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Reland "h264: bail out early when failing to parse SPS/PPS ids"

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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}
This commit is contained in:
Danil Chapovalov 2024-05-03 09:09:52 +00:00 committed by WebRTC LUCI CQ
parent b1a71aa7c3
commit 8a5f807313
2 changed files with 71 additions and 41 deletions
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74
modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
View File

@ -133,45 +133,47 @@ absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
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.";
}
if (result == 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);
}
// 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);
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;
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;
}
if (sps) {
parsed_payload->video_header.width = sps->width;
parsed_payload->video_header.height = sps->height;
nalu.sps_id = sps->id;
} else {
RTC_LOG(LS_WARNING) << "Failed to parse SPS id from SPS slice.";
}
parsed_payload->video_header.frame_type =
VideoFrameType::kVideoFrameKey;
break;
}
case H264::NaluType::kPps: {
@ -185,6 +187,7 @@ absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
} else {
RTC_LOG(LS_WARNING)
<< "Failed to parse PPS id and SPS id from PPS slice.";
return absl::nullopt;
}
break;
}
@ -200,6 +203,7 @@ absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
} else {
RTC_LOG(LS_WARNING) << "Failed to parse PPS id from slice of type: "
<< static_cast<int>(nalu.type);
return absl::nullopt;
}
break;
}

38
modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc
View File

@ -396,12 +396,6 @@ TEST(VideoRtpDepacketizerH264Test, TruncationJustAfterSingleStapANalu) {
EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
}
TEST(VideoRtpDepacketizerH264Test, ShortSpsPacket) {
const uint8_t kPayload[] = {0x27, 0x80, 0x00};
VideoRtpDepacketizerH264 depacketizer;
EXPECT_TRUE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
}
TEST(VideoRtpDepacketizerH264Test, SeiPacket) {
const uint8_t kPayload[] = {
kSei, // F=0, NRI=0, Type=6.
@ -421,5 +415,37 @@ TEST(VideoRtpDepacketizerH264Test, SeiPacket) {
EXPECT_EQ(h264.nalus[0].pps_id, -1);
}
TEST(VideoRtpDepacketizerH264Test, ShortSpsPacket) {
const uint8_t kPayload[] = {0x27, 0x80, 0x00};
VideoRtpDepacketizerH264 depacketizer;
EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
}
TEST(VideoRtpDepacketizerH264Test, BadSps) {
const uint8_t kPayload[] = {
kSps, 0x42, 0x41, 0x2a, 0xd3, 0x93, 0xd3, 0x3b // Payload.
};
VideoRtpDepacketizerH264 depacketizer;
EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
}
TEST(VideoRtpDepacketizerH264Test, BadPps) {
const uint8_t kPayload[] = {
kPps,
0x00 // Payload.
};
VideoRtpDepacketizerH264 depacketizer;
EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
}
TEST(VideoRtpDepacketizerH264Test, BadSlice) {
const uint8_t kPayload[] = {
kIdr,
0xc0 // Payload.
};
VideoRtpDepacketizerH264 depacketizer;
EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
}
} // namespace
} // namespace webrtc