=audio # a group chronos is in and root isn't
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/external/webrtc/talk/media/base/ |
H A D | audioframe.h | 41 AudioFrame(int16_t* audio, size_t audio_length, int sample_freq, bool stereo) argument 42 : audio10ms_(audio),
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/external/webrtc/webrtc/common_audio/vad/mock/ |
H A D | mock_vad.h | 26 enum Activity(const int16_t* audio,
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/external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
H A D | audio_encoder_pcm.cc | 82 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
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/external/webrtc/webrtc/tools/e2e_quality/audio/ |
H A D | audio_e2e_harness.cc | 11 // 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));
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/external/autotest/client/site_tests/audio_AudioCorruption/ |
H A D | audio_AudioCorruption.py | 14 """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.
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/external/webrtc/webrtc/common_audio/vad/ |
H A D | vad.cc | 28 Activity VoiceActivity(const int16_t* audio, 31 int ret = WebRtcVad_Process(handle_, sample_rate_hz, audio, num_samples);
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