chromium/third_party/webrtc/audio/channel_send.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 AUDIO_CHANNEL_SEND_H_
#define AUDIO_CHANNEL_SEND_H_

#include <memory>
#include <string>
#include <vector>

#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/crypto/crypto_options.h"
#include "api/environment/environment.h"
#include "api/frame_transformer_interface.h"
#include "api/function_view.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "modules/rtp_rtcp/source/rtp_sender_audio.h"

namespace webrtc {

class FrameEncryptorInterface;
class RtpTransportControllerSendInterface;

struct CallSendStatistics { … };

namespace voe {

class ChannelSendInterface { … };

std::unique_ptr<ChannelSendInterface> CreateChannelSend(
    const Environment& env,
    Transport* rtp_transport,
    RtcpRttStats* rtcp_rtt_stats,
    FrameEncryptorInterface* frame_encryptor,
    const webrtc::CryptoOptions& crypto_options,
    bool extmap_allow_mixed,
    int rtcp_report_interval_ms,
    uint32_t ssrc,
    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
    RtpTransportControllerSendInterface* transport_controller);

}  // namespace voe
}  // namespace webrtc

#endif  // AUDIO_CHANNEL_SEND_H_