webrtc_m130/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
Danil Chapovalov 8a5f807313 Reland "h264: bail out early when failing to parse SPS/PPS ids"
This reverts commit e1607ed3a619ae30cf8564ce401df5e03dd7bf4b.

Reason for revert: downstream project adjusted

Original change's description:
> Revert "h264: bail out early when failing to parse SPS/PPS ids"
>
> This reverts commit 4344eb713bb9a6d04d922d00fb492dfb31c9111f.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > h264: bail out early when failing to parse SPS/PPS ids
> >
> > This currently gets caught later in the process by the H264 SPS/PPS
> > tracker but can be rejected explicitly here. The network observable
> > behavior should be similar and request a key frame after a 200ms delay, at least for entities that send such bad bitstreams
> >
> > BUG=webrtc:337076010
> >
> > Change-Id: I239c64efa7db631460ef9e9986d283335303df5f
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349060
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Philipp Hancke <phancke@meta.com>
> > Cr-Commit-Position: refs/heads/main@{#42211}
>
> Bug: webrtc:337076010
> Change-Id: I15b815c69f1d25e41fb222d46359655242589fba
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349661
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42217}

Bug: webrtc:337076010
Change-Id: Ibe5a960b9b5fdf9a35e5dfffb47b78ade36b0cec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349700
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42223}
2024-05-03 11:33:45 +00:00

