diff --git a/examples/src/bin/appsink.rs b/examples/src/bin/appsink.rs index 0f734c537..132690474 100644 --- a/examples/src/bin/appsink.rs +++ b/examples/src/bin/appsink.rs @@ -1,3 +1,15 @@ +// This example demonstrates the use of the appsink element. +// It operates the following pipeline: + +// {audiotestsrc} - {appsink} + +// The application specifies what format it wants to handle. This format +// is applied by calling set_caps on the appsink. Now it's the audiotestsrc's +// task to provide this data format. If the element connected to the appsink's +// sink-pad were not able to provide what we ask them to, this would fail. +// This is the format we request: +// Audio / Signed 16bit / 1 channel / arbitrary sample rate + #[macro_use] extern crate gstreamer as gst; use gst::prelude::*; @@ -54,6 +66,10 @@ fn create_pipeline() -> Result { .dynamic_cast::() .expect("Sink element is expected to be an appsink!"); + // Tell the appsink what format we want. It will then be the audiotestsrc's job to + // provide the format we request. + // This can be set after linking the two objects, because format negotiation between + // both elements will happen during pre-rolling of the pipeline. appsink.set_caps(&gst::Caps::new_simple( "audio/x-raw", &[ @@ -64,9 +80,13 @@ fn create_pipeline() -> Result { ], )); + // Getting data out of the appsink is done by setting callbacks on it. + // The appsink will then call those handlers, as soon as data is available. appsink.set_callbacks( gst_app::AppSinkCallbacks::new() + // Add a handler to the "new-sample" signal. .new_sample(|appsink| { + // Pull the sample in question out of the appsink's buffer. let sample = match appsink.pull_sample() { None => return gst::FlowReturn::Eos, Some(sample) => sample, @@ -84,6 +104,13 @@ fn create_pipeline() -> Result { return gst::FlowReturn::Error; }; + // At this point, buffer is only a reference to an existing memory region somewhere. + // When we want to access its content, we have to map it while requesting the required + // mode of access (read, read/write). + // This type of abstraction is necessary, because the buffer in question might not be + // on the machine's main memory itself, but rather in the GPU's memory. + // So mapping the buffer makes the underlying memory region accessible to us. + // See: https://gstreamer.freedesktop.org/documentation/plugin-development/advanced/allocation.html let map = if let Some(map) = buffer.map_readable() { map } else { @@ -96,6 +123,9 @@ fn create_pipeline() -> Result { return gst::FlowReturn::Error; }; + // We know what format the data in the memory region has, since we requested + // it by setting the appsink's caps. So what we do here is interpret the + // memory region we mapped as an array of signed 16 bit integers. let samples = if let Ok(samples) = map.as_slice_of::() { samples } else { @@ -108,6 +138,8 @@ fn create_pipeline() -> Result { return gst::FlowReturn::Error; }; + // For buffer (= chunk of samples), we calculate the root mean square: + // (https://en.wikipedia.org/wiki/Root_mean_square) let sum: f64 = samples .iter() .map(|sample| { diff --git a/examples/src/bin/appsrc.rs b/examples/src/bin/appsrc.rs index 3c459a0ff..7ab65d4e5 100644 --- a/examples/src/bin/appsrc.rs +++ b/examples/src/bin/appsrc.rs @@ -1,3 +1,15 @@ +// This example shows how to use the appsrc element. +// It operates the following pipeline: + +// {appsrc} - {videoconvert} - {autovideosink} + +// The application itself provides the video-data for the pipeline, by providing +// it in the callback of the appsrc element. Videoconvert makes sure that the +// format the application provides can be displayed by the autovideosink +// at the end of the pipeline. +// The application provides data of the following format: +// Video / BGRx (4 bytes) / 2 fps + extern crate gstreamer as gst; use gst::prelude::*; extern crate gstreamer_app as gst_app; @@ -53,12 +65,15 @@ fn create_pipeline() -> Result { .dynamic_cast::() .expect("Source element is expected to be an appsrc!"); - let info = gst_video::VideoInfo::new(gst_video::VideoFormat::Bgrx, WIDTH as u32, HEIGHT as u32) - .fps(gst::Fraction::new(2, 1)) - .build() - .expect("Failed to create video info"); + // Specify the format we want to provide as application into the pipeline + // by creating a video info with the given format and creating caps from it for the appsrc element. + let video_info = + gst_video::VideoInfo::new(gst_video::VideoFormat::Bgrx, WIDTH as u32, HEIGHT as u32) + .fps(gst::Fraction::new(2, 1)) + .build() + .expect("Failed to create video info"); - appsrc.set_caps(&info.to_caps().unwrap()); + appsrc.set_caps(&video_info.to_caps().unwrap()); appsrc.set_property_format(gst::Format::Time); // Our frame counter, that is stored in the mutable environment @@ -70,8 +85,15 @@ fn create_pipeline() -> Result { // need-data callback. let mut i = 0; appsrc.set_callbacks( + // Since our appsrc element operates in pull mode (it asks us to provide data), + // we add a handler for the need-data callback and provide new data from there. + // In our case, we told gstreamer that we do 2 frames per second. While the + // buffers of all elements of the pipeline are still empty, this will be called + // a couple of times until all of them are filled. After this initial period, + // this handler will be called (on average) twice per second. gst_app::AppSrcCallbacks::new() .need_data(move |appsrc, _| { + // We only produce 100 frames if i == 100 { let _ = appsrc.end_of_stream(); return; @@ -83,11 +105,19 @@ fn create_pipeline() -> Result { let g = if i % 3 == 0 { 0 } else { 255 }; let b = if i % 5 == 0 { 0 } else { 255 }; - let mut buffer = gst::Buffer::with_size(WIDTH * HEIGHT * 4).unwrap(); + // Create the buffer that can hold exactly one BGRx frame. + let mut buffer = gst::Buffer::with_size(video_info.size()).unwrap(); { let buffer = buffer.get_mut().unwrap(); + // For each frame we produce, we set the timestamp when it should be displayed + // (pts = presentation time stamp) + // The autovideosink will use this information to display the frame at the right time. buffer.set_pts(i * 500 * gst::MSECOND); + // At this point, buffer is only a reference to an existing memory region somewhere. + // When we want to access its content, we have to map it while requesting the required + // mode of access (read, read/write). + // See: https://gstreamer.freedesktop.org/documentation/plugin-development/advanced/allocation.html let mut data = buffer.map_writable().unwrap(); for p in data.as_mut_slice().chunks_mut(4) { diff --git a/examples/src/bin/decodebin.rs b/examples/src/bin/decodebin.rs index efb707f52..cd8a44a0e 100644 --- a/examples/src/bin/decodebin.rs +++ b/examples/src/bin/decodebin.rs @@ -1,3 +1,34 @@ +// This example demonstrates the use of the decodebin element +// The decodebin element tries to automatically detect the incoming +// format and to autoplug the appropriate demuxers / decoders to handle it. +// and decode it to raw audio, video or subtitles. +// Before the pipeline hasn't been prerolled, the decodebin can't possibly know what +// format it gets as its input. So at first, the pipeline looks like this: + +// {filesrc} - {decodebin} + +// As soon as the decodebin has detected the stream format, it will try to decode every +// contained stream to its raw format. +// The application connects a signal-handler to decodebin's pad-added signal, which tells us +// whenever the decodebin provided us with another contained (raw) stream from the input file. + +// This application supports audio and video streams. Video streams are +// displayed using an autovideosink, and audiostreams are played back using autoaudiosink. +// So for a file that contains one audio and one video stream, +// the pipeline looks like the following: + +// /-[audio]-{audioconvert}-{audioresample}-{autoaudiosink} +// {filesrc}-{decodebin}-| +// \-[video]-{viceoconvert}-{videoscale}-{autovideosink} + +// Both auto-sinks at the end automatically select the best available (actual) sink. Since the +// selection of available actual sinks is platform specific +// (like using pulseaudio for audio output on linux, e.g.), +// we need to add the audioconvert and audioresample elements before handing the stream to the +// autoaudiosink, because we need to make sure, that the stream is always supported by the actual sink. +// Especially Windows APIs tend to be quite picky about samplerate and sample-format. +// The same applies to videostreams. + #[macro_use] extern crate gstreamer as gst; use gst::prelude::*; @@ -66,18 +97,36 @@ fn example_main() -> Result<(), Error> { let decodebin = gst::ElementFactory::make("decodebin", None).ok_or(MissingElement("decodebin"))?; + // Tell the filesrc what file to load src.set_property("location", &uri)?; pipeline.add_many(&[&src, &decodebin])?; gst::Element::link_many(&[&src, &decodebin])?; + // Need to move a new reference into the closure. + // !!ATTENTION!!: + // It might seem appealing to use pipeline.clone() here, because that greatly + // simplifies the code within the callback. What this actually does, however, is creating + // a memory leak. The clone of a pipeline is a new strong reference on the pipeline. + // Storing this strong reference of the pipeline within the callback (we are moving it in!), + // which is in turn stored in another strong reference on the pipeline is creating a + // reference cycle. + // DO NOT USE pipeline.clone() TO USE THE PIPELINE WITHIN A CALLBACK let pipeline_weak = pipeline.downgrade(); + // Connect to decodebin's pad-added signal, that is emitted whenever + // it found another stream from the input file and found a way to decode it to its raw format. + // decodebin automatically adds a src-pad for this raw stream, which + // we can use to build the follow-up pipeline. decodebin.connect_pad_added(move |dbin, src_pad| { + // Here we temporarily retrieve a strong reference on the pipeline from the weak one + // we moved into this callback. let pipeline = match pipeline_weak.upgrade() { Some(pipeline) => pipeline, None => return, }; + // Try to detect whether the raw stream decodebin provided us with + // just now is either audio or video (or none of both, e.g. subtitles). let (is_audio, is_video) = { let media_type = src_pad.get_current_caps().and_then(|caps| { caps.get_structure(0).map(|s| { @@ -100,8 +149,14 @@ fn example_main() -> Result<(), Error> { } }; + // We create a closure here, calling it directly below it, because this greatly + // improves readability for error-handling. Like this, we can simply use the + // ?