Category: GNOME

Over the last few weeks, GStreamer’s RTP stack got a couple of new and quite useful features. As it is difficult to configure, mostly because there being so many different possible configurations, I decided to write about this a bit with some example code.

The features are RFC 6051-style rapid synchronization of RTP streams, which can be used for inter-stream (e.g. audio/video) synchronization as well as inter-device (i.e. network) synchronization, and the ability to easily retrieve absolute sender clock times per packet on the receiver side.

Note that each of this was already possible before with GStreamer via different mechanisms with different trade-offs. Obviously, not being able to have working audio/video synchronization would be simply not acceptable and I previously talked about how to do inter-device synchronization with GStreamer before, for example at the GStreamer Conference 2015 in Düsseldorf.

The example code below will make use of the GStreamer RTSP Server library but can be applied to any kind of RTP workflow, including WebRTC, and are written in Rust but the same can also be achieved in any other language. The full code can be found in this repository.

And for reference, the merge requests to enable all this are [1], [2] and [3]. You probably don’t want to backport those to an older version of GStreamer though as there are dependencies on various other changes elsewhere. All of the following needs at least GStreamer from the git main branch as of today, or the upcoming 1.22 release.

Baseline Sender / Receiver Code

The starting point of the example code can be found here in the baseline branch. All the important steps are commented so it should be relatively self-explanatory.

Sender

The sender is starting an RTSP server on the local machine on port 8554 and provides a media with H264 video and Opus audio on the mount point /test. It can be started with

$ cargo run -p rtp-rapid-sync-example-send

After starting the server it can be accessed via GStreamer with e.g. gst-play-1.0 rtsp://127.0.0.1:8554/test or similarly via VLC or any other software that supports RTSP.

This does not do anything special yet but lays the foundation for the following steps. It creates an RTSP server instance with a custom RTSP media factory, which in turn creates custom RTSP media instances. All this is not needed at this point yet but will allow for the necessary customization later.

One important aspect here is that the base time of the media’s pipeline is set to zero

pipeline.set_base_time(gst::ClockTime::ZERO);
pipeline.set_start_time(gst::ClockTime::NONE);

This allows the timeoverlay element that is placed in the video part of the pipeline to render the clock time over the video frames. We’re going to use this later to confirm on the receiver that the clock time on the sender and the one retrieved on the receiver are the same.

let video_overlay = gst::ElementFactory::make("timeoverlay", None)
    .context("Creating timeoverlay")?;
[...]
video_overlay.set_property_from_str("time-mode", "running-time");

It actually only supports rendering the running time of each buffer, but in a live pipeline with the base time set to zero the running time and pipeline clock time are the same. See the documentation for some more details about the time concepts in GStreamer.

Overall this creates the following RTSP stream producer bin, which will be used also in all the following steps:

Receiver

The receiver is a simple playbin pipeline that plays an RTSP URI given via command-line parameters and runs until the stream is finished or an error has happened.

It can be run with the following once the sender is started

$ cargo run -p rtp-rapid-sync-example-send -- "rtsp://192.168.1.101:8554/test"

Please don’t forget to replace the IP with the IP of the machine that is actually running the server.

All the code should be familiar to anyone who ever wrote a GStreamer application in Rust, except for one part that might need a bit more explanation

pipeline.connect_closure(
    "source-setup",
    false,
    glib::closure!(|_playbin: &gst::Pipeline, source: &gst::Element| {
        source.set_property("latency", 40u32);
    }),
);

playbin is going to create an rtspsrc, and at that point it will emit the source-setup signal so that the application can do any additional configuration of the source element. Here we’re connecting a signal handler to that signal to do exactly that.

By default rtspsrc introduces a latency of 2 seconds of latency, which is a lot more than what is usually needed. For live, non-VOD RTSP streams this value should be around the network jitter and here we’re configuring that to 40 milliseconds.

Retrieval of absolute sender clock times

Now as the first step we’re going to retrieve the absolute sender clock times for each video frame on the receiver. They will be rendered by the receiver at the bottom of each video frame and will also be printed to stdout. The changes between the previous version of the code and this version can be seen here and the final code here in the sender-clock-time-retrieval branch.

When running the sender and receiver as before, the video from the receiver should look similar to the following

The upper time that is rendered on the video frames is rendered by the sender, the bottom time is rendered by the receiver and both should always be the same unless something is broken here. Both times are the pipeline clock time when the sender created/captured the video frame.

In this configuration the absolute clock times of the sender are provided to the receiver via the NTP / RTP timestamp mapping provided by the RTCP Sender Reports. That’s also the reason why it takes about 5s for the receiver to know the sender’s clock time as RTCP packets are not scheduled very often and only after about 5s by default. The RTCP interval can be configured on rtpbin together with many other things.

Sender

On the sender-side the configuration changes are rather small and not even absolutely necessary.

rtpbin.set_property_from_str("ntp-time-source", "clock-time");

By default the RTP NTP time used in the RTCP packets is based on the local machine’s walltime clock converted to the NTP epoch. While this works fine, this is not the clock that is used for synchronizing the media and as such there will be drift between the RTP timestamps of the media and the NTP time from the RTCP packets, which will be reset every time the receiver receives a new RTCP Sender Report from the sender.

Instead, we configure rtpbin here to use the pipeline clock as the source for the NTP timestamps used in the RTCP Sender Reports. This doesn’t give us (by default at least, see later) an actual NTP timestamp but it doesn’t have the drift problem mentioned before. Without further configuration, in this pipeline the used clock is the monotonic system clock.

rtpbin.set_property("rtcp-sync-send-time", false);

rtpbin normally uses the time when a packet is sent out for the NTP / RTP timestamp mapping in the RTCP Sender Reports. This is changed with this property to instead use the time when the video frame / audio sample was captured, i.e. it does not include all the latency introduced by encoding and other processing in the sender pipeline.

This doesn’t make any big difference in this scenario but usually one would be interested in the capture clock times and not the send clock times.

Receiver

On the receiver-side there are a few more changes. First of all we have to opt-in to rtpjitterbuffer putting a reference timestamp metadata on every received packet with the sender’s absolute clock time.

pipeline.connect_closure(
    "source-setup",
    false,
    glib::closure!(|_playbin: &gst::Pipeline, source: &gst::Element| {
        source.set_property("latency", 40u32);
        source.set_property("add-reference-timestamp-meta", true);
    }),
);

rtpjitterbuffer will start putting the metadata on packets once it knows the NTP / RTP timestamp mapping, i.e. after the first RTCP Sender Report is received in this case. Between the Sender Reports it is going to interpolate the clock times. The normal timestamps (PTS) on each packet are not affected by this and are still based on whatever clock is used locally by the receiver for synchronization.

To actually make use of the reference timestamp metadata we add a timeoverlay element as video-filter on the receiver:

let timeoverlay =
    gst::ElementFactory::make("timeoverlay", None).context("Creating timeoverlay")?;

timeoverlay.set_property_from_str("time-mode", "reference-timestamp");
timeoverlay.set_property_from_str("valignment", "bottom");

pipeline.set_property("video-filter", &timeoverlay);

This will then render the sender’s absolute clock times at the bottom of each video frame, as seen in the screenshot above.

And last we also add a pad probe on the sink pad of the timeoverlay element to retrieve the reference timestamp metadata of each video frame and then printing the sender’s clock time to stdout:

let sinkpad = timeoverlay
    .static_pad("video_sink")
    .expect("Failed to get timeoverlay sinkpad");
sinkpad
    .add_probe(gst::PadProbeType::BUFFER, |_pad, info| {
        if let Some(gst::PadProbeData::Buffer(ref buffer)) = info.data {
            if let Some(meta) = buffer.meta::<gst::ReferenceTimestampMeta>() {
                println!("Have sender clock time {}", meta.timestamp());
            } else {
                println!("Have no sender clock time");
            }
        }

        gst::PadProbeReturn::Ok
    })
    .expect("Failed to add pad probe");

Rapid synchronization via RTP header extensions

The main problem with the previous code is that the sender’s clock times are only known once the first RTCP Sender Report is received by the receiver. There are many ways to configure rtpbin to make this happen faster (e.g. by reducing the RTCP interval or by switching to the AVPF RTP profile) but in any case the information would be transmitted outside the actual media data flow and it can’t be guaranteed that it is actually known on the receiver from the very first received packet onwards. This is of course not a problem in every use-case, but for the cases where it is there is a solution for this problem.

RFC 6051 defines an RTP header extension that allows to transmit the NTP timestamp that corresponds an RTP packet directly together with this very packet. And that’s what the next changes to the code are making use of.

The changes between the previous version of the code and this version can be seen here and the final code here in the rapid-synchronization branch.

Sender

To add the header extension on the sender-side it is only necessary to add an instance of the corresponding header extension implementation to the payloaders.

let hdr_ext = gst_rtp::RTPHeaderExtension::create_from_uri(
    "urn:ietf:params:rtp-hdrext:ntp-64",
    )
    .context("Creating NTP 64-bit RTP header extension")?;
hdr_ext.set_id(1);
video_pay.emit_by_name::<()>("add-extension", &[&hdr_ext]);

This first instantiates the header extension based on the uniquely defined URI for it, then sets its ID to 1 (see RFC 5285) and then adds it to the video payloader. The same is then done for the audio payloader.

By default this will add the header extension to every RTP packet that has a different RTP timestamp than the previous one. In other words: on the first packet that corresponds to an audio or video frame. Via properties on the header extension this can be configured but generally the default should be sufficient.

Receiver

On the receiver-side no changes would actually be necessary. The use of the header extension is signaled via the SDP (see RFC 5285) and it will be automatically made use of inside rtpbin as another source of NTP / RTP timestamp mappings in addition to the RTCP Sender Reports.

However, we configure one additional property on rtpbin

source.connect_closure(
    "new-manager",
    false,
    glib::closure!(|_rtspsrc: &gst::Element, rtpbin: &gst::Element| {
        rtpbin.set_property("min-ts-offset", gst::ClockTime::from_mseconds(1));
    }),
);

Inter-stream audio/video synchronization

The reason for configuring the min-ts-offset property on the rtpbin is that the NTP / RTP timestamp mapping is not only used for providing the reference timestamp metadata but it is also used for inter-stream synchronization by default. That is, for getting correct audio / video synchronization.

With RTP alone there is no mechanism to synchronize multiple streams against each other as the packet’s RTP timestamps of different streams have no correlation to each other. This is not too much of a problem as usually the packets for audio and video are received approximately at the same time but there’s still some inaccuracy in there.

One approach to fix this is to use the NTP / RTP timestamp mapping for each stream, either from the RTCP Sender Reports or from the RTP header extension, and that’s what is made use of here. And because the mapping is provided very often via the RTP header extension but the RTP timestamps are only accurate up to clock rate (1/90000s for video and 1/48000s) for audio in this case, we configure a threshold of 1ms for adjusting the inter-stream synchronization. Without this it would be adjusted almost continuously by a very small amount back and forth.

Other approaches for inter-stream synchronization are provided by RTSP itself before streaming starts (via the RTP-Info header), but due to a bug this is currently not made use of by GStreamer.

Yet another approach would be via the clock information provided by RFC 7273, about which I already wrote previously and which is also supported by GStreamer. This also allows inter-device, network synchronization and used for that purpose as part of e.g. AES67, Ravenna, SMPTE 2022 / 2110 and many other protocols.

Inter-device network synchronization

Now for the last part, we’re going to add actual inter-device synchronization to this example. The changes between the previous version of the code and this version can be seen here and the final code here in the network-sync branch. This does not use the clock information provided via RFC 7273 (which would be another option) but uses the same NTP / RTP timestamp mapping that was discussed above.

When starting the receiver multiple times on different (or the same) machines, each of them should play back the media synchronized to each other and exactly 2 seconds after the corresponding audio / video frames are produced on the sender.

For this, both sender and all receivers are using an NTP clock (pool.ntp.org in this case) instead of the local monotonic system clock for media synchronization (i.e. as the pipeline clock). Instead of an NTP clock it would also be possible to any other mechanism for network clock synchronization, e.g. PTP or the GStreamer netclock.

println!("Syncing to NTP clock");
clock
    .wait_for_sync(gst::ClockTime::from_seconds(5))
    .context("Syncing NTP clock")?;
println!("Synced to NTP clock");

This code instantiates a GStreamer NTP clock and then synchronously waits up to 5 seconds for it to synchronize. If that fails then the application simply exits with an error.

Sender

On the sender side all that is needed is to configure the RTSP media factory, and as such the pipeline used inside it, to use the NTP clock

factory.set_clock(Some(&clock));

This causes all media inside the sender’s pipeline to be synchronized according to this NTP clock and to also use it for the NTP timestamps in the RTCP Sender Reports and the RTP header extension.

Receiver

On the receiver side the same has to happen

pipeline.use_clock(Some(&clock));

In addition a couple more settings have to be configured on the receiver though. First of all we configure a static latency of 2 seconds on the receiver’s pipeline.

pipeline.set_latency(gst::ClockTime::from_seconds(2));

This is necessary as GStreamer can’t know the latency of every receiver (e.g. different decoders might be used), and also because the sender latency can’t be automatically known. Each audio / video frame will be timestamped on the receiver with the NTP timestamp when it was captured / created, but since then all the latency of the sender, the network and the receiver pipeline has passed and for this some compensation must happen.

Which value to use here depends a lot on the overall setup, but 2 seconds is a (very) safe guess in this case. The value only has to be larger than the sum of sender, network and receiver latency and in the end has the effect that the receiver is showing the media exactly that much later than the sender has produced it.

