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acfd279a14
Implements a two-pass approach to packetization which creates
packets of an even size similar to RtpPacketizer::SplitAboutEqually.
This improves the bandwidth estimation.
The algorithm does a first pass with the existing packetizer, then
iterates through the resulting packet sizes and sums up the bytes left unused in each packet.
It then calculates a new maximum packet length as
configured_max_packet_len - ((unused_bytes - packets + 1) / packets)
adjusts for the overhead and re-runs the packetization algorithm.
For example, a list of OBUs with sizes
{1206, 1476, 1431}
currently gets packetized greedily as payload sizes
{1200, 1200, 1200, 523}
With this change, it gets packetized as
{1032, 1032, 1032, 1028}
This change is guarded by the field trial
WebRTC-Video-AV1EvenPayloadSizes
which is acting as a rollout flag.
BUG=webrtc:15927
Co-authored-by: Shyam Sadhwani <shyamsadhwani@meta.com>
Change-Id: I4f0b3c27de6f06104908dd769c4dd1f34115712c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348100
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42203}
78 lines
2.7 KiB
C++
78 lines
2.7 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_AV1_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_AV1_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <vector>
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#include "api/array_view.h"
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#include "api/video/video_frame_type.h"
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#include "modules/rtp_rtcp/source/rtp_format.h"
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namespace webrtc {
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class RtpPacketizerAv1 : public RtpPacketizer {
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public:
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RtpPacketizerAv1(rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits,
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VideoFrameType frame_type,
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bool is_last_frame_in_picture,
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bool even_distribution);
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~RtpPacketizerAv1() override = default;
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size_t NumPackets() const override { return packets_.size() - packet_index_; }
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bool NextPacket(RtpPacketToSend* packet) override;
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private:
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struct Obu {
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uint8_t header;
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uint8_t extension_header; // undefined if (header & kXbit) == 0
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rtc::ArrayView<const uint8_t> payload;
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int size; // size of the header and payload combined.
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};
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struct Packet {
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explicit Packet(int first_obu_index) : first_obu(first_obu_index) {}
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// Indexes into obus_ vector of the first and last obus that should put into
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// the packet.
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int first_obu;
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int num_obu_elements = 0;
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int first_obu_offset = 0;
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int last_obu_size;
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// Total size consumed by the packet.
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int packet_size = 0;
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};
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// Parses the payload into serie of OBUs.
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static std::vector<Obu> ParseObus(rtc::ArrayView<const uint8_t> payload);
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// Returns the number of additional bytes needed to store the previous OBU
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// element if an additonal OBU element is added to the packet.
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static int AdditionalBytesForPreviousObuElement(const Packet& packet);
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// Packetize and try to distribute the payload evenly across packets.
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static std::vector<Packet> PacketizeAboutEqually(
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rtc::ArrayView<const Obu> obus,
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PayloadSizeLimits limits);
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static std::vector<Packet> Packetize(rtc::ArrayView<const Obu> obus,
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PayloadSizeLimits limits);
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uint8_t AggregationHeader() const;
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const VideoFrameType frame_type_;
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const std::vector<Obu> obus_;
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const std::vector<Packet> packets_;
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const bool is_last_frame_in_picture_;
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size_t packet_index_ = 0;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_AV1_H_
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