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gst-plugins-rs/net/webrtc/src/webrtcsink/imp.rs
Thibault Saunier 6b11284e8a webrtcsink: Make the turn-server prop a turn-servers list 
So that we can simply specify several turn servers at once

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/973>
2022-11-16 14:48:16 +00:00

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// SPDX-License-Identifier: MPL-2.0
use anyhow::Context;
use gst::glib;
use gst::prelude::*;
use gst::subclass::prelude::*;
use gst_rtp::prelude::*;
use gst_utils::StreamProducer;
use gst_video::subclass::prelude::*;
use gst_webrtc::WebRTCDataChannel;
use async_std::task;
use futures::prelude::*;
use anyhow::{anyhow, Error};
use once_cell::sync::Lazy;
use std::collections::HashMap;
use std::ops::Mul;
use std::sync::Mutex;
use super::homegrown_cc::CongestionController;
use super::{WebRTCSinkCongestionControl, WebRTCSinkError, WebRTCSinkMitigationMode};
use crate::signaller::Signaller;
use std::collections::BTreeMap;
static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
gst::DebugCategory::new(
"webrtcsink",
gst::DebugColorFlags::empty(),
Some("WebRTC sink"),
)
});
const CUDA_MEMORY_FEATURE: &str = "memory:CUDAMemory";
const GL_MEMORY_FEATURE: &str = "memory:GLMemory";
const NVMM_MEMORY_FEATURE: &str = "memory:NVMM";
const RTP_TWCC_URI: &str =
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
const DEFAULT_STUN_SERVER: Option<&str> = Some("stun://stun.l.google.com:19302");
const DEFAULT_MIN_BITRATE: u32 = 1000;
/* I have found higher values to cause packet loss *somewhere* in
* my local network, possibly related to chrome's pretty low UDP
* buffer sizes */
const DEFAULT_MAX_BITRATE: u32 = 8192000;
const DEFAULT_CONGESTION_CONTROL: WebRTCSinkCongestionControl =
WebRTCSinkCongestionControl::GoogleCongestionControl;
const DEFAULT_DO_FEC: bool = true;
const DEFAULT_DO_RETRANSMISSION: bool = true;
const DEFAULT_ENABLE_DATA_CHANNEL_NAVIGATION: bool = false;
const DEFAULT_START_BITRATE: u32 = 2048000;
/* Start adding some FEC when the bitrate > 2Mbps as we found experimentally
* that it is not worth it below that threshold */
const DO_FEC_THRESHOLD: u32 = 2000000;
#[derive(Debug, Clone, Copy)]
struct CCInfo {
heuristic: WebRTCSinkCongestionControl,
min_bitrate: u32,
max_bitrate: u32,
start_bitrate: u32,
}
/// User configuration
struct Settings {
video_caps: gst::Caps,
audio_caps: gst::Caps,
turn_servers: gst::Array,
stun_server: Option<String>,
cc_info: CCInfo,
do_fec: bool,
do_retransmission: bool,
enable_data_channel_navigation: bool,
meta: Option<gst::Structure>,
}
/// Represents a codec we can offer
#[derive(Debug, Clone)]
struct Codec {
encoder: gst::ElementFactory,
payloader: gst::ElementFactory,
caps: gst::Caps,
payload: i32,
}
impl Codec {
fn is_video(&self) -> bool {
self.encoder
.has_type(gst::ElementFactoryType::VIDEO_ENCODER)
}
}
/// Wrapper around our sink pads
#[derive(Debug, Clone)]
struct InputStream {
sink_pad: gst::GhostPad,
producer: Option<StreamProducer>,
/// The (fixed) caps coming in
in_caps: Option<gst::Caps>,
/// The caps we will offer, as a set of fixed structures
out_caps: Option<gst::Caps>,
/// Pace input data
clocksync: Option<gst::Element>,
}
/// Wrapper around webrtcbin pads
#[derive(Clone)]
struct WebRTCPad {
pad: gst::Pad,
/// The (fixed) caps of the corresponding input stream
in_caps: gst::Caps,
/// The m= line index in the SDP
media_idx: u32,
ssrc: u32,
/// The name of the corresponding InputStream's sink_pad
stream_name: String,
/// The payload selected in the answer, None at first
payload: Option<i32>,
}
/// Wrapper around GStreamer encoder element, keeps track of factory
