chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

/*
 *  Copyright (c) 2012 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 MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_

#include <stddef.h>
#include <stdint.h>

#include <memory>

#include "absl/strings/string_view.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
#include "modules/rtp_rtcp/source/dtmf_queue.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "rtc_base/one_time_event.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"

namespace webrtc {

class RTPSenderAudio { … };

}  // namespace webrtc

#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_