mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-19 16:47:50 +01:00
e1e94ad4c8
With this cl, packets that are discarded in RtpTransport now notifies Call, so that they can be part of BWE even if they are dropped. These packets have been recevied on the transport, and has bin decrypted and parsed and thus can be accounted for. The un demuxable packets are forwarded to Call similarly how RTCP packets are forwarded. Bug: webrtc:14928 Change-Id: Ia53349c7b316c4442a3c7aac085a85ec4f4ab9ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299262 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39727}
110 lines
4.2 KiB
C++
110 lines
4.2 KiB
C++
/*
|
|
* Copyright 2017 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 PC_RTP_TRANSPORT_INTERNAL_H_
|
|
#define PC_RTP_TRANSPORT_INTERNAL_H_
|
|
|
|
#include <string>
|
|
|
|
#include "call/rtp_demuxer.h"
|
|
#include "p2p/base/ice_transport_internal.h"
|
|
#include "pc/session_description.h"
|
|
#include "rtc_base/network_route.h"
|
|
#include "rtc_base/ssl_stream_adapter.h"
|
|
#include "rtc_base/third_party/sigslot/sigslot.h"
|
|
|
|
namespace rtc {
|
|
class CopyOnWriteBuffer;
|
|
struct PacketOptions;
|
|
} // namespace rtc
|
|
|
|
namespace webrtc {
|
|
|
|
// This class is an internal interface; it is not accessible to API consumers
|
|
// but is accessible to internal classes in order to send and receive RTP and
|
|
// RTCP packets belonging to a single RTP session. Additional convenience and
|
|
// configuration methods are also provided.
|
|
class RtpTransportInternal : public sigslot::has_slots<> {
|
|
public:
|
|
virtual ~RtpTransportInternal() = default;
|
|
|
|
virtual void SetRtcpMuxEnabled(bool enable) = 0;
|
|
|
|
virtual const std::string& transport_name() const = 0;
|
|
|
|
// Sets socket options on the underlying RTP or RTCP transports.
|
|
virtual int SetRtpOption(rtc::Socket::Option opt, int value) = 0;
|
|
virtual int SetRtcpOption(rtc::Socket::Option opt, int value) = 0;
|
|
|
|
virtual bool rtcp_mux_enabled() const = 0;
|
|
|
|
virtual bool IsReadyToSend() const = 0;
|
|
|
|
// Called whenever a transport's ready-to-send state changes. The argument
|
|
// is true if all used transports are ready to send. This is more specific
|
|
// than just "writable"; it means the last send didn't return ENOTCONN.
|
|
sigslot::signal1<bool> SignalReadyToSend;
|
|
|
|
// Called whenever an RTCP packet is received. There is no equivalent signal
|
|
// for demuxable RTP packets because they would be forwarded to the
|
|
// BaseChannel through the RtpDemuxer callback.
|
|
sigslot::signal2<rtc::CopyOnWriteBuffer*, int64_t> SignalRtcpPacketReceived;
|
|
|
|
// Called whenever a RTP packet that can not be demuxed by the transport is
|
|
// received.
|
|
sigslot::signal<const webrtc::RtpPacketReceived&>
|
|
SignalUnDemuxableRtpPacketReceived;
|
|
|
|
// Called whenever the network route of the P2P layer transport changes.
|
|
// The argument is an optional network route.
|
|
sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
|
|
|
|
// Called whenever a transport's writable state might change. The argument is
|
|
// true if the transport is writable, otherwise it is false.
|
|
sigslot::signal1<bool> SignalWritableState;
|
|
|
|
sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
|
|
|
|
virtual bool IsWritable(bool rtcp) const = 0;
|
|
|
|
// TODO(zhihuang): Pass the `packet` by copy so that the original data
|
|
// wouldn't be modified.
|
|
virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
|
|
const rtc::PacketOptions& options,
|
|
int flags) = 0;
|
|
|
|
virtual bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
|
|
const rtc::PacketOptions& options,
|
|
int flags) = 0;
|
|
|
|
// This method updates the RTP header extension map so that the RTP transport
|
|
// can parse the received packets and identify the MID. This is called by the
|
|
// BaseChannel when setting the content description.
|
|
//
|
|
// TODO(zhihuang): Merging and replacing following methods handling header
|
|
// extensions with SetParameters:
|
|
// UpdateRtpHeaderExtensionMap,
|
|
// UpdateSendEncryptedHeaderExtensionIds,
|
|
// UpdateRecvEncryptedHeaderExtensionIds,
|
|
// CacheRtpAbsSendTimeHeaderExtension,
|
|
virtual void UpdateRtpHeaderExtensionMap(
|
|
const cricket::RtpHeaderExtensions& header_extensions) = 0;
|
|
|
|
virtual bool IsSrtpActive() const = 0;
|
|
|
|
virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
|
|
RtpPacketSinkInterface* sink) = 0;
|
|
|
|
virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // PC_RTP_TRANSPORT_INTERNAL_H_
|