/// name in order to provide a unified set / get bitrate API, also
/// tracks a raw capsfilter used to resize / decimate the input video
/// stream according to the bitrate, thresholds hardcoded for now
pub struct VideoEncoder {
factory_name: String,
codec_name: String,
element: gst::Element,
filter: gst::Element,
halved_framerate: gst::Fraction,
video_info: gst_video::VideoInfo,
session_id: String,
mitigation_mode: WebRTCSinkMitigationMode,
pub transceiver: gst_webrtc::WebRTCRTPTransceiver,
}
struct Session {
id: String,
pipeline: gst::Pipeline,
webrtcbin: gst::Element,
rtprtxsend: Option<gst::Element>,
webrtc_pads: HashMap<u32, WebRTCPad>,
peer_id: String,
encoders: Vec<VideoEncoder>,
// Our Homegrown controller (if cc_info.heuristic == Homegrown)
congestion_controller: Option<CongestionController>,
// Our BandwidthEstimator (if cc_info.heuristic == GoogleCongestionControl)
rtpgccbwe: Option<gst::Element>,
sdp: Option<gst_sdp::SDPMessage>,
stats: gst::Structure,
cc_info: CCInfo,
links: HashMap<u32, gst_utils::ConsumptionLink>,
stats_sigid: Option<glib::SignalHandlerId>,
}
#[derive(PartialEq)]
enum SignallerState {
Started,
Stopped,
}
#[derive(Debug, serde::Deserialize)]
struct NavigationEvent {
mid: Option<String>,
#[serde(flatten)]
event: gst_video::NavigationEvent,
}
/* Our internal state */
struct State {
signaller: Box<dyn super::SignallableObject>,
signaller_state: SignallerState,
sessions: HashMap<String, Session>,
codecs: BTreeMap<i32, Codec>,
/// Used to abort codec discovery
codecs_abort_handle: Option<futures::future::AbortHandle>,
/// Used to wait for the discovery task to fully stop
codecs_done_receiver: Option<futures::channel::oneshot::Receiver<()>>,
/// Used to determine whether we can start the signaller when going to Playing,
/// or whether we should wait
codec_discovery_done: bool,
audio_serial: u32,
video_serial: u32,
streams: HashMap<String, InputStream>,
navigation_handler: Option<NavigationEventHandler>,
mids: HashMap<String, String>,
}
fn create_navigation_event(sink: &super::WebRTCSink, msg: &str) {
let event: Result<NavigationEvent, _> = serde_json::from_str(msg);
if let Ok(event) = event {
gst::log!(CAT, obj: sink, "Processing navigation event: {:?}", event);
if let Some(mid) = event.mid {
let this = sink.imp();
let state = this.state.lock().unwrap();
if let Some(stream_name) = state.mids.get(&mid) {
if let Some(stream) = state.streams.get(stream_name) {
let event = gst::event::Navigation::new(event.event.structure());
if !stream.sink_pad.push_event(event.clone()) {
gst::info!(CAT, "Could not send event: {:?}", event);
}
}
}
} else {
let this = sink.imp();
let state = this.state.lock().unwrap();
let event = gst::event::Navigation::new(event.event.structure());
state.streams.iter().for_each(|(_, stream)| {
if stream.sink_pad.name().starts_with("video_") {
gst::log!(CAT, "Navigating to: {:?}", event);
if !stream.sink_pad.push_event(event.clone()) {
gst::info!(CAT, "Could not send event: {:?}", event);
}
}
});
}
} else {
gst::error!(CAT, "Invalid navigation event: {:?}", msg);
}
}
/// Wrapper around `gst::ElementFactory::make` with a better error
/// message
pub fn make_element(element: &str, name: Option<&str>) -> Result<gst::Element, Error> {
let mut builder = gst::ElementFactory::make(element);
if let Some(name) = name {
builder = builder.name(name);
}
builder
.build()
.with_context(|| format!("Failed to make element {}", element))
}
/// Simple utility for tearing down a pipeline cleanly
struct PipelineWrapper(gst::Pipeline);
// Structure to generate GstNavigation event from a WebRTCDataChannel
// This is simply used to hold references to the inner items.
