chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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_INCLUDE_RTP_RTCP_DEFINES_H_
#define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_

#include <stddef.h>

#include <array>
#include <cstddef>
#include <cstdint>
#include <memory>
#include <utility>
#include <vector>

#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_format.h"
#include "api/rtp_headers.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h"
#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
#include "system_wrappers/include/clock.h"

#define RTCP_CNAME_SIZE …
#define IP_PACKET_SIZE …

namespace webrtc {
class RtpPacket;
class RtpPacketToSend;
namespace rtcp {
class TransportFeedback;
}

const int kVideoPayloadTypeFrequency = …;

// TODO(bugs.webrtc.org/6458): Remove this when all the depending projects are
// updated to correctly set rtp rate for RtcpSender.
const int kBogusRtpRateForAudioRtcp = …;

// Minimum RTP header size in bytes.
const uint8_t kRtpHeaderSize = …;

bool IsLegalMidName(absl::string_view name);
bool IsLegalRsidName(absl::string_view name);

// This enum must not have any gaps, i.e., all integers between
// kRtpExtensionNone and kRtpExtensionNumberOfExtensions must be valid enum
// entries.
enum RTPExtensionType : int { … };

enum RTCPAppSubTypes { … };

// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
enum RTCPPacketType : uint32_t { … };

enum class KeyFrameReqMethod : uint8_t { … };

enum RtxMode { … };

const size_t kRtxHeaderSize = …;

struct RtpState { … };

class RtcpIntraFrameObserver { … };

// Observer for incoming LossNotification RTCP messages.
// See the documentation of LossNotification for details.
class RtcpLossNotificationObserver { … };

// Interface to watch incoming rtcp packets related to the link in general.
// All message handlers have default empty implementation. This way users only
// need to implement the ones they are interested in.
// All message handles pass `receive_time` parameter, which is receive time
// of the rtcp packet that triggered the update.
class NetworkLinkRtcpObserver { … };

// NOTE! `kNumMediaTypes` must be kept in sync with RtpPacketMediaType!
static constexpr size_t kNumMediaTypes = …;
enum class RtpPacketMediaType : size_t { … };

struct RtpPacketSendInfo { … };

class NetworkStateEstimateObserver { … };

class TransportFeedbackObserver { … };

// Interface for PacketRouter to send rtcp feedback on behalf of
// congestion controller.
// TODO(bugs.webrtc.org/8239): Remove and use RtcpTransceiver directly
// when RtcpTransceiver always present in rtp transport.
class RtcpFeedbackSenderInterface { … };

class StreamFeedbackObserver { … };

class StreamFeedbackProvider { … };

class RtcpRttStats { … };

struct RtpPacketCounter { … };

// Data usage statistics for a (rtp) stream.
struct StreamDataCounters { … };

class RtpSendRates { … };

// Callback, called whenever byte/packet counts have been updated.
class StreamDataCountersCallback { … };

// Information exposed through the GetStats api.
struct RtpReceiveStats { … };

// Callback, used to notify an observer whenever new rates have been estimated.
class BitrateStatisticsObserver { … };

// Callback, used to notify an observer whenever a packet is sent to the
// transport.
class SendPacketObserver { … };

}  // namespace webrtc
#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_