Use meson built in support for dependency to threads lib #10

Revisit when we do all the final changes around rtpsession and writing of TWCC seqnums.

Be a bit more consistent with how we log ssrcs

ptdemux can't know which of the payloadtypes the lost-event is actually meant for, and taking a guess of it being the "previous" one is going to be wrong in many cases.

And some cleanup/warning fixes.

...it crashes our libvpx...

Need to sort out this mess later on!

It will take the GstVideoRegionOfInterestMeta, and turn one of its specified types into a RTP header-extension. Co-authored-by: Camilo Celis Guzman <[email protected]>

Basically, all frames are "i-frames" here.

Can't have a mix of GstValueArray and GValueArray here.

As per https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2

The state change function will release the lock before emitting the property change event and relock afterwards. This simplifies the flow of calling code dramatically. Co-authored-by: John-Mark Bell <[email protected]>

Co-authored-by: John-Mark Bell <[email protected]>

Additionally, reduce the numbers.

Co-authored-by: John-Mark Bell <[email protected]>

commit a8b63ae927b6dddeeeae5d83803a0e8e1c7b2a3f Author: John Bassett <[email protected]> Date: Tue May 21 18:23:05 2019 +0100 Fix deadlock during shutdown Deadlock was caused by one thread calling close on a socket whilst another thread was blocked on a call to recvmsg. In this scenario the recv thread would wake up after the recvmsg had timed out but then attempt another read from the now closed socket. Normally this would result in an error, but if the file descriptor had been reused in the meantime the recvmsg call could end up blocking forever trying to read from the wrong fd. - Make sure the socket fds used are reset to -1 before closing to prevent other threads attempting to read from a socket that has already been closed. - Call shutdown on sockets before closing to guarantee other threads reading from the socket are unblocked immediately. - Check for return of 0 when reading from a socket to handle sockets that have been cleanly closed from another thread.

commit 3a3cb88c388919b9348b828becb8576a580eccc4 Author: John-Mark Bell <[email protected]> Date: Thu Aug 15 22:22:59 2019 +0100 Forcibly cancel iterator thread

commit 7d050e2140efd73b408eecac5f4cede116b7f1f1 Author: John-Mark Bell <[email protected]> Date: Sat Aug 17 19:15:30 2019 +0100 Remove iterator thread This is not necessary a world using AF_CONN only. Fix the one place where an iterator would be used to clean up directly in this case.

…m as conflicts with cPython bindings' commit ecf2de543c3157e507e9fc2e7c8319bc443ede39 Author: Thomas Williams <[email protected]> Date: Thu Jun 17 14:36:26 2021 +0100 init_random rename to init_usrsctp_random as conflicts with cPython bindings

Co-authored-by: John-Mark Bell <[email protected]>

Changes lost in monorepo merge; originally from 7bbcd29

Although 60 seconds is often enough, we have seen cases where the harnessed pipeline could taken longer than that to process a buffer. This is especially true when we run under additional tools like Valgrind or Debug builds.

For "limiting" this means using the token bucket algorithm, allowing a sender to control the bitrate being consumed by rtx-packets, in case of an "out-of-control" receiver asking for way too much packets. 'max-kbps' sets the maximum kilobit per second that the RTX element will give out based on requests, limiting the ability for a receiver to congest the available network bandwidth. 'max-bucket-size' is related to the token bucket algorithm, and will allow control of the "burstiness" of the RTX requests, preventing a huge spike in the network. For "stuffing" this means adding RTX packets as stuffing to meet the total target bitrate 'stuffing-kbps'. There are cases when output rate sent on the network needs to be near constant, but the media isn't (Encoders producing variable bitrates) Also, for probing the network for available bitrate, the stuffing can be used to create redundant information at a certain bitrate. Co-authored-by: Tulio Beloqui <[email protected]> Co-authored-by: Stian Selnes <[email protected]>