#[derive(Debug)]
struct NavigationEventHandler((glib::SignalHandlerId, WebRTCDataChannel));
/// Our instance structure
#[derive(Default)]
pub struct WebRTCSink {
state: Mutex<State>,
settings: Mutex<Settings>,
}
impl Default for Settings {
fn default() -> Self {
Self {
video_caps: ["video/x-vp8", "video/x-h264", "video/x-vp9", "video/x-h265"]
.iter()
.map(|s| gst::Structure::new_empty(s))
.collect::<gst::Caps>(),
audio_caps: ["audio/x-opus"]
.iter()
.map(|s| gst::Structure::new_empty(s))
.collect::<gst::Caps>(),
stun_server: DEFAULT_STUN_SERVER.map(String::from),
turn_servers: gst::Array::new(Vec::new() as Vec<glib::SendValue>),
cc_info: CCInfo {
heuristic: WebRTCSinkCongestionControl::GoogleCongestionControl,
min_bitrate: DEFAULT_MIN_BITRATE,
max_bitrate: DEFAULT_MAX_BITRATE as u32,
start_bitrate: DEFAULT_START_BITRATE,
},
do_fec: DEFAULT_DO_FEC,
do_retransmission: DEFAULT_DO_RETRANSMISSION,
enable_data_channel_navigation: DEFAULT_ENABLE_DATA_CHANNEL_NAVIGATION,
meta: None,
}
}
}
impl Default for State {
fn default() -> Self {
let signaller = Signaller::default();
Self {
signaller: Box::new(signaller),
signaller_state: SignallerState::Stopped,
sessions: HashMap::new(),
codecs: BTreeMap::new(),
codecs_abort_handle: None,
codecs_done_receiver: None,
codec_discovery_done: false,
audio_serial: 0,
video_serial: 0,
streams: HashMap::new(),
navigation_handler: None,
mids: HashMap::new(),
}
}
}
fn make_converter_for_video_caps(caps: &gst::Caps) -> Result<gst::Element, Error> {
assert!(caps.is_fixed());
let video_info = gst_video::VideoInfo::from_caps(caps)?;
let ret = gst::Bin::default();
let (head, mut tail) = {
if let Some(feature) = caps.features(0) {
if feature.contains(CUDA_MEMORY_FEATURE) {
let cudaupload = make_element("cudaupload", None)?;
let cudaconvert = make_element("cudaconvert", None)?;
let cudascale = make_element("cudascale", None)?;
ret.add_many(&[&cudaupload, &cudaconvert, &cudascale])?;
gst::Element::link_many(&[&cudaupload, &cudaconvert, &cudascale])?;
(cudaupload, cudascale)
} else if feature.contains(GL_MEMORY_FEATURE) {
let glupload = make_element("glupload", None)?;
let glconvert = make_element("glcolorconvert", None)?;
let glscale = make_element("glcolorscale", None)?;
ret.add_many(&[&glupload, &glconvert, &glscale])?;
gst::Element::link_many(&[&glupload, &glconvert, &glscale])?;
(glupload, glscale)
} else if feature.contains(NVMM_MEMORY_FEATURE) {
let queue = make_element("queue", None)?;
let nvconvert = make_element("nvvideoconvert", None)?;
nvconvert.set_property("compute-hw", 0);
nvconvert.set_property("nvbuf-memory-type", 0);
ret.add_many(&[&queue, &nvconvert])?;
gst::Element::link_many(&[&queue, &nvconvert])?;
(queue, nvconvert)
} else {
let convert = make_element("videoconvert", None)?;
let scale = make_element("videoscale", None)?;
ret.add_many(&[&convert, &scale])?;
gst::Element::link_many(&[&convert, &scale])?;
(convert, scale)
}
} else {
let convert = make_element("videoconvert", None)?;
let scale = make_element("videoscale", None)?;
