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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/call/audio_sink.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/call/audio_sink.h')
-rw-r--r-- | third_party/libwebrtc/api/call/audio_sink.h | 48 |
1 files changed, 48 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/call/audio_sink.h b/third_party/libwebrtc/api/call/audio_sink.h new file mode 100644 index 0000000000..fec26593a6 --- /dev/null +++ b/third_party/libwebrtc/api/call/audio_sink.h @@ -0,0 +1,48 @@ +/* + * Copyright (c) 2015 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_CALL_AUDIO_SINK_H_ +#define API_CALL_AUDIO_SINK_H_ + +#include <stddef.h> +#include <stdint.h> + +namespace webrtc { + +// Represents a simple push audio sink. +class AudioSinkInterface { + public: + virtual ~AudioSinkInterface() {} + + struct Data { + Data(const int16_t* data, + size_t samples_per_channel, + int sample_rate, + size_t channels, + uint32_t timestamp) + : data(data), + samples_per_channel(samples_per_channel), + sample_rate(sample_rate), + channels(channels), + timestamp(timestamp) {} + + const int16_t* data; // The actual 16bit audio data. + size_t samples_per_channel; // Number of frames in the buffer. + int sample_rate; // Sample rate in Hz. + size_t channels; // Number of channels in the audio data. + uint32_t timestamp; // The RTP timestamp of the first sample. + }; + + virtual void OnData(const Data& audio) = 0; +}; + +} // namespace webrtc + +#endif // API_CALL_AUDIO_SINK_H_ |