Squashed commit of the following: commit c2e3ed1 Author: Camilo Celis Guzman <[email protected]> Date: Tue Nov 2 22:30:12 2021 +0900 rtphdrextroi: correct max. number of roi-types commit ac3c6c4 Author: Camilo Celis Guzman <[email protected]> Date: Fri Oct 29 23:00:14 2021 +0900 rtphdrextroi: add test for multiple roi-types commit 1fd0bc8 Author: Camilo Celis Guzman <[email protected]> Date: Thu Oct 28 19:42:04 2021 +0900 rtphdrextroi: WIP: multiple roi-types support commit 4b9f2d8 Author: Camilo Celis Guzman <[email protected]> Date: Thu Oct 28 17:19:57 2021 +0900 rtphdrextroi: add test for case of multiple RoI metas with the same roi-type The expected behaviour is that only one of them is payloaded commit b17f460 Author: Camilo Celis Guzman <[email protected]> Date: Thu Oct 28 17:03:55 2021 +0900 rtphdrextroi: make roi-type construct only

Fix suppresion to support release and debug builds. Here is debug build call stack: ``` ==10707== by 0x48B5520: g_malloc (gmem.c:106) ==10707== by 0x48D19DC: g_slice_alloc (gslice.c:1069) ==10707== by 0x48D3947: g_slist_copy_deep (gslist.c:619) ==10707== by 0x48D38B8: g_slist_copy (gslist.c:567) ==10707== by 0x4ADC90B: gst_debug_remove_with_compare_func (gstinfo.c:1504) ``` In release build `g_slist_copy (gslist.c:567)` got inlined: ``` ==15419== by 0x48963E0: g_malloc (gmem.c:106) ==15419== by 0x48AA382: g_slice_alloc (gslice.c:1069) ==15419== by 0x48AB732: g_slist_copy_deep (gslist.c:619) ==15419== by 0x4A39B8F: gst_debug_remove_with_compare_func (gstinfo.c:1504) ```

doing a sync point request all_headers set to FALSE stops the FIR request in gstrtpsession.

Prevents logical error as if defined as a stack variable it could contain garbage data if uninitialized.

disabled If the checks are not built, it will leave behind some "unused" variables, so this warnings will always trigger.

This allows user-level signal handler to catch this Co-authored-by: Frederik Vestre <[email protected]>

Enables device enumeration and monitor using the ACameraManager API

The association can be NULL if the element was never started, or if the association ID was already in use. Be symmetrical with SCTPEnc and check this.

execute regardless of the provider being stopped.

Calling destroy_session() without first calling finish() deadlocks almost every time, due to the explanation given in gst_vtenc_finish_encoding()

…ze() - Decrementing number_of_associations in dispose() can be troublesome, as as per documentation, the dispose function can be called many times. - Also removed extra check if dispose is present, as parent class (GObject) ensures it is available.

supported

Remove this before releasing please

Also specify the alignment as part of caps always.

These made one of our vpx test fail. GST_DEBUG=*vp*:LOG GST_CHECKS=vpx_enc_rate_limit_key_frames python3 /pexip/external/meson/meson.py test vpx -vv Look into why and how more closely!

The notify is already on a timer, and notifying again outside this breaks the contract of not receiving more stats.

The initialization of the iterators locks depended on SCTP_ITERATOR definition, but sctp_pcb_finish() calls destroy on them without that check. Note that also these locks are accessed without any check on SCTP_ITERATOR so doing the initialization here fixes that problem too.