-operator within the closure, and handle the actual error down below where + // we call the insert_sink(..) closure. let insert_sink = |is_audio, is_video| -> Result<(), Error> { if is_audio { + // decodebin found a raw audiostream, so we build the follow-up pipeline to + // play it on the default audio playback device (using autoaudiosink). let queue = gst::ElementFactory::make("queue", None).ok_or(MissingElement("queue"))?; let convert = gst::ElementFactory::make("audioconvert", None) @@ -115,13 +170,21 @@ fn example_main() -> Result<(), Error> { pipeline.add_many(elements)?; gst::Element::link_many(elements)?; + // !!ATTENTION!!: + // This is quite important and people forget it often. Without making sure that + // the new elements have the same state as the pipeline, things will fail later. + // They would still be in Null state and can't process data. for e in elements { e.sync_state_with_parent()?; } + // Get the queue element's sink pad and link the decodebin's newly created + // src pad for the audio stream to it. let sink_pad = queue.get_static_pad("sink").expect("queue has no sinkpad"); src_pad.link(&sink_pad).into_result()?; } else if is_video { + // decodebin found a raw videostream, so we build the follow-up pipeline to + // display it using the autovideosink. let queue = gst::ElementFactory::make("queue", None).ok_or(MissingElement("queue"))?; let convert = gst::ElementFactory::make("videoconvert", None) @@ -139,6 +202,8 @@ fn example_main() -> Result<(), Error> { e.sync_state_with_parent()? } + // Get the queue element's sink pad and link the decodebin's newly created + // src pad for the video stream to it. let sink_pad = queue.get_static_pad("sink").expect("queue has no sinkpad"); src_pad.link(&sink_pad).into_result()?; } @@ -146,7 +211,17 @@ fn example_main() -> Result<(), Error> { Ok(()) }; + // When adding and linking new elements in a callback fails, error information is often sparse. + // GStreamer's built-in debugging can be hard to link back to the exact position within the code + // that failed. Since callbacks are called from random threads within the pipeline, it can get hard + // to get good error information. The macros used in the following can solve that. With the use + // of those, one can send arbitrary rust types (using the pipeline's bus) into the mainloop. + // What we send here is unpacked down below, in the iteration-code over sent bus-messages. + // Because we are using the failure crate for error details here, we even get a backtrace for + // where the error was constructed. (If RUST_BACKTRACE=1 is set) if let Err(err) = insert_sink(is_audio, is_video) { + // The following sends a message of type Error on the bus, containing our detailed + // error information. #[cfg(feature = "v1_10")] gst_element_error!( dbin, @@ -174,6 +249,10 @@ fn example_main() -> Result<(), Error> { .get_bus() .expect("Pipeline without bus. Shouldn't happen!"); + // This code iterates over all messages that are sent across our pipeline's bus. + // In the callback ("pad-added" on the decodebin), we sent better error information + // using a bus message. This is the position where we get those messages and log + // the contained information. while let Some(msg) = bus.timed_pop(gst::CLOCK_TIME_NONE) { use gst::MessageView; @@ -185,6 +264,11 @@ fn example_main() -> Result<(), Error> { #[cfg(feature = "v1_10")] { match err.get_details() { + // This bus-message of type error contained our custom error-details struct + // that we sent in the pad-added callback above. So we unpack it and log + // the detailed error information here. details contains a glib::SendValue. + // The unpacked error is the converted to a Result::Err, stopping the + // application's execution. Some(details) if details.get_name() == "error-details" => details .get::<&ErrorValue>("error") .and_then(|v| v.0.lock().unwrap().take()) diff --git a/examples/src/bin/discoverer.rs b/examples/src/bin/discoverer.rs index e46e4163a..ab4cef969 100644 --- a/examples/src/bin/discoverer.rs +++ b/examples/src/bin/discoverer.rs @@ -1,3 +1,13 @@ +// This example uses gstreamer's discoverer api +// https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/GstDiscoverer.html +// To detect as much information from a given URI. +// The amount of time that the discoverer is allowed to use is limited by a timeout. +// This allows to handle e.g. network problems gracefully. When the timeout hits before +// discoverer was able to detect anything, discoverer will report an error. +// In this example, we catch this error and stop the application. +// Discovered information could for example contain the stream's duration or whether it is +// seekable (filesystem) or not (some http servers). + extern crate gstreamer as gst; extern crate gstreamer_pbutils as pbutils; diff --git a/examples/src/bin/encodebin.rs b/examples/src/bin/encodebin.rs index 213457394..8e5fb417d 100644 --- a/examples/src/bin/encodebin.rs +++ b/examples/src/bin/encodebin.rs @@ -1,3 +1,17 @@ +// This example demonstrates the use of the encodebin element. +// The example takes an arbitrary URI as input, which it will try to decode +// and finally reencode using the encodebin element. +// For more information about how the decodebin element works, have a look at +// the decodebin-example. +// Since we tell the encodebin what format we want to get out of it from the start, +// it provides the correct caps and we can link it before starting the pipeline. +// After the decodebin has found all streams and we piped them into the encodebin, +// the operated pipeline looks as follows: + +// /-{queue}-{audioconvert}-{audioresample}-\ +// {uridecodebin} -| {encodebin}-{filesink} +// \-{queue}-{videoconvert}-{videoscale}----/ + #[macro_use] extern crate gstreamer as gst; use gst::prelude::*; @@ -54,16 +68,24 @@ impl glib::subclass::boxed::BoxedType for ErrorValue { glib_boxed_derive_traits!(ErrorValue); fn configure_encodebin(encodebin: &gst::Element) -> Result<(), Error> { + // To tell the encodebin what we want it to produce, we create an EncodingProfile + // https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/GstEncodingProfile.html + // This profile consists of information about the contained audio and video formats + // as well as the container format we want everything to be combined into. + + // Every audiostream piped into the encodebin should be encoded using vorbis. let audio_profile = gst_pbutils::EncodingAudioProfileBuilder::new() .format(&gst::Caps::new_simple("audio/x-vorbis", &[])) .presence(0) .build()?; + // Every videostream piped into the encodebin should be encoded using theora. let video_profile = gst_pbutils::EncodingVideoProfileBuilder::new() .format(&gst::Caps::new_simple("video/x-theora", &[])) .presence(0) .build()?; + // All streams are then finally combined into a matroska container. let container_profile = gst_pbutils::EncodingContainerProfileBuilder::new() .name("container") .format(&gst::Caps::new_simple("video/x-matroska", &[])) @@ -71,6 +93,7 @@ fn configure_encodebin(encodebin: &gst::Element) -> Result<(), Error> { .add_profile(&(audio_profile)) .build()?; + // Finally, apply the EncodingProfile onto our encodebin element. encodebin .set_property("profile", &container_profile) .expect("set profile property failed"); @@ -105,17 +128,38 @@ fn example_main() -> Result<(), Error> { sink.set_property("location", &output_file) .expect("setting location property failed"); + // Configure the encodebin. + // Here we tell the bin what format we expect it to create at its output. configure_encodebin(&encodebin)?; pipeline .add_many(&[&src, &encodebin, &sink]) .expect("failed to add elements to pipeline"); + // It is clear from the start, that encodebin has only one src pad, so we can + // directly link it to our filesink without problems. + // The caps of encodebin's src-pad are set after we configured the encoding-profile. + // (But filesink doesn't really care about the caps at its input anyway) gst::Element::link_many(&[&encodebin, &sink])?; - // Need to move a new reference into the closure - let pipeline_clone = pipeline.clone(); + // Need to move a new reference into the closure. + // !!ATTENTION!!: + // It might seem appealing to use pipeline.clone() here, because that greatly + // simplifies the code within the callback. What this actually does, however, is creating + // a memory leak. The clone of a pipeline is a new strong reference on the pipeline. + // Storing this strong reference of the pipeline within the callback (we are moving it in!), + // which is in turn stored in another strong reference on the pipeline is creating a + // reference cycle. + // DO NOT USE pipeline.clone() TO USE THE PIPELINE WITHIN A CALLBACK + let pipeline_weak = pipeline.downgrade(); + // Much of the following is the same code as in the decodebin example + // so if you want more information on that front, have a look there. src.connect_pad_added(move |dbin, dbin_src_pad| { - let pipeline = &pipeline_clone; + // Here we temporarily retrieve a strong reference on the pipeline from the weak one + // we moved into this callback. + let pipeline = match pipeline_weak.upgrade() { + Some(pipeline) => pipeline, + None => return, + }; let (is_audio, is_video) = { let media_type = dbin_src_pad.get_current_caps().and_then(|caps| { @@ -157,6 +201,9 @@ fn example_main() -> Result<(), Error> { .expect("failed to add audio elements to pipeline"); gst::Element::link_many(elements)?; + // Request a sink pad from our encodebin, that can handle a raw audiostream. + // The encodebin will then automatically create an internal pipeline, that encodes + // the audio stream in the format we specified in the EncodingProfile. let enc_sink_pad = encodebin .get_request_pad("audio_%u") .expect("Could not get audio pad from encodebin"); @@ -169,6 +216,8 @@ fn example_main() -> Result<(), Error> { e.sync_state_with_parent()?; } + // Get the queue element's sink pad and link the decodebin's newly created + // src pad for the audio stream to it. let sink_pad = queue.get_static_pad("sink").expect("queue has no sinkpad"); dbin_src_pad.link(&sink_pad).into_result()?; } else if is_video { @@ -185,6 +234,9 @@ fn example_main() -> Result<(), Error> { .expect("failed to add video elements to pipeline"); gst::Element::link_many(elements)?; + // Request a sink pad from our encodebin, that can handle a raw videostream. + // The encodebin will then automatically create an internal pipeline, that encodes + // the audio stream in the format we specified in the EncodingProfile. let enc_sink_pad = encodebin .get_request_pad("video_%u") .expect("Could not get video pad from encodebin"); @@ -197,6 +249,8 @@ fn example_main() -> Result<(), Error> { e.sync_state_with_parent()? } + // Get the queue element's sink pad and link the decodebin's newly created + // src pad for the video stream to