And last we also have to tell rtpbin that

1. sender and receiver clock are synchronized to each other, i.e. in this case both are using exactly the same NTP clock, and that no translation to the pipeline’s clock is necessary, and
2. that the outgoing timestamps on the receiver should be exactly the sender timestamps and that this conversion should happen based on the NTP / RTP timestamp mapping

source.set_property_from_str("buffer-mode", "synced");
source.set_property("ntp-sync", true);

And that’s it.

A careful reader will also have noticed that all of the above would also work without the RTP header extension, but then the receivers would only be synchronized once the first RTCP Sender Report is received. That’s what the test-netclock.c / test-netclock-client.c example from the GStreamer RTSP server is doing.

As usual with RTP, the above is by far not the only way of doing this and GStreamer also supports various other synchronization mechanisms. Which one is the correct one for a specific use-case depends on a lot of factors.

It has been quite a while since the last status update for the GStreamer Rust bindings and the GStreamer Rust plugins, so the new releases last week make for a good opportunity to do so now.

Bindings

I won’t write too much about the bindings this time. The latest version as of now is 0.16.1, which means that since I started working on the bindings there were 8 major releases. In that same time there were 45 contributors working on the bindings, which seems quite a lot and really makes me happy.

Just as before, I don’t think any major APIs are missing from the bindings anymore, even for implementing subclasses of the various GStreamer types. The wide usage of the bindings in Free Software projects and commercial products also shows both the interest in writing GStreamer applications and plugins in Rust as well as that the bindings are complete enough and production-ready.

Most of the changes since the last status update involve API cleanups, usability improvements, various bugfixes and addition of minor API that was not included before. The details of all changes can be read in the changelog.

The bindings work with any GStreamer version since 1.8 (released more than 4 years ago), support APIs up to GStreamer 1.18 (to be released soon) and work with Rust 1.40 or newer.

Plugins

The biggest progress probably happened with the GStreamer Rust plugins.

There also was a new release last week, 0.6.0, which was the first release where selected plugins were also uploaded to the Rust package (“crate”) database crates.io. This makes it easy for Rust applications to embed any of these plugins statically instead of depending on them to be available on the system.

Overall there are now 40 GStreamer elements in 18 plugins by 28 contributors available as part of the gst-plugins-rs repository, one tutorial plugin with 4 elements and various plugins in external locations.

These 40 GStreamer elements are the following:

Audio

• rsaudioecho: Port of the audioecho element from gst-plugins-good
• rsaudioloudnorm: Live audio loudness normalization element based on the FFmpeg af_loudnorm filter
• claxondec: FLAC lossless audio codec decoder element based on the pure-Rust claxon implementation
• csoundfilter: Audio filter that can use any filter defined via the Csound audio programming language
• lewtondec: Vorbis audio decoder element based on the pure-Rust lewton implementation

Video

• cdgdec/cdgparse: Decoder and parser for the CD+G video codec based on a pure-Rust CD+G implementation, used for example by karaoke CDs
• cea608overlay: CEA-608 Closed Captions overlay element
• cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT subtitles) converter
• tttocea608: CEA-608 Closed Captions from timed-text converter
• mccenc/mccparse: MacCaption Closed Caption format encoder and parser
• sccenc/sccparse: Scenarist Closed Caption format encoder and parser
• dav1dec: AV1 video decoder based on the dav1d decoder implementation by the VLC project
• rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e encoder implementation
• rsflvdemux: Alternative to the flvdemux FLV demuxer element from gst-plugins-good, not feature-equivalent yet
• rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust implementations by the image-rs project

Text

• textwrap: Element for line-wrapping timed text (e.g. subtitles) for better screen-fitting, including hyphenation support for some languages

Network

• reqwesthttpsrc: HTTP(S) source element based on the Rust reqwest/hyper HTTP implementations and almost feature-equivalent with the main GStreamer HTTP source souphttpsrc
• s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
• awstranscriber: Live audio to timed text transcription element using the Amazon AWS Transcribe API

Generic

• sodiumencrypter/sodiumdecrypter: Encryption/decryption element based on libsodium/NaCl
• togglerecord: Recording element that allows to pause/resume recordings easily and considers keyframe boundaries
• fallbackswitch/fallbacksrc: Elements for handling potentially failing (network) sources, restarting them on errors/timeout and showing a fallback stream instead
• threadshare: Set of elements that provide alternatives for various existing GStreamer elements but allow to share the streaming threads between each other to reduce the number of threads
• rsfilesrc/rsfilesink: File source/sink elements as replacements for the existing filesrc/filesink elements

When talking to various people at conferences in the last year or at conferences, a recurring topic was that they believed that the GTK Rust bindings are not ready for use yet.

I don’t know where that perception comes from but if it was true, there wouldn’t have been applications like Fractal, Podcasts or Shortwave using GTK from Rust, or I wouldn’t be able to do a workshop about desktop application development in Rust with GTK and GStreamer at the Linux Application Summit in Barcelona this Friday (code can be found here already) or earlier this year at GUADEC.

One reason I sometimes hear is that there is not support for creating subclasses of GTK types in Rust yet. While that was true, it is not true anymore nowadays. But even more important: unless you want to create your own special widgets, you don’t need that. Many examples and tutorials in other languages make use of inheritance/subclassing for the applications’ architecture, but that’s because it is the idiomatic pattern in those languages. However, in Rust other patterns are more idiomatic and even for those examples and tutorials in other languages it wouldn’t be the one and only option to design applications.

Almost everything is included in the bindings at this point, so seriously consider writing your next GTK UI application in Rust. While some minor features are still missing from the bindings, none of those should prevent you from successfully writing your application.

And if something is actually missing for your use-case or something is not working as expected, please let us know. We’d be happy to make your life easier!

P.S.

Some people are already experimenting with new UI development patterns on top of the GTK Rust bindings. So if you want to try developing an UI application but want to try something different than the usual signal/callback spaghetti code, also take a look at those.

After almost 6 months, a new release of the GStreamer Rust bindings and the GStreamer plugin writing infrastructure for Rust is out. As usual this was coinciding with the release of all the gtk-rs crates to make use of all the new features they contain.

Thanks to all the contributors of both gtk-rs and the GStreamer bindings for all the nice changes that happened over the last 6 months!

And as usual, if you find any bugs please report them and if you have any questions let me know.

GStreamer Bindings

For the full changelog check here.

Most changes this time were internally, especially because many user-facing changes (like Debug impls for various types) were already backported to the minor releases in the 0.11 release series.

WebRTC

The biggest change this time is probably the inclusion of bindings for the GStreamer WebRTC library.

This allows using building all kinds of WebRTC applications outside the browser (or providing a WebRTC implementation for a browser), and while not as full-featured as Google’s own implementation, this interoperates well with the various browsers and generally works much better on embedded devices.

A small example application in Rust is available here.

Serde

Optionally, serde trait implementations for the Serialize and Deserialize trait can be enabled for various fundamental GStreamer types, including caps, buffers, events, messages and tag lists. This allows serializing them into any format that can be handled by serde (which are many!), and deserializing them back to normal Rust structs.

Generic Tag API

Previously only a strongly-typed tag API was exposed that made it impossible to use the wrong data type for a specific tag, e.g. code that tries to store a string for the track number or an integer for the title would simply not compile:

let mut tags = gst::TagList::new();
{
    let tags = tags.get_mut().unwrap();
    tags.add::<Title>(&"some title", gst::TagMergeMode::Append);
    tags.add::<TrackNumber>(&12, gst::TagMergeMode::Append);
}

While this is convenient, it made it rather complicated to work with tag lists if you only wanted to handle them in a generic way. For example by iterating over the tag list and simply checking what kind of tags are available. To solve that, a new generic API was added in addition. This works on glib::Values, which can store any kind of type, and using the wrong type for a specific tag would simply cause an error at runtime instead of compile-time.

let mut tags = gst::TagList::new();
{
    let tags = tags.get_mut().unwrap();
    tags.add_generic(&gst::tags::TAG_TITLE, &"some title", gst::TagMergeMode::Append)
.expect("wrong type for title tag");
    tags.add_generic(&gst::tags::TAG_TRACK_NUMBER, &12, gst::TagMergeMode::Append)
.expect("wrong type for track number tag");
}

This also greatly simplified the serde serialization/deserialization for tag lists.

GStreamer Plugins

For the full changelog check here.

gobject-subclass

The main change this time is that all the generic GObject subclassing infrastructure was moved out of the gst-plugin crate and moved to its own gobject-subclass crate as part of the gtk-rs organization.

As part of this, some major refactoring has happened that allows subclassing more different types but also makes it simpler to add new types. There are also experimental crates for adding some subclassing support to gio and gtk, and a PR for autogenerating part of the code via the gir code generator.

More classes!

The other big addition this time is that it’s now possible to subclass GStreamer Pads and GhostPads, to implement the ChildProxy interface and to subclass the Aggregator and AggregatorPad class.

This now allows to write custom mixer/muxer-style elements (or generally elements that have multiple sink pads) in Rust via the Aggregator base class, and to have custom pad types for elements to allow for setting custom properties on the pads (e.g. to control the opacity of a single video mixer input).

There is currently no example for such an element, but I’ll add a very simple video mixer to the repository some time in the next weeks and will also write a blog post about it for explaining all the steps.

For one of our customers at Centricular we were working on a quite interesting project. Their use-case was basically to receive an as-high-as-possible number of audio RTP streams over UDP, transcode them, and then send them out via UDP again. Due to how GStreamer usually works, they were running into some performance issues.

This blog post will describe the first set of improvements that were implemented for this use-case, together with a minimal benchmark and the results. My colleague Mathieu will follow up with one or two other blog posts with the other improvements and a more full-featured benchmark.

The short version is that CPU usage decreased by about 65-75%, i.e. allowing 3-4x more streams with the same CPU usage. Also parallelization works better and usage of different CPU cores is more controllable, allowing for better scalability. And a fixed, but configurable number of threads is used, which is independent of the number of streams.

The code for this blog post can be found here.

Table of Contents

1. GStreamer & Threads
2. Thread-Sharing GStreamer Elements
3. Available Elements
4. Little Benchmark
5. Conclusion

GStreamer & Threads

In GStreamer, by default each source is running from its own OS thread. Additionally, for receiving/sending RTP, there will be another thread in the RTP jitterbuffer, yet another thread for receiving RTCP (another source) and a last thread for sending RTCP at the right times. And RTCP has to be received and sent for the receiver and sender side part of the pipeline, so the number of threads doubles. In the sum this gives at least 1 + 1 + (1 + 1) * 2 = 6 threads per RTP stream in this scenario. In a normal audio scenario, there will be one packet received/sent e.g. every 20ms on each stream, and every now and then an RTCP packet. So most of the time all these threads are only waiting.

Apart from the obvious waste of OS resources (1000 streams would be 6000 threads), this also brings down performance as all the time threads are being woken up. This means that context switches have to happen basically all the time.

To solve this we implemented a mechanism to share threads, and in the end as a result we have a fixed, but configurable number of threads that is independent from the number of streams. And can run e.g. 500 streams just fine on a single thread with a single core, which was completely impossible before. In addition we also did some work to reduce the number of allocations for each packet, so that after startup no additional allocations happen per packet anymore for buffers. See Mathieu’s upcoming blog post for details.

In this blog post, I’m going to write about a generic mechanism for sources, queues and similar elements to share their threads between each other. For the RTP related bits (RTP jitterbuffer and RTCP timer) this was not used due to reuse of existing C codebases.

Thread-Sharing GStreamer Elements

The code in question can be found here, a small benchmark is in the examples directory and it is going to be used for the results later. A full-featured benchmark will come in Mathieu’s blog post.

This is a new GStreamer plugin, written in Rust and around the Tokio crate for asynchronous IO and generally a “task scheduler”.

While this could certainly also have been written in C around something like libuv, doing this kind of work in Rust is simply more productive and fun due to its safety guarantees and the strong type system, which definitely reduced the amount of debugging a lot. And in addition “modern” language features like closures, which make working with futures much more ergonomic.

When using these elements it is important to have full control over the pipeline and its elements, and the dataflow inside the pipeline has to be carefully considered to properly configure how to share threads. For example the following two restrictions should be kept in mind all the time:

1. Downstream of such an element, the streaming thread must never ever block for considerable amounts of time. Otherwise all other elements inside the same thread-group would be blocked too, even if they could do any work now
2. This generally all works better in live pipelines, where media is produced in real-time and not as fast as possible

Available Elements

So this repository currently contains the generic infrastructure (see the src/iocontext.rs source file) and a couple of elements:

• an UDP source: ts-udpsrc, a replacement for udpsrc
• an app source: ts-appsrc, a replacement for appsrc to inject packets into the pipeline from the application
• a queue: ts-queue, a replacement for queue that is useful for adding buffering to a pipeline part. The upstream side of the queue will block if not called from another thread-sharing element, but if called from another thread-sharing element it will pause the current task asynchronously. That is, stop the upstream task from producing more data.
• a proxysink/src element: ts-proxysrc, ts-proxysink, replacements for proxysink/proxysrc for connecting two pipelines with each other. This basically works like the queue, but split into two elements.
• a tone generator source around spandsp: ts-tonesrc, a replacement for tonegeneratesrc. This also contains some minimal FFI bindings for that part of the spandsp C library.

All these elements have more or less the same API as their non-thread-sharing counterparts.

