webrtc/api/call/transport.h
Markus Handell c8c4a282a6 Introduce support for video packet batching.
This CL introduces a new feature enabling video packet send batches.
The feature is enabled via
PeerConnectionInterface
::RTCConfiguration
::MediaConfig
::enable_send_packet_batching.

PacketOptions have been augmented with attribute "batchable" (set for
all video packets) and attribute "last_packet_in_batch" which gives
injected AsyncPacketSockets a chance to understand when a batch begins
and ends.

When the feature is on, packets are collected in RtpSenderEgress. On
reception of OnBatchComplete from PacingController, RtpSenderEgress
sends the collected batch, setting "last_packet_in_batch" to true
in the last packet.

Bug: chromium:1439830
Change-Id: I1846b9d4a8a0efd227d617691213a2e048bdc8a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303720
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40012}
2023-05-08 16:24:03 +00:00

58 lines
1.7 KiB
C++

/*
* Copyright (c) 2013 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 API_CALL_TRANSPORT_H_
#define API_CALL_TRANSPORT_H_
#include <stddef.h>
#include <stdint.h>
#include "api/ref_counted_base.h"
#include "api/scoped_refptr.h"
namespace webrtc {
// TODO(holmer): Look into unifying this with the PacketOptions in
// asyncpacketsocket.h.
struct PacketOptions {
PacketOptions();
PacketOptions(const PacketOptions&);
~PacketOptions();
// A 16 bits positive id. Negative ids are invalid and should be interpreted
// as packet_id not being set.
int packet_id = -1;
// Additional data bound to the RTP packet for use in application code,
// outside of WebRTC.
rtc::scoped_refptr<rtc::RefCountedBase> additional_data;
// Whether this is a retransmission of an earlier packet.
bool is_retransmit = false;
bool included_in_feedback = false;
bool included_in_allocation = false;
// Whether this packet can be part of a packet batch at lower levels.
bool batchable = false;
// Whether this packet is the last of a batch.
bool last_packet_in_batch = false;
};
class Transport {
public:
virtual bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) = 0;
virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
protected:
virtual ~Transport() {}
};
} // namespace webrtc
#endif // API_CALL_TRANSPORT_H_