- Move the cast macro to do a basic cast instead of check gtype instance as that one access a global lock. - Configure and cleanup the sctp association only on state change from NULL->READY and READY->NULL. - Moved the start/stop of the srcpad task in the activate_mode function, rather than the state change. This fixes the task thread accessing invalid data. Co-Authored-By: Håvard Graff <[email protected]>

gst_sctp_association_get can call g_object_ref on a destroyed sctp association because the unref calls do not hold the associations_lock. To solve this we add an API to unref the association while holding the lock. Moved the calls to usrsctp_init/finish to get/unref() respectively to not depend on GObjects behavior, keep the counting of the associations in the init/finalize. Also added g_hash_table_destroy() calls to the association hash table after the last association is destroyed.

gst_element_add_pad() - Fixes a deadlock in which we will hold the lock inside gst_element_add_pad() and then later in gst_element_post_message_default(). This is triggered by calling gst_pad_start_task() from the pads activate_mode() function. (cherry picked from commit 22cc976)

This fixes a potential NULL access on the pads parent inside gst_sctp_data_srcpad_loop (cherry picked from commit fc409b9)

…Provider

Just for convenience

…r getting the clock-rate" This reverts commit 608b4ee.

RTP caps from payloaders use seqnum-offset, hence the jitterbuffer should use the same. Same for rtspsrc.

Given a constant source of jitter, the total-avg-jitter will be the "correct" amount of jitter experienced. Comparing this number to the running average for many long-running tests, having a slightly larger window seems to approximate the jitter much better. In addition we avoid weighting 0 too much when the initial calculations are done by ramping up the window and not counting the very first measurement.

To avoid the GQuark global lock per entry.

As demonstrated by the test, you can end up in situations where your packet-rate gets insanly high, and you produce thousands of RTX-timers /lost events, when the big-gap logic is there to prevent exactly that.

So what happens here is that at some point we receive packet #10508, which triggers the big gap logic, placing that buffer inside the gap_packets GList: gst_rtp_jitter_buffer_chain: expected #10529, got #10508, gap of -21 handle_big_gap_buffer: buffer too old -21 < -20, first one - waiting Then packet #10550 arrives: gst_rtp_jitter_buffer_chain: expected #10530, got #10550, gap of 20 This causes too many timers to be pending, causing a reset: gst_rtp_jitter_buffer_chain: 182 pending timers > 180 - resetting gst_rtp_jitter_buffer_reset:< flush and reset jitterbuffer Inside the reset() code, the next_seqnum is set to the first buffer in gap_packets, #10508: gst_rtp_jitter_buffer_reset: setting next_seqnum to #10508 And reset() calls chain() with all its gap_packets, pushing it out: gst_rtp_jitter_buffer_chain: Received packet #10508 at time 123:45:05.364650711, discont 0, rtx 1 However, #10508 is actually an RTX packet, so it gets dropped: gst_rtp_jitter_buffer_chain:<rtpjitterbuffer0> Unsolicited RTX packet #10508 detected, dropping Next #10550 is pushed from reset(): gst_rtp_jitter_buffer_chain:<rtpjitterbuffer0> Received packet #10550 at time 123:45:05.516136868, discont 0, rtx 0 But because next_seqnum is set to #10508, we get this: handle_next_buffer: Sequence number GAP detected: expected 10508 instead of 10550 (42 missing) So now the jitterbuffer is completely stalled, all buffer arriving now will just queue up until it sees the promised #10508, which will be ~65000 packets from now, and we will trigger the "full" logic before then, causing a deadlock.

Too dangerous.

…ons. Co-authored-by: Håvard Graff <[email protected]>

Ignore sequential packets with the same rtp time stamp during jitter estimation.

This to unbreak simulcast tests

…lushed - Fixes a java exception of flushing the encoder in an ilegal state

GstPhotography interface

It is possible to log out an uninitialized SSRC. Reported as by clang's scan build.

It is possible to do a comparison when the ASSERT checks are disabled

appearance

To help debug memory bugs

Changed code to have gst_sctp_association_change_state() and unlocked() version.

We need to ensure that members like state, sctp_assoc_id and sctp_ass_sock are accessed with a lock.

To be consistent with the rest of the code.

…ectly instead

This fixes a very nasty double-free on the SCTP association in which a SCTP association weak-reference will be dangling around and the elements unref() will land in an unlocked gst_sctp_association_finalize() call. Other changes: - Added locking around the SCTP association in both encoder and decoder. - Added ref() API to the SCTP association, to be consistent with unref(). - Calling SCTP associations API with a ref() of the current association to safely avoid finalizing a last reference. - Moved maybe_set_state_to_ready to an unlocked() version, since in the contexts it is called there is no need to release and acquire the lock.