it. let sink_pad = queue.get_static_pad("sink").expect("queue has no sinkpad"); dbin_src_pad.link(&sink_pad).into_result()?; } diff --git a/examples/src/bin/events.rs b/examples/src/bin/events.rs index ad364f9a8..e1cc57a1f 100644 --- a/examples/src/bin/events.rs +++ b/examples/src/bin/events.rs @@ -1,3 +1,24 @@ +// This example demonstrates how events can be created and sent to the pipeline. +// What this example does is scheduling a timeout on the main loop, and +// sending an EOS message on the bus from there - telling the pipeline +// to shut down. Once that event is processed by everything, the EOS message +// is going to be sent and we catch that one to shut down everything. + +// GStreamer's bus is an abstraction layer above an arbitrary main loop. +// This makes sure that GStreamer can be used in conjunction with any existing +// other framework (GUI frameworks, mostly) that operate their own main loops. +// Main idea behind the bus is the simplification between the application and +// GStreamer, because GStreamer is heavily threaded underneath. + +// Any thread can post messages to the bus, which is essentially a thread-safe +// queue of messages to process. When a new message was sent to the bus, it +// will wake up the main loop implementation underneath it (which will then +// process the pending messages from the main loop thread). + +// An application itself can post messages to the bus aswell. +// This makes it possible, e.g., to schedule an arbitrary piece of code +// to run in the main loop thread - avoiding potential threading issues. + extern crate gstreamer as gst; use gst::prelude::*; @@ -11,14 +32,31 @@ fn example_main() { let main_loop = glib::MainLoop::new(None, false); + // This creates a pipeline by parsing the gst-launch pipeline syntax. let pipeline = gst::parse_launch("audiotestsrc ! fakesink").unwrap(); let bus = pipeline.get_bus().unwrap(); let ret = pipeline.set_state(gst::State::Playing); assert_ne!(ret, gst::StateChangeReturn::Failure); + // Need to move a new reference into the closure. + // !!ATTENTION!!: + // It might seem appealing to use pipeline.clone() here, because that greatly + // simplifies the code within the callback. What this actually does, however, is creating + // a memory leak. The clone of a pipeline is a new strong reference on the pipeline. + // Storing this strong reference of the pipeline within the callback (we are moving it in!), + // which is in turn stored in another strong reference on the pipeline is creating a + // reference cycle. + // DO NOT USE pipeline.clone() TO USE THE PIPELINE WITHIN A CALLBACK let pipeline_weak = pipeline.downgrade(); + // Add a timeout to the main loop. This closure will be executed + // in an interval of 5 seconds. The return value of the handler function + // determines whether the handler still wants to be called: + // - glib::Continue(false) - stop calling this handler, remove timeout + // - glib::Continue(true) - continue calling this handler glib::timeout_add_seconds(5, move || { + // Here we temporarily retrieve a strong reference on the pipeline from the weak one + // we moved into this callback. let pipeline = match pipeline_weak.upgrade() { Some(pipeline) => pipeline, None => return glib::Continue(false), @@ -26,15 +64,32 @@ fn example_main() { println!("sending eos"); + // We create an EndOfStream event here, that tells all elements to drain + // their internal buffers to their following elements, essentially draining the + // whole pipeline (front to back). It ensuring that no data is left unhandled and potentially + // headers were rewritten (e.g. when using something like an MP4 or Matroska muxer). + // The EOS event is handled directly from this very thread until the first + // queue element is reached during pipeline-traversal, where it is then queued + // up and later handled from the queue's streaming thread for the elements + // following that queue. + // Once all sinks are done handling the EOS event (and all buffers that were before the + // EOS event in the pipeline already), the pipeline would post an EOS message on the bus, + // essentially telling the application that the pipeline is completely drained. let ev = gst::Event::new_eos().build(); pipeline.send_event(ev); + // Remove this handler, the pipeline will shutdown anyway, now that we + // sent the EOS event. glib::Continue(false) }); //bus.add_signal_watch(); //bus.connect_message(move |_, msg| { let main_loop_clone = main_loop.clone(); + // This sets the bus's signal handler (don't be mislead by the "add", there can only be one). + // Every message from the bus is passed through this function. Its returnvalue determines + // whether the handler wants to be called again. If glib::Continue(false) is returned, the + // handler is removed and will never be called again. The mainloop still runs though. bus.add_watch(move |_, msg| { use gst::MessageView; @@ -42,6 +97,8 @@ fn example_main() { match msg.view() { MessageView::Eos(..) => { println!("received eos"); + // An EndOfStream event was sent to the pipeline, so we tell our main loop + // to stop execution here. main_loop.quit() } MessageView::Error(err) => { @@ -56,14 +113,21 @@ fn example_main() { _ => (), }; + // Tell the mainloop to continue executing this callback. glib::Continue(true) }); + // Operate GStreamer's bus, facilliating GLib's mainloop here. + // This function call will block until you tell the mainloop to quit + // (see above for how to do this). main_loop.run(); let ret = pipeline.set_state(gst::State::Null); assert_ne!(ret, gst::StateChangeReturn::Failure); + // Remove the watch function from the bus. + // Again: There can always only be one watch function. + // Thus we don't have to tell him which function to remove. bus.remove_watch(); } diff --git a/examples/src/bin/futures.rs b/examples/src/bin/futures.rs index ffc59bfdb..79249c7f6 100644 --- a/examples/src/bin/futures.rs +++ b/examples/src/bin/futures.rs @@ -1,3 +1,8 @@ +// This example demonstrates how to use the gstreamer crate in conjunction +// with the future trait. The example waits for either an error to occur, +// or for an EOS message. When a message notifying about either of both +// is received, the future is resolved. + extern crate gstreamer as gst; use gst::prelude::*; @@ -11,20 +16,28 @@ use std::env; mod examples_common; fn example_main() { + // Read the pipeline to launch from the commandline, using the launch syntax. let pipeline_str = env::args().collect::>()[1..].join(" "); gst::init().unwrap(); + // Create a pipeline from the launch-syntax given on the cli. let pipeline = gst::parse_launch(&pipeline_str).unwrap(); let bus = pipeline.get_bus().unwrap(); let ret = pipeline.set_state(gst::State::Playing); assert_ne!(ret, gst::StateChangeReturn::Failure); + // BusStream implements the Stream trait, but Stream::for_each is + // calling a closure for each item and returns a Future that resolves + // when the stream is done or an error has happened let messages = gst::BusStream::new(&bus) .for_each(|msg| { use gst::MessageView; + // Determine whether we want to resolve the future, or we still have + // to wait. The future is resolved when either an error occurs, or the + // pipeline succeeded execution (got an EOS event). let quit = match msg.view() { MessageView::Eos(..) => true, MessageView::Error(err) => { @@ -40,13 +53,16 @@ fn example_main() { }; if quit { - Err(()) + Err(()) // This resolves the future that is returned by for_each + // FIXME: At the moment, EOS messages also result in the future to be resolved + // by an error. This should probably be changed in the future. } else { - Ok(()) + Ok(()) // Continue - do not resolve the future yet. } }) .and_then(|_| Ok(())); + // Synchronously wait on the future we created above. let _ = block_on(messages); let ret = pipeline.set_state(gst::State::Null); diff --git a/examples/src/bin/ges.rs b/examples/src/bin/ges.rs index a98722453..21304d392 100644 --- a/examples/src/bin/ges.rs +++ b/examples/src/bin/ges.rs @@ -1,3 +1,42 @@ +// HELP: New to GES. Is everything here correct? + +// This example demonstrates how to use the gstreamer editing services. +// This is gstreamer's framework to implement non-linear editing. +// It provides a timeline API that internally manages a dynamically changing +// pipeline. (e.g.: alternating video streams in second 1, 2, and 3) +// Timeline: +// _________________________________________________ +// | 00:01 | 00:02 | 00:03 | +// ================================================= +// Layer0: || ###CLIP####|| || ###CLIP###|| +// || ####00#####|| || ####01####|| +// ================================================= +// Layer1: || ###CLIP#### || || +// || ####00##### || || +// ================================================= + +// - Assets are the base of most components in GES. One asset essentially represents +// one resource (e.g. a file). Different files and filetypes can contain different +// types of things. Thus - you can extract different high-level types from an +// asset. If you created an asset from a video file, you could for example "extract" +// a GESClip from it. Same goes for audio files. +// - There even is the GESProject subclass of GESAsset, which can be used to load a whole +// previously saved project. And since GESProject essentially is a GESAsset itself, you +// can then extract the stored components (like the timeline e.g.) from it. +// - Clips are the high-level types (above assets), managing multimedia elements (such as +// videos or audio clips). Within the timeline, they are arranged in layers. +// Those layers essentially behave like in common photo editing software: They specify +// the order in which they are composited, and can therefore overlay each other. +// Clips are essentially wrappers around the underlying GStreamer elements needed +// to work with them. They also provide high-level APIs to add effects into the +// clip's internal pipeline. +// Multiple clips can also be grouped together (even across layers!) to one, making it +// possible to work with all of them as if they were one. +// - Like noted above, Layers specify the order in which the different layers are composited. +// This is specified by their priority. Layers with higher priority (lower number) trump +// those with lowers (higher number). Thus, Layers with higher priority are "in the front". +// - The timeline is the enclosing element, grouping all layers and providing a timeframe. + extern crate gstreamer as gst; use gst::prelude::*; @@ -19,14 +58,18 @@ extern crate glib; fn main_loop(uri: &str) -> Result<(), glib::BoolError> { ges::init()?; + // Begin by creating a timeline with audio and video tracks let timeline = ges::Timeline::new_audio_video(); + // Create