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https://gitlab.freedesktop.org/gstreamer/gstreamer-rs.git
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Add basic-tutorial-8
Fixes https://github.com/sdroege/gstreamer-rs/pull/52
This commit is contained in:
Guillaume Desmottes
Sebastian Dröge
parent
c9636bca2e
commit
82681d2b24
2 changed files with 268 additions and 0 deletions
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@ -8,8 +8,11 @@ glib = { version = "0.3", git = "https://github.com/gtk-rs/glib" }
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gdk = { version = "0.6", git = "https://github.com/gtk-rs/gdk", optional = true }
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gtk = { version = "0.2", git = "https://github.com/gtk-rs/gtk", features = ["v3_10"], optional = true }
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gstreamer = { path = "../gstreamer" }
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gstreamer-audio = { path = "../gstreamer-audio" }
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gstreamer-video = { path = "../gstreamer-video" }
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gstreamer-app = { path = "../gstreamer-app" }
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send-cell = "0.1"
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byte-slice-cast = "0.1"
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[badges]
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travis-ci = { repository = "sdroege/gstreamer-rs", branch = "master" }
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265
tutorials/src/bin/basic-tutorial-8.rs
Normal file
265
tutorials/src/bin/basic-tutorial-8.rs
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@ -0,0 +1,265 @@
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use std::sync::{Arc, Mutex};
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extern crate byte_slice_cast;
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use byte_slice_cast::*;
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extern crate gstreamer as gst;
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use gst::prelude::*;
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extern crate gstreamer_audio as gst_audio;
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use gst_audio::AudioInfo;
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extern crate gstreamer_app as gst_app;
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use gst_app::{AppSink, AppSrc};
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extern crate glib;
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use glib::source::SourceId;
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const CHUNK_SIZE: usize = 1024; // Amount of bytes we are sending in each buffer
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const SAMPLE_RATE: u32 = 44100; // Samples per second we are sending
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#[derive(Debug)]
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struct CustomData {
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source_id: Option<SourceId>,
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num_samples: u64, // Number of samples generated so far (for timestamp generation)
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// For waveform generation
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a: f64,
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b: f64,
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c: f64,
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d: f64,
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appsrc: AppSrc,
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appsink: AppSink,
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}
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impl CustomData {
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fn new(appsrc: &AppSrc, appsink: &AppSink) -> CustomData {
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CustomData {
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source_id: None,
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num_samples: 0,
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a: 0.0,
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b: 1.0,
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c: 0.0,
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d: 1.0,
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appsrc: appsrc.clone(),
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appsink: appsink.clone(),
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}
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}
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}
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fn main() {
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// Initialize GStreamer
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if let Err(err) = gst::init() {
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eprintln!("Failed to initialize Gst: {}", err);
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return;
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}
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let appsrc = gst::ElementFactory::make("appsrc", "audio_source").unwrap();
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let tee = gst::ElementFactory::make("tee", "tee").unwrap();
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let audio_queue = gst::ElementFactory::make("queue", "audio_queue").unwrap();
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let audio_convert1 = gst::ElementFactory::make("audioconvert", "audio_convert1").unwrap();
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let audio_resample = gst::ElementFactory::make("audioresample", "audio_resample").unwrap();
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let audio_sink = gst::ElementFactory::make("autoaudiosink", "audio_sink").unwrap();
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let video_queue = gst::ElementFactory::make("queue", "video_queue").unwrap();
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let audio_convert2 = gst::ElementFactory::make("audioconvert", "audio_convert2").unwrap();
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let visual = gst::ElementFactory::make("wavescope", "visual").unwrap();
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let video_convert = gst::ElementFactory::make("videoconvert", "video_convert").unwrap();
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let video_sink = gst::ElementFactory::make("autovideosink", "video_sink").unwrap();
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let app_queue = gst::ElementFactory::make("queue", "app_queue").unwrap();
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let appsink = gst::ElementFactory::make("appsink", "app_sink").unwrap();
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let pipeline = gst::Pipeline::new("test-pipeline");
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visual.set_property_from_str("shader", "none");
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visual.set_property_from_str("style", "lines");
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pipeline
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.add_many(&[
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&appsrc,
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&tee,
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&audio_queue,
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&audio_convert1,
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&audio_resample,
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&audio_sink,
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&video_queue,
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&audio_convert2,
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&visual,
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&video_convert,
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&video_sink,
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&app_queue,
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&appsink,
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])
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.unwrap();
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gst::Element::link_many(&[&appsrc, &tee]).unwrap();
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gst::Element::link_many(&[
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&audio_queue,
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&audio_convert1,
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&audio_resample,
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&audio_sink,
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]).unwrap();
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gst::Element::link_many(&[
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&video_queue,
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&audio_convert2,
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&visual,
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&video_convert,
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&video_sink,
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]).unwrap();
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gst::Element::link_many(&[&app_queue, &appsink]).unwrap();
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let tee_audio_pad = tee.get_request_pad("src_%u").unwrap();
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println!(
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"Obtained request pad {} for audio branch",
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tee_audio_pad.get_name()
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);
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let queue_audio_pad = audio_queue.get_static_pad("sink").unwrap();
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tee_audio_pad.link(&queue_audio_pad).into_result().unwrap();
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let tee_video_pad = tee.get_request_pad("src_%u").unwrap();