315 lines
12 KiB
C++

/*
* Copyright (c) 2020 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.
*/
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h"
#include <cstddef>
#include <cstdint>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "common_video/h264/h264_common.h"
#include "common_video/h264/pps_parser.h"
#include "common_video/h264/sps_parser.h"
#include "common_video/h264/sps_vui_rewriter.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_format_h264.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace {
constexpr size_t kNalHeaderSize = 1;
constexpr size_t kFuAHeaderSize = 2;
constexpr size_t kLengthFieldSize = 2;
constexpr size_t kStapAHeaderSize = kNalHeaderSize + kLengthFieldSize;
// TODO(pbos): Avoid parsing this here as well as inside the jitter buffer.
bool ParseStapAStartOffsets(const uint8_t* nalu_ptr,
size_t length_remaining,
std::vector<size_t>* offsets) {
size_t offset = 0;
while (length_remaining > 0) {
// Buffer doesn't contain room for additional nalu length.
if (length_remaining < sizeof(uint16_t))
return false;
uint16_t nalu_size = ByteReader<uint16_t>::ReadBigEndian(nalu_ptr);
nalu_ptr += sizeof(uint16_t);
length_remaining -= sizeof(uint16_t);
if (nalu_size > length_remaining)
return false;
nalu_ptr += nalu_size;
length_remaining -= nalu_size;
offsets->push_back(offset + kStapAHeaderSize);
offset += kLengthFieldSize + nalu_size;
}
return true;
}
absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
rtc::CopyOnWriteBuffer rtp_payload) {
const uint8_t* const payload_data = rtp_payload.cdata();
absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
absl::in_place);
bool modified_buffer = false;
parsed_payload->video_payload = rtp_payload;
parsed_payload->video_header.width = 0;
parsed_payload->video_header.height = 0;
parsed_payload->video_header.codec = kVideoCodecH264;
parsed_payload->video_header.simulcastIdx = 0;
parsed_payload->video_header.is_first_packet_in_frame = true;
auto& h264_header = parsed_payload->video_header.video_type_header
.emplace<RTPVideoHeaderH264>();
const uint8_t* nalu_start = payload_data + kNalHeaderSize;
const size_t nalu_length = rtp_payload.size() - kNalHeaderSize;
uint8_t nal_type = payload_data[0] & kH264TypeMask;
std::vector<size_t> nalu_start_offsets;
if (nal_type == H264::NaluType::kStapA) {
// Skip the StapA header (StapA NAL type + length).
if (rtp_payload.size() <= kStapAHeaderSize) {
RTC_LOG(LS_ERROR) << "StapA header truncated.";
return absl::nullopt;
}
if (!ParseStapAStartOffsets(nalu_start, nalu_length, &nalu_start_offsets)) {
RTC_LOG(LS_ERROR) << "StapA packet with incorrect NALU packet lengths.";
return absl::nullopt;
}
h264_header.packetization_type = kH264StapA;
nal_type = payload_data[kStapAHeaderSize] & kH264TypeMask;
} else {
h264_header.packetization_type = kH264SingleNalu;
nalu_start_offsets.push_back(0);
}
h264_header.nalu_type = nal_type;
parsed_payload->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
nalu_start_offsets.push_back(rtp_payload.size() +
kLengthFieldSize); // End offset.
for (size_t i = 0; i < nalu_start_offsets.size() - 1; ++i) {
size_t start_offset = nalu_start_offsets[i];
// End offset is actually start offset for next unit, excluding length field
// so remove that from this units length.
size_t end_offset = nalu_start_offsets[i + 1] - kLengthFieldSize;
if (end_offset - start_offset < H264::kNaluTypeSize) {
RTC_LOG(LS_ERROR) << "STAP-A packet too short";
return absl::nullopt;
}
NaluInfo nalu;
nalu.type = payload_data[start_offset] & kH264TypeMask;
nalu.sps_id = -1;
nalu.pps_id = -1;
start_offset += H264::kNaluTypeSize;
switch (nalu.type) {
case H264::NaluType::kSps: {
// Check if VUI is present in SPS and if it needs to be modified to
// avoid
// excessive decoder latency.
// Copy any previous data first (likely just the first header).
rtc::Buffer output_buffer;
if (start_offset)
output_buffer.AppendData(payload_data, start_offset);
absl::optional<SpsParser::SpsState> sps;
SpsVuiRewriter::ParseResult result = SpsVuiRewriter::ParseAndRewriteSps(
&payload_data[start_offset], end_offset - start_offset, &sps,
nullptr, &output_buffer, SpsVuiRewriter::Direction::kIncoming);
switch (result) {
case SpsVuiRewriter::ParseResult::kFailure:
RTC_LOG(LS_WARNING) << "Failed to parse SPS NAL unit.";
return absl::nullopt;
case SpsVuiRewriter::ParseResult::kVuiRewritten:
if (modified_buffer) {
RTC_LOG(LS_WARNING)
<< "More than one H264 SPS NAL units needing "
"rewriting found within a single STAP-A packet. "
"Keeping the first and rewriting the last.";
}
// Rewrite length field to new SPS size.
if (h264_header.packetization_type == kH264StapA) {
size_t length_field_offset =
start_offset - (H264::kNaluTypeSize + kLengthFieldSize);
// Stap-A Length includes payload data and type header.
size_t rewritten_size =