a new layer that will contain our timed clips. let layer = timeline.append_layer(); let pipeline = ges::Pipeline::new(); pipeline.set_timeline(&timeline); + // Load a clip from the given uri and add it to the layer. let clip = ges::UriClip::new(uri); layer.add_clip(&clip); + // Add an effect to the clip's video stream. let effect = ges::Effect::new("agingtv"); clip.add(&effect).unwrap(); @@ -38,6 +81,8 @@ fn main_loop(uri: &str) -> Result<(), glib::BoolError> { .unwrap() ); + // Retrieve the asset that was automatically used behind the scenes, to + // extract the clip from. let asset = clip.get_asset().unwrap(); let duration = asset .downcast::() @@ -50,6 +95,10 @@ fn main_loop(uri: &str) -> Result<(), glib::BoolError> { duration / 4 ); + // The inpoint specifies where in the clip we start, the duration specifies + // how much we play from that point onwards. Setting the inpoint to something else + // than 0, or the duration something smaller than the clip's actual duration will + // cut the clip. clip.set_inpoint(duration / 2); clip.set_duration(duration / 4); diff --git a/examples/src/bin/gtksink.rs b/examples/src/bin/gtksink.rs index 6997de007..6f94029d5 100644 --- a/examples/src/bin/gtksink.rs +++ b/examples/src/bin/gtksink.rs @@ -1,3 +1,15 @@ +// This example demonstrates how to use gstreamer in conjunction with the gtk widget toolkit. +// This example shows the video produced by a videotestsrc within a small gtk gui. +// For this, the gtkglsink is used, which creates a gtk widget one can embed the gtk gui. +// For this, there multiple types of widgets. gtkglsink uses OpenGL to render frames, and +// gtksink uses the CPU to render the frames (which is way slower). +// So the example application first tries to use OpenGL, and when that fails, fall back. +// The pipeline looks like the following: + +// gtk-gui: {gtkglsink}-widget +// (|) +// {videotestsrc} - {glsinkbin} + extern crate gstreamer as gst; use gst::prelude::*; @@ -15,16 +27,32 @@ use std::env; fn create_ui(app: >k::Application) { let pipeline = gst::Pipeline::new(None); let src = gst::ElementFactory::make("videotestsrc", None).unwrap(); + // Create the gtk sink and retrieve the widget from it. The sink element will be used + // in the pipeline, and the widget will be embedded in our gui. + // Gstreamer then displays frames in the gtk widget. + // First, we try to use the OpenGL version - and if that fails, we fall back to non-OpenGL. let (sink, widget) = if let Some(gtkglsink) = gst::ElementFactory::make("gtkglsink", None) { + // Using the OpenGL widget succeeded, so we are in for a nice playback experience with + // low cpu usage. :) + // The gtkglsink essentially allocates an OpenGL texture on the GPU, that it will display. + // Now we create the glsinkbin element, which is responsible for conversions and for uploading + // video frames to our texture (if they are not already in the GPU). Now we tell the OpenGL-sink + // about our gtkglsink element, form where it will retrieve the OpenGL texture to fill. let glsinkbin = gst::ElementFactory::make("glsinkbin", None).unwrap(); glsinkbin .set_property("sink", >kglsink.to_value()) .unwrap(); - + // The gtkglsink creates the gtk widget for us. This is accessible through a property. + // So we get it and use it later to add it to our gui. let widget = gtkglsink.get_property("widget").unwrap(); (glsinkbin, widget.get::().unwrap()) } else { + // Unfortunately, using the OpenGL widget didn't work out, so we will have to render + // our frames manually, using the CPU. An example why this may fail is, when + // the PC doesn't have proper graphics drivers installed. let sink = gst::ElementFactory::make("gtksink", None).unwrap(); + // The gtksink creates the gtk widget for us. This is accessible through a property. + // So we get it and use it later to add it to our gui. let widget = sink.get_property("widget").unwrap(); (sink, widget.get::().unwrap()) }; @@ -32,9 +60,11 @@ fn create_ui(app: >k::Application) { pipeline.add_many(&[&src, &sink]).unwrap(); src.link(&sink).unwrap(); + // Create a simple gtk gui window to place our widget into. let window = gtk::Window::new(gtk::WindowType::Toplevel); window.set_default_size(320, 240); let vbox = gtk::Box::new(gtk::Orientation::Vertical, 0); + // Add our widget to the gui vbox.pack_start(&widget, true, true, 0); let label = gtk::Label::new("Position: 00:00:00"); vbox.pack_start(&label, true, true, 5); @@ -43,32 +73,39 @@ fn create_ui(app: >k::Application) { app.add_window(&window); + // Need to move a new reference into the closure. + // !!ATTENTION!!: + // It might seem appealing to use pipeline.clone() here, because that greatly + // simplifies the code within the callback. What this actually does, however, is creating + // a memory leak. The clone of a pipeline is a new strong reference on the pipeline. + // Storing this strong reference of the pipeline within the callback (we are moving it in!), + // which is in turn stored in another strong reference on the pipeline is creating a + // reference cycle. + // DO NOT USE pipeline.clone() TO USE THE PIPELINE WITHIN A CALLBACK let pipeline_weak = pipeline.downgrade(); + // Add a timeout to the main loop that will periodically (every 500ms) be + // executed. This will query the current position within the stream from + // the underlying pipeline, and display it in our gui. + // Since this closure is called by the mainloop thread, we are allowed + // to modify the gui widgets here. let timeout_id = gtk::timeout_add(500, move || { + // Here we temporarily retrieve a strong reference on the pipeline from the weak one + // we moved into this callback. let pipeline = match pipeline_weak.upgrade() { Some(pipeline) => pipeline, None => return glib::Continue(true), }; + // Query the current playing position from the underlying pipeline. let position = pipeline .query_position::() .unwrap_or_else(|| 0.into()); + // Display the playing position in the gui. label.set_text(&format!("Position: {:.0}", position)); - + // Tell the callback to continue calling this closure. glib::Continue(true) }); - let app_weak = app.downgrade(); - window.connect_delete_event(move |_, _| { - let app = match app_weak.upgrade() { - Some(app) => app, - None => return Inhibit(false), - }; - - app.quit(); - Inhibit(false) - }); - let bus = pipeline.get_bus().unwrap(); let ret = pipeline.set_state(gst::State::Playing); @@ -115,6 +152,7 @@ fn create_ui(app: >k::Application) { } fn main() { + // Initialize gstreamer and the gtk widget toolkit libraries. gst::init().unwrap(); gtk::init().unwrap(); diff --git a/examples/src/bin/gtkvideooverlay.rs b/examples/src/bin/gtkvideooverlay.rs index 0906a3672..9f866d8ae 100644 --- a/examples/src/bin/gtkvideooverlay.rs +++ b/examples/src/bin/gtkvideooverlay.rs @@ -1,3 +1,22 @@ +// This example demonstrates another type of combination of gtk and gstreamer, +// in comparision to the gtksink example. +// This example uses regions that are managed by the window system, and uses +// the window system's api to insert a videostream into these regions. +// So essentially, the window system of the system overlays our gui with +// the video frames - within the region that we tell it to use. +// Disadvantage of this method is, that it's highly platform specific, since +// the big platforms all have their own window system. Thus, this example +// has special code to handle differences between platforms. +// Windows could theoretically be supported by this example, but is not yet implemented. +// One of the very few (if not the single one) platform, that can not provide the API +// needed for this are Linux desktops using Wayland. +// TODO: Add Windows support +// In this case, a testvideo is displayed within our gui, using the +// following pipeline: + +// {videotestsrc} - {xvimagesink(on linux)} +// {videotestsrc} - {glimagesink(on mac)} + extern crate gstreamer as gst; use gst::prelude::*; @@ -28,9 +47,17 @@ fn create_ui(app: >k::Application) { let pipeline = gst::Pipeline::new(None); let src = gst::ElementFactory::make("videotestsrc", None).unwrap(); + // Since using the window system to overlay our gui window is making + // direct contact with the windowing system, this is highly platform- + // specific. This example supports Linux and Mac (using X11 and Quartz). let sink = if cfg!(feature = "gtkvideooverlay-x11") { + // When we are on linux with the Xorg display server, we use the + // X11 protocol's XV extension, which allows to overlay regions + // with video streams. For this, we use the xvimagesink element. gst::ElementFactory::make("xvimagesink", None).unwrap() } else if cfg!(feature = "gtkvideooverlay-quartz") { + // On Mac, this is done by overlaying a window region with an + // OpenGL-texture, using the glimagesink element. gst::ElementFactory::make("glimagesink", None).unwrap() } else { unreachable!() @@ -39,26 +66,49 @@ fn create_ui(app: >k::Application) { pipeline.add_many(&[&src, &sink]).unwrap(); src.link(&sink).unwrap(); + // First, we create our gtk window - which will contain a region where + // our overlayed video will be displayed in. let window = gtk::Window::new(gtk::WindowType::Toplevel); window.set_default_size(320, 240); let vbox = gtk::Box::new(gtk::Orientation::Vertical, 0); + // This creates the widget we will display our overlay in. + // Later, we will try to tell our window system about this region, so + // it can overlay it with our video stream. let video_window = gtk::DrawingArea::new(); video_window.set_size_request(320, 240); + + // Use the platform-specific sink to create our overlay. + // Since we only use the video_overlay in the closure below, we need a weak reference. + // !!ATTENTION!!: + // It might seem appealing to use .clone() here, because that greatly + // simplifies the code within the callback. What this actually does, however, is creating + // a memory leak. let video_overlay = sink .clone() .dynamic_cast::() .unwrap() .downgrade(); + // Connect to this widget's realize signal, which will be emitted + // after its display has been initialized. This is neccessary, because + // the window system doesn't know about our region until it was initialized. video_window.connect_realize(move |video_window| { + // Here we temporarily retrieve a strong reference on the video-overlay from the + // weak reference that we moved into the closure. let video_overlay = match video_overlay.upgrade() { Some(video_overlay) => video_overlay, None => return, }; + // Gtk uses gdk under the hood, to handle its drawing. Drawing regions are + // called gdk windows. We request this underlying drawing region from the + // widget we will overlay with our video. let gdk_window = video_window.get_window().unwrap(); + // This is where we tell our window system about the drawing-region we + // want it to overlay. Most often, the window system would only know + // about our most outer region (or: our window). if !gdk_window.ensure_native() { println!