15 Commits

Author SHA1 Message Date
Philipp Hancke
3753c8190e h264: fix first_packet_in_frame logic for multislice in a single rtp packet
a frame must be (or should be) first when it contains either SPS (but not just PPS),
is an IDR or is a slice with first_mb_in_slice == 0.

Fixes an edge case where a STAP-A with SPS, PPS and multiple slices of an IDR fit
into a single RTP packet which can happen with small 320x196 frames

BUG=webrtc:352379280,webrtc:346608838

Change-Id: Ic6dea6c81db759d0d7ddd4054407103fd791f6c5
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357121
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42652}
2024-07-19 08:49:24 +00:00
Sergio Garcia Murillo
45e5e385f3 Use ArrayView on H264 bitstream parsing
No-Try: true
Bug: webrtc:42225170
Change-Id: I4682f400054fee5c86ea24bebf6d703fb90074da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42650}
2024-07-19 08:16:11 +00:00
Sergio Garcia Murillo
e19ce9b3db Fix is_first_packet_in_frame when receiving multiple slices per H264 frame
Bug: webrtc:346608838
Change-Id: I70ad3a952f37dde878f77d35c959c6973d283b9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354460
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42497}
2024-06-17 11:31:52 +00:00
Jeremy Leconte
a0b22af9e1 Revert "Temporary add 'RTPVideoHeaderH264::nalus_length'."
This reverts commit 04dd95fcac549fbdc330cee1de65074961db5934.

Reason for revert: code has been updated

Original change's description:
> Temporary add 'RTPVideoHeaderH264::nalus_length'.
>
> This is a forward fix for https://webrtc-review.googlesource.com/c/src/+/354622 that breaks client code using nalus_length.
>
> No-Try: true
> Change-Id: Ic0fc41696e408adefe4eb8792150a64b1eab49da
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354840
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Owners-Override: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#42493}

Bug: None
Change-Id: I1b65fe94ca07efdb8c7643e2ac46517050095018
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354860
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42496}
2024-06-17 11:08:33 +00:00
Jeremy Leconte
04dd95fcac Temporary add 'RTPVideoHeaderH264::nalus_length'.
This is a forward fix for https://webrtc-review.googlesource.com/c/src/+/354622 that breaks client code using nalus_length.

No-Try: true
Change-Id: Ic0fc41696e408adefe4eb8792150a64b1eab49da
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354840
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42493}
2024-06-17 08:07:16 +00:00
Sergio Garcia Murillo
469e69800f Remove kMaxNalusPerPacket hard limit for H264 frames
Bug: webrtc:346608838
Change-Id: I067401250994bc57897edff8e8a18c3088d96b08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354622
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42487}
2024-06-14 16:29:42 +00:00
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
Mirko Bonadei
e1607ed3a6 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}
2024-05-03 08:02:31 +00:00
Philipp Hancke
4344eb713b 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}
2024-05-02 16:15:18 +00:00
Byoungchan Lee
6c2ac2ea6b Fix math involving enums in C++20
(-Wdeprecated-anon-enum-enum-conversion)
- Replace enum with constexpr if necessary.
- Merge multiple definitions for H.264 NalDefs and FuDefs and apply
  constexpr.

Bug: chromium:1284275
Change-Id: I4a4d95ed6aba258e7c19c3ae6251c8b78caf84ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276561
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38215}
2022-09-27 06:55:31 +00:00
Danil Chapovalov
46cc32d89f Replace ABSL_FALLTHROUGH_INTENDED with c++17 attribute
the new spelling is more standard and more compact, in particular doesn't need extra include and thus dependency

Bug: None
Change-Id: Iaea69d2154e4d9eff2468514f5734cb3fe016ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35709}
2022-01-17 14:55:02 +00:00
Danil Chapovalov
e15dc58f32 Use rtc::CopyOnWriteBuffer::MutableData through webrtc
where mutable access is required.

Bug: webrtc:12334
Change-Id: I4b2b74f836aaf7f12278c3569d0d49936297716b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32936}
2021-01-11 11:31:33 +00:00
Danil Chapovalov
64f1f3f04e Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED
Bug: None
Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 15:20:35 +00:00
Danil Chapovalov
61d6471912 Change H264 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: If5169f47d85918356fa66e2bf3422d722044aa1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165581
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30264}
2020-01-15 12:26:55 +00:00
Danil Chapovalov
b42aeaa3fb Move RtpDepacketizerH264 into own files
Bug: webrtc:11152
Change-Id: Iab4975e9f378b177a2abf34559f9b74752e69843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165582
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30212}
2020-01-10 15:33:54 +00:00