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Struct AudioFrame | MixedReality-WebRTC Documentation
https://microsoft.github.io/MixedReality-WebRTC/api/Microsoft.MixedReality.WebRTC.AudioFrame.html
Number of consecutive samples in the audio data buffer. WebRTC generally delivers frames in 10ms chunks, so for e.g. a 16 kHz sample rate the sample count would be 1000. Declaration public uint sampleCount Field Value | Improve this Doc View Source sampleRate Sample rate, in Hz. Generally in the range 8-48 kHz. Declaration public uint sampleRate
MixedReality-WebRTC/AudioFrame.cs at master - GitHub
https://github.com/microsoft/MixedReality-WebRTC/blob/master/libs/Microsoft.MixedReality.WebRTC/AudioFrame.cs/
/// Number of consecutive samples in the audio data buffer. /// WebRTC generally delivers frames in 10ms chunks, so for e.g. a 16 kHz /// sample rate the sample count would be 1000. /// </summary> public uint sampleCount; } /// <summary> /// Delegate used for events when an audio frame has been produced /// and is ready for consumption.
Struct AudioFrame | MixedReality-WebRTC Documentation
https://microsoft.github.io/MixedReality-WebRTC/versions/release/1.0/api/Microsoft.MixedReality.WebRTC.AudioFrame.html
Struct AudioFrame Single raw uncompressed audio frame. Namespace: Microsoft.MixedReality.WebRTC Assembly: Microsoft.MixedReality.WebRTC.dll Syntax. public struct AudioFrame. Remarks. The use of ref struct is an optimization to avoid heap allocation on each frame while having a nicer-to-use container to pass a frame accross methods.
Issue 2965203002: Let NetEq reset the AudioFrame ... - WebRTC
https://codereview.webrtc.org/2965203002
Issue 2965203002: Let NetEq reset the AudioFrame during muted state (Closed) Created 3 years, 5 months ago by hlundin-webrtc Modified 3 years, 5 months ago Reviewers: minyue-webrtc Base URL: Comments: 0
Breakout Box: How to generate timestamp for AudioFrame?
https://groups.google.com/g/discuss-webrtc/c/i65s454tcco
This specification issue https://github.com/WICG/web-codecs/issues/107 has not been resolved. I filed https://bugs.chromium.org/p/chromium/issues/detail?id=1185755 ...
Codecs used by WebRTC - Web media technologies | MDN
https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs
The WebRTC API makes it possible to construct web sites and apps that let users communicate in real time, using audio and/or video as well as optional data and other information. To communicate, the two devices need to be able to agree upon a mutually-understood codec for each track so they can successfully communicate and present the shared media. This guide …
c++ - Feeding input stream from PortAudio to webrtc ...
https://stackoverflow.com/questions/42609432/feeding-input-stream-from-portaudio-to-webrtcaudioprocessing
When I try to instantiate an AudioFrame for the first method like so: AudioFrame frame; I get the following error: main.cpp:161:22: error: aggregate ‘webrtc::AudioFrame frame’ has incomplete type and cannot be defined webrtc::AudioFrame frame; The second and third methods call for the data to be in the format "const float* const* src".
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