("Can't create native window for widget"); process::exit(-1); @@ -75,7 +125,14 @@ fn create_ui(app: >k::Application) { ) -> *mut c_void; } + // This is unsafe because the "window handle" we pass here is basically like a raw pointer. + // If a wrong value were to be passed here (and you can pass any integer), then the window + // system will most likely cause the application to crash. unsafe { + // Here we ask gdk what native window handle we got assigned for + // our video region from the window system, and then we will + // pass this unique identifier to the overlay provided by our + // sink - so the sink can then arrange the overlay. let xid = gdk_x11_window_get_xid(gdk_window.to_glib_none().0); video_overlay.set_window_handle(xid as usize); } @@ -91,7 +148,14 @@ fn create_ui(app: >k::Application) { ) -> *mut c_void; } + // This is unsafe because the "window handle" we pass here is basically like a raw pointer. + // If a wrong value were to be passed here (and you can pass any integer), then the window + // system will most likely cause the application to crash. unsafe { + // Here we ask gdk what native window handle we got assigned for + // our video region from the windowing system, and then we will + // pass this unique identifier to the overlay provided by our + // sink - so the sink can then arrange the overlay. let window = gdk_quartz_window_get_nsview(gdk_window.to_glib_none().0); video_overlay.set_window_handle(window as usize); } @@ -112,32 +176,39 @@ fn create_ui(app: >k::Application) { app.add_window(&window); + // Need to move a new reference into the closure. + // !!ATTENTION!!: + // It might seem appealing to use pipeline.clone() here, because that greatly + // simplifies the code within the callback. What this actually does, however, is creating + // a memory leak. The clone of a pipeline is a new strong reference on the pipeline. + // Storing this strong reference of the pipeline within the callback (we are moving it in!), + // which is in turn stored in another strong reference on the pipeline is creating a + // reference cycle. + // DO NOT USE pipeline.clone() TO USE THE PIPELINE WITHIN A CALLBACK let pipeline_weak = pipeline.downgrade(); + // Add a timeout to the main loop that will periodically (every 500ms) be + // executed. This will query the current position within the stream from + // the underlying pipeline, and display it in our gui. + // Since this closure is called by the mainloop thread, we are allowed + // to modify the gui widgets here. let timeout_id = gtk::timeout_add(500, move || { + // Here we temporarily retrieve a strong reference on the pipeline from the weak one + // we moved into this callback. let pipeline = match pipeline_weak.upgrade() { Some(pipeline) => pipeline, - None => return glib::Continue(true), + None => return glib::Continue(false), }; + // Query the current playing position from the underlying pipeline. let position = pipeline .query_position::() .unwrap_or_else(|| 0.into()); + // Display the playing position in the gui. label.set_text(&format!("Position: {:.0}", position)); - + // Tell the timeout to continue calling this callback. glib::Continue(true) }); - let app_weak = app.downgrade(); - window.connect_delete_event(move |_, _| { - let app = match app_weak.upgrade() { - Some(app) => app, - None => return Inhibit(false), - }; - - app.quit(); - Inhibit(false) - }); - let bus = pipeline.get_bus().unwrap(); let ret = pipeline.set_state(gst::State::Playing); @@ -190,6 +261,7 @@ fn main() { process::exit(-1); } + // Initialize gstreamer and the gtk widget toolkit libraries. gst::init().unwrap(); gtk::init().unwrap(); diff --git a/examples/src/bin/iterator.rs b/examples/src/bin/iterator.rs index 64dc347a4..41fa86ee6 100644 --- a/examples/src/bin/iterator.rs +++ b/examples/src/bin/iterator.rs @@ -1,3 +1,7 @@ +// This example demonstrates how to use GStreamer's iteration APIs. +// This is used at multiple occassions - for example to iterate an +// element's pads. + extern crate gstreamer as gst; use gst::prelude::*; @@ -7,15 +11,28 @@ mod examples_common; fn example_main() { gst::init().unwrap(); + // Create and use an identity element here. + // This element does nothing, really. We also never add it to a pipeline. + // We just want to iterate the identity element's pads. let identity = gst::ElementFactory::make("identity", None).unwrap(); + // Get an iterator over all pads of the identity-element. let mut iter = identity.iterate_pads(); loop { + // In an endless-loop, we use the iterator until we either reach the end + // or we hit an error. match iter.next() { Ok(Some(pad)) => println!("Pad: {}", pad.get_name()), Ok(None) => { + // We reached the end of the iterator, there are no more pads println!("Done"); break; } + // It is very important to handle this resync error by calling resync + // on the iterator. This error happens, when the container that is iterated + // changed during iteration. (e.g. a pad was added while we used the + // iterator to iterate over all of an element's pads). + // After calling resync on the iterator, iteration will start from the beginning + // again. So the application should be able to handle that. Err(gst::IteratorError::Resync) => { println!("Iterator resync"); iter.resync(); diff --git a/examples/src/bin/launch.rs b/examples/src/bin/launch.rs index be8248ed5..f7acbd124 100644 --- a/examples/src/bin/launch.rs +++ b/examples/src/bin/launch.rs @@ -1,3 +1,8 @@ +// This is a simplified rust-reimplementation of the gst-launch- +// cli tool. It has no own parameters and simply parses the cli arguments +// as launch syntax. +// When the parsing succeeded, the pipeline is run until the stream ends or an error happens. + extern crate gstreamer as gst; use gst::prelude::*; @@ -8,10 +13,20 @@ use std::process; mod examples_common; fn example_main() { + // Get a string containing the passed pipeline launch syntax let pipeline_str = env::args().collect::>()[1..].join(" "); gst::init().unwrap(); + // Let GStreamer create a pipeline from the parsed launch syntax on the cli. + // In comparision to the launch_glib_main example, this is using the advanced launch syntax + // parsing API of GStreamer. The function returns a Result, handing us the pipeline if + // parsing and creating succeeded, and hands us detailed error information if something + // went wrong. The error is passed as gst::ParseError. In this example, we separately + // handle the NoSuchElement error, that GStreamer uses to notify us about elements + // used within the launch syntax, that are not available (not installed). + // Especially GUIs should probably handle this case, to tell users that they need to + // install the corresponding gstreamer plugins. let mut context = gst::ParseContext::new(); let pipeline = match gst::parse_launch_full(&pipeline_str, Some(&mut context), gst::ParseFlags::NONE) { diff --git a/examples/src/bin/launch_glib_main.rs b/examples/src/bin/launch_glib_main.rs index 02293864d..e8136ac79 100644 --- a/examples/src/bin/launch_glib_main.rs +++ b/examples/src/bin/launch_glib_main.rs @@ -1,3 +1,12 @@ +// This is a simplified rust-reimplementation of the gst-launch- +// cli tool. It has no own parameters and simply parses the cli arguments +// as launch syntax. +// When the parsing succeeded, the pipeline is run until it exits. +// Main difference between this example and the launch example is the use of +// GLib's main loop to operate GStreamer's bus. This allows to also do other +// things from the main loop (timeouts, UI events, socket events, ...) instead +// of just handling messages from GStreamer's bus. + extern crate gstreamer as gst; use gst::prelude::*; @@ -9,12 +18,15 @@ use std::env; mod examples_common; fn example_main() { + // Get a string containing the passed pipeline launch syntax let pipeline_str = env::args().collect::>()[1..].join(" "); gst::init().unwrap(); + // Like teasered above, we use GLib's main loop to operate GStreamer's bus. let main_loop = glib::MainLoop::new(None, false); + // Let GStreamer create a pipeline from the parsed launch syntax on the cli. let pipeline = gst::parse_launch(&pipeline_str).unwrap(); let bus = pipeline.get_bus().unwrap(); @@ -51,6 +63,9 @@ fn example_main() { let ret = pipeline.set_state(gst::State::Null); assert_ne!(ret, gst::StateChangeReturn::Failure); + // Here we remove the bus watch we added above. This avoids a memory leak, that might + // otherwise happen because we moved a strong reference (clone of main_loop) into the + // callback closure above. bus.remove_watch(); } diff --git a/examples/src/bin/pad_probes.rs b/examples/src/bin/pad_probes.rs index 493860280..d690287f1 100644 --- a/examples/src/bin/pad_probes.rs +++ b/examples/src/bin/pad_probes.rs @@ -1,3 +1,12 @@ +// This example demonstrates the use of GStreamer's pad probe APIs. +// Probes are callbacks that can be installed by the application and will notify +// the application about the states of the dataflow. Those are mostly used for +// changing pipelines dynamically at runtime or for inspecting/modifying buffers or events + +// |-[probe] +// / +// {audiotestsrc} - {fakesink} + extern crate gstreamer as gst; use gst::prelude::*; extern crate gstreamer_audio as gst_audio; @@ -13,6 +22,9 @@ mod examples_common; fn example_main() { gst::init().unwrap(); + // Parse the pipeline we want to probe from a static in-line string. + // Here we give our audiotestsrc a name, so we can retrieve that element + // from the resulting pipeline. let pipeline = gst::parse_launch(&format!