ret.add_many(&[&convert, &scale])?;
gst::Element::link_many(&[&convert, &scale])?;
(convert, scale)
}
};
ret.add_pad(
&gst::GhostPad::with_target(Some("sink"), &head.static_pad("sink").unwrap()).unwrap(),
)
.unwrap();
if video_info.fps().numer() != 0 {
let vrate = make_element("videorate", None)?;
vrate.set_property("drop-only", true);
vrate.set_property("skip-to-first", true);
ret.add(&vrate)?;
tail.link(&vrate)?;
tail = vrate;
}
ret.add_pad(
&gst::GhostPad::with_target(Some("src"), &tail.static_pad("src").unwrap()).unwrap(),
)
.unwrap();
Ok(ret.upcast())
}
/// Default configuration for known encoders, can be disabled
/// by returning True from an encoder-setup handler.
fn configure_encoder(enc: &gst::Element, start_bitrate: u32) {
if let Some(factory) = enc.factory() {
match factory.name().as_str() {
"vp8enc" | "vp9enc" => {
enc.set_property("deadline", 1i64);
enc.set_property("target-bitrate", start_bitrate as i32);
enc.set_property("cpu-used", -16i32);
enc.set_property("keyframe-max-dist", 2000i32);
enc.set_property_from_str("keyframe-mode", "disabled");
enc.set_property_from_str("end-usage", "cbr");
enc.set_property("buffer-initial-size", 100i32);
enc.set_property("buffer-optimal-size", 120i32);
enc.set_property("buffer-size", 150i32);
enc.set_property("max-intra-bitrate", 250i32);
enc.set_property_from_str("error-resilient", "default");
enc.set_property("lag-in-frames", 0i32);
}
"x264enc" => {
enc.set_property("bitrate", start_bitrate / 1000);
enc.set_property_from_str("tune", "zerolatency");
enc.set_property_from_str("speed-preset", "ultrafast");
enc.set_property("threads", 12u32);
enc.set_property("key-int-max", 2560u32);
enc.set_property("b-adapt", false);
enc.set_property("vbv-buf-capacity", 120u32);
}
"nvh264enc" => {
enc.set_property("bitrate", start_bitrate / 1000);
enc.set_property("gop-size", 2560i32);
enc.set_property_from_str("rc-mode", "cbr-ld-hq");
enc.set_property("zerolatency", true);
}
"vaapih264enc" | "vaapivp8enc" => {
enc.set_property("bitrate", start_bitrate / 1000);
enc.set_property("keyframe-period", 2560u32);
enc.set_property_from_str("rate-control", "cbr");
}
"nvv4l2h264enc" => {
enc.set_property("bitrate", start_bitrate);
enc.set_property_from_str("preset-level", "UltraFastPreset");
enc.set_property("maxperf-enable", true);
enc.set_property("insert-vui", true);
enc.set_property("idrinterval", 256u32);
enc.set_property("insert-sps-pps", true);
enc.set_property("insert-aud", true);
enc.set_property_from_str("control-rate", "constant_bitrate");
}
"nvv4l2vp8enc" => {
enc.set_property("bitrate", start_bitrate);
enc.set_property_from_str("preset-level", "UltraFastPreset");
enc.set_property("maxperf-enable", true);
enc.set_property("idrinterval", 256u32);
enc.set_property_from_str("control-rate", "constant_bitrate");
}
_ => (),
}
}
}
/// Bit of an awkward function, but the goal here is to keep
/// most of the encoding code for consumers in line with
/// the codec discovery code, and this gets the job done.
