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This is a reland of 49470c2ac4
Tentative reland to rule-out bot flakiness.
Original change's description:
> Add AV1 RtpDepacketizer class
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> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
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> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29814}
TBR=saza@webrtc.org,philipel@webrtc.org
Bug: webrtc:11042
Change-Id: Ibd672ce685bcab86960500740465539ed70fcdf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159941
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29819}
162 lines
5.9 KiB
C++
162 lines
5.9 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 "modules/rtp_rtcp/source/rtp_depacketizer_av1.h"
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#include <stddef.h>
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#include <stdint.h>
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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#include "rtc_base/byte_buffer.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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namespace {
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// AV1 format:
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//
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// RTP payload syntax:
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// 0 1 2 3 4 5 6 7
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// +-+-+-+-+-+-+-+-+
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// |Z|Y| W |-|-|-|-| (REQUIRED)
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// +=+=+=+=+=+=+=+=+ (REPEATED W-1 times, or any times if W = 0)
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// |1| |
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// +-+ OBU fragment|
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// |1| | (REQUIRED, leb128 encoded)
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// +-+ size |
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// |0| |
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// +-+-+-+-+-+-+-+-+
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// | OBU fragment |
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// | ... |
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// +=+=+=+=+=+=+=+=+
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// | ... |
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// +=+=+=+=+=+=+=+=+ if W > 0, last fragment MUST NOT have size field
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// | OBU fragment |
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// | ... |
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// +=+=+=+=+=+=+=+=+
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//
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//
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// OBU syntax:
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// 0 1 2 3 4 5 6 7
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// +-+-+-+-+-+-+-+-+
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// |0| type |X|S|-| (REQUIRED)
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// +-+-+-+-+-+-+-+-+
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// X: | TID |SID|-|-|-| (OPTIONAL)
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// +-+-+-+-+-+-+-+-+
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// |1| |
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// +-+ OBU payload |
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// S: |1| | (OPTIONAL, variable length leb128 encoded)
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// +-+ size |
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// |0| |
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// +-+-+-+-+-+-+-+-+
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// | OBU payload |
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// | ... |
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constexpr int kObuTypeSequenceHeader = 1;
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int ObuType(uint8_t obu_header) {
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return (obu_header & 0b0'1111'000u) >> 3;
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}
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bool RtpStartsWithFragment(uint8_t aggregation_header) {
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return aggregation_header & 0b1000'0000u;
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}
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bool RtpEndsWithFragment(uint8_t aggregation_header) {
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return aggregation_header & 0b0100'0000u;
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}
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int RtpNumObus(uint8_t aggregation_header) { // 0 for any number of obus.
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return (aggregation_header & 0b0011'0000u) >> 4;
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}
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} // namespace
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bool RtpDepacketizerAv1::Parse(ParsedPayload* parsed_payload,
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const uint8_t* payload_data,
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size_t payload_data_length) {
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RTC_DCHECK(parsed_payload);
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if (payload_data_length == 0) {
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RTC_DLOG(LS_ERROR) << "Empty rtp payload.";
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return false;
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}
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// To assemble frame, all of the rtp payload is required, including
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// aggregation header.
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parsed_payload->payload = payload_data;
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parsed_payload->payload_length = payload_data_length;
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rtc::ByteBufferReader payload(reinterpret_cast<const char*>(payload_data),
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payload_data_length);
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uint8_t aggregation_header;
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RTC_CHECK(payload.ReadUInt8(&aggregation_header));
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// TODO(danilchap): Set AV1 codec when there is such enum value
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parsed_payload->video.codec = VideoCodecType::kVideoCodecGeneric;
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// These are not accurate since frame may consist of several packet aligned
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// chunks of obus, but should be good enough for most cases. It might produce
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// frame that do not map to any real frame, but av1 decoder should be able to
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// handle it since it promise to handle individual obus rather than full
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// frames.
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parsed_payload->video.is_first_packet_in_frame =
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!RtpStartsWithFragment(aggregation_header);
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parsed_payload->video.is_last_packet_in_frame =
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!RtpEndsWithFragment(aggregation_header);
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parsed_payload->video.frame_type = VideoFrameType::kVideoFrameDelta;
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// If packet starts a frame, check if it contains Sequence Header OBU.
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// In that case treat it as key frame packet.
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if (parsed_payload->video.is_first_packet_in_frame) {
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int num_expected_obus = RtpNumObus(aggregation_header);
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// The only OBU that can preceed SequenceHeader is a TemporalDelimiter OBU,
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// so check no more than two OBUs while searching for SH.
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for (int obu_index = 1; payload.Length() > 0 && obu_index <= 2;
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++obu_index) {
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uint64_t fragment_size;
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// When num_expected_obus > 0, last OBU (fragment) is not preceeded by
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// the size field. See W field in
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// https://aomediacodec.github.io/av1-rtp-spec/#43-av1-aggregation-header
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bool has_fragment_size = (obu_index != num_expected_obus);
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if (has_fragment_size) {
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if (!payload.ReadUVarint(&fragment_size)) {
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RTC_DLOG(LS_WARNING)
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<< "Failed to read OBU fragment size for OBU#" << obu_index;
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return false;
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}
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if (fragment_size > payload.Length()) {
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RTC_DLOG(LS_WARNING) << "OBU fragment size " << fragment_size
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<< " exceeds remaining payload size "
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<< payload.Length() << " for OBU#" << obu_index;
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// Malformed input: written size is larger than remaining buffer.
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return false;
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}
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} else {
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fragment_size = payload.Length();
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}
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// Though it is inpractical to pass empty fragments, it is allowed.
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if (fragment_size == 0) {
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RTC_LOG(LS_WARNING)
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<< "Weird obu of size 0 at offset "
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<< (payload_data_length - payload.Length()) << ", skipping.";
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continue;
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}
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uint8_t obu_header = *reinterpret_cast<const uint8_t*>(payload.Data());
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if (ObuType(obu_header) == kObuTypeSequenceHeader) {
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// TODO(bugs.webrtc.org/11042): Check frame_header OBU and/or frame OBU
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// too for other conditions of the start of a new coded video sequence.
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// For proper checks checking single packet might not be enough. See
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// https://aomediacodec.github.io/av1-spec/av1-spec.pdf section 7.5
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parsed_payload->video.frame_type = VideoFrameType::kVideoFrameKey;
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break;
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}
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payload.Consume(fragment_size);
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}
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}
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return true;
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}
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} // namespace webrtc
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