blob: eda47af568ac193fb196404d0cd5defc5e9f085f [file] [log] [blame]
* Copyright 2022 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.
#include <stddef.h>
#include <stdint.h>
#include <functional>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/audio_options.h"
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/video/recordable_encoded_frame.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/dscp.h"
#include "rtc_base/logging.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/socket.h"
#include "rtc_base/thread_annotations.h"
// This file contains the base classes for classes that implement
// the channel interfaces.
// These implementation classes used to be the exposed interface names,
// but this is in the process of being changed.
namespace cricket {
// The `MediaChannelUtil` class provides functionality that is used by
// multiple MediaChannel-like objects, of both sending and receiving
// types.
class MediaChannelUtil {
MediaChannelUtil(webrtc::TaskQueueBase* network_thread,
bool enable_dscp = false);
virtual ~MediaChannelUtil();
// Returns the absolute sendtime extension id value from media channel.
virtual int GetRtpSendTimeExtnId() const;
webrtc::Transport* transport() { return &transport_; }
// Base methods to send packet using MediaChannelNetworkInterface.
// These methods are used by some tests only.
bool SendPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options);
bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options);
int SetOption(MediaChannelNetworkInterface::SocketType type,
rtc::Socket::Option opt,
int option);
// Functions that form part of one or more interface classes.
// Not marked override, since this class does not inherit from the
// interfaces.
// Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
// Set to true if it's allowed to mix one- and two-byte RTP header extensions
// in the same stream. The setter and getter must only be called from
// worker_thread.
void SetExtmapAllowMixed(bool extmap_allow_mixed);
bool ExtmapAllowMixed() const;
void SetInterface(MediaChannelNetworkInterface* iface);
// Returns `true` if a non-null MediaChannelNetworkInterface pointer is held.
// Must be called on the network thread.
bool HasNetworkInterface() const;
bool DscpEnabled() const;
void SetPreferredDscp(rtc::DiffServCodePoint new_dscp);
// Implementation of the webrtc::Transport interface required
// by Call().
class TransportForMediaChannels : public webrtc::Transport {
TransportForMediaChannels(webrtc::TaskQueueBase* network_thread,
bool enable_dscp);
virtual ~TransportForMediaChannels();
// Implementation of webrtc::Transport
bool SendRtp(rtc::ArrayView<const uint8_t> packet,
const webrtc::PacketOptions& options) override;
bool SendRtcp(rtc::ArrayView<const uint8_t> packet) override;
// Not implementation of webrtc::Transport
void SetInterface(MediaChannelNetworkInterface* iface);
int SetOption(MediaChannelNetworkInterface::SocketType type,
rtc::Socket::Option opt,
int option);
bool DoSendPacket(rtc::CopyOnWriteBuffer* packet,
bool rtcp,
const rtc::PacketOptions& options);
bool HasNetworkInterface() const {
return network_interface_ != nullptr;
bool DscpEnabled() const { return enable_dscp_; }
void SetPreferredDscp(rtc::DiffServCodePoint new_dscp);
// This is the DSCP value used for both RTP and RTCP channels if DSCP is
// enabled. It can be changed at any time via `SetPreferredDscp`.
rtc::DiffServCodePoint PreferredDscp() const {
return preferred_dscp_;
// Apply the preferred DSCP setting to the underlying network interface RTP
// and RTCP channels. If DSCP is disabled, then apply the default DSCP
// value.
void UpdateDscp() RTC_RUN_ON(network_thread_);
int SetOptionLocked(MediaChannelNetworkInterface::SocketType type,
rtc::Socket::Option opt,
int option) RTC_RUN_ON(network_thread_);
const rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> network_safety_
webrtc::TaskQueueBase* const network_thread_;
const bool enable_dscp_;
MediaChannelNetworkInterface* network_interface_
RTC_GUARDED_BY(network_thread_) = nullptr;
rtc::DiffServCodePoint preferred_dscp_ RTC_GUARDED_BY(network_thread_) =
bool extmap_allow_mixed_ = false;
TransportForMediaChannels transport_;
} // namespace cricket