API-wise, each of these elements has a set of properties for controlling how it is sharing threads with other elements, and with which elements:

• context: A string that defines in which group this element is. All elements with the same context are running on the same thread or group of threads,
• context-threads: Number of threads to use in this context. -1 means exactly one thread, 1 and above used N+1 threads (1 thread for polling fds, N worker threads) and 0 sets N to the number of available CPU cores. As long as no considerable work is done in these threads, -1 has shown to be the most efficient. See also this tokio GitHub issue
• context-wait: Number of milliseconds that the threads will wait on each iteration. This allows to reduce CPU usage even further by handling all events/packets that arrived during that timespan to be handled all at once instead of waking up the thread every time a little event happens, thus reducing context switches again

The elements are all pushing data downstream from a tokio thread whenever data is available, assuming that downstream does not block. If downstream is another thread-sharing element and it would have to block (e.g. a full queue), it instead returns a new future to upstream so that upstream can asynchronously wait on that future before producing more output. By this, back-pressure is implemented between different GStreamer elements without ever blocking any of the tokio threads. All this is implemented around the normal GStreamer data-flow mechanisms, there is no “tokio fast-path” between elements.

Little Benchmark

As mentioned above, there’s a small benchmark application in the examples directory. This basically sets up a configurable number of streams and directly connects them to a fakesink, throwing away all packets. Additionally there is another thread that is sending all these packets. As such, this is really the most basic benchmark and not very realistic but nonetheless it shows the same performance improvement as the real application. Again, see Mathieu’s upcoming blog post for a more realistic and complete benchmark.

When running it, make sure that your user can create enough fds. The benchmark will just abort if not enough fds can be allocated. You can control this with ulimit -n SOME_NUMBER, and allowing a couple of thousands is generally a good idea. The benchmarks below were running with 10000.

After running cargo build –release to build the plugin itself, you can run the benchmark with:

cargo run --release --example udpsrc-benchmark -- 1000 ts-udpsrc -1 1 20

and in another shell the UDP sender with

cargo run --release --example udpsrc-benchmark-sender -- 1000

This runs 1000 streams, uses ts-udpsrc (alternative would be udpsrc), configures exactly one thread -1, 1 context, and a wait time of 20ms. See above for what these settings mean. You can check CPU usage with e.g. top. Testing was done on an Intel i7-4790K, with Rust 1.25 and GStreamer 1.14. One packet is sent every 20ms for each stream.

Source	Streams	Threads	Contexts	Wait	CPU
udpsrc	1000	1000	x	x	44%
ts-udpsrc	1000	-1	1	0	18%
ts-udpsrc	1000	-1	1	20	13%
ts-udpsrc	1000	-1	2	20	15%
ts-udpsrc	1000	2	1	20	16%
ts-udpsrc	1000	2	2	20	27%
Source	Streams	Threads	Contexts	Wait	CPU
udpsrc	2000	2000	x	x	95%
ts-udpsrc	2000	-1	1	20	29%
ts-udpsrc	2000	-1	2	20	31%
Source	Streams	Threads	Contexts	Wait	CPU
ts-udpsrc	3000	-1	1	20	36%
ts-udpsrc	3000	-1	2	20	47%

Results for 3000 streams for the old udpsrc are not included as starting up that many threads needs too long.

The best configuration is apparently a single thread per context (see this tokio GitHub issue) and waiting 20ms for every iterations. Compared to the old udpsrc, CPU usage is about one third in that setting, and generally it seems to parallelize well. It’s not clear to me why the last test has 11% more CPU with two contexts, while in every other test the number of contexts does not really make a difference, and also not for that many streams in the real test-case.

The waiting does not reduce CPU usage a lot in this benchmark, but on the real test-case it does. The reason is most likely that this benchmark basically sends all packets at once, then waits for the remaining time, then sends the next packets.

Take these numbers with caution, the real test-case in Mathieu’s blog post will show the improvements in the bigger picture, where it was generally a quarter of CPU usage and almost perfect parallelization when increasing the number of contexts.

Conclusion

Generally this was a fun exercise and we’re quite happy with the results, especially the real results. It took me some time to understand how tokio works internally so that I can implement all kinds of customizations on top of it, but for normal usage of tokio that should not be required and the overall design makes a lot of sense to me, as well as the way how futures are implemented in Rust. It requires some learning and understanding how exactly the API can be used and behaves, but once that point is reached it seems like a very productive and performant solution for asynchronous IO. And modelling asynchronous IO problems based on the Rust-style futures seems a nice and intuitive fit.

The performance measurements also showed that GStreamer’s default usage of threads is not always optimal, and a model like in upipe or pipewire (or rather SPA) can provide better performance. But as this also shows, it is possible to implement something like this on top of GStreamer and for the common case, using threads like in GStreamer reduces the cognitive load on the developer a lot.

For a future version of GStreamer, I don’t think we should make the threading “manual” like in these two other projects, but instead provide some API additions that make it nicer to implement thread-sharing elements and to add ways in the GStreamer core to make streaming threads non-blocking. All this can be implemented already, but it could be nicer.

All this “only” improved the number of threads, and thus the threading and context switching overhead. Many other optimizations in other areas are still possible on top of this, for example optimizing receive performance and reducing the number of memory copies inside the pipeline even further. If that’s something you would be interested in, feel free to get in touch.

And with that: Read Mathieu’s upcoming blog posts about the other parts, RTP jitterbuffer / RTCP timer thread sharing, and no allocations, and the full benchmark.

Following the GStreamer 1.14 release and the new round of gtk-rs releases, there are also new releases for the GStreamer Rust bindings (0.11) and the plugin writing infrastructure (0.2).

Thanks also to all the contributors for making these releases happen and adding lots of valuable changes and API additions.

GStreamer Rust Bindings

The main changes in the Rust bindings were the update to GStreamer 1.14 (which brings in quite some new API, like GstPromise), a couple of API additions (GstBufferPool specifically) and the addition of the GstRtspServer and GstPbutils crates. The former allows writing a full RTSP server in a couple of lines of code (with lots of potential for customizations), the latter provides access to the GstDiscoverer helper object that allows inspecting files and streams for their container format, codecs, tags and all kinds of other metadata.

The GstPbutils crate will also get other features added in the near future, like encoding profile bindings to allow using the encodebin GStreamer element (a helper element for automatically selecting/configuring encoders and muxers) from Rust.

But the biggest changes in my opinion is some refactoring that was done to the Event, Message and Query APIs. Previously you would have to use a view on a newly created query to be able to use the type-specific functions on it

let mut q = gst::Query::new_position(gst::Format::Time);
if pipeline.query(q.get_mut().unwrap()) {
    match q.view() {
        QueryView::Position(ref p) => Some(p.get_result()),
        _ => None,
    }
} else {
    None
}

Now you can directly use the type-specific functions on a newly created query

let mut q = gst::Query::new_position(gst::Format::Time);
if pipeline.query(&mut q) {
    Some(q.get_result())
} else {
    None
}

In addition, the views can now dereference directly to the event/message/query itself and provide access to their API, which simplifies some code even more.

Plugin Writing Infrastructure

While the plugin writing infrastructure did not see that many changes apart from a couple of bugfixes and updating to the new versions of everything else, this does not mean that development on it stalled. Quite the opposite. The existing code works very well already and there was just no need for adding anything new for the projects I and others did on top of it, most of the required API additions were in the GStreamer bindings.

So the status here is the same as last time, get started writing GStreamer plugins in Rust. It works well!

A bit later than anticipated, this is now part two of the blog post series about writing GStreamer elements in Rust. Part one can be found here, and I’ll assume that everything written there is known already.

In this part, a raw audio sine wave source element is going to be written. It will be similar to the one Mathieu was writing in his blog post about writing such a GStreamer element in Python. Various details will be different though, but more about that later.

The final code can be found here.

Table of Contents

1. Boilerplate
2. Caps Negotiation
3. Query Handling
4. Buffer Creation
5. (Pseudo) Live Mode
6. Unlocking
7. Seeking

Boilerplate

The first part here will be all the boilerplate required to set up the element. You can safely skip this if you remember all this from the previous blog post.

Our sine wave element is going to produce raw audio, with a number of channels and any possible sample rate with both 32 bit and 64 bit floating point samples. It will produce a simple sine wave with a configurable frequency, volume/mute and number of samples per audio buffer. In addition it will be possible to configure the element in (pseudo) live mode, meaning that it will only produce data in real-time according to the pipeline clock. And it will be possible to seek to any time/sample position on our source element. It will basically be a more simply version of the audiotestsrc element from gst-plugins-base.

So let’s get started with all the boilerplate. This time our element will be based on the BaseSrc base class instead of BaseTransform.

use glib;
use gst;
use gst::prelude::*;
use gst_base::prelude::*;
use gst_audio;

use byte_slice_cast::*;

use gst_plugin::properties::*;
use gst_plugin::object::*;
use gst_plugin::element::*;
use gst_plugin::base_src::*;

use std::{i32, u32};
use std::sync::Mutex;
use std::ops::Rem;

use num_traits::float::Float;
use num_traits::cast::NumCast;

// Default values of properties
const DEFAULT_SAMPLES_PER_BUFFER: u32 = 1024;
const DEFAULT_FREQ: u32 = 440;
const DEFAULT_VOLUME: f64 = 0.8;
const DEFAULT_MUTE: bool = false;
const DEFAULT_IS_LIVE: bool = false;

// Property value storage
#[derive(Debug, Clone, Copy)]
struct Settings {
    samples_per_buffer: u32,
    freq: u32,
    volume: f64,
    mute: bool,
    is_live: bool,
}

impl Default for Settings {
    fn default() -> Self {
        Settings {
            samples_per_buffer: DEFAULT_SAMPLES_PER_BUFFER,
            freq: DEFAULT_FREQ,
            volume: DEFAULT_VOLUME,
            mute: DEFAULT_MUTE,
            is_live: DEFAULT_IS_LIVE,
        }
    }
}

// Metadata for the properties
static PROPERTIES: [Property; 5] = [
    Property::UInt(
        "samples-per-buffer",
        "Samples Per Buffer",
        "Number of samples per output buffer",
        (1, u32::MAX),
        DEFAULT_SAMPLES_PER_BUFFER,
        PropertyMutability::ReadWrite,
    ),
    Property::UInt(
        "freq",
        "Frequency",
        "Frequency",
        (1, u32::MAX),
        DEFAULT_FREQ,
        PropertyMutability::ReadWrite,
    ),
    Property::Double(
        "volume",
        "Volume",
        "Output volume",
        (0.0, 10.0),
        DEFAULT_VOLUME,
        PropertyMutability::ReadWrite,
    ),
    Property::Boolean(
        "mute",
        "Mute",
        "Mute",
        DEFAULT_MUTE,
        PropertyMutability::ReadWrite,
    ),
    Property::Boolean(
        "is-live",
        "Is Live",
        "(Pseudo) live output",
        DEFAULT_IS_LIVE,
        PropertyMutability::ReadWrite,
    ),
];

// Stream-specific state, i.e. audio format configuration
// and sample offset
struct State {
    info: Option<gst_audio::AudioInfo>,
    sample_offset: u64,
    sample_stop: Option<u64>,
    accumulator: f64,
}

impl Default for State {
    fn default() -> State {
        State {
            info: None,
            sample_offset: 0,
            sample_stop: None,
            accumulator: 0.0,
        }
    }
}

// Struct containing all the element data
struct SineSrc {
    cat: gst::DebugCategory,
    settings: Mutex<Settings>,
    state: Mutex<State>,
}

impl SineSrc {
    // Called when a new instance is to be created
    fn new(element: &BaseSrc) -> Box<BaseSrcImpl<BaseSrc>> {
        // Initialize live-ness and notify the base class that
        // we'd like to operate in Time format
        element.set_live(DEFAULT_IS_LIVE);
        element.set_format(gst::Format::Time);

        Box::new(Self {
            cat: gst::DebugCategory::new(
                "rssinesrc",
                gst::DebugColorFlags::empty(),
                "Rust Sine Wave Source",
            ),
            settings: Mutex::new(Default::default()),
            state: Mutex::new(Default::default()),
        })
    }

    // Called exactly once when registering the type. Used for
    // setting up metadata for all instances, e.g. the name and
    // classification and the pad templates with their caps.
    //
    // Actual instances can create pads based on those pad templates
    // with a subset of the caps given here. In case of basesrc,
    // a "src" and "sink" pad template are required here and the base class
    // will automatically instantiate pads for them.
    //
    // Our element here can output f32 and f64
    fn class_init(klass: &mut BaseSrcClass) {
        klass.set_metadata(
            "Sine Wave Source",
            "Source/Audio",
            "Creates a sine wave",
            "Sebastian Dröge <sebastian@centricular.com>",
        );

        // On the src pad, we can produce F32/F64 with any sample rate
        // and any number of channels
        let caps = gst::Caps::new_simple(
            "audio/x-raw",
            &[
                (
                    "format",
                    &gst::List::new(&[
                        &gst_audio::AUDIO_FORMAT_F32.to_string(),
                        &gst_audio::AUDIO_FORMAT_F64.to_string(),
                    ]),
                ),
                ("layout", &"interleaved"),
                ("rate", &gst::IntRange::<i32>::new(1, i32::MAX)),
                ("channels", &gst::IntRange::<i32>::new(1, i32::MAX)),
            ],
        );
        // The src pad template must be named "src" for basesrc
        // and specific a pad that is always there
        let src_pad_template = gst::PadTemplate::new(
            "src",
            gst::PadDirection::Src,
            gst::PadPresence::Always,
            &caps,
        );
        klass.add_pad_template(src_pad_template);