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println!(
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"Obtained request pad {} for video branch",
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tee_video_pad.get_name()
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);
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let queue_video_pad = video_queue.get_static_pad("sink").unwrap();
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tee_video_pad.link(&queue_video_pad).into_result().unwrap();
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let tee_app_pad = tee.get_request_pad("src_%u").unwrap();
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let queue_app_pad = app_queue.get_static_pad("sink").unwrap();
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tee_app_pad.link(&queue_app_pad).into_result().unwrap();
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// configure appsrc
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let info = AudioInfo::new(gst_audio::AudioFormat::S16le, SAMPLE_RATE, 1)
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.build()
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.unwrap();
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let audio_caps = info.to_caps().unwrap();
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let appsrc = appsrc
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.clone()
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.dynamic_cast::<AppSrc>()
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.expect("Source element is expected to be an appsrc!");
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appsrc.set_caps(&audio_caps);
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appsrc.set_property_format(gst::Format::Time);
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let appsink = appsink
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.clone()
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.dynamic_cast::<AppSink>()
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.expect("Sink element is expected to be an appsink!");
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let data: Arc<Mutex<CustomData>> = Arc::new(Mutex::new(CustomData::new(&appsrc, &appsink)));
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let data_clone = data.clone();
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appsrc.connect_need_data(move |_, _size| {
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let data = &data_clone;
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let mut d = data.lock().unwrap();
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if d.source_id.is_none() {
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println!("start feeding");
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let data_clone = data.clone();
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d.source_id = Some(glib::source::idle_add(move || {
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let data = &data_clone;
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let (appsrc, buffer) = {
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let mut data = data.lock().unwrap();
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let mut buffer = gst::Buffer::with_size(CHUNK_SIZE).unwrap();
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let num_samples = CHUNK_SIZE / 2; /* Each sample is 16 bits */
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let pts = gst::SECOND
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.mul_div_floor(data.num_samples, SAMPLE_RATE as u64)
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.expect("u64 overflow");
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let duration = gst::SECOND
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.mul_div_floor(CHUNK_SIZE as u64, SAMPLE_RATE as u64)
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.expect("u64 overflow");
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{
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let buffer = buffer.get_mut().unwrap();
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{
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let mut samples = buffer.map_writable().unwrap();
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let samples = samples.as_mut_slice().as_mut_slice_of::<i16>().unwrap();
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// Generate some psychodelic waveforms
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data.c += data.d;
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data.d -= data.c / 1000.0;
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let freq = 1100.0 + 1000.0 * data.d;
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for sample in samples.iter_mut() {
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data.a += data.b;
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data.b -= data.a / freq;
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*sample = 500 * (data.a as i16);
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}
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data.num_samples += num_samples as u64;
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}
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buffer.set_pts(pts);
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buffer.set_duration(duration);
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}
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(data.appsrc.clone(), buffer)
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};
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match appsrc.push_buffer(buffer) {
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gst::FlowReturn::Ok => glib::Continue(true),
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_ => glib::Continue(false),
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}
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}));
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}
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});
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let data_clone = data.clone();
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appsrc.connect_enough_data(move |_| {
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let data = &data_clone;
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let mut data = data.lock().unwrap();
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if let Some(source) = data.source_id.take() {
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println!("stop feeding");
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glib::source::source_remove(source);
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}
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});
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// configure appsink
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appsink.set_emit_signals(true);
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appsink.set_caps(&audio_caps);
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let data_clone = data.clone();
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appsink.connect_new_sample(move |_| {
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let appsink = {
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let data = &data_clone.lock().unwrap();
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data.appsink.clone()
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};
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if let Some(_sample) = appsink.pull_sample() {
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// The only thing we do in this example is print a * to indicate a received buffer
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print!("*");
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}
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gst::FlowReturn::Ok
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});
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pipeline
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.set_state(gst::State::Playing)
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.into_result()
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.expect("Unable to set the pipeline to the Playing state.");
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let main_loop = glib::MainLoop::new(None, false);
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let main_loop_clone = main_loop.clone();
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let bus = pipeline.get_bus().unwrap();
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bus.connect_message(move |_, msg| match msg.view() {
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gst::MessageView::Error(err) => {
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let main_loop = &main_loop_clone;
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eprintln!(
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"Error received from element {:?}: {}",
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msg.get_src().map(|s| s.get_path_string()),
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err.get_error()
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);
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eprintln!("Debugging information: {:?}", err.get_debug());
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main_loop.quit();
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}
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_ => (),
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});
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bus.add_signal_watch();
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main_loop.run();
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pipeline
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.set_state(gst::State::Null)
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.into_result()
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.expect("Unable to set the pipeline to the Null state.");
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}
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