output_buffer.size() - start_offset + H264::kNaluTypeSize;
ByteWriter<uint16_t>::WriteBigEndian(
&output_buffer[length_field_offset], rewritten_size);
}
parsed_payload->video_payload.SetData(output_buffer.data(),
output_buffer.size());
// Append rest of packet.
parsed_payload->video_payload.AppendData(
&payload_data[end_offset],
nalu_length + kNalHeaderSize - end_offset);
modified_buffer = true;
[[fallthrough]];
case SpsVuiRewriter::ParseResult::kVuiOk:
RTC_DCHECK(sps);
nalu.sps_id = sps->id;
parsed_payload->video_header.width = sps->width;
parsed_payload->video_header.height = sps->height;
parsed_payload->video_header.frame_type =
VideoFrameType::kVideoFrameKey;
break;
}
break;
}
case H264::NaluType::kPps: {
uint32_t pps_id;
uint32_t sps_id;
if (PpsParser::ParsePpsIds(&payload_data[start_offset],
end_offset - start_offset, &pps_id,
&sps_id)) {
nalu.pps_id = pps_id;
nalu.sps_id = sps_id;
} else {
RTC_LOG(LS_WARNING)
<< "Failed to parse PPS id and SPS id from PPS slice.";
return absl::nullopt;
}
break;
}
case H264::NaluType::kIdr:
parsed_payload->video_header.frame_type =
VideoFrameType::kVideoFrameKey;
[[fallthrough]];
case H264::NaluType::kSlice: {
absl::optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
&payload_data[start_offset], end_offset - start_offset);
if (pps_id) {
nalu.pps_id = *pps_id;
} else {
RTC_LOG(LS_WARNING) << "Failed to parse PPS id from slice of type: "
<< static_cast<int>(nalu.type);
return absl::nullopt;
}
break;
}
// Slices below don't contain SPS or PPS ids.
case H264::NaluType::kAud:
case H264::NaluType::kEndOfSequence:
case H264::NaluType::kEndOfStream:
case H264::NaluType::kFiller:
case H264::NaluType::kSei:
break;
case H264::NaluType::kStapA:
case H264::NaluType::kFuA:
RTC_LOG(LS_WARNING) << "Unexpected STAP-A or FU-A received.";
return absl::nullopt;
}
if (h264_header.nalus_length == kMaxNalusPerPacket) {
RTC_LOG(LS_WARNING)
<< "Received packet containing more than " << kMaxNalusPerPacket
<< " NAL units. Will not keep track sps and pps ids for all of them.";
} else {
h264_header.nalus[h264_header.nalus_length++] = nalu;
}
}
return parsed_payload;
}
absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuaNalu(
rtc::CopyOnWriteBuffer rtp_payload) {
if (rtp_payload.size() < kFuAHeaderSize) {
RTC_LOG(LS_ERROR) << "FU-A NAL units truncated.";
return absl::nullopt;
}
absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
absl::in_place);
uint8_t fnri = rtp_payload.cdata()[0] & (kH264FBit | kH264NriMask);
uint8_t original_nal_type = rtp_payload.cdata()[1] & kH264TypeMask;
bool first_fragment = (rtp_payload.cdata()[1] & kH264SBit) > 0;
NaluInfo nalu;
nalu.type = original_nal_type;
nalu.sps_id = -1;
nalu.pps_id = -1;
if (first_fragment) {
absl::optional<uint32_t> pps_id =
PpsParser::ParsePpsIdFromSlice(rtp_payload.cdata() + 2 * kNalHeaderSize,
rtp_payload.size() - 2 * kNalHeaderSize);
if (pps_id) {
nalu.pps_id = *pps_id;
} else {
RTC_LOG(LS_WARNING)
<< "Failed to parse PPS from first fragment of FU-A NAL "
"unit with original type: "
<< static_cast<int>(nalu.type);
}
uint8_t original_nal_header = fnri | original_nal_type;
rtp_payload =
rtp_payload.Slice(kNalHeaderSize, rtp_payload.size() - kNalHeaderSize);
rtp_payload.MutableData()[0] = original_nal_header;
parsed_payload->video_payload = std::move(rtp_payload);
} else {
parsed_payload->video_payload =
rtp_payload.Slice(kFuAHeaderSize, rtp_payload.size() - kFuAHeaderSize);
}
if (original_nal_type == H264::NaluType::kIdr) {
parsed_payload->video_header.frame_type = VideoFrameType::kVideoFrameKey;
} else {
parsed_payload->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
}
parsed_payload->video_header.width = 0;
parsed_payload->video_header.height = 0;
parsed_payload->video_header.codec = kVideoCodecH264;
parsed_payload->video_header.simulcastIdx = 0;
parsed_payload->video_header.is_first_packet_in_frame = first_fragment;
auto& h264_header = parsed_payload->video_header.video_type_header
.emplace<RTPVideoHeaderH264>();
h264_header.packetization_type = kH264FuA;
h264_header.nalu_type = original_nal_type;
if (first_fragment) {
h264_header.nalus[h264_header.nalus_length] = nalu;
h264_header.nalus_length = 1;
}
return parsed_payload;
}
} // namespace
absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerH264::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
if (rtp_payload.size() == 0) {
RTC_LOG(LS_ERROR) << "Empty payload.";
return absl::nullopt;
}
uint8_t nal_type = rtp_payload.cdata()[0] & kH264TypeMask;
if (nal_type == H264::NaluType::kFuA) {
// Fragmented NAL units (FU-A).
return ParseFuaNalu(std::move(rtp_payload));
} else {
// We handle STAP-A and single NALU's the same way here. The jitter buffer
// will depacketize the STAP-A into NAL units later.
return ProcessStapAOrSingleNalu(std::move(rtp_payload));
}
}
} // namespace webrtc