No need to treat them as lost when we can accurately report the reordering in the TWCC feedback message.

Basically being able to jump "back" and report older seqnums, as well as reporting the same packet multiple times.

* introduce a configurable window in which to compute various stats * bugfixes in the twcc report generating code * optimizations around using less lists of packets * much improved handling of reordered packets

…00ms" This reverts commit 9434083.

Because of the nature of not having a clear ownership between the association and the elements (encoder or decoder) is not possible to rely on the associations signals to have a valid reference on the encoder. This also applies for the decoder. Since g_signal_connect_object is not thread-safe (more on this in the documentation) and we can't hold the association mutex while emitting a signal, it is completely possible -as our stress tests demonstrate- that we ran into data races. To fix this situation, I introduce proper callback structures per element context and inside the association we always keep a reference to the element while calling the related callback.

(cherry picked from commit 1550d22)

Comparing with wireshark, this gives the correct size per packet on the wire vs. what is used internally. Also add the original RTP sequence-number to the logging for easy correlation.

In order to limit the amount of packets that is used for stuffing, we both introduce a hard limit of how many packets a "burst" can consist of, but also use padding to make packets slightly bigger, and hence reduce the need to send quite so many.

IF we are going to forward events on non-active pads, we need to send our sticky events prior to this event to make it valid. However, this will lead to the situation of getting a caps change downstream for any event being received on either pad in the funnel. In the case of a decoder sitting downsstream, this would mean potentially replacing that decoder on a caps-change, meaning an "irrelevant" event received on a non-active pad would cause a decoder to lose all state, potentially needing an I-frame or similar.

This reverts commit 7284f57.

Subclasses can now start DTX by calling gst_audio_decoder_start_dtx(), this will spawn a task on the decoders srcpad and generate silent buffers until a new audio packet or a gap event arrives. An API for checking if the decoder is running DTX has been also added in case subclasses need context.

This bug stops us from updating the "send" timestamp for sent packets and cause the log to be spammed with warnings for each packet in cases where the receiver stops sending us the twcc feedback report. (cherry picked from commit 2749324)

- use RTP time instead of pts, the former is based on the sender clock and so provides a more precise metric. - instead of averaging spacings, update the packet spacing when we receive two consecutive identical spacings, discarding decreasing RTP times. This new heuristic should provide a more precise and constant packet spacing on audio streams. It will allow us to support discontinuous transmission (DTX) as the large spacings introduced by DTX will no longer increase the actual packet spacing. Co-Authored-By: Tulio Beloqui <[email protected]> (cherry picked from commit f68e9b9)

disconnecting when on_receive() is called from usrsctplib (cherry picked from commit b50426e)

Workaround for sctplab/usrsctp#405: - Since the sctp socket can outlive the sctp assoacition, we need to only register/deregister the association when creating/closing the socket. This prevents invalid calls to sctp_packet_out() and receive_cb() in potential invalid states. Workaround for sctplab/usrsctp#383: - Retry usrsctp_finish() for 5seconds. This fixes a race condition between usrsctp_close() and usrsctp_finish() in which a dead socket will be accessed by the SCTP thread. (cherry picked from commit 9cf8ac2)

The ID can be zero.

…onPacketReceivedCb Follow a consistent pattern when executing callbacks in handle_message().

…ssociationPacketReceivedCb" This reverts commit a120437. We might not need this change after all.

