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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}
74 lines
3.1 KiB
C++
74 lines
3.1 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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#include <stddef.h>
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#include <stdint.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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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "modules/rtp_rtcp/source/rtp_packetizer_av1.h"
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#include "rtc_base/checks.h"
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#include "test/fuzzers/fuzz_data_helper.h"
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namespace webrtc {
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void FuzzOneInput(const uint8_t* data, size_t size) {
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test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
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RtpPacketizer::PayloadSizeLimits limits;
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limits.max_payload_len = 1200;
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// Read uint8_t to be sure reduction_lens are much smaller than
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// max_payload_len and thus limits structure is valid.
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limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
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limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
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limits.single_packet_reduction_len =
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fuzz_input.ReadOrDefaultValue<uint8_t>(0);
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const VideoFrameType kFrameTypes[] = {VideoFrameType::kVideoFrameKey,
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VideoFrameType::kVideoFrameDelta};
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VideoFrameType frame_type = fuzz_input.SelectOneOf(kFrameTypes);
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// Main function under test: RtpPacketizerAv1's constructor.
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// "even distribution" is transitional and still exercises the other code path
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// so does not require another fuzzer.
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RtpPacketizerAv1 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
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limits, frame_type,
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/*is_last_frame_in_picture=*/true,
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/*even_distribution=*/true);
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size_t num_packets = packetizer.NumPackets();
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if (num_packets == 0) {
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return;
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}
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// When packetization was successful, validate NextPacket function too.
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// While at it, check that packets respect the payload size limits.
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RtpPacketToSend rtp_packet(nullptr);
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// Single packet.
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if (num_packets == 1) {
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RTC_CHECK(packetizer.NextPacket(&rtp_packet));
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RTC_CHECK_LE(rtp_packet.payload_size(),
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limits.max_payload_len - limits.single_packet_reduction_len);
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return;
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}
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// First packet.
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RTC_CHECK(packetizer.NextPacket(&rtp_packet));
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RTC_CHECK_LE(rtp_packet.payload_size(),
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limits.max_payload_len - limits.first_packet_reduction_len);
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// Middle packets.
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for (size_t i = 1; i < num_packets - 1; ++i) {
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RTC_CHECK(packetizer.NextPacket(&rtp_packet))
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<< "Failed to get packet#" << i;
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RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
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<< "Packet #" << i << " exceeds it's limit";
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}
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// Last packet.
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RTC_CHECK(packetizer.NextPacket(&rtp_packet));
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RTC_CHECK_LE(rtp_packet.payload_size(),
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limits.max_payload_len - limits.last_packet_reduction_len);
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}
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} // namespace webrtc
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