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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/audio/audio_frame_processor.h | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/audio/audio_frame_processor.h')
-rw-r--r-- | third_party/libwebrtc/api/audio/audio_frame_processor.h | 43 |
1 files changed, 43 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio/audio_frame_processor.h b/third_party/libwebrtc/api/audio/audio_frame_processor.h new file mode 100644 index 0000000000..cb65c4817e --- /dev/null +++ b/third_party/libwebrtc/api/audio/audio_frame_processor.h @@ -0,0 +1,43 @@ +/* + * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_AUDIO_FRAME_PROCESSOR_H_ +#define API_AUDIO_AUDIO_FRAME_PROCESSOR_H_ + +#include <functional> +#include <memory> + +namespace webrtc { + +class AudioFrame; + +// If passed into PeerConnectionFactory, will be used for additional +// processing of captured audio frames, performed before encoding. +// Implementations must be thread-safe. +class AudioFrameProcessor { + public: + using OnAudioFrameCallback = std::function<void(std::unique_ptr<AudioFrame>)>; + virtual ~AudioFrameProcessor() = default; + + // Processes the frame received from WebRTC, is called by WebRTC off the + // realtime audio capturing path. AudioFrameProcessor must reply with + // processed frames by calling `sink_callback` if it was provided in SetSink() + // call. `sink_callback` can be called in the context of Process(). + virtual void Process(std::unique_ptr<AudioFrame> frame) = 0; + + // Atomically replaces the current sink with the new one. Before the + // first call to this function, or if the provided `sink_callback` is nullptr, + // processed frames are simply discarded. + virtual void SetSink(OnAudioFrameCallback sink_callback) = 0; +}; + +} // namespace webrtc + +#endif // API_AUDIO_AUDIO_FRAME_PROCESSOR_H_ |