        // Install all our properties
        klass.install_properties(&PROPERTIES);
    }
}

impl ObjectImpl<BaseSrc> for SineSrc {
    // Called whenever a value of a property is changed. It can be called
    // at any time from any thread.
    fn set_property(&self, obj: &glib::Object, id: u32, value: &glib::Value) {
        let prop = &PROPERTIES[id as usize];
        let element = obj.clone().downcast::<BaseSrc>().unwrap();

        match *prop {
            Property::UInt("samples-per-buffer", ..) => {
                let mut settings = self.settings.lock().unwrap();
                let samples_per_buffer = value.get().unwrap();
                gst_info!(
                    self.cat,
                    obj: &element,
                    "Changing samples-per-buffer from {} to {}",
                    settings.samples_per_buffer,
                    samples_per_buffer
                );
                settings.samples_per_buffer = samples_per_buffer;
                drop(settings);

                let _ =
                    element.post_message(&gst::Message::new_latency().src(Some(&element)).build());
            }
            Property::UInt("freq", ..) => {
                let mut settings = self.settings.lock().unwrap();
                let freq = value.get().unwrap();
                gst_info!(
                    self.cat,
                    obj: &element,
                    "Changing freq from {} to {}",
                    settings.freq,
                    freq
                );
                settings.freq = freq;
            }
            Property::Double("volume", ..) => {
                let mut settings = self.settings.lock().unwrap();
                let volume = value.get().unwrap();
                gst_info!(
                    self.cat,
                    obj: &element,
                    "Changing volume from {} to {}",
                    settings.volume,
                    volume
                );
                settings.volume = volume;
            }
            Property::Boolean("mute", ..) => {
                let mut settings = self.settings.lock().unwrap();
                let mute = value.get().unwrap();
                gst_info!(
                    self.cat,
                    obj: &element,
                    "Changing mute from {} to {}",
                    settings.mute,
                    mute
                );
                settings.mute = mute;
            }
            Property::Boolean("is-live", ..) => {
                let mut settings = self.settings.lock().unwrap();
                let is_live = value.get().unwrap();
                gst_info!(
                    self.cat,
                    obj: &element,
                    "Changing is-live from {} to {}",
                    settings.is_live,
                    is_live
                );
                settings.is_live = is_live;
            }
            _ => unimplemented!(),
        }
    }

    // Called whenever a value of a property is read. It can be called
    // at any time from any thread.
    fn get_property(&self, _obj: &glib::Object, id: u32) -> Result<glib::Value, ()> {
        let prop = &PROPERTIES[id as usize];

        match *prop {
            Property::UInt("samples-per-buffer", ..) => {
                let settings = self.settings.lock().unwrap();
                Ok(settings.samples_per_buffer.to_value())
            }
            Property::UInt("freq", ..) => {
                let settings = self.settings.lock().unwrap();
                Ok(settings.freq.to_value())
            }
            Property::Double("volume", ..) => {
                let settings = self.settings.lock().unwrap();
                Ok(settings.volume.to_value())
            }
            Property::Boolean("mute", ..) => {
                let settings = self.settings.lock().unwrap();
                Ok(settings.mute.to_value())
            }
            Property::Boolean("is-live", ..) => {
                let settings = self.settings.lock().unwrap();
                Ok(settings.is_live.to_value())
            }
            _ => unimplemented!(),
        }
    }
}

// Virtual methods of gst::Element. We override none
impl ElementImpl<BaseSrc> for SineSrc { }

impl BaseSrcImpl<BaseSrc> for SineSrc {
    // Called when starting, so we can initialize all stream-related state to its defaults
    fn start(&self, element: &BaseSrc) -> bool {
        // Reset state
        *self.state.lock().unwrap() = Default::default();

        gst_info!(self.cat, obj: element, "Started");

        true
    }

    // Called when shutting down the element so we can release all stream-related state
    fn stop(&self, element: &BaseSrc) -> bool {
        // Reset state
        *self.state.lock().unwrap() = Default::default();

        gst_info!(self.cat, obj: element, "Stopped");

        true
    }
}

struct SineSrcStatic;

// The basic trait for registering the type: This returns a name for the type and registers the
// instance and class initializations functions with the type system, thus hooking everything
// together.
impl ImplTypeStatic<BaseSrc> for SineSrcStatic {
    fn get_name(&self) -> &str {
        "SineSrc"
    }

    fn new(&self, element: &BaseSrc) -> Box<BaseSrcImpl<BaseSrc>> {
        SineSrc::new(element)
    }

    fn class_init(&self, klass: &mut BaseSrcClass) {
        SineSrc::class_init(klass);
    }
}

// Registers the type for our element, and then registers in GStreamer under
// the name "sinesrc" for being able to instantiate it via e.g.
// gst::ElementFactory::make().
pub fn register(plugin: &gst::Plugin) {
    let type_ = register_type(SineSrcStatic);
    gst::Element::register(plugin, "rssinesrc", 0, type_);
}

If any of this needs explanation, please see the previous blog post and the comments in the code. The explanation for all the structs fields and what they’re good for will follow in the next sections.

With all of the above and a small addition to src/lib.rs this should compile now.

mod sinesrc;
[...]

fn plugin_init(plugin: &gst::Plugin) -> bool {
    [...]
    sinesrc::register(plugin);
    true
}

Also a couple of new crates have to be added to Cargo.toml and src/lib.rs, but you best check the code in the repository for details.

Caps Negotiation

The first part that we have to implement, just like last time, is caps negotiation. We already notified the base class about any caps that we can potentially handle via the caps in the pad template in class_init but there are still two more steps of behaviour left that we have to implement.

First of all, we need to get notified whenever the caps that our source is configured for are changing. This will happen once in the very beginning and then whenever the pipeline topology or state changes and new caps would be more optimal for the new situation. This notification happens via the BaseTransform::set_caps virtual method.

    fn set_caps(&self, element: &BaseSrc, caps: &gst::CapsRef) -> bool {
        use std::f64::consts::PI;

        let info = match gst_audio::AudioInfo::from_caps(caps) {
            None => return false,
            Some(info) => info,
        };

        gst_debug!(self.cat, obj: element, "Configuring for caps {}", caps);

        element.set_blocksize(info.bpf() * (*self.settings.lock().unwrap()).samples_per_buffer);

        let settings = *self.settings.lock().unwrap();
        let mut state = self.state.lock().unwrap();

        // If we have no caps yet, any old sample_offset and sample_stop will be
        // in nanoseconds
        let old_rate = match state.info {
            Some(ref info) => info.rate() as u64,
            None => gst::SECOND_VAL,
        };

        // Update sample offset and accumulator based on the previous values and the
        // sample rate change, if any
        let old_sample_offset = state.sample_offset;
        let sample_offset = old_sample_offset
            .mul_div_floor(info.rate() as u64, old_rate)
            .unwrap();

        let old_sample_stop = state.sample_stop;
        let sample_stop =
            old_sample_stop.map(|v| v.mul_div_floor(info.rate() as u64, old_rate).unwrap());

        let accumulator =
            (sample_offset as f64).rem(2.0 * PI * (settings.freq as f64) / (info.rate() as f64));

        *state = State {
            info: Some(info),
            sample_offset: sample_offset,
            sample_stop: sample_stop,
            accumulator: accumulator,
        };

        drop(state);

        let _ = element.post_message(&gst::Message::new_latency().src(Some(element)).build());

        true
    }

In here we parse the caps into a AudioInfo and then store that in our internal state, while updating various fields. We tell the base class about the number of bytes each buffer is usually going to hold, and update our current sample position, the stop sample position (when a seek with stop position happens, we need to know when to stop) and our accumulator. This happens by scaling both positions by the old and new sample rate. If we don’t have an old sample rate, we assume nanoseconds (this will make more sense once seeking is implemented). The scaling is done with the help of the muldiv crate, which implements scaling of integer types by a fraction with protection against overflows by doing up to 128 bit integer arithmetic for intermediate values.

The accumulator is the updated based on the current phase of the sine wave at the current sample position.

As a last step we post a new LATENCY message on the bus whenever the sample rate has changed. Our latency (in live mode) is going to be the duration of a single buffer, but more about that later.

BaseSrc is by default already selecting possible caps for us, if there are multiple options. However these defaults might not be (and often are not) ideal and we should override the default behaviour slightly. This is done in the BaseSrc::fixate virtual method.

    fn fixate(&self, element: &BaseSrc, caps: gst::Caps) -> gst::Caps {
        // Fixate the caps. BaseSrc will do some fixation for us, but
        // as we allow any rate between 1 and MAX it would fixate to 1. 1Hz
        // is generally not a useful sample rate.
        //
        // We fixate to the closest integer value to 48kHz that is possible
        // here, and for good measure also decide that the closest value to 1
        // channel is good.
        let mut caps = gst::Caps::truncate(caps);
        {
            let caps = caps.make_mut();
            let s = caps.get_mut_structure(0).unwrap();
            s.fixate_field_nearest_int("rate", 48_000);
            s.fixate_field_nearest_int("channels", 1);
        }

        // Let BaseSrc fixate anything else for us. We could've alternatively have
        // called Caps::fixate() here
        element.parent_fixate(caps)
    }

Here we take the caps that are passed in, truncate them (i.e. remove all but the very first Structure) and then manually fixate the sample rate to the closest value to 48kHz. By default, caps fixation would result in the lowest possible sample rate but this is usually not desired.

For good measure, we also fixate the number of channels to the closest value to 1, but this would already be the default behaviour anyway. And then chain up to the parent class’ implementation of fixate, which for now basically does the same as Caps::fixate(). After this, the caps are fixated, i.e. there is only a single Structure left and all fields have concrete values (no ranges or sets).

Query Handling

As our source element will work by generating a new audio buffer from a specific offset, and especially works in Time format, we want to notify downstream elements that we don’t want to run in Pull mode, only in Push mode. In addition would prefer sequential reading. However we still allow seeking later. For a source that does not know about Time, e.g. a file source, the format would be configured as Bytes. Other values than Time and Bytes generally don’t make any sense.

The main difference here is that otherwise the base class would ask us to produce data for arbitrary Byte offsets, and we would have to produce data for that. While possible in our case, it’s a bit annoying and for other audio sources it’s not easily possible at all.

Downstream elements will try to query this very information from us, so we now have to override the default query handling of BaseSrc and handle the SCHEDULING query differently. Later we will also handle other queries differently.

    fn query(&self, element: &BaseSrc, query: &mut gst::QueryRef) -> bool {
        use gst::QueryView;

        match query.view_mut() {
            // We only work in Push mode. In Pull mode, create() could be called with
            // arbitrary offsets and we would have to produce for that specific offset
            QueryView::Scheduling(ref mut q) => {
                q.set(gst::SchedulingFlags::SEQUENTIAL, 1, -1, 0);
                q.add_scheduling_modes(&[gst::PadMode::Push]);
                return true;
            }
            _ => (),
        }
        BaseSrcBase::parent_query(element, query)
    }

To handle the SCHEDULING query specifically, we first have to match on a view (mutable because we want to modify the view) of the query check the type of the query. If it indeed is a scheduling query, we can set the SEQUENTIAL flag and specify that we handle only Push mode, then return true directly as we handled the query already.

In all other cases we fall back to the parent class’ implementation of the query virtual method.

Buffer Creation

Now we have everything in place for a working element, apart from the virtual method to actually generate the raw audio buffers with the sine wave. From a high-level BaseSrc works by calling the create virtual method over and over again to let the subclass produce a buffer until it returns an error or signals the end of the stream.

Let’s first talk about how to generate the sine wave samples themselves. As we want to operate on 32 bit and 64 bit floating point numbers, we implement a generic function for generating samples and storing them in a mutable byte slice. This is done with the help of the num_traits crate, which provides all kinds of useful traits for abstracting over numeric types. In our case we only need the Float and NumCast traits.

Instead of writing a generic implementation with those traits, it would also be possible to do the same with a simple macro that generates a function for both types. Which approach is nicer is a matter of taste in the end, the compiler output should be equivalent for both cases.

    fn process<F: Float + FromByteSlice>(
        data: &mut [u8],
        accumulator_ref: &mut f64,
        freq: u32,
        rate: u32,
        channels: u32,
        vol: f64,
    ) {
        use std::f64::consts::PI;

        // Reinterpret our byte-slice as a slice containing elements of the type
        // we're interested in. GStreamer requires for raw audio that the alignment
        // of memory is correct, so this will never ever fail unless there is an
        // actual bug elsewhere.
        let data = data.as_mut_slice_of::<F>().unwrap();

        // Convert all our parameters to the target type for calculations
        let vol: F = NumCast::from(vol).unwrap();
        let freq = freq as f64;
        let rate = rate as f64;
        let two_pi = 2.0 * PI;

        // We're carrying a accumulator with up to 2pi around instead of working
        // on the sample offset. High sample offsets cause too much inaccuracy when
        // converted to floating point numbers and then iterated over in 1-steps
        let mut accumulator = *accumulator_ref;
        let step = two_pi * freq / rate;

        for chunk in data.chunks_mut(channels as usize) {
            let value = vol * F::sin(NumCast::from(accumulator).unwrap());
            for sample in chunk {
                *sample = value;
            }

            accumulator += step;
            if accumulator >= two_pi {
                accumulator -= two_pi;
            }
        }

        *accumulator_ref = accumulator;
    }

This function takes the mutable byte slice from our buffer as argument, as well as the current value of the accumulator and the relevant settings for generating the sine wave.

As a first step, we “cast” the byte slice to one of the target type (f32 or f64) with the help of the byte_slice_cast crate. This ensures that alignment and sizes are all matching and returns a mutable slice of our target type if successful. In case of GStreamer, the buffer alignment is guaranteed to be big enough for our types here and we allocate the buffer of a correct size later.

Now we convert all the parameters to the types we will use later, and store them together with the current accumulator value in local variables. Then we iterate over the whole floating point number slice in chunks with all channels, and fill each channel with the current value of our sine wave.

The sine wave itself is calculated by val = volume * sin(2 * PI * frequency * (i + accumulator) / rate), but we actually calculate it by simply increasing the accumulator by 2 * PI * frequency / rate for every sample instead of doing the multiplication for each sample. We also make sure that the accumulator always stays between 0 and 2 * PI to prevent any inaccuracies from floating point numbers to affect our produced samples.