( "audiotestsrc name=src ! audio/x-raw,format={},channels=1 ! fakesink", gst_audio::AUDIO_FORMAT_S16.to_string() @@ -20,18 +32,35 @@ fn example_main() { .unwrap(); let pipeline = pipeline.dynamic_cast::().unwrap(); + // Get the audiotestsrc element from the pipeline that GStreamer + // created for us while parsing the launch syntax above. let src = pipeline.get_by_name("src").unwrap(); + // Get the audiotestsrc's src-pad. let src_pad = src.get_static_pad("src").unwrap(); + // Add a probe handler on the audiotestsrc's src-pad. + // This handler gets called for every buffer that passes the pad we probe. src_pad.add_probe(gst::PadProbeType::BUFFER, |_, probe_info| { + // Interpret the data sent over the pad as one buffer if let Some(gst::PadProbeData::Buffer(ref buffer)) = probe_info.data { + // At this point, buffer is only a reference to an existing memory region somewhere. + // When we want to access its content, we have to map it while requesting the required + // mode of access (read, read/write). + // This type of abstraction is necessary, because the buffer in question might not be + // on the machine's main memory itself, but rather in the GPU's memory. + // So mapping the buffer makes the underlying memory region accessible to us. + // See: https://gstreamer.freedesktop.org/documentation/plugin-development/advanced/allocation.html let map = buffer.map_readable().unwrap(); + // We know what format the data in the memory region has, since we requested + // it by setting the appsink's caps. So what we do here is interpret the + // memory region we mapped as an array of signed 16 bit integers. let samples = if let Ok(samples) = map.as_slice_of::() { samples } else { return gst::PadProbeReturn::Ok; }; + // For buffer (= chunk of samples), we calculate the root mean square: let sum: f64 = samples .iter() .map(|sample| { diff --git a/examples/src/bin/pango-cairo.rs b/examples/src/bin/pango-cairo.rs index 840c615db..cd648fc22 100644 --- a/examples/src/bin/pango-cairo.rs +++ b/examples/src/bin/pango-cairo.rs @@ -1,3 +1,16 @@ +// This example demonstrates how to overlay a video using the cairo +// library. For this, the cairooverlay element is used on a video stream. +// Additionally, this example uses functionality of the pango library, which handles +// text layouting. The pangocairo crate is a nice wrapper combining both libraries +// into a nice interface. +// The drawing surface which the cairooverlay element creates internally can then +// normally be drawn on using the cairo library. +// The operated pipeline looks like this: + +// {videotestsrc} - {cairooverlay} - {capsfilter} - {videoconvert} - {autovideosink} +// The capsfilter element allows us to dictate the video resolution we want for the +// videotestsrc and the cairooverlay element. + extern crate glib; extern crate gstreamer as gst; @@ -79,28 +92,54 @@ fn create_pipeline() -> Result { pipeline.add_many(&[&src, &overlay, &capsfilter, &videoconvert, &sink])?; gst::Element::link_many(&[&src, &overlay, &capsfilter, &videoconvert, &sink])?; + // Plug in a capsfilter element that will force the videotestsrc and the cairooverlay to work + // with images of the size 800x800. let caps = gst::Caps::builder("video/x-raw") .field("width", &800i32) .field("height", &800i32) .build(); capsfilter.set_property("caps", &caps).unwrap(); + // The videotestsrc supports multiple test patterns. In this example, we will use the + // pattern with a white ball moving around the video's center point. src.set_property_from_str("pattern", "ball"); + // The PangoFontMap represents the set of fonts available for a particular rendering system. let fontmap = pangocairo::FontMap::new().unwrap(); + // Create a new pango layouting context for the fontmap. let context = fontmap.create_context().unwrap(); + // Create a pango layout object. This object is a string of text we want to layout. + // It is wrapped in a LayoutWrapper (defined above) to be able to send it across threads. let layout = LayoutWrapper(pango::Layout::new(&context)); + // Select the text content and the font we want to use for the piece of text. let font_desc = pango::FontDescription::from_string("Sans Bold 26"); layout.set_font_description(&font_desc); layout.set_text("GStreamer"); + // The following is a context struct (containing the pango layout and the configured video info). + // We have to wrap it in an Arc (or Rc) to get reference counting, that is: to be able to have + // shared ownership of it in multiple different places (the two signal handlers here). + // We have to wrap it in a Mutex because Rust's type-system can't know that both signals are + // only ever called from a single thread (the streaming thread). It would be enough to have + // something that is Send in theory but that's not how signal handlers are generated unfortunately. + // The Mutex (or otherwise if we didn't need the Sync bound we could use a RefCell) is to implement + // interior mutability (see Rust docs). Via this we can get a mutable reference to the contained + // data which is checked at runtime for uniqueness (blocking in case of mutex, panic in case + // of refcell) instead of compile-time (like with normal references). let drawer = Arc::new(Mutex::new(DrawingContext { layout: glib::SendUniqueCell::new(layout).unwrap(), info: None, })); let drawer_clone = drawer.clone(); + // Connect to the cairooverlay element's "draw" signal, which is emitted for + // each videoframe piped through the element. Here we have the possibility to + // draw on top of the frame (overlay it), using the cairo render api. + // Signals connected with the connect(, ...) API get their arguments + // passed as array of glib::Value. For a documentation about the actual arguments + // it is always a good idea to either check the element's signals using either + // gst-inspect, or the online documentation. overlay .connect("draw", false, move |args| { use std::f64::consts::PI; @@ -108,7 +147,10 @@ fn create_pipeline() -> Result { let drawer = &drawer_clone; let drawer = drawer.lock().unwrap(); + // Get the signal's arguments let _overlay = args[0].get::().unwrap(); + // This is the cairo context. This is the root of all of cairo's + // drawing functionality. let cr = args[1].get::().unwrap(); let timestamp = args[2].get::().unwrap(); let _duration = args[3].get::().unwrap(); @@ -121,10 +163,22 @@ fn create_pipeline() -> Result { * ((timestamp % (10 * gst::SECOND)).unwrap() as f64 / (10.0 * gst::SECOND_VAL as f64)); + // The image we draw (the text) will be static, but we will change the + // transformation on the drawing context, which rotates and shifts everything + // that we draw afterwards. Like this, we have no complicated calulations + // in the actual drawing below. + // Calling multiple transformation methods after each other will apply the + // new transformation on top. If you repeat the cr.rotate(angle) line below + // this a second time, everything in the canvas will rotate twice as fast. cr.translate(info.width() as f64 / 2.0, info.height() as f64 / 2.0); cr.rotate(angle); + // This loop will render 10 times the string "GStreamer" in a circle for i in 0..10 { + // Cairo, like most rendering frameworks, is using a stack for transformations + // with this, we push our current transformation onto this stack - allowing us + // to make temporary changes / render something / and then returning to the + // previous transformations. cr.save(); let angle = (360. * i as f64) / 10.0; @@ -132,14 +186,23 @@ fn create_pipeline() -> Result { cr.set_source_rgb(red, 0.0, 1.0 - red); cr.rotate(angle * PI / 180.0); + // Update the text layout. This function is only updating pango's internal state. + // So e.g. that after a 90 degree rotation it knows that what was previously going + // to end up as a 200x100 rectangle would now be 100x200. pangocairo::functions::update_layout(&cr, &layout); let (width, _height) = layout.get_size(); + // Using width and height of the text, we can properly possition it within + // our canvas. cr.move_to( -(width as f64 / pango::SCALE as f64) / 2.0, -(info.height() as f64) / 2.0, ); + // After telling the layout object where to draw itself, we actually tell + // it to draw itself into our cairo context. pangocairo::functions::show_layout(&cr, &layout); + // Here we go one step up in our stack of transformations, removing any + // changes we did to them since the last call to cr.save(); cr.restore(); } @@ -148,6 +211,13 @@ fn create_pipeline() -> Result { .unwrap(); let drawer_clone = drawer.clone(); + // Add a signal handler to the overlay's "caps-changed" signal. This could e.g. + // be called when the sink that we render to does not support resizing the image + // itself - but the user just changed the window-size. The element after the overlay + // will then change its caps and we use the notification about this change to + // resize our canvas's size. + // Another possibility for when this might happen is, when our video is a network + // stream that dynamically changes resolution when enough bandwith is available. overlay .connect("caps-changed", false, move |args| { let _overlay = args[0].get::().unwrap(); diff --git a/examples/src/bin/playbin.rs b/examples/src/bin/playbin.rs index 5d0bb0c7a..d52cb53ca 100644 --- a/examples/src/bin/playbin.rs +++ b/examples/src/bin/playbin.rs @@ -1,3 +1,14 @@ +// This example demonstrates GStreamer's playbin element. +// This element takes an arbitrary URI as parameter, and if there is a source +// element within gstreamer, that supports this uri, the playbin will try +// to automatically create a pipeline that properly plays this media source. +// For this, the playbin internally relies on more bin elements, like the +// autovideosink and the decodebin. +// Essentially, this element is a single-element pipeline able to play +// any format from any uri-addressable source that gstreamer supports. +// Much of the playbin's behavior can be controlled by so-called flags, as well +// as the playbin's properties and signals. + extern crate gstreamer as gst; use gst::prelude::*; @@ -19,12 +30,15 @@ fn example_main() { std::process::exit(-1) }; + // Create a new playbin element, and tell it what uri to play back. let playbin = gst::ElementFactory::make("playbin", None).unwrap(); playbin .set_property("uri", &glib::Value::from(uri)) .unwrap(); // For flags handling + // With flags, one can configure playbin's behavior such as whether it + // should play back contained video streams, or if it should render subtitles. // let flags = playbin.get_property("flags").unwrap(); // let flags_class = FlagsClass::new(flags.type_()).unwrap(); // let flags = flags_class.builder_with_value(flags).unwrap() @@ -34,13 +48,35 @@ fn example_main() { // .unwrap(); // playbin.set_property("flags", &flags).unwrap(); + // The playbin also provides any kind of metadata that it found in the played stream. + // For this, the playbin provides signals notifying about changes in the metadata. + // Doing this with a signal makes sense for multiple reasons. + // - The metadata is only found after the pipeline has been started + // - Live streams (such as internet radios) update this metadata during the stream + // Note that this signal will be emitted from the streaming threads usually, + // not the application's threads! playbin .connect("audio-tags-changed", false, |values| { + // The metadata of any of the contained audio streams changed + // In the case of a live-stream from an internet radio, this could for example + // mark the beginning of a new track, or a new DJ. let playbin = values[0].get::().unwrap(); + // This gets the index of the stream that changed. This is neccessary, since + // there could e.g. be multiple audio streams (english, spanish, ...). let idx = values[1].get::().unwrap(); println!