| /external/webrtc/src/modules/audio_processing/ |
| audio_processing_impl.cc | 45 AudioProcessingImpl* apm = new AudioProcessingImpl(id); local
46 if (apm->Initialize() != kNoError) {
47 delete apm;
48 apm = NULL;
51 return apm;
54 void AudioProcessing::Destroy(AudioProcessing* apm) {
55 delete static_cast<AudioProcessingImpl*>(apm);
|
| /hardware/libhardware_legacy/audio/ |
| audio_policy_hal.cpp | 50 AudioPolicyInterface *apm; member in struct:android_audio_legacy::legacy_audio_policy
70 return lap->apm->setDeviceConnectionState(
82 return (audio_policy_dev_state_t)lap->apm->getDeviceConnectionState(
91 lap->apm->setPhoneState((int) state);
107 lap->apm->setForceUse((AudioSystem::force_use)usage,
117 return (audio_policy_forced_cfg_t)lap->apm->getForceUse(
127 lap->apm->setSystemProperty("ro.camera.sound.forced", can_mute ? "0" : "1");
133 return lap->apm->initCheck();
147 return lap->apm->getOutput((AudioSystem::stream_type)stream,
157 return lap->apm->startOutput(output, (AudioSystem::stream_type)stream
[all...] |
| /hardware/qcom/audio/legacy/alsa_sound/ |
| audio_policy_hal.cpp | 50 AudioPolicyInterface *apm; member in struct:android_audio_legacy::qcom_audio_policy
70 return qap->apm->setDeviceConnectionState(
82 return (audio_policy_dev_state_t)qap->apm->getDeviceConnectionState(
91 qap->apm->setPhoneState((int) state);
107 qap->apm->setForceUse((AudioSystem::force_use)usage,
117 return (audio_policy_forced_cfg_t)qap->apm->getForceUse(
127 qap->apm->setSystemProperty("ro.camera.sound.forced", can_mute ? "0" : "1");
133 return qap->apm->initCheck();
146 return qap->apm->getOutput((AudioSystem::stream_type)stream,
155 return qap->apm->startOutput(output, (AudioSystem::stream_type)stream
[all...] |
| /external/webrtc/src/modules/audio_processing/test/ |
| process_test.cc | 144 AudioProcessing* apm = AudioProcessing::Create(0); local
145 ASSERT_TRUE(apm != NULL);
171 ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
201 ASSERT_EQ(apm->kNoError,
202 apm->set_sample_rate_hz(sample_rate_hz));
211 ASSERT_EQ(apm->kNoError,
212 apm->set_num_channels(num_capture_input_channels,
220 ASSERT_EQ(apm->kNoError,
221 apm->set_num_reverse_channels(num_render_channels))
[all...] |
| /frameworks/av/media/libeffects/preprocessing/ |
| PreProcessing.cpp | 104 webrtc::AudioProcessing* apm; // handle on webRTC audio processing module (APM) member in struct:preproc_session_s
106 uint32_t apmSamplingRate; // webRTC APM sampling rate (8/16 or 32 kHz)
281 webrtc::GainControl *agc = effect->session->apm->gain_control();
444 webrtc::EchoControlMobile *aec = effect->session->apm->echo_control_mobile();
469 *(uint32_t *)pValue = 1000 * effect->session->apm->stream_delay_ms();
489 status = effect->session->apm->set_stream_delay_ms(value/1000);
567 webrtc::NoiseSuppression *ns = effect->session->apm->noise_suppression();
761 session->apm = NULL;
778 session->apm = webrtc::AudioProcessing::Create(session->io)
[all...] |