fn setup_encoding(
pipeline: &gst::Pipeline,
src: &gst::Element,
input_caps: &gst::Caps,
codec: &Codec,
ssrc: Option<u32>,
twcc: bool,
) -> Result<(gst::Element, gst::Element, gst::Element), Error> {
let conv = match codec.is_video() {
true => make_converter_for_video_caps(input_caps)?.upcast(),
false => gst::parse_bin_from_description("audioresample ! audioconvert", true)?.upcast(),
};
let conv_filter = make_element("capsfilter", None)?;
let enc = codec
.encoder
.create()
.build()
.with_context(|| format!("Creating encoder {}", codec.encoder.name()))?;
let pay = codec
.payloader
.create()
.build()
.with_context(|| format!("Creating payloader {}", codec.payloader.name()))?;
let parse_filter = make_element("capsfilter", None)?;
pay.set_property("mtu", 1200_u32);
pay.set_property("pt", codec.payload as u32);
if let Some(ssrc) = ssrc {
pay.set_property("ssrc", ssrc);
}
pipeline
.add_many(&[&conv, &conv_filter, &enc, &parse_filter, &pay])
.unwrap();
gst::Element::link_many(&[src, &conv, &conv_filter, &enc])
.with_context(|| "Linking encoding elements")?;
let codec_name = codec.caps.structure(0).unwrap().name();
if let Some(parser) = if codec_name == "video/x-h264" {
Some(make_element("h264parse", None)?)
} else if codec_name == "video/x-h265" {
Some(make_element("h265parse", None)?)
} else {
None
} {
pipeline.add(&parser).unwrap();
gst::Element::link_many(&[&enc, &parser, &parse_filter])
.with_context(|| "Linking encoding elements")?;
} else {
gst::Element::link_many(&[&enc, &parse_filter])
.with_context(|| "Linking encoding elements")?;
}
let conv_caps = if codec.is_video() {
let mut structure_builder = gst::Structure::builder("video/x-raw")
.field("pixel-aspect-ratio", gst::Fraction::new(1, 1));
if codec.encoder.name() == "nvh264enc" {
// Quirk: nvh264enc can perform conversion from RGB formats, but
// doesn't advertise / negotiate colorimetry correctly, leading
// to incorrect color display in Chrome (but interestingly not in
// Firefox). In any case, restrict to exclude RGB formats altogether,
// and let videoconvert do the conversion properly if needed.
structure_builder =
structure_builder.field("format", gst::List::new(["NV12", "YV12", "I420"]));
}
gst::Caps::builder_full_with_any_features()
.structure(structure_builder.build())
.build()
} else {
gst::Caps::builder("audio/x-raw").build()
};
match codec.encoder.name().as_str() {
"vp8enc" | "vp9enc" => {
pay.set_property_from_str("picture-id-mode", "15-bit");
}
_ => (),
}
/* We only enforce TWCC in the offer caps, once a remote description
* has been set it will get automatically negotiated. This is necessary
* because the implementor in Firefox had apparently not understood the
* concept of *transport-wide* congestion control, and firefox doesn't
* provide feedback for audio packets.