Now that this is done, we need to implement the BaseSrc::create virtual method for actually allocating the buffer, setting timestamps and other metadata and it and calling our above function.

    fn create(
        &self,
        element: &BaseSrc,
        _offset: u64,
        _length: u32,
    ) -> Result<gst::Buffer, gst::FlowReturn> {
        // Keep a local copy of the values of all our properties at this very moment. This
        // ensures that the mutex is never locked for long and the application wouldn't
        // have to block until this function returns when getting/setting property values
        let settings = *self.settings.lock().unwrap();

        // Get a locked reference to our state, i.e. the input and output AudioInfo
        let mut state = self.state.lock().unwrap();
        let info = match state.info {
            None => {
                gst_element_error!(element, gst::CoreError::Negotiation, ["Have no caps yet"]);
                return Err(gst::FlowReturn::NotNegotiated);
            }
            Some(ref info) => info.clone(),
        };

        // If a stop position is set (from a seek), only produce samples up to that
        // point but at most samples_per_buffer samples per buffer
        let n_samples = if let Some(sample_stop) = state.sample_stop {
            if sample_stop <= state.sample_offset {
                gst_log!(self.cat, obj: element, "At EOS");
                return Err(gst::FlowReturn::Eos);
            }

            sample_stop - state.sample_offset
        } else {
            settings.samples_per_buffer as u64
        };

        // Allocate a new buffer of the required size, update the metadata with the
        // current timestamp and duration and then fill it according to the current
        // caps
        let mut buffer =
            gst::Buffer::with_size((n_samples as usize) * (info.bpf() as usize)).unwrap();
        {
            let buffer = buffer.get_mut().unwrap();

            // Calculate the current timestamp (PTS) and the next one,
            // and calculate the duration from the difference instead of
            // simply the number of samples to prevent rounding errors
            let pts = state
                .sample_offset
                .mul_div_floor(gst::SECOND_VAL, info.rate() as u64)
                .unwrap()
                .into();
            let next_pts: gst::ClockTime = (state.sample_offset + n_samples)
                .mul_div_floor(gst::SECOND_VAL, info.rate() as u64)
                .unwrap()
                .into();
            buffer.set_pts(pts);
            buffer.set_duration(next_pts - pts);

            // Map the buffer writable and create the actual samples
            let mut map = buffer.map_writable().unwrap();
            let data = map.as_mut_slice();

            if info.format() == gst_audio::AUDIO_FORMAT_F32 {
                Self::process::<f32>(
                    data,
                    &mut state.accumulator,
                    settings.freq,
                    info.rate(),
                    info.channels(),
                    settings.volume,
                );
            } else {
                Self::process::<f64>(
                    data,
                    &mut state.accumulator,
                    settings.freq,
                    info.rate(),
                    info.channels(),
                    settings.volume,
                );
            }
        }
        state.sample_offset += n_samples;
        drop(state);

        gst_debug!(self.cat, obj: element, "Produced buffer {:?}", buffer);

        Ok(buffer)
    }

Just like last time, we start with creating a copy of our properties (settings) and keeping a mutex guard of the internal state around. If the internal state has no AudioInfo yet, we error out. This would mean that no caps were negotiated yet, which is something we can’t handle and is not really possible in our case.

Next we calculate how many samples we have to generate. If a sample stop position was set by a seek event, we have to generate samples up to at most that point. Otherwise we create at most the number of samples per buffer that were set via the property. Then we allocate a buffer of the corresponding size, with the help of the bpf field of the AudioInfo, and then set its metadata and fill the samples.

The metadata that is set is the timestamp (PTS), and the duration. The duration is calculated from the difference of the following buffer’s timestamp and the current buffer’s. By this we ensure that rounding errors are not causing the next buffer’s timestamp to have a different timestamp than the sum of the current’s and its duration. While this would not be much of a problem in GStreamer (inaccurate and jitterish timestamps are handled just fine), we can prevent it here and do so.

Afterwards we call our previously defined function on the writably mapped buffer and fill it with the sample values.

With all this, the element should already work just fine in any GStreamer-based application, for example gst-launch-1.0. Don’t forget to set the GST_PLUGIN_PATH environment variable correctly like last time. Before running this, make sure to turn down the volume of your speakers/headphones a bit.

export GST_PLUGIN_PATH=<code>{{EJS73}}</code>/target/debug
gst-launch-1.0 rssinesrc freq=440 volume=0.9 ! audioconvert ! autoaudiosink

You should hear a 440Hz sine wave now.

(Pseudo) Live Mode

Many audio (and video) sources can actually only produce data in real-time and data is produced according to some clock. So far our source element can produce data as fast as downstream is consuming data, but we optionally can change that. We simulate a live source here now by waiting on the pipeline clock, but with a real live source you would only ever be able to have the data in real-time without any need to wait on a clock. And usually that data is produced according to a different clock than the pipeline clock, in which case translation between the two clocks is needed but we ignore this aspect for now. For details check the GStreamer documentation.

For working in live mode, we have to add a few different parts in various places. First of all, we implement waiting on the clock in the create function.

    fn create(...
        [...]
        state.sample_offset += n_samples;
        drop(state);

        // If we're live, we are waiting until the time of the last sample in our buffer has
        // arrived. This is the very reason why we have to report that much latency.
        // A real live-source would of course only allow us to have the data available after
        // that latency, e.g. when capturing from a microphone, and no waiting from our side
        // would be necessary..
        //
        // Waiting happens based on the pipeline clock, which means that a real live source
        // with its own clock would require various translations between the two clocks.
        // This is out of scope for the tutorial though.
        if element.is_live() {
            let clock = match element.get_clock() {
                None => return Ok(buffer),
                Some(clock) => clock,
            };

            let segment = element
                .get_segment()
                .downcast::<gst::format::Time>()
                .unwrap();
            let base_time = element.get_base_time();
            let running_time = segment.to_running_time(buffer.get_pts() + buffer.get_duration());

            // The last sample's clock time is the base time of the element plus the
            // running time of the last sample
            let wait_until = running_time + base_time;
            if wait_until.is_none() {
                return Ok(buffer);
            }

            let id = clock.new_single_shot_id(wait_until).unwrap();

            gst_log!(
                self.cat,
                obj: element,
                "Waiting until {}, now {}",
                wait_until,
                clock.get_time()
            );
            let (res, jitter) = id.wait();
            gst_log!(
                self.cat,
                obj: element,
                "Waited res {:?} jitter {}",
                res,
                jitter
            );
        }

        gst_debug!(self.cat, obj: element, "Produced buffer {:?}", buffer);

        Ok(buffer)
    }

To be able to wait on the clock, we first of all need to calculate the clock time until when we want to wait. In our case that will be the clock time right after the end of the last sample in the buffer we just produced. Simply because you can’t capture a sample before it was produced.

We calculate the running time from the PTS and duration of the buffer with the help of the currently configured segment and then add the base time of the element on this to get the clock time as result. Please check the GStreamer documentation for details, but in short the running time of a pipeline is the time since the start of the pipeline (or the last reset of the running time) and the running time of a buffer can be calculated from its PTS and the segment, which provides the information to translate between the two. The base time is the clock time when the pipeline went to the Playing state, so just an offset.

Next we wait and then return the buffer as before.

Now we also have to tell the base class that we’re running in live mode now. This is done by calling set_live(true) on the base class before changing the element state from Ready to Paused. For this we override the Element::change_state virtual method.

impl ElementImpl<BaseSrc> for SineSrc {
    fn change_state(
        &self,
        element: &BaseSrc,
        transition: gst::StateChange,
    ) -> gst::StateChangeReturn {
        // Configure live'ness once here just before starting the source
        match transition {
            gst::StateChange::ReadyToPaused => {
                element.set_live(self.settings.lock().unwrap().is_live);
            }
            _ => (),
        }

        element.parent_change_state(transition)
    }
}

And as a last step, we also need to notify downstream elements about our latency. Live elements always have to report their latency so that synchronization can work correctly. As the clock time of each buffer is equal to the time when it was created, all buffers would otherwise arrive late in the sinks (they would appear as if they should’ve been played already at the time when they were created). So all the sinks will have to compensate for the latency that it took from capturing to the sink, and they have to do that in a coordinated way (otherwise audio and video would be out of sync if both have different latencies). For this the pipeline is querying each sink for the latency on its own branch, and then configures a global latency on all sinks according to that.

This querying is done with the LATENCY query, which we will now also have to handle.

    fn query(&self, element: &BaseSrc, query: &mut gst::QueryRef) -> bool {
        use gst::QueryView;

        match query.view_mut() {
            // We only work in Push mode. In Pull mode, create() could be called with
            // arbitrary offsets and we would have to produce for that specific offset
            QueryView::Scheduling(ref mut q) => {
                [...]
            }
            // In Live mode we will have a latency equal to the number of samples in each buffer.
            // We can't output samples before they were produced, and the last sample of a buffer
            // is produced that much after the beginning, leading to this latency calculation
            QueryView::Latency(ref mut q) => {
                let settings = *self.settings.lock().unwrap();
                let state = self.state.lock().unwrap();

                if let Some(ref info) = state.info {
                    let latency = gst::SECOND
                        .mul_div_floor(settings.samples_per_buffer as u64, info.rate() as u64)
                        .unwrap();
                    gst_debug!(self.cat, obj: element, "Returning latency {}", latency);
                    q.set(settings.is_live, latency, gst::CLOCK_TIME_NONE);
                    return true;
                } else {
                    return false;
                }
            }
            _ => (),
        }
        BaseSrcBase::parent_query(element, query)
    }

The latency that we report is the duration of a single audio buffer, because we’re simulating a real live source here. A real live source won’t be able to output the buffer before the last sample of it is captured, and the difference between when the first and last sample were captured is exactly the latency that we add here. Other elements further downstream that introduce further latency would then add their own latency on top of this.

Inside the latency query we also signal that we are indeed a live source, and additionally how much buffering we can do (in our case, infinite) until data would be lost. The last part is important if e.g. the video branch has a higher latency, causing the audio sink to have to wait some additional time (so that audio and video stay in sync), which would then require the whole audio branch to buffer some data. As we have an artificial live source, we can always generate data for the next time but a real live source would only have a limited buffer and if no data is read and forwarded once that runs full, data would get lost.

You can test this again with e.g. gst-launch-1.0 by setting the is-live property to true. It should write in the output now that the pipeline is live.

In Mathieu’s blog post this was implemented without explicit waiting and the usage of the get_times virtual method, but as this is only really useful for pseudo live sources like this one I decided to explain how waiting on the clock can be achieved correctly and even more important how that relates to the next section.

Unlocking

With the addition of the live mode, the create function is now blocking and waiting on the clock for some time. This is suboptimal as for example a (flushing) seek would have to wait now until the clock waiting is done, or when shutting down the application would have to wait.

To prevent this, all waiting/blocking in GStreamer streaming threads should be interruptible/cancellable when requested. And for example the ClockID that we got from the clock for waiting can be cancelled by calling unschedule() on it. We only have to do it from the right place and keep it accessible. The right place is the BaseSrc::unlock virtual method.

struct ClockWait {
    clock_id: Option<gst::ClockId>,
    flushing: bool,
}

struct SineSrc {
    cat: gst::DebugCategory,
    settings: Mutex<Settings>,
    state: Mutex<State>,
    clock_wait: Mutex<ClockWait>,
}

[...]

    fn unlock(&self, element: &BaseSrc) -> bool {
        // This should unblock the create() function ASAP, so we
        // just unschedule the clock it here, if any.
        gst_debug!(self.cat, obj: element, "Unlocking");
        let mut clock_wait = self.clock_wait.lock().unwrap();
        if let Some(clock_id) = clock_wait.clock_id.take() {
            clock_id.unschedule();
        }
        clock_wait.flushing = true;

        true
    }

We store the clock ID in our struct, together with a boolean to signal whether we’re supposed to flush already or not. And then inside unlock unschedule the clock ID and set this boolean flag to true.

Once everything is unlocked, we need to reset things again so that data flow can happen in the future. This is done in the unlock_stop virtual method.

    fn unlock_stop(&self, element: &BaseSrc) -> bool {
        // This signals that unlocking is done, so we can reset
        // all values again.
        gst_debug!(self.cat, obj: element, "Unlock stop");
        let mut clock_wait = self.clock_wait.lock().unwrap();
        clock_wait.flushing = false;

        true
    }

To make sure that this struct is always initialized correctly, we also call unlock from stop, and unlock_stop from start.

Now as a last step, we need to actually make use of the new struct we added around the code where we wait for the clock.

            // Store the clock ID in our struct unless we're flushing anyway.
            // This allows to asynchronously cancel the waiting from unlock()
            // so that we immediately stop waiting on e.g. shutdown.
            let mut clock_wait = self.clock_wait.lock().unwrap();
            if clock_wait.flushing {
                gst_debug!(self.cat, obj: element, "Flushing");
                return Err(gst::FlowReturn::Flushing);
            }

            let id = clock.new_single_shot_id(wait_until).unwrap();
            clock_wait.clock_id = Some(id.clone());
            drop(clock_wait);

            gst_log!(
                self.cat,
                obj: element,
                "Waiting until {}, now {}",
                wait_until,
                clock.get_time()
            );
            let (res, jitter) = id.wait();
            gst_log!(
                self.cat,
                obj: element,
                "Waited res {:?} jitter {}",
                res,
                jitter
            );
            self.clock_wait.lock().unwrap().clock_id.take();

            // If the clock ID was unscheduled, unlock() was called
            // and we should return Flushing immediately.
            if res == gst::ClockReturn::Unscheduled {
                gst_debug!(self.cat, obj: element, "Flushing");
                return Err(gst::FlowReturn::Flushing);
            }

The important part in this code is that we first have to check if we are already supposed to unlock, before even starting to wait. Otherwise we would start waiting without anybody ever being able to unlock. Then we need to store the clock id in the struct and make sure to drop the mutex guard so that the unlock function can take it again for unscheduling the clock ID. And once waiting is done, we need to remove the clock id from the struct again and in case of ClockReturn::Unscheduled we directly return FlowReturn::Flushing instead of the error.