Co-authored-by: John Bassett <[email protected]> (cherry picked from commit 03e16b7)

A set of unrelated but minor changes to VP8 and VP9 payloaders: VP8: - avoid resetting PictureID if NO_PICTURE_ID mode is set. VP9: - minor refactor of PictureID logic to bring the logic in line with the vp8pay. - add picture-id-offset property. - remove unused struct and variable on a test. VPX: - rename PictureID enum to be more inline with each other. - add picture-id property. - increment rather than resetting the PictureID on a FLUSH_START to maximize interop. Squashed commit of the following (upstream): commit 0cee3cd Author: Camilo Celis Guzman <[email protected]> Date: Thu May 4 02:41:09 2023 +0900 rtpvp8pay: rtpvp9pay: access picture_id property atomically Atomically set and get the picture_id. This changeset only atomically gets the picture-id when such property is queried on the element, on every other place where it is accessed internally it is accessed directly. This is because there is no MT scenario where we would be modifying this value and reading it internally in parallel. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530> commit e4d8cda Author: Camilo Celis Guzman <[email protected]> Date: Wed May 3 20:53:41 2023 +0900 rtpvp8pay, rtpvp9pay: increment PictureID on FLUSH_START In recent versions of Chrome (M106) a change on their jitter buffer means that they are very susceptible to PictureID discontinuities. Then avoid at all cost resetting the PictureID. Moreover, according to the RFCs for VP8 and VP9 payloads; the PictureID can start off at any random value. So there is no logical problem of incrementing it here rather than resetting it, as long as it is a different PictureID. WebRTC's recent corruption issue: https://bugs.chromium.org/p/webrtc/issues/detail?id=15101 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530> commit f159fd8 Author: Camilo Celis Guzman <[email protected]> Date: Wed May 3 00:34:55 2023 +0900 rtpvp8pay, rtpvp9pay: expose picture-id as a property Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530> commit 38d5899 Author: Camilo Celis Guzman <[email protected]> Date: Tue May 2 22:21:34 2023 +0900 rtpvp9pay: tests: remove unused struct and argument on test Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530> commit 11187a8 Author: Camilo Celis Guzman <[email protected]> Date: Tue May 2 21:45:48 2023 +0900 rtpvp9pay: add picture-id-offset property Bring the VP9 payloader in sync in this regard to the VP8 payloader Allowing setting the picture id to a known value is useful when testing. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530> commit 7cffb40 Author: Camilo Celis Guzman <[email protected]> Date: Tue Apr 25 15:36:45 2023 +0900 rtpvp9pay: minor refactor of PictureID logic This brings the logic inline with the vp8pay Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530> commit a79616e Author: Camilo Celis Guzman <[email protected]> Date: Tue Apr 25 15:25:57 2023 +0900 rtpvp8pay: avoid reseting PictureID if NO_PICTURE_ID mode is set There is no logical change here, as `& (1 << nbits) - 1` would produce also 0 when NO_PICTURE_ID mode is choosen. However, this avoid computing a random integer that is actually unused. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530> commit 7dd6375 Author: Camilo Celis Guzman <[email protected]> Date: Tue Apr 25 15:11:10 2023 +0900 rtpvp8pay, rtpvp9pay: use GType like name for PictureIDMode Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530> (cherry picked from commit 9814555)

When using perfect-rtptime, rtpbasepayload needs to have access to buffer offsets to calculate the RTP timestamps. (cherry picked from commit 064bfba) (cherry picked from commit c2106ac)

Set a default offset on buffers: the delta between this buffer's RTP timestamp and the first RTP timestamp. This can be used when forwarding (depayloading/repayloading) media for codecs without specific offset logic in their payloader subclasses (e.g. Opus). The default offset allows the symmetric payloader to reconstruct the original RTP timestamp deltas for its output RTP timestamps, rather than having to rely on the (potentially altered) buffer PTS. (cherry picked from commit a97f44f) (cherry picked from commit 52edb23)

The gist of the issue and fix are somewhat obscure and relate with how a buffer is mapped into before on different platforms. The test uses a va_args method to create, set and push an RTP buffer to the depayloader. However, due to the nature of the fields that can be set, and the use of the va_args. The ordering of this fields matter. This is because some of this fields are only set of a GstRTPBuffer and other are public GstBuffer properties. The problem lies when we set a GstRTPBuffer field, which requires a map of the buffer and **then** try to set a public property on the raw GstBuffer that has already been mapped! Fix this by: 1). correct the order of fields set on the buffer. 2). assert hard when trying to set a public property on the raw GstBuffer after it has been mapped. (cherry picked from commit 38571db)

Controls weather or not we would want to parse the bitstream in order to extract partition information, which is currently only used if no custom meta is present.