*/
if twcc {
let twcc_extension = gst_rtp::RTPHeaderExtension::create_from_uri(RTP_TWCC_URI).unwrap();
twcc_extension.set_id(1);
pay.emit_by_name::<()>("add-extension", &[&twcc_extension]);
}
conv_filter.set_property("caps", conv_caps);
let parse_caps = if codec_name == "video/x-h264" {
gst::Caps::builder(codec_name)
.field("stream-format", "avc")
.field("profile", "constrained-baseline")
.build()
} else if codec_name == "video/x-h265" {
gst::Caps::builder(codec_name)
.field("stream-format", "hvc1")
.build()
} else {
gst::Caps::new_any()
};
parse_filter.set_property("caps", parse_caps);
gst::Element::link_many(&[&parse_filter, &pay]).with_context(|| "Linking encoding elements")?;
Ok((enc, conv_filter, pay))
}
impl VideoEncoder {
fn new(
element: gst::Element,
filter: gst::Element,
video_info: gst_video::VideoInfo,
peer_id: &str,
codec_name: &str,
transceiver: gst_webrtc::WebRTCRTPTransceiver,
) -> Self {
let halved_framerate = video_info.fps().mul(gst::Fraction::new(1, 2));
Self {
factory_name: element.factory().unwrap().name().into(),
codec_name: codec_name.to_string(),
element,
filter,
halved_framerate,
video_info,
session_id: peer_id.to_string(),
mitigation_mode: WebRTCSinkMitigationMode::NONE,
transceiver,
}
}
pub fn bitrate(&self) -> i32 {
match self.factory_name.as_str() {
"vp8enc" | "vp9enc" => self.element.property::<i32>("target-bitrate"),
"x264enc" | "nvh264enc" | "vaapih264enc" | "vaapivp8enc" => {
(self.element.property::<u32>("bitrate") * 1000) as i32
}
"nvv4l2h264enc" | "nvv4l2vp8enc" => (self.element.property::<u32>("bitrate")) as i32,
factory => unimplemented!("Factory {} is currently not supported", factory),
}
}
pub fn scale_height_round_2(&self, height: i32) -> i32 {
let ratio = gst_video::calculate_display_ratio(
self.video_info.width(),
self.video_info.height(),
self.video_info.par(),
gst::Fraction::new(1, 1),
)
.unwrap();
let width = height.mul_div_ceil(ratio.numer(), ratio.denom()).unwrap();
(width + 1) & !1
}
pub fn set_bitrate(&mut self, element: &super::WebRTCSink, bitrate: i32) {
match self.factory_name.as_str() {
"vp8enc" | "vp9enc" => self.element.set_property("target-bitrate", bitrate),
"x264enc" | "nvh264enc" | "vaapih264enc" | "vaapivp8enc" => self
.element
.set_property("bitrate", (bitrate / 1000) as u32),
"nvv4l2h264enc" | "nvv4l2vp8enc" => {
self.element.set_property("bitrate", bitrate as u32)
}
factory => unimplemented!("Factory {} is currently not supported", factory),
}
let current_caps = self.filter.property::<gst::Caps>("caps");
let mut s = current_caps.structure(0).unwrap().to_owned();
// Hardcoded thresholds, may be tuned further in the future, and
// adapted according to the codec in use
if bitrate < 500000 {
let height = 360i32.min(self.video_info.height() as i32);
let width = self.scale_height_round_2(height);
s.set("height", height);
s.set("width", width);
if self.halved_framerate.numer() != 0 {
s.set("framerate", self.halved_framerate);
}
self.mitigation_mode =
WebRTCSinkMitigationMode::DOWNSAMPLED | WebRTCSinkMitigationMode::DOWNSCALED;
} else if bitrate < 1000000 {
let height = 360i32.min(self.video_info.height() as i32);
let width = self.scale_height_round_2(height);
s.set("height", height);
s.set("width", width);
s.remove_field("framerate");
self.mitigation_mode = WebRTCSinkMitigationMode::DOWNSCALED;
} else if bitrate < 2000000 {
let height = 720i32.min(self.video_info.height() as i32);
let width = self.scale_height_round_2(height);
s.set("height", height);
s.set("width", width);
s.remove_field("framerate");
self.mitigation_mode = WebRTCSinkMitigationMode::DOWNSCALED;
} else {
s.remove_field("height");
s.remove_field("width");