Similarly when using other blocking APIs it is important that they are woken up in a similar way when unlock is called. Otherwise the application developer’s and thus user experience will be far from ideal.

Seeking

As a last feature we implement seeking on our source element. In our case that only means that we have to update the sample_offset and sample_stop fields accordingly, other sources might have to do more work than that.

Seeking is implemented in the BaseSrc::do_seek virtual method, and signalling whether we can actually seek in the is_seekable virtual method.

    fn is_seekable(&self, _element: &BaseSrc) -> bool {
        true
    }

    fn do_seek(&self, element: &BaseSrc, segment: &mut gst::Segment) -> bool {
        // Handle seeking here. For Time and Default (sample offset) seeks we can
        // do something and have to update our sample offset and accumulator accordingly.
        //
        // Also we should remember the stop time (so we can stop at that point), and if
        // reverse playback is requested. These values will all be used during buffer creation
        // and for calculating the timestamps, etc.

        if segment.get_rate() < 0.0 {
            gst_error!(self.cat, obj: element, "Reverse playback not supported");
            return false;
        }

        let settings = *self.settings.lock().unwrap();
        let mut state = self.state.lock().unwrap();

        // We store sample_offset and sample_stop in nanoseconds if we
        // don't know any sample rate yet. It will be converted correctly
        // once a sample rate is known.
        let rate = match state.info {
            None => gst::SECOND_VAL,
            Some(ref info) => info.rate() as u64,
        };

        if let Some(segment) = segment.downcast_ref::<gst::format::Time>() {
            use std::f64::consts::PI;

            let sample_offset = segment
                .get_start()
                .unwrap()
                .mul_div_floor(rate, gst::SECOND_VAL)
                .unwrap();

            let sample_stop = segment
                .get_stop()
                .map(|v| v.mul_div_floor(rate, gst::SECOND_VAL).unwrap());

            let accumulator =
                (sample_offset as f64).rem(2.0 * PI * (settings.freq as f64) / (rate as f64));

            gst_debug!(
                self.cat,
                obj: element,
                "Seeked to {}-{:?} (accum: {}) for segment {:?}",
                sample_offset,
                sample_stop,
                accumulator,
                segment
            );

            *state = State {
                info: state.info.clone(),
                sample_offset: sample_offset,
                sample_stop: sample_stop,
                accumulator: accumulator,
            };

            true
        } else if let Some(segment) = segment.downcast_ref::<gst::format::Default>() {
            use std::f64::consts::PI;

            if state.info.is_none() {
                gst_error!(
                    self.cat,
                    obj: element,
                    "Can only seek in Default format if sample rate is known"
                );
                return false;
            }

            let sample_offset = segment.get_start().unwrap();
            let sample_stop = segment.get_stop().0;

            let accumulator =
                (sample_offset as f64).rem(2.0 * PI * (settings.freq as f64) / (rate as f64));

            gst_debug!(
                self.cat,
                obj: element,
                "Seeked to {}-{:?} (accum: {}) for segment {:?}",
                sample_offset,
                sample_stop,
                accumulator,
                segment
            );

            *state = State {
                info: state.info.clone(),
                sample_offset: sample_offset,
                sample_stop: sample_stop,
                accumulator: accumulator,
            };

            true
        } else {
            gst_error!(
                self.cat,
                obj: element,
                "Can't seek in format {:?}",
                segment.get_format()
            );

            false
        }
    }

Currently no support for reverse playback is implemented here, that is left as an exercise for the reader. So as a first step we check if the segment has a negative rate, in which case we just fail and return false.

Afterwards we again take a copy of the settings, keep a mutable mutex guard of our state and then start handling the actual seek.

If no caps are known yet, i.e. the AudioInfo is None, we assume a rate of 1 billion. That is, we just store the time in nanoseconds for now and let the set_caps function take care of that (which we already implemented accordingly) once the sample rate is known.

Then, if a Time seek is performed, we convert the segment start and stop position from time to sample offsets and save them. And then update the accumulator in a similar way as in the set_caps function. If a seek is in Default format (i.e. sample offsets for raw audio), we just have to store the values and update the accumulator but only do so if the sample rate is known already. A sample offset seek does not make any sense until the sample rate is known, so we just fail here to prevent unexpected surprises later.

Try the following pipeline for testing seeking. You should be able to seek the current time drawn over the video, and with the left/right cursor key you can seek. Also this shows that we create a quite nice sine wave.

gst-launch-1.0 rssinesrc ! audioconvert ! monoscope ! timeoverlay ! navseek ! glimagesink

And with that all features are implemented in our sine wave raw audio source.

This is part one of a series of blog posts that I’ll write in the next weeks, as previously announced in the GStreamer Rust bindings 0.10.0 release blog post. Since the last series of blog posts about writing GStreamer plugins in Rust ([1] [2] [3] [4]) a lot has changed, and the content of those blog posts has only historical value now, as the journey of experimentation to what exists now.

In this first part we’re going to write a plugin that contains a video filter element. The video filter can convert from RGB to grayscale, either output as 8-bit per pixel grayscale or 32-bit per pixel RGB. In addition there’s a property to invert all grayscale values, or to shift them by up to 255 values. In the end this will allow you to watch Big Bucky Bunny, or anything else really that can somehow go into a GStreamer pipeline, in grayscale. Or encode the output to a new video file, send it over the network via WebRTC or something else, or basically do anything you want with it.

This will show the basics of how to write a GStreamer plugin and element in Rust: the basic setup for registering a type and implementing it in Rust, and how to use the various GStreamer API and APIs from the Rust standard library to do the processing.

The final code for this plugin can be found here, and it is based on the 0.1 version of the gst-plugin crate and the 0.10 version of the gstreamer crate. At least Rust 1.20 is required for all this. I’m also assuming that you have GStreamer (at least version 1.8) installed for your platform, see e.g. the GStreamer bindings installation instructions.

Table of Contents

1. Project Structure
2. Plugin Initialization
3. Type Registration
4. Type Class & Instance Initialization
5. Caps & Pad Templates
6. Caps Handling Part 1
7. Caps Handling Part 2
8. Conversion of BGRx Video Frames to Grayscale
9. Testing the new element
10. Properties
11. What next?

Project Structure

We’ll create a new cargo project with cargo init –lib –name gst-plugin-tutorial. This will create a basically empty Cargo.toml and a corresponding src/lib.rs. We will use this structure: lib.rs will contain all the plugin related code, separate modules will contain any GStreamer plugins that are added.

The empty Cargo.toml has to be updated to list all the dependencies that we need, and to define that the crate should result in a cdylib, i.e. a C library that does not contain any Rust-specific metadata. The final Cargo.toml looks as follows

[package]
name = "gst-plugin-tutorial"
version = "0.1.0"
authors = ["Sebastian Dröge <sebastian@centricular.com>"]
repository = "https://github.com/sdroege/gst-plugin-rs"
license = "MIT/Apache-2.0"

[dependencies]
glib = "0.4"
gstreamer = "0.10"
gstreamer-base = "0.10"
gstreamer-video = "0.10"
gst-plugin = "0.1"

[lib]
name = "gstrstutorial"
crate-type = ["cdylib"]
path = "src/lib.rs"

We’re depending on the gst-plugin crate, which provides all the basic infrastructure for implementing GStreamer plugins and elements. In addition we depend on the gstreamer, gstreamer-base and gstreamer-video crates for various GStreamer API that we’re going to use later, and the glib crate to be able to use some GLib API that we’ll need. GStreamer is building upon GLib, and this leaks through in various places.

With the basic project structure being set-up, we should be able to compile the project with cargo build now, which will download and build all dependencies and then creates a file called target/debug/libgstrstutorial.so (or .dll on Windows, .dylib on macOS). This is going to be our GStreamer plugin.

To allow GStreamer to find our new plugin and make it available in every GStreamer-based application, we could install it into the system- or user-wide GStreamer plugin path or simply point the GST_PLUGIN_PATH environment variable to the directory containing it:

export GST_PLUGIN_PATH=<code>{{EJS90}}</code>/target/debug

If you now run the gst-inspect-1.0 tool on the libgstrstutorial.so, it will not yet print all information it can extract from the plugin but for now just complains that this is not a valid GStreamer plugin. Which is true, we didn’t write any code for it yet.

Plugin Initialization

Let’s start editing src/lib.rs to make this an actual GStreamer plugin. First of all, we need to add various extern crate directives to be able to use our dependencies and also mark some of them #[macro_use] because we’re going to use macros defined in some of them. This looks like the following

extern crate glib;
#[macro_use]
extern crate gstreamer as gst;
extern crate gstreamer_base as gst_base;
extern crate gstreamer_video as gst_video;
#[macro_use]
extern crate gst_plugin;

Next we make use of the plugin_define! macro from the gst-plugin crate to set-up the static metadata of the plugin (and make the shared library recognizeable by GStreamer to be a valid plugin), and to define the name of our entry point function (plugin_init) where we will register all the elements that this plugin provides.

plugin_define!(
    b"rstutorial\0",
    b"Rust Tutorial Plugin\0",
    plugin_init,
    b"1.0\0",
    b"MIT/X11\0",
    b"rstutorial\0",
    b"rstutorial\0",
    b"https://github.com/sdroege/gst-plugin-rs\0",
    b"2017-12-30\0"
);

This is unfortunately not very beautiful yet due to a) GStreamer requiring this information to be statically available in the shared library, not returned by a function (starting with GStreamer 1.14 it can be a function), and b) Rust not allowing raw strings (b”blabla) to be concatenated with a macro like the std::concat macro (so that the b and \0 parts could be hidden away). Expect this to become better in the future.

The static plugin metadata that we provide here is

1. name of the plugin
2. short description for the plugin
3. name of the plugin entry point function
4. version number of the plugin
5. license of the plugin (only a fixed set of licenses is allowed here, see)
6. source package name
7. binary package name (only really makes sense for e.g. Linux distributions)
8. origin of the plugin
9. release date of this version

In addition we’re defining an empty plugin entry point function that just returns true

fn plugin_init(plugin: &gst::Plugin) -> bool {
    true
}

With all that given, gst-inspect-1.0 should print exactly this information when running on the libgstrstutorial.so file (or .dll on Windows, or .dylib on macOS)

gst-inspect-1.0 target/debug/libgstrstutorial.so

Type Registration

As a next step, we’re going to add another module rgb2gray to our project, and call a function called register from our plugin_init function.

mod rgb2gray;

fn plugin_init(plugin: &gst::Plugin) -> bool {
    rgb2gray::register(plugin);
    true
}

With that our src/lib.rs is complete, and all following code is only in src/rgb2gray.rs. At the top of the new file we first need to add various use-directives to import various types and functions we’re going to use into the current module’s scope

use glib;
use gst;
use gst::prelude::*;
use gst_video;

use gst_plugin::properties::*;
use gst_plugin::object::*;
use gst_plugin::element::*;
use gst_plugin::base_transform::*;

use std::i32;
use std::sync::Mutex;

GStreamer is based on the GLib object system (GObject). C (just like Rust) does not have built-in support for object orientated programming, inheritance, virtual methods and related concepts, and GObject makes these features available in C as a library. Without language support this is a quite verbose endeavour in C, and the gst-plugin crate tries to expose all this in a (as much as possible) Rust-style API while hiding all the details that do not really matter.

So, as a next step we need to register a new type for our RGB to Grayscale converter GStreamer element with the GObject type system, and then register that type with GStreamer to be able to create new instances of it. We do this with the following code

struct Rgb2GrayStatic;

impl ImplTypeStatic<BaseTransform> for Rgb2GrayStatic {
    fn get_name(&self) -> &str {
        "Rgb2Gray"
    }

    fn new(&self, element: &BaseTransform) -> Box<BaseTransformImpl<BaseTransform>> {
        Rgb2Gray::new(element)
    }

    fn class_init(&self, klass: &mut BaseTransformClass) {
        Rgb2Gray::class_init(klass);
    }
}

pub fn register(plugin: &gst::Plugin) {
    let type_ = register_type(Rgb2GrayStatic);
    gst::Element::register(plugin, "rsrgb2gray", 0, type_);
}

This defines a zero-sized struct Rgb2GrayStatic that is used to implement the ImplTypeStatic<BaseTransform> trait on it for providing static information about the type to the type system. In our case this is a zero-sized struct, but in other cases this struct might contain actual data (for example if the same element code is used for multiple elements, e.g. when wrapping a generic codec API that provides support for multiple decoders and then wanting to register one element per decoder). By implementing ImplTypeStatic<BaseTransform> we also declare that our element is going to be based on the GStreamer BaseTransform base class, which provides a relatively simple API for 1:1 transformation elements like ours is going to be.

ImplTypeStatic provides functions that return a name for the type, and functions for initializing/returning a new instance of our element (new) and for initializing the class metadata (class_init, more on that later). We simply let those functions proxy to associated functions on the Rgb2Gray struct that we’re going to define at a later time.

In addition, we also define a register function (the one that is already called from our plugin_init function) and in there first register the Rgb2GrayStatic type metadata with the GObject type system to retrieve a type ID, and then register this type ID to GStreamer to be able to create new instances of it with the name “rsrgb2gray” (e.g. when using gst::ElementFactory::make).

