| /external/chromium_org/third_party/webrtc/voice_engine/include/ |
| voe_video_sync.h | 14 // - Playout delay tuning to synchronize the voice with video.
15 // - Playout delay monitoring.
58 // Gets the current sound card buffer size (playout delay).
65 // computes based on inter-arrival times and its playout mode.
68 // Sets an initial delay for the playout jitter buffer. The playout of the
72 // playout mode.
83 // playout mode. NetEq maintains this latency unless a higher value is
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| voe_hardware.h | 58 // Gets the number of audio devices available for playout.
67 // Gets the name of a specific playout device given by an |index|.
69 // (GUID) for the playout device.
77 // Sets the audio device used for playout.
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| voe_file.h | 62 // playout.
73 // playout.
112 // Starts recording the mixed playout audio.
118 // Stops recording the mixed playout audio.
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| voe_dtmf.h | 74 // Enables or disables local tone playout for received DTMF events
78 // Gets the DTMF playout status.
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| /external/icu/icu4c/source/samples/layout/ |
| pflow.h | 14 #include "layout/playout.h"
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| /external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/ |
| before_streaming_fixture.h | 38 // Stops all sending and playout.
41 // Resumes all sending and playout.
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| /external/icu/icu4c/source/layoutex/ |
| layoutex.vcxproj.filters | 24 <ClCompile Include="playout.cpp">
48 <CustomBuild Include="layout\playout.h">
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| audio_coding_module_impl.h | 144 // Get current playout frequency.
166 // Minimum playout delay.
169 // Maximum playout delay.
175 // Impose an initial delay on playout. ACM plays silence until |delay_ms|
179 // TODO(turajs): DTMF playout is always activated in NetEq these APIs should
182 // Configure Dtmf playout status i.e on/off playout the incoming outband Dtmf
186 // Get Dtmf playout status.
194 // Set playout mode voice, fax.
197 // Get playout mode voice, fax
[all...] |
| initial_delay_manager.h | 68 // Get playout timestamp.
85 // Update playout timestamps. While buffering, this is about
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| /external/chromium_org/media/cast/receiver/ |
| frame_receiver.h | 32 // FrameReceiver also includes logic for computing the playout time for each
33 // frame, accounting for a constant targeted playout delay. The purpose of the
34 // playout delay is to provide a fixed window of time between the capture event
35 // on the sender and the playout on the receiver. This is important because
92 // Computes the playout time for a frame with the given |rtp_timestamp|.
168 // Lip-sync values used to compute the playout time of each frame from its RTP
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| mediastreamprovider.h | 47 // Enable/disable the audio playout of a remote audio track with |ssrc|.
56 // Sets the audio playout volume of a remote audio track with |ssrc|.
71 // Enable/disable the video playout of a remote video track with |ssrc|.
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| APITest.h | 53 // Set Min delay, get delay, playout timestamp
59 // Playout Mode, background noise mode.
60 // Receiver Frequency, playout frequency.
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| /external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
| file_before_streaming_test.cc | 101 // 3. no output if playout is not started.
105 TEST_LOG("Playout is not started. File will not be played out.\n");
112 TEST_LOG("Playout is now started. File will be played out.\n");
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| dtmf_test.cc | 56 TEST_LOG("Disabling DTMF playout (no tone should be heard) \n");
61 TEST_LOG("Enabling DTMF playout (tone should be heard) \n");
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| file_test.cc | 60 webrtc::test::OutputPath() + "playout.wav";
62 TEST_LOG("Recording playout to %s.\n", recording_filename.c_str());
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| hardware_before_streaming_test.cc | 40 "Either you have no recording / playout device "
99 // Check playout side (see recording side test for more info on GUIDs).
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| /external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
| audio_track_jni.cc | 156 // and the max playout volume
210 "%s: Playout thread shutdown timed out, cannot "
317 " Playout already initialized");
346 // Try to initialize the playout side
373 " Playout already started");
380 " Playout device is not specified");
387 " Playout already initialized");
468 " Playout not initialized");
475 " Playout already started");
514 // Signal to playout thread that we want to star
[all...] |
| audio_track_jni.h | 27 const uint32_t N_PLAY_CHANNELS = 1; // default is mono playout
135 JNIEnv* _jniEnvPlay; // The JNI env for playout thread
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| /external/chromium_org/third_party/webrtc/voice_engine/test/android/android_test/src/org/webrtc/voiceengine/test/ |
| AndroidTest.java | 437 * false)) { WebrtcLog("VoE Recording Playout failed"); }
438 * WebrtcLog("VoE start Recording Playout end");
440 // Start playout
442 WebrtcLog("VoE start playout failed");
446 // Start playout file
449 // WebrtcLog("VoE start playout file failed");
510 // WebrtcLog("VoE stop Recording Playout failed");
512 // WebrtcLog("VoE stop Recording Playout ended");
520 // Stop playout file
522 // WebrtcLog("VoE stop playout file failed");
[all...] |
| /external/chromium_org/media/cast/logging/ |
| logging_defines.h | 68 // Render / playout delay. Only set for FRAME_PLAYOUT events.
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| /external/chromium_org/third_party/webrtc/modules/audio_device/test/ |
| README.txt | 8 Repeat this test for different selections of playout and recording devices.
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| /external/chromium_org/third_party/webrtc/modules/audio_device/ios/ |
| audio_device_ios.cc | 198 _id, " Playout device is not specified");
464 available = false; // Stereo playout not supported on iOS
475 " Stereo playout is not supported on this platform");
577 " Playout already initialized");
674 // Overrides the receiver playout route to speaker instead. See
724 // Try to initialize the playout side
767 " Playout already started");
773 " Playout already initialized");
779 " Playout device is not specified");
857 " Playout already initialized - InitPlayOrRecord() "
[all...] |
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| voe_hardware_impl.cc | 442 // Store state about activated playout to be able to restore it after the
443 // playout device has been modified.
447 "SetPlayoutDevice() device is modified while playout is "
453 "SetPlayoutDevice() unable to stop playout");
483 "SetPlayoutDevice() unable to set the playout device");
500 "SetPlayoutDevice() failed to set stereo playout mode");
503 // Restore playout if it was enabled already when calling this function.
510 "SetPlayoutDevice() playout is now being restored...");
515 "SetPlayoutDevice() failed to initialize playout");
522 "SetPlayoutDevice() failed to start playout");
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_sync_module.cc | 174 LOG(LS_ERROR) << "voe_sync_interface_ NULL, can't set playout delay.";
178 // Setting initial playout delay to voice engine (video engine is updated via
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/interface/ |
| audio_coding_module.h | 703 // Set a minimum for the playout delay, used for lip-sync. NetEq maintains
717 // Set a maximum for the playout delay
730 // is computed based on inter-arrival times and playout mode of NetEq. The
738 // Configure DTMF playout, i.e. whether out-of-band
742 // -enable : if true to enable playout out-of-band DTMF tones,
746 // -1 if the method fails, e.g. DTMF playout is not supported.
753 // Get Dtmf playout status.
757 // false if playout of Dtmf tones is disabled.
792 // Call this API to set the playout mode. Playout mode could be optimize
[all...] |