("audio tags of audio stream {} changed:", idx); + // HELP: is this correct? + // We were only notified about the change of metadata. If we want to do + // something with it, we first need to actually query the metadata from the playbin. + // We do this by facilliating the get-audio-tags action-signal on playbin. + // Sending an action-signal to an element essentially is a function call on the element. + // It is done that way, because elements do not have their own function API, they are + // relying on GStreamer and GLib's API. The only way an element can communicate with an + // application is via properties, signals or action signals (or custom messages, events, queries). + // So what the following code does, is essentially asking playbin to tell us its already + // internally stored tag list for this stream index. let tags = playbin .emit("get-audio-tags", &[&idx.to_value()]) .unwrap() @@ -63,6 +99,8 @@ fn example_main() { }) .unwrap(); + // The playbin element itself is a playbin, so it can be used as one, despite being + // created from an element factory. let bus = playbin.get_bus().unwrap(); let ret = playbin.set_state(gst::State::Playing); diff --git a/examples/src/bin/player.rs b/examples/src/bin/player.rs index f60f3fd04..529021768 100644 --- a/examples/src/bin/player.rs +++ b/examples/src/bin/player.rs @@ -1,3 +1,10 @@ +// This example shows how to use the GstPlayer API. +// The GstPlayer API is a convenience API to allow implement playback applications +// without having to write too much code. +// Most of the tasks a player needs to support (such as seeking and switching +// audio / subtitle streams or changing the volume) are all supported by simple +// one-line function calls on the GstPlayer. + extern crate gstreamer as gst; use gst::prelude::*; @@ -28,11 +35,14 @@ fn main_loop(uri: &str) -> Result<(), Error> { Some(&dispatcher.upcast::()), ); - player.set_property("uri", &glib::Value::from(uri))?; + // Tell the player what uri to play. + player.set_uri(uri); let error = Arc::new(Mutex::new(Ok(()))); let main_loop_clone = main_loop.clone(); + // Connect to the player's "end-of-stream" signal, which will tell us when the + // currently played media stream reached its end. player.connect_end_of_stream(move |player| { let main_loop = &main_loop_clone; player.stop(); @@ -41,6 +51,8 @@ fn main_loop(uri: &str) -> Result<(), Error> { let main_loop_clone = main_loop.clone(); let error_clone = Arc::clone(&error); + // Connect to the player's "error" signal, which will inform us about eventual + // errors (such as failing to retrieve a http stream). player.connect_error(move |player, err| { let main_loop = &main_loop_clone; let error = &error_clone; diff --git a/examples/src/bin/queries.rs b/examples/src/bin/queries.rs index cb5cc79f8..3b93bfde2 100644 --- a/examples/src/bin/queries.rs +++ b/examples/src/bin/queries.rs @@ -1,3 +1,17 @@ +// This example demonstrates how to use GStreamer's query functionality. +// These are a way to query information from either elements or pads. +// Such information could for example be the current position within +// the stream (i.e. the playing time). Queries can traverse the pipeline +// (both up and downstream). This functionality is essential, since most +// queries can only answered by specific elements in a pipeline (such as the +// stream's duration, which often can only be answered by the demuxer). +// Since gstreamer has many elements that itself contain other elements that +// we don't know of, we can simply send a query for the duration into the +// pipeline and the query is passed along until an element feels capable +// of answering. +// For convenience, the API has a set of pre-defined queries, but also +// allows custom queries (which can be defined and used by your own elements). + extern crate gstreamer as gst; use gst::prelude::*; @@ -9,22 +23,35 @@ use std::env; mod examples_common; fn example_main() { + // Get a string containing the passed pipeline launch syntax let pipeline_str = env::args().collect::>()[1..].join(" "); gst::init().unwrap(); let main_loop = glib::MainLoop::new(None, false); + // Let GStreamer create a pipeline from the parsed launch syntax on the cli. let pipeline = gst::parse_launch(&pipeline_str).unwrap(); let bus = pipeline.get_bus().unwrap(); let ret = pipeline.set_state(gst::State::Playing); assert_ne!(ret, gst::StateChangeReturn::Failure); - let main_loop_clone = main_loop.clone(); - + // Need to move a new reference into the closure. + // !!ATTENTION!!: + // It might seem appealing to use pipeline.clone() here, because that greatly + // simplifies the code within the callback. What this actually dose, however, is creating + // a memory leak. The clone of a pipeline is a new strong reference on the pipeline. + // Storing this strong reference of the pipeline within the callback (we are moving it in!), + // which is in turn stored in another strong reference on the pipeline is creating a + // reference cycle. + // DO NOT USE pipeline.clone() TO USE THE PIPELINE WITHIN A CALLBACK let pipeline_weak = pipeline.downgrade(); + // Add a timeout to the main loop. This closure will be executed + // in an interval of 1 second. let timeout_id = glib::timeout_add_seconds(1, move || { + // Here we temporarily retrieve a strong reference on the pipeline from the weak one + // we moved into this callback. let pipeline = match pipeline_weak.upgrade() { Some(pipeline) => pipeline, None => return glib::Continue(true), @@ -33,6 +60,9 @@ fn example_main() { //let pos = pipeline.query_position(gst::Format::Time).unwrap_or(-1); //let dur = pipeline.query_duration(gst::Format::Time).unwrap_or(-1); let pos = { + // Create a new position query and send it to the pipeline. + // This will traverse all elements in the pipeline, until one feels + // capable of answering the query. let mut q = gst::Query::new_position(gst::Format::Time); if pipeline.query(&mut q) { Some(q.get_result()) @@ -44,6 +74,9 @@ fn example_main() { .unwrap(); let dur = { + // Create a new duration query and send it to the pipeline. + // This will traverse all elements in the pipeline, until one feels + // capable of answering the query. let mut q = gst::Query::new_duration(gst::Format::Time); if pipeline.query(&mut q) { Some(q.get_result()) @@ -59,6 +92,8 @@ fn example_main() { glib::Continue(true) }); + // Need to move a new reference into the closure. + let main_loop_clone = main_loop.clone(); //bus.add_signal_watch(); //bus.connect_message(move |_, msg| { bus.add_watch(move |_, msg| { diff --git a/examples/src/bin/rtsp-server-record.rs b/examples/src/bin/rtsp-server-record.rs index ea0dbd27d..250920491 100644 --- a/examples/src/bin/rtsp-server-record.rs +++ b/examples/src/bin/rtsp-server-record.rs @@ -1,3 +1,9 @@ +// This example demonstrates how to set up a rtsp server using GStreamer. +// While the "rtsp-server" example is about streaming media to connecting +// clients, this example is mainly about recording media that clients +// send to the server. For this, the launch syntax pipeline, that is passed +// to this example's cli is spawned and the client's media is streamed into it. + extern crate failure; extern crate gio; extern crate glib; @@ -37,13 +43,26 @@ fn main_loop() -> Result<(), Error> { return Err(Error::from(UsageError(args[0].clone()))); } + // Mostly analog to the rtsp-server example, the server is created + // and the factory for our test mount is configured. let main_loop = glib::MainLoop::new(None, false); let server = RTSPServer::new(); - let factory = RTSPMediaFactory::new(); + // Much like HTTP servers, RTSP servers have multiple endpoints that + // provide or take different streams. Here, we ask our server to give + // us a reference to its list of endpoints, so we can add our + // test endpoint. let mounts = server.get_mount_points().ok_or(NoMountPoints)?; + // Next, we create a factory for the endpoint we want to create. + // The job of the factory is to create a new pipeline for each client that + // connects, or (if configured to do so) to reuse an existing pipeline. + let factory = RTSPMediaFactory::new(); + // Here we configure a method of authentication that we want the + // server to require from clients. let auth = RTSPAuth::new(); let token = RTSPToken::new(&[(*RTSP_TOKEN_MEDIA_FACTORY_ROLE, &"user")]); let basic = RTSPAuth::make_basic("user", "password"); + // For propery authentication, we want to use encryption. And there's no + // encryption without a certificate! let cert = gio::TlsCertificate::new_from_pem( "-----BEGIN CERTIFICATE-----\ MIICJjCCAY+gAwIBAgIBBzANBgkqhkiG9w0BAQUFADCBhjETMBEGCgmSJomT8ixk\ @@ -71,6 +90,8 @@ fn main_loop() -> Result<(), Error> { )?; // Bindable versions were added in b1f515178a363df0322d7adbd5754e1f6e2083c9 + // This declares that the user "user" (once authenticated) has a role that + // allows them to access and construct media factories. unsafe { ffi::gst_rtsp_media_factory_add_role( factory.to_glib_none().0, @@ -87,13 +108,35 @@ fn main_loop() -> Result<(), Error> { auth.set_tls_certificate(&cert); auth.add_basic(basic.as_str(), &token); + // Here, we tell the RTSP server about the authentication method we + // configured above. server.set_auth(&auth); + factory.set_launch(args[1].as_str()); + // Tell the RTSP server that we want to work in RECORD mode (clients send) + // data to us. factory.set_transport_mode(RTSPTransportMode::RECORD); + // The RTSP protocol allows a couple of different profiles for the actually + // used protocol of data-transmission. With this, we can limit the selection + // from which connecting clients have to choose. + // SAVP/SAVPF are via SRTP (encrypted), that's what the S is for. + // The F in the end is for feedback (an extension that allows more bidirectional + // feedback between sender and receiver). AV is just Audio/Video, P is Profile :) + // The default, old RTP profile is AVP factory.set_profiles(RTSPProfile::SAVP | RTSPProfile::SAVPF); + // Now we add a new mount-point and tell the RTSP server to use the factory + // we configured beforehand. This factory will take on the job of creating + // a pipeline, which will take on the incoming data of connected clients. mounts.add_factory("/test", &factory); + // Attach the server to our main context. + // A main context is the thing where other stuff is registering itself for its + // events (e.g. sockets, GStreamer bus, ...) and the main loop is something that + // polls the main context for its events and dispatches them to whoever is + // interested in them. In this example, we only do have one, so we can + // leave the context parameter empty, it will automatically select + // the default one. let id = server.attach(None); println!