Type Class & Instance Initialization

As a next step we declare the Rgb2Gray struct and implement the new and class_init functions on it. In the first version, this struct is almost empty but we will later use it to store all state of our element.

struct Rgb2Gray {
    cat: gst::DebugCategory,
}

impl Rgb2Gray {
    fn new(_transform: &BaseTransform) -> Box<BaseTransformImpl<BaseTransform>> {
        Box::new(Self {
            cat: gst::DebugCategory::new(
                "rsrgb2gray",
                gst::DebugColorFlags::empty(),
                "Rust RGB-GRAY converter",
            ),
        })
    }

    fn class_init(klass: &mut BaseTransformClass) {
        klass.set_metadata(
            "RGB-GRAY Converter",
            "Filter/Effect/Converter/Video",
            "Converts RGB to GRAY or grayscale RGB",
            "Sebastian Dröge <sebastian@centricular.com>",
        );

        klass.configure(BaseTransformMode::NeverInPlace, false, false);
    }
}

In the new function we return a boxed (i.e. heap-allocated) version of our struct, containing a newly created GStreamer debug category of name “rsrgb2gray”. We’re going to use this debug category later for making use of GStreamer’s debug logging system for logging the state and changes of our element.

In the class_init function we, again, set up some metadata for our new element. In this case these are a description, a classification of our element, a longer description and the author. The metadata can later be retrieved and made use of via the Registry and PluginFeature/ElementFactory API. We also configure the BaseTransform class and define that we will never operate in-place (producing our output in the input buffer), and that we don’t want to work in passthrough mode if the input/output formats are the same.

Additionally we need to implement various traits on the Rgb2Gray struct, which will later be used to override virtual methods of the various parent classes of our element. For now we can keep the trait implementations empty. There is one trait implementation required per parent class.

impl ObjectImpl<BaseTransform> for Rgb2Gray {}
impl ElementImpl<BaseTransform> for Rgb2Gray {}
impl BaseTransformImpl<BaseTransform> for Rgb2Gray {}

With all this defined, gst-inspect-1.0 should be able to show some more information about our element already but will still complain that it’s not complete yet.

Caps & Pad Templates

Data flow of GStreamer elements is happening via pads, which are the input(s) and output(s) (or sinks and sources) of an element. Via the pads, buffers containing actual media data, events or queries are transferred. An element can have any number of sink and source pads, but our new element will only have one of each.

To be able to declare what kinds of pads an element can create (they are not necessarily all static but could be created at runtime by the element or the application), it is necessary to install so-called pad templates during the class initialization. These pad templates contain the name (or rather “name template”, it could be something like src_%u for e.g. pad templates that declare multiple possible pads), the direction of the pad (sink or source), the availability of the pad (is it always there, sometimes added/removed by the element or to be requested by the application) and all the possible media types (called caps) that the pad can consume (sink pads) or produce (src pads).

In our case we only have always pads, one sink pad called “sink”, on which we can only accept RGB (BGRx to be exact) data with any width/height/framerate and one source pad called “src”, on which we will produce either RGB (BGRx) data or GRAY8 (8-bit grayscale) data. We do this by adding the following code to the class_init function.

        let caps = gst::Caps::new_simple(
            "video/x-raw",
            &[
                (
                    "format",
                    &gst::List::new(&[
                        &gst_video::VideoFormat::Bgrx.to_string(),
                        &gst_video::VideoFormat::Gray8.to_string(),
                    ]),
                ),
                ("width", &gst::IntRange::<i32>::new(0, i32::MAX)),
                ("height", &gst::IntRange::<i32>::new(0, i32::MAX)),
                (
                    "framerate",
                    &gst::FractionRange::new(
                        gst::Fraction::new(0, 1),
                        gst::Fraction::new(i32::MAX, 1),
                    ),
                ),
            ],
        );

        let src_pad_template = gst::PadTemplate::new(
            "src",
            gst::PadDirection::Src,
            gst::PadPresence::Always,
            &caps,
        );
        klass.add_pad_template(src_pad_template);

        let caps = gst::Caps::new_simple(
            "video/x-raw",
            &[
                ("format", &gst_video::VideoFormat::Bgrx.to_string()),
                ("width", &gst::IntRange::<i32>::new(0, i32::MAX)),
                ("height", &gst::IntRange::<i32>::new(0, i32::MAX)),
                (
                    "framerate",
                    &gst::FractionRange::new(
                        gst::Fraction::new(0, 1),
                        gst::Fraction::new(i32::MAX, 1),
                    ),
                ),
            ],
        );

        let sink_pad_template = gst::PadTemplate::new(
            "sink",
            gst::PadDirection::Sink,
            gst::PadPresence::Always,
            &caps,
        );
        klass.add_pad_template(sink_pad_template);

The names “src” and “sink” are pre-defined by the BaseTransform class and this base-class will also create the actual pads with those names from the templates for us whenever a new element instance is created. Otherwise we would have to do that in our new function but here this is not needed.

If you now run gst-inspect-1.0 on the rsrgb2gray element, these pad templates with their caps should also show up.

Caps Handling Part 1

As a next step we will add caps handling to our new element. This involves overriding 4 virtual methods from the BaseTransformImpl trait, and actually storing the configured input and output caps inside our element struct. Let’s start with the latter

struct State {
    in_info: gst_video::VideoInfo,
    out_info: gst_video::VideoInfo,
}

struct Rgb2Gray {
    cat: gst::DebugCategory,
    state: Mutex<Option<State>>,
}

impl Rgb2Gray {
    fn new(_transform: &BaseTransform) -> Box<BaseTransformImpl<BaseTransform>> {
        Box::new(Self {
            cat: gst::DebugCategory::new(
                "rsrgb2gray",
                gst::DebugColorFlags::empty(),
                "Rust RGB-GRAY converter",
            ),
            state: Mutex::new(None),
        })
    }
}

We define a new struct State that contains the input and output caps, stored in a VideoInfo. VideoInfo is a struct that contains various fields like width/height, framerate and the video format and allows to conveniently with the properties of (raw) video formats. We have to store it inside a Mutex in our Rgb2Gray struct as this can (in theory) be accessed from multiple threads at the same time.

Whenever input/output caps are configured on our element, the set_caps virtual method of BaseTransform is called with both caps (i.e. in the very beginning before the data flow and whenever it changes), and all following video frames that pass through our element should be according to those caps. Once the element is shut down, the stop virtual method is called and it would make sense to release the State as it only contains stream-specific information. We’re doing this by adding the following to the BaseTransformImpl trait implementation

impl BaseTransformImpl<BaseTransform> for Rgb2Gray {
    fn set_caps(&self, element: &BaseTransform, incaps: &gst::Caps, outcaps: &gst::Caps) -> bool {
        let in_info = match gst_video::VideoInfo::from_caps(incaps) {
            None => return false,
            Some(info) => info,
        };
        let out_info = match gst_video::VideoInfo::from_caps(outcaps) {
            None => return false,
            Some(info) => info,
        };

        gst_debug!(
            self.cat,
            obj: element,
            "Configured for caps {} to {}",
            incaps,
            outcaps
        );

        *self.state.lock().unwrap() = Some(State {
            in_info: in_info,
            out_info: out_info,
        });

        true
    }

    fn stop(&self, element: &BaseTransform) -> bool {
        // Drop state
        let _ = self.state.lock().unwrap().take();

        gst_info!(self.cat, obj: element, "Stopped");

        true
    }
}

This code should be relatively self-explanatory. In set_caps we’re parsing the two caps into a VideoInfo and then store this in our State, in stop we drop the State and replace it with None. In addition we make use of our debug category here and use the gst_info! and gst_debug! macros to output the current caps configuration to the GStreamer debug logging system. This information can later be useful for debugging any problems once the element is running.

Next we have to provide information to the BaseTransform base class about the size in bytes of a video frame with specific caps. This is needed so that the base class can allocate an appropriately sized output buffer for us, that we can then fill later. This is done with the get_unit_size virtual method, which is required to return the size of one processing unit in specific caps. In our case, one processing unit is one video frame. In the case of raw audio it would be the size of one sample multiplied by the number of channels.

impl BaseTransformImpl<BaseTransform> for Rgb2Gray {
    fn get_unit_size(&self, _element: &BaseTransform, caps: &gst::Caps) -> Option<usize> {
        gst_video::VideoInfo::from_caps(caps).map(|info| info.size())
    }
}

We simply make use of the VideoInfo API here again, which conveniently gives us the size of one video frame already.

Instead of get_unit_size it would also be possible to implement the transform_size virtual method, which is getting passed one size and the corresponding caps, another caps and is supposed to return the size converted to the second caps. Depending on how your element works, one or the other can be easier to implement.

Caps Handling Part 2

We’re not done yet with caps handling though. As a very last step it is required that we implement a function that is converting caps into the corresponding caps in the other direction. For example, if we receive BGRx caps with some width/height on the sinkpad, we are supposed to convert this into new caps with the same width/height but BGRx or GRAY8. That is, we can convert BGRx to BGRx or GRAY8. Similarly, if the element downstream of ours can accept GRAY8 with a specific width/height from our source pad, we have to convert this to BGRx with that very same width/height.

This has to be implemented in the transform_caps virtual method, and looks as following

impl BaseTransformImpl<BaseTransform> for Rgb2Gray {
    fn transform_caps(
        &self,
        element: &BaseTransform,
        direction: gst::PadDirection,
        caps: gst::Caps,
        filter: Option<&gst::Caps>,
    ) -> gst::Caps {
        let other_caps = if direction == gst::PadDirection::Src {
            let mut caps = caps.clone();

            for s in caps.make_mut().iter_mut() {
                s.set("format", &gst_video::VideoFormat::Bgrx.to_string());
            }

            caps
        } else {
            let mut gray_caps = gst::Caps::new_empty();

            {
                let gray_caps = gray_caps.get_mut().unwrap();

                for s in caps.iter() {
                    let mut s_gray = s.to_owned();
                    s_gray.set("format", &gst_video::VideoFormat::Gray8.to_string());
                    gray_caps.append_structure(s_gray);
                }
                gray_caps.append(caps.clone());
            }

            gray_caps
        };

        gst_debug!(
            self.cat,
            obj: element,
            "Transformed caps from {} to {} in direction {:?}",
            caps,
            other_caps,
            direction
        );

        if let Some(filter) = filter {
            filter.intersect_with_mode(&other_caps, gst::CapsIntersectMode::First)
        } else {
            other_caps
        }
    }
}

This caps conversion happens in 3 steps. First we check if we got caps for the source pad. In that case, the caps on the other pad (the sink pad) are going to be exactly the same caps but no matter if the caps contained BGRx or GRAY8 they must become BGRx as that’s the only format that our sink pad can accept. We do this by creating a clone of the input caps, then making sure that those caps are actually writable (i.e. we’re having the only reference to them, or a copy is going to be created) and then iterate over all the structures inside the caps and then set the “format” field to BGRx. After this, all structures in the new caps will be with the format field set to BGRx.

Similarly, if we get caps for the sink pad and are supposed to convert it to caps for the source pad, we create new caps and in there append a copy of each structure of the input caps (which are BGRx) with the format field set to GRAY8. In the end we append the original caps, giving us first all caps as GRAY8 and then the same caps as BGRx. With this ordering we signal to GStreamer that we would prefer to output GRAY8 over BGRx.

In the end the caps we created for the other pad are filtered against optional filter caps to reduce the potential size of the caps. This is done by intersecting the caps with that filter, while keeping the order (and thus preferences) of the filter caps (gst::CapsIntersectMode::First).

Conversion of BGRx Video Frames to Grayscale

Now that all the caps handling is implemented, we can finally get to the implementation of the actual video frame conversion. For this we start with defining a helper function bgrx_to_gray that converts one BGRx pixel to a grayscale value. The BGRx pixel is passed as a &[u8] slice with 4 elements and the function returns another u8 for the grayscale value.

impl Rgb2Gray {
    #[inline]
    fn bgrx_to_gray(in_p: &[u8]) -> u8 {
        // See https://en.wikipedia.org/wiki/YUV#SDTV_with_BT.601
        const R_Y: u32 = 19595; // 0.299 * 65536
        const G_Y: u32 = 38470; // 0.587 * 65536
        const B_Y: u32 = 7471; // 0.114 * 65536

        assert_eq!(in_p.len(), 4);

        let b = u32::from(in_p[0]);
        let g = u32::from(in_p[1]);
        let r = u32::from(in_p[2]);

        let gray = ((r * R_Y) + (g * G_Y) + (b * B_Y)) / 65536;
        (gray as u8)
    }
}

This function works by extracting the blue, green and red components from each pixel (remember: we work on BGRx, so the first value will be blue, the second green, the third red and the fourth unused), extending it from 8 to 32 bits for a wider value-range and then converts it to the Y component of the YUV colorspace (basically what your grandparents’ black & white TV would’ve displayed). The coefficients come from the Wikipedia page about YUV and are normalized to unsigned 16 bit integers so we can keep some accuracy, don’t have to work with floating point arithmetic and stay inside the range of 32 bit integers for all our calculations. As you can see, the green component is weighted more than the others, which comes from our eyes being more sensitive to green than to other colors.

Note: This is only doing the actual conversion from linear RGB to grayscale (and in BT.601 colorspace). To do this conversion correctly you need to know your colorspaces and use the correct coefficients for conversion, and also do gamma correction. See this about why it is important.

