chromium/third_party/webrtc/audio/audio_send_stream.h

/*
 *  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 AUDIO_AUDIO_SEND_STREAM_H_
#define AUDIO_AUDIO_SEND_STREAM_H_

#include <memory>
#include <utility>
#include <vector>

#include "absl/functional/any_invocable.h"
#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "audio/audio_level.h"
#include "audio/channel_send.h"
#include "call/audio_send_stream.h"
#include "call/audio_state.h"
#include "call/bitrate_allocator.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "rtc_base/experiments/struct_parameters_parser.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/synchronization/mutex.h"

namespace webrtc {
class RtcpRttStats;
class RtpTransportControllerSendInterface;

struct AudioAllocationConfig { … };
namespace internal {
class AudioState;

class AudioSendStream final : public webrtc::AudioSendStream,
                              public webrtc::BitrateAllocatorObserver { … };
}  // namespace internal
}  // namespace webrtc

#endif  // AUDIO_AUDIO_SEND_STREAM_H_