...or else we have to carry rt_lib as a dependency

This should be squashed with the previous window-work for twcc, to avoid those size/delay properties ever being introduced.

With the reordering-patch, we started keeping the last packet of the previous report around IF there was any packetloss reported in the previous report, and then start the new report with this packet. We found that this is exactly what WebRTC are doing, and we decided to mimic their behavior here. However, ever since the big "reordering-patch" went in, we have seen some strange discrepancies around what the netsim bitrate have been set to drop, and what has *actually* been reported as dropped. After a long investigation, we found that the "missing_run" variable in the RecvPacket struct could have contained a number from the previous run, but no this number was no longer valid for the next report, BUT WAS STILL BEING USED! This basically means that we were encoding a wildly different set of packets than we actually intended (and actually received), and hence the discrepancies in bitrates...

A good enough version is available from macport. The version supplied by gstreamer is only for x86_64

To avoid deadlock when shutting down.

This will produce a single I-frame at the start and then after that only on request. This makes much more sense than "0", which means producing an I-frame for every single frame!

The idea is basically that every n second, we delay each packet within a X ms window, so that they all burst out at the same time at the end of that window, recreating a throttling effect for a network, but without actually losing any packets.

…gsttask

regression from 63005d (cherry picked from commit eb702a5)

…packets - Removed the upstream latency as well, not needed anymore.

Fixes g_param_spec_float: assertion 'default_value >= minimum && default_value <= maximum' failed

- kVTInvalidSessionErr: common if the application goes between fg and bg mode. - kVTVideoEncoderMalfunctionErr: if the encoder has an internal error.

Having _invoke_on_main defined in separate plugins doesn't work if you want to built the statically.

than dispatch_async on set_window_handle

CreateMacOSSurface should only be called from the main thread.

In our dynamic build winsock2 symbols are missing.

gstvkwindow_cocoa: call close on the internal window Fixes window leak.

- Moved internal gst_ks_video_device_parse_win32_error into a common function ks_parse_win32_error to reuse and add more logging in other places. - Removed unused function gst_ks_video_src_get_device_name_values. - Removed priority list, it doesn't make much sense for cameras. By doing so we also fixed a strange issue with CreateFile failing on a second attempt. Co-Authored-By: Knut Saastad <[email protected]>

To fix calling usrsctp_conninput() on a dead socket. user_socketvar.h: added KASSERT to make sure we do not try to reference a dead socket

the same effect

So we don't have to involve C++ in gst land! Also, link dcsctp statically

to avoid mixing C++ and glib allocators

implemented timeouts single main loop for every sctp association associations lifetime is now only managed by the factory

…s finalize()

- to enable aggressive-heartbeat

- renamed gst_sctp_association_start to connect(), to keep consistency with disconnect - handling and logging SctpSocket_Error, closing the socket on error - configure the socket for aggressive-heartbeat, not quite working.. yet!

…ebug

-lpthread directly just didn't work on my setup. Using the meson built-in dependency seems to be the way other libs do it.

Commits on Oct 27, 2022

gstleaks: FIXUP remove unused variable

havardgraff committed Oct 27, 2022

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Commits on Nov 25, 2022

More videoenc lock fixes

frederik-vestre-pexip committed Nov 25, 2022

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gstvideorate: Fix of buffer duration.

amberik1 committed Dec 7, 2022

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rtptwcc: fixing leaks

tbeloqui committed Jan 4, 2023

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started transport implementation

tbeloqui committed Sep 25, 2023

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Use meson built in support for dependency to threads lib #10

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Use meson built in support for dependency to threads lib #10

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