Afterwards we have to actually call this function on every pixel. For this the transform virtual method is implemented, which gets a input and output buffer passed and we’re supposed to read the input buffer and fill the output buffer. The implementation looks as follows, and is going to be our biggest function for this element

impl BaseTransformImpl<BaseTransform> for Rgb2Gray {
    fn transform(
        &self,
        element: &BaseTransform,
        inbuf: &gst::Buffer,
        outbuf: &mut gst::BufferRef,
    ) -> gst::FlowReturn {
        let mut state_guard = self.state.lock().unwrap();
        let state = match *state_guard {
            None => {
                gst_element_error!(element, gst::CoreError::Negotiation, ["Have no state yet"]);
                return gst::FlowReturn::NotNegotiated;
            }
            Some(ref mut state) => state,
        };

        let in_frame = match gst_video::VideoFrameRef::from_buffer_ref_readable(
            inbuf.as_ref(),
            &state.in_info,
        ) {
            None => {
                gst_element_error!(
                    element,
                    gst::CoreError::Failed,
                    ["Failed to map input buffer readable"]
                );
                return gst::FlowReturn::Error;
            }
            Some(in_frame) => in_frame,
        };

        let mut out_frame =
            match gst_video::VideoFrameRef::from_buffer_ref_writable(outbuf, &state.out_info) {
                None => {
                    gst_element_error!(
                        element,
                        gst::CoreError::Failed,
                        ["Failed to map output buffer writable"]
                    );
                    return gst::FlowReturn::Error;
                }
                Some(out_frame) => out_frame,
            };

        let width = in_frame.width() as usize;
        let in_stride = in_frame.plane_stride()[0] as usize;
        let in_data = in_frame.plane_data(0).unwrap();
        let out_stride = out_frame.plane_stride()[0] as usize;
        let out_format = out_frame.format();
        let out_data = out_frame.plane_data_mut(0).unwrap();

        if out_format == gst_video::VideoFormat::Bgrx {
            assert_eq!(in_data.len() % 4, 0);
            assert_eq!(out_data.len() % 4, 0);
            assert_eq!(out_data.len() / out_stride, in_data.len() / in_stride);

            let in_line_bytes = width * 4;
            let out_line_bytes = width * 4;

            assert!(in_line_bytes <= in_stride);
            assert!(out_line_bytes <= out_stride);

            for (in_line, out_line) in in_data
                .chunks(in_stride)
                .zip(out_data.chunks_mut(out_stride))
            {
                for (in_p, out_p) in in_line[..in_line_bytes]
                    .chunks(4)
                    .zip(out_line[..out_line_bytes].chunks_mut(4))
                {
                    assert_eq!(out_p.len(), 4);

                    let gray = Rgb2Gray::bgrx_to_gray(in_p);
                    out_p[0] = gray;
                    out_p[1] = gray;
                    out_p[2] = gray;
                }
            }
        } else if out_format == gst_video::VideoFormat::Gray8 {
            assert_eq!(in_data.len() % 4, 0);
            assert_eq!(out_data.len() / out_stride, in_data.len() / in_stride);

            let in_line_bytes = width * 4;
            let out_line_bytes = width;

            assert!(in_line_bytes <= in_stride);
            assert!(out_line_bytes <= out_stride);

            for (in_line, out_line) in in_data
                .chunks(in_stride)
                .zip(out_data.chunks_mut(out_stride))
            {
                for (in_p, out_p) in in_line[..in_line_bytes]
                    .chunks(4)
                    .zip(out_line[..out_line_bytes].iter_mut())
                {
                    let gray = Rgb2Gray::bgrx_to_gray(in_p);
                    *out_p = gray;
                }
            }
        } else {
            unimplemented!();
        }

        gst::FlowReturn::Ok
    }
}

What happens here is that we first of all lock our state (the input/output VideoInfo) and error out if we don’t have any yet (which can’t really happen unless other elements have a bug, but better safe than sorry). After that we map the input buffer readable and the output buffer writable with the VideoFrameRef API. By mapping the buffers we get access to the underlying bytes of them, and the mapping operation could for example make GPU memory available or just do nothing and give us access to a normally allocated memory area. We have access to the bytes of the buffer until the VideoFrameRef goes out of scope.

Instead of VideoFrameRef we could’ve also used the gst::Buffer::map_readable() and gst::Buffer::map_writable() API, but different to those the VideoFrameRef API also extracts various metadata from the raw video buffers and makes them available. For example we can directly access the different planes as slices without having to calculate the offsets ourselves, or we get directly access to the width and height of the video frame.

After mapping the buffers, we store various information we’re going to need later in local variables to save some typing later. This is the width (same for input and output as we never changed the width in transform_caps), the input and out (row-) stride (the number of bytes per row/line, which possibly includes some padding at the end of each line for alignment reasons), the output format (which can be BGRx or GRAY8 because of how we implemented transform_caps) and the pointers to the first plane of the input and output (which in this case also is the only plane, BGRx and GRAY8 both have only a single plane containing all the RGB/gray components).

Then based on whether the output is BGRx or GRAY8, we iterate over all pixels. The code is basically the same in both cases, so I’m only going to explain the case where BGRx is output.

We start by iterating over each line of the input and output, and do so by using the chunks iterator to give us chunks of as many bytes as the (row-) stride of the video frame is, do the same for the other frame and then zip both iterators together. This means that on each iteration we get exactly one line as a slice from each of the frames and can then start accessing the actual pixels in each line.

To access the individual pixels in each line, we again use the chunks iterator the same way, but this time to always give us chunks of 4 bytes from each line. As BGRx uses 4 bytes for each pixel, this gives us exactly one pixel. Instead of iterating over the whole line, we only take the actual sub-slice that contains the pixels, not the whole line with stride number of bytes containing potential padding at the end. Now for each of these pixels we call our previously defined bgrx_to_gray function and then fill the B, G and R components of the output buffer with that value to get grayscale output. And that’s all.

Using Rust high-level abstractions like the chunks iterators and bounds-checking slice accesses might seem like it’s going to cause quite some performance penalty, but if you look at the generated assembly most of the bounds checks are completely optimized away and the resulting assembly code is close to what one would’ve written manually (especially when using the newly-added exact_chunks iterators). Here you’re getting safe and high-level looking code with low-level performance!

You might’ve also noticed the various assertions in the processing function. These are there to give further hints to the compiler about properties of the code, and thus potentially being able to optimize the code better and moving e.g. bounds checks out of the inner loop and just having the assertion outside the loop check for the same. In Rust adding assertions can often improve performance by allowing further optimizations to be applied, but in the end always check the resulting assembly to see if what you did made any difference.

Testing the new element

Now we implemented almost all functionality of our new element and could run it on actual video data. This can be done now with the gst-launch-1.0 tool, or any application using GStreamer and allowing us to insert our new element somewhere in the video part of the pipeline. With gst-launch-1.0 you could run for example the following pipelines

# Run on a test pattern
gst-launch-1.0 videotestsrc ! rsrgb2gray ! videoconvert ! autovideosink

# Run on some video file, also playing the audio
gst-launch-1.0 playbin uri=file:///path/to/some/file video-filter=rsrgb2gray

Note that you will likely want to compile with cargo build –release and add the target/release directory to GST_PLUGIN_PATH instead. The debug build might be too slow, and generally the release builds are multiple orders of magnitude (!) faster.

Properties

The only feature missing now are the properties I mentioned in the opening paragraph: one boolean property to invert the grayscale value and one integer property to shift the value by up to 255. Implementing this on top of the previous code is not a lot of work. Let’s start with defining a struct for holding the property values and defining the property metadata.

const DEFAULT_INVERT: bool = false;
const DEFAULT_SHIFT: u32 = 0;

#[derive(Debug, Clone, Copy)]
struct Settings {
    invert: bool,
    shift: u32,
}

impl Default for Settings {
    fn default() -> Self {
        Settings {
            invert: DEFAULT_INVERT,
            shift: DEFAULT_SHIFT,
        }
    }
}

static PROPERTIES: [Property; 2] = [
    Property::Boolean(
        "invert",
        "Invert",
        "Invert grayscale output",
        DEFAULT_INVERT,
        PropertyMutability::ReadWrite,
    ),
    Property::UInt(
        "shift",
        "Shift",
        "Shift grayscale output (wrapping around)",
        (0, 255),
        DEFAULT_SHIFT,
        PropertyMutability::ReadWrite,
    ),
];

struct Rgb2Gray {
    cat: gst::DebugCategory,
    settings: Mutex<Settings>,
    state: Mutex<Option<State>>,
}

impl Rgb2Gray {
    fn new(_transform: &BaseTransform) -> Box<BaseTransformImpl<BaseTransform>> {
        Box::new(Self {
            cat: gst::DebugCategory::new(
                "rsrgb2gray",
                gst::DebugColorFlags::empty(),
                "Rust RGB-GRAY converter",
            ),
            settings: Mutex::new(Default::default()),
            state: Mutex::new(None),
        })
    }
}

This should all be rather straightforward: we define a Settings struct that stores the two values, implement the Default trait for it, then define a two-element array with property metadata (names, description, ranges, default value, writability), and then store the default value of our Settings struct inside another Mutex inside the element struct.

In the next step we have to make use of these: we need to tell the GObject type system about the properties, and we need to implement functions that are called whenever a property value is set or get.

impl Rgb2Gray {
    fn class_init(klass: &mut BaseTransformClass) {
        [...]
        klass.install_properties(&PROPERTIES);
        [...]
    }
}

impl ObjectImpl<BaseTransform> for Rgb2Gray {
    fn set_property(&self, obj: &glib::Object, id: u32, value: &glib::Value) {
        let prop = &PROPERTIES[id as usize];
        let element = obj.clone().downcast::<BaseTransform>().unwrap();

        match *prop {
            Property::Boolean("invert", ..) => {
                let mut settings = self.settings.lock().unwrap();
                let invert = value.get().unwrap();
                gst_info!(
                    self.cat,
                    obj: &element,
                    "Changing invert from {} to {}",
                    settings.invert,
                    invert
                );
                settings.invert = invert;
            }
            Property::UInt("shift", ..) => {
                let mut settings = self.settings.lock().unwrap();
                let shift = value.get().unwrap();
                gst_info!(
                    self.cat,
                    obj: &element,
                    "Changing shift from {} to {}",
                    settings.shift,
                    shift
                );
                settings.shift = shift;
            }
            _ => unimplemented!(),
        }
    }

    fn get_property(&self, _obj: &glib::Object, id: u32) -> Result<glib::Value, ()> {
        let prop = &PROPERTIES[id as usize];

        match *prop {
            Property::Boolean("invert", ..) => {
                let settings = self.settings.lock().unwrap();
                Ok(settings.invert.to_value())
            }
            Property::UInt("shift", ..) => {
                let settings = self.settings.lock().unwrap();
                Ok(settings.shift.to_value())
            }
            _ => unimplemented!(),
        }
    }
}

Property values can be changed from any thread at any time, that’s why the Mutex is needed here to protect our struct. And we’re using a new mutex to be able to have it locked only for the shorted possible amount of time: we don’t want to keep it locked for the whole time of the transform function, otherwise applications trying to set/get values would block for up to one frame.

In the property setter/getter functions we are working with a glib::Value. This is a dynamically typed value type that can contain values of any type, together with the type information of the contained value. Here we’re using it to handle an unsigned integer (u32) and a boolean for our two properties. To know which property is currently set/get, we get an identifier passed which is the index into our PROPERTIES array. We then simply match on the name of that to decide which property was meant

With this implemented, we can already compile everything, see the properties and their metadata in gst-inspect-1.0 and can also set them on gst-launch-1.0 like this

# Set invert to true and shift to 128
gst-launch-1.0 videotestsrc ! rsrgb2gray invert=true shift=128 ! videoconvert ! autovideosink

If we set GST_DEBUG=rsrgb2gray:6 in the environment before running that, we can also see the corresponding debug output when the values are changing. The only thing missing now is to actually make use of the property values for the processing. For this we add the following changes to bgrx_to_gray and the transform function

impl Rgb2Gray {
    #[inline]
    fn bgrx_to_gray(in_p: &[u8], shift: u8, invert: bool) -> u8 {
        [...]

        let gray = ((r * R_Y) + (g * G_Y) + (b * B_Y)) / 65536;
        let gray = (gray as u8).wrapping_add(shift);

        if invert {
            255 - gray
        } else {
            gray
        }
    }
}

impl BaseTransformImpl<BaseTransform> for Rgb2Gray {
    fn transform(
        &self,
        element: &BaseTransform,
        inbuf: &gst::Buffer,
        outbuf: &mut gst::BufferRef,
    ) -> gst::FlowReturn {
        let settings = *self.settings.lock().unwrap();
        [...]
                    let gray = Rgb2Gray::bgrx_to_gray(in_p, settings.shift as u8, settings.invert);
        [...]
    }
}

And that’s all. If you run the element in gst-launch-1.0 and change the values of the properties you should also see the corresponding changes in the video output.

Note that we always take a copy of the Settings struct at the beginning of the transform function. This ensures that we take the mutex only the shorted possible amount of time and then have a local snapshot of the settings for each frame.

Also keep in mind that the usage of the property values in the bgrx_to_gray function is far from optimal. It means the addition of another condition to the calculation of each pixel, thus potentially slowing it down a lot. Ideally this condition would be moved outside the inner loops and the bgrx_to_gray function would made generic over that. See for example this blog post about “branchless Rust” for ideas how to do that, the actual implementation is left as an exercise for the reader.

What next?

I hope the code walkthrough above was useful to understand how to implement GStreamer plugins and elements in Rust. If you have any questions, feel free to ask them here in the comments.

The same approach also works for audio filters or anything that can be handled in some way with the API of the BaseTransform base class. You can find another filter, an audio echo filter, using the same approach here.

In the next blog post in this series I’ll show how to use another base class to implement another kind of element, but for the time being you can also check the GIT repository for various other element implementations.