| /frameworks/base/telecomm/java/android/telecom/ |
| AudioState.aidl | 22 parcelable AudioState;
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| AudioState.java | 33 public class AudioState implements Parcelable {
60 public AudioState(boolean muted, int route, int supportedRouteMask) {
66 public AudioState(AudioState state) {
72 public AudioState(CallAudioState state) {
83 if (!(obj instanceof AudioState)) {
86 AudioState state = (AudioState) obj;
94 "[AudioState isMuted: %b, route: %s, supportedRouteMask: %s]",
130 * Responsible for creating AudioState objects for deserialized Parcels
[all...] |
| Phone.java | 44 * @param audioState The new {@link AudioState}.
49 public void onAudioStateChanged(Phone phone, AudioState audioState) { }
308 public final AudioState getAudioState() {
309 return new AudioState(mCallAudioState);
333 private void fireCallAudioStateChanged(CallAudioState audioState) {
335 listener.onCallAudioStateChanged(this, audioState);
336 listener.onAudioStateChanged(this, new AudioState(audioState));
[all...] |
| InCallService.java | 205 public void onAudioStateChanged(Phone phone, AudioState audioState) {
206 InCallService.this.onAudioStateChanged(audioState);
315 public final AudioState getAudioState() {
386 * @param audioState The new {@link AudioState}.
391 public void onAudioStateChanged(AudioState audioState) {
397 * @param audioState The new {@link CallAudioState}.
399 public void onCallAudioStateChanged(CallAudioState audioState) {
[all...] |
| CallAudioState.java | 87 public CallAudioState(AudioState state) {
109 "[AudioState isMuted: %b, route: %s, supportedRouteMask: %s]",
166 * Responsible for creating AudioState objects for deserialized Parcels.
194 * Writes AudioState object into a serializeable Parcel.
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| Conference.java | 231 public final AudioState getAudioState() {
232 return new AudioState(mCallAudioState);
323 public void onAudioStateChanged(AudioState state) {}
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| /external/webrtc/webrtc/ |
| audio_state.h | 26 // AudioState holds the state which must be shared between multiple instances of
28 class AudioState : public rtc::RefCountInterface {
32 // AudioState will tickle the VoE refcount to keep it alive for as long as
33 // the AudioState itself.
41 static rtc::scoped_refptr<AudioState> Create(
42 const AudioState::Config& config);
44 virtual ~AudioState() {}
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| call.h | 83 // AudioState which is possibly shared between multiple calls.
85 rtc::scoped_refptr<AudioState> audio_state;
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| /external/webrtc/webrtc/audio/ |
| audio_state.h | 24 class AudioState final : public webrtc::AudioState,
27 explicit AudioState(const AudioState::Config& config);
28 ~AudioState() override;
43 const webrtc::AudioState::Config config_;
56 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
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| audio_state.cc | 21 AudioState::AudioState(const AudioState::Config& config)
24 // Only one AudioState should be created per VoiceEngine.
28 AudioState::~AudioState() {
33 VoiceEngine* AudioState::voice_engine() {
38 bool AudioState::typing_noise_detected() const {
45 int AudioState::AddRef() const {
50 int AudioState::Release() const
[all...] |
| audio_state_unittest.cc | 29 AudioState::Config& config() { return config_; }
34 AudioState::Config config_;
40 rtc::scoped_refptr<AudioState> audio_state =
41 AudioState::Create(helper.config());
47 rtc::scoped_ptr<internal::AudioState> audio_state(
48 new internal::AudioState(helper.config()));
53 rtc::scoped_ptr<internal::AudioState> audio_state(
54 new internal::AudioState(helper.config()));
60 rtc::scoped_ptr<internal::AudioState> audio_state(
61 new internal::AudioState(helper.config()))
[all...] |
| audio_send_stream.h | 31 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
53 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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| audio_receive_stream.h | 33 const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
58 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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| audio_send_stream.cc | 61 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
201 internal::AudioState* audio_state =
202 static_cast<internal::AudioState*>(audio_state_.get());
214 internal::AudioState* audio_state =
215 static_cast<internal::AudioState*>(audio_state_.get());
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| audio_send_stream_unittest.cc | 67 AudioState::Config config;
69 audio_state_ = AudioState::Create(config);
108 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
158 rtc::scoped_refptr<AudioState> audio_state_;
235 internal::AudioState* internal_audio_state =
236 static_cast<internal::AudioState*>(helper.audio_state().get());
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| audio_receive_stream.cc | 86 const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
249 internal::AudioState* audio_state =
250 static_cast<internal::AudioState*>(audio_state_.get());
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| audio_receive_stream_unittest.cc | 78 AudioState::Config config;
80 audio_state_ = AudioState::Create(config);
119 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
159 rtc::scoped_refptr<AudioState> audio_state_;
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| /cts/suite/audio_quality/lib/include/audio/ |
| AudioLocal.h | 77 enum AudioState{
83 volatile AudioState mState;
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| /external/webrtc/talk/media/base/ |
| mediaengine.h | 78 virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
153 virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
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| /external/webrtc/webrtc/call/ |
| call_unittest.cc | 23 webrtc::AudioState::Config audio_state_config;
26 config.audio_state = webrtc::AudioState::Create(audio_state_config);
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| /packages/services/Telecomm/src/com/android/server/telecom/ |
| CallsManagerListenerBase.java | 19 import android.telecom.AudioState;
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| CallAudioRouteStateMachine.java | 44 * The eight states are all instances of the abstract base class, {@link AudioState}. Each state
48 * Messages are processed first by the processMessage method in the base class, AudioState.
49 * Any messages not completely handled by AudioState are further processed by the same method in
172 abstract class AudioState extends State {
363 abstract class EarpieceRoute extends AudioState {
540 abstract class HeadsetRoute extends AudioState {
734 abstract class BluetoothRoute extends AudioState {
[all...] |
| /external/webrtc/talk/media/webrtc/ |
| webrtcvoiceengine.h | 70 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
130 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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| /packages/services/Telephony/sip/src/com/android/services/telephony/sip/ |
| SipConnection.java | 23 import android.telecom.AudioState;
75 public void onAudioStateChanged(AudioState state) {
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| /external/webrtc/webrtc/test/ |
| call_test.cc | 49 AudioState::Config audio_state_config;
51 send_config.audio_state = AudioState::Create(audio_state_config);
57 AudioState::Config audio_state_config;
59 recv_config.audio_state = AudioState::Create(audio_state_config);
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