webrtc/call/audio_sender.h
Tim Na b8c775aeaf Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
Bug: webrtc:11251
Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30238}
2020-01-13 18:31:30 +00:00

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/*
* 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 CALL_AUDIO_SENDER_H_
#define CALL_AUDIO_SENDER_H_
#include <memory>
#include "api/audio/audio_frame.h"
namespace webrtc {
class AudioSender {
public:
// Encode and send audio.
virtual void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) = 0;
virtual ~AudioSender() = default;
};
} // namespace webrtc
#endif // CALL_AUDIO_SENDER_H_