( @@ -101,6 +144,8 @@ fn main_loop() -> Result<(), Error> { server.get_bound_port() ); + // Start the mainloop. From this point on, the server will start to take + // incoming connections from clients. main_loop.run(); glib::source_remove(id); diff --git a/examples/src/bin/rtsp-server.rs b/examples/src/bin/rtsp-server.rs index 1d0bcb466..c470ea200 100644 --- a/examples/src/bin/rtsp-server.rs +++ b/examples/src/bin/rtsp-server.rs @@ -1,3 +1,8 @@ +// This example demonstrates how to set up a rtsp server using GStreamer. +// For this, the example parses an arbitrary pipeline in launch syntax +// from the cli and provides this pipeline's output as stream, served +// using GStreamers rtsp server. + use std::env; extern crate gstreamer as gst; @@ -33,14 +38,42 @@ fn main_loop() -> Result<(), Error> { let main_loop = glib::MainLoop::new(None, false); let server = gst_rtsp_server::RTSPServer::new(); - let factory = gst_rtsp_server::RTSPMediaFactory::new(); + // Much like HTTP servers, RTSP servers have multiple endpoints that + // provide different streams. Here, we ask our server to give + // us a reference to his list of endpoints, so we can add our + // test endpoint, providing the pipeline from the cli. let mounts = server.get_mount_points().ok_or(NoMountPoints)?; + // Next, we create a factory for the endpoint we want to create. + // The job of the factory is to create a new pipeline for each client that + // connects, or (if configured to do so) to reuse an existing pipeline. + let factory = gst_rtsp_server::RTSPMediaFactory::new(); + // Here we tell the media factory the media we want to serve. + // This is done in the launch syntax. When the first client connects, + // the factory will use this syntax to create a new pipeline instance. factory.set_launch(args[1].as_str()); + // This setting specifies whether each connecting client gets the output + // of a new instance of the pipeline, or whether all connected clients share + // the output of the same pipeline. + // If you want to stream a fixed video you have stored on the server to any + // client, you would not set this to shared here (since every client wants + // to start at the beginning of the video). But if you want to distribute + // a live source, you will probably want to set this to shared, to save + // computing and memory capacity on the server. factory.set_shared(true); + // Now we add a new mount-point and tell the RTSP server to serve the content + // provided by the factory we configured above, when a client connects to + // this specific path. mounts.add_factory("/test", &factory); + // Attach the server to our main context. + // A main context is the thing where other stuff is registering itself for its + // events (e.g. sockets, GStreamer bus, ...) and the main loop is something that + // polls the main context for its events and dispatches them to whoever is + // interested in them. In this example, we only do have one, so we can + // leave the context parameter empty, it will automatically select + // the default one. let id = server.attach(None); println!( @@ -48,6 +81,8 @@ fn main_loop() -> Result<(), Error> { server.get_bound_port() ); + // Start the mainloop. From this point on, the server will start to serve + // our quality content to connecting clients. main_loop.run(); glib::source_remove(id); diff --git a/examples/src/bin/tagsetter.rs b/examples/src/bin/tagsetter.rs index 3bd520e5f..f6727516a 100644 --- a/examples/src/bin/tagsetter.rs +++ b/examples/src/bin/tagsetter.rs @@ -1,3 +1,23 @@ +// This example demonstrates how to set and store metadata using +// GStreamer. Some elements support setting tags on a media stream. +// An example would be id3v2mux. The element signals this by implementing +// The GstTagsetter interface. You can query any element implementing this +// interface from the pipeline, and then tell the returned implementation +// of GstTagsetter what tags to apply to the media stream. +// This example's pipeline creates a new flac file from the testaudiosrc +// that the example application will add tags to using GstTagsetter. +// The operated pipeline looks like this: + +// {audiotestsrc} - {flacenc} - {filesink} + +// For example for pipelines that transcode a multimedia file, the input +// already has tags. For cases like this, the GstTagsetter has the merge +// setting, which the application can configure to tell the element +// implementing the interface whether to merge newly applied tags to the +// already existing ones, or if all existing ones should replace, etc. +// (More modes of operation are possible, see: gst::TagMergeMode) +// This merge-mode can also be supplied to any method that adds new tags. + extern crate gstreamer as gst; use gst::prelude::*; @@ -34,6 +54,7 @@ struct ErrorMessage { fn example_main() -> Result<(), Error> { gst::init()?; + // Parse the pipeline we want to probe from a static in-line string. let mut context = gst::ParseContext::new(); let pipeline = match gst::parse_launch_full( "audiotestsrc wave=white-noise num-buffers=100 ! flacenc ! filesink location=test.flac", @@ -54,6 +75,8 @@ fn example_main() -> Result<(), Error> { .downcast::() .map_err(|_| failure::err_msg("Generated pipeline is no pipeline"))?; + // Query the pipeline for elements implementing the GstTagsetter interface. + // In our case, this will return the flacenc element. let tagsetter = pipeline .get_by_interface(gst::TagSetter::static_type()) .ok_or_else(|| failure::err_msg("No TagSetter found"))?; @@ -61,7 +84,12 @@ fn example_main() -> Result<(), Error> { .dynamic_cast::() .map_err(|_| failure::err_msg("No TagSetter found"))?; + // Tell the element implementing the GstTagsetter interface how to handle already existing + // metadata. tagsetter.set_tag_merge_mode(gst::TagMergeMode::KeepAll); + // Set the "title" tag to "Special randomized white-noise". + // The second parameter gst::TagMergeMode::Append tells the tagsetter to append this title + // if there already is one. tagsetter.add::(&"Special randomized white-noise", gst::TagMergeMode::Append); let bus = pipeline.get_bus().unwrap(); diff --git a/examples/src/bin/toc.rs b/examples/src/bin/toc.rs index e2e527530..faf3d6d63 100644 --- a/examples/src/bin/toc.rs +++ b/examples/src/bin/toc.rs @@ -1,3 +1,14 @@ +// This example demonstrates the use of GStreamer's ToC API. This API is used +// to manage a table of contents contained in the handled media stream. +// Chapters within a matroska file would be an example of a scenario for using +// this API. Elements that can parse ToCs from a stream (such as matroskademux) +// notify all elements in the pipeline when they encountered a ToC. +// For this, the example operates the following pipeline: + +// /-{queue} - {fakesink} +// {filesrc} - {decodebin} - {queue} - {fakesink} +// \- ... + extern crate gstreamer as gst; use gst::prelude::*; @@ -29,13 +40,29 @@ fn example_main() { pipeline.add_many(&[&src, &decodebin]).unwrap(); gst::Element::link_many(&[&src, &decodebin]).unwrap(); - // Need to move a new reference into the closure + // Need to move a new reference into the closure. + // !!ATTENTION!!: + // It might seem appealing to use pipeline.clone() here, because that greatly + // simplifies the code within the callback. What this actually dose, however, is creating + // a memory leak. The clone of a pipeline is a new strong reference on the pipeline. + // Storing this strong reference of the pipeline within the callback (we are moving it in!), + // which is in turn stored in another strong reference on the pipeline is creating a + // reference cycle. + // DO NOT USE pipeline.clone() TO USE THE PIPELINE WITHIN A CALLBACK let pipeline_weak = pipeline.downgrade(); + // Connect to decodebin's pad-added signal, that is emitted whenever it found another stream + // from the input file and found a way to decode it to its raw format. decodebin.connect_pad_added(move |_, src_pad| { + // Here we temporarily retrieve a strong reference on the pipeline from the weak one + // we moved into this callback. let pipeline = match pipeline_weak.upgrade() { Some(pipeline) => pipeline, None => return, }; + + // In this example, we are only interested about parsing the ToC, so + // we simply pipe every encountered stream into a fakesink, essentially + // throwing away the data. let queue = gst::ElementFactory::make("queue", None).unwrap(); let sink = gst::ElementFactory::make("fakesink", None).unwrap(); @@ -58,6 +85,11 @@ fn example_main() { let bus = pipeline.get_bus().unwrap(); + // Instead of using a main loop (like GLib's), we manually iterate over + // GStreamer's bus messages in this example. We don't need any special + // functionality like timeouts or GLib socket notifications, so this is sufficient. + // The bus is manually operated by repeatedly calling timed_pop on the bus with + // the desired timeout for when to stop waiting for new messages. (None = Wait forever) while let Some(msg) = bus.timed_pop(gst::CLOCK_TIME_NONE) { use gst::MessageView; @@ -73,27 +105,43 @@ fn example_main() { break; } MessageView::Toc(msg_toc) => { + // Some element found a ToC in the current media stream and told + // us by posting a message to GStreamer's bus. let (toc, updated) = msg_toc.get_toc(); println!( "\nReceived toc: {:?} - updated: {}", toc.get_scope(), updated ); + // Get a list of tags that are ToC specific. if let Some(tags) = toc.get_tags() { println!("- tags: {}", tags.to_string()); } + // ToCs do not have a fixed structure. Depending on the format that + // they were parsed from, they might have different tree-like structures, + // so applications that want to support ToCs (for example in the form + // of jumping between chapters in a video) have to try parsing and + // interpreting the ToC manually. + // In this example, we simply want to print the ToC structure, so + // we iterate everything and don't try to interpret anything. for toc_entry in toc.get_entries() { + // Every entry in a ToC has its own type. One type could for + // example be Chapter. println!( "\t{:?} - {}", toc_entry.get_entry_type(), toc_entry.get_uid() ); + // Every ToC entry can have a set of timestamps (start, stop). if let Some((start, stop)) = toc_entry.get_start_stop_times() { println!("\t- start: {}, stop: {}", start, stop); } + // Every ToC entry can have tags to it. if let Some(tags) = toc_entry.get_tags() { println!("\t- tags: {}", tags.to_string()); } + // Every ToC entry can have a set of child entries. + // With this structure, you can create trees of arbitrary depth. for toc_sub_entry in toc_entry.get_sub_entries() { println!( "\n\t\t{:?} - {}",