s.remove_field("framerate");
self.mitigation_mode = WebRTCSinkMitigationMode::NONE;
}
let caps = gst::Caps::builder_full_with_any_features()
.structure(s)
.build();
if !caps.is_strictly_equal(&current_caps) {
gst::log!(
CAT,
obj: element,
"session {}: setting bitrate {} and caps {} on encoder {:?}",
self.session_id,
bitrate,
caps,
self.element
);
self.filter.set_property("caps", caps);
}
}
fn gather_stats(&self) -> gst::Structure {
gst::Structure::builder("application/x-webrtcsink-video-encoder-stats")
.field("bitrate", self.bitrate())
.field("mitigation-mode", self.mitigation_mode)
.field("codec-name", self.codec_name.as_str())
.field(
"fec-percentage",
self.transceiver.property::<u32>("fec-percentage"),
)
.build()
}
}
impl State {
fn finalize_session(
&mut self,
element: &super::WebRTCSink,
session: &mut Session,
signal: bool,
) {
gst::info!(CAT, "Ending session {}", session.id);
session.pipeline.debug_to_dot_file_with_ts(
gst::DebugGraphDetails::all(),
format!("removing-peer-{}-", session.peer_id,),
);
for ssrc in session.webrtc_pads.keys() {
session.links.remove(ssrc);
}
session.pipeline.call_async(|pipeline| {
let _ = pipeline.set_state(gst::State::Null);
});
if signal {
self.signaller.session_ended(element, &session.peer_id);
}
}
fn end_session(
&mut self,
element: &super::WebRTCSink,
session_id: &str,
signal: bool,
) -> Option<Session> {
if let Some(mut session) = self.sessions.remove(session_id) {
self.finalize_session(element, &mut session, signal);
Some(session)
} else {
None
}
}
fn maybe_start_signaller(&mut self, element: &super::WebRTCSink) {
if self.signaller_state == SignallerState::Stopped
&& element.current_state() >= gst::State::Paused
&& self.codec_discovery_done
{
if let Err(err) = self.signaller.start(element) {
gst::error!(CAT, obj: element, "error: {}", err);
gst::element_error!(
element,
gst::StreamError::Failed,
["Failed to start signaller {}", err]
);
} else {
gst::info!(CAT, "Started signaller");
self.signaller_state = SignallerState::Started;
}
}
}
fn maybe_stop_signaller(&mut self, element: &super::WebRTCSink) {
if self.signaller_state == SignallerState::Started {
self.signaller.stop(element);
self.signaller_state = SignallerState::Stopped;
gst::info!(CAT, "Stopped signaller");
}
}
}
impl Session {
fn new(
id: String,
pipeline: gst::Pipeline,
webrtcbin: gst::Element,
peer_id: String,
congestion_controller: Option<CongestionController>,
rtpgccbwe: Option<gst::Element>,
cc_info: CCInfo,
) -> Self {
Self {
id,
pipeline,
webrtcbin,
peer_id,
cc_info,
rtprtxsend: None,
congestion_controller,
rtpgccbwe,
stats: gst::Structure::new_empty("application/x-webrtc-stats"),
sdp: None,
webrtc_pads: HashMap::new(),
encoders: Vec::new(),
links: HashMap::new(),
stats_sigid: None,
}
}
fn gather_stats(&self) -> gst::Structure {
let mut ret = self.stats.to_owned();
let encoder_stats: Vec<_> = self
.encoders
.iter()
.map(VideoEncoder::gather_stats)
.map(|s| s.to_send_value())
.collect();
let our_stats = gst::Structure::builder("application/x-webrtcsink-consumer-stats")
.field("video-encoders", gst::Array::from(encoder_stats))
.build();
ret.set("consumer-stats", our_stats);
ret
}
fn generate_ssrc(&self) -> u32 {
loop {
let ret = fastrand::u32(..);
if !self.webrtc_pads.contains_key(&ret) {
return ret;
}
}
}
/// Request a sink pad on our webrtcbin, and set its transceiver's codec_preferences
fn request_webrtcbin_pad(
&mut self,
element: &super::WebRTCSink,
settings: &Settings,
stream: &InputStream,
) {
let ssrc = self.generate_ssrc();
let media_idx = self.webrtc_pads.len() as i32;
let mut payloader_caps = stream.out_caps.as_ref().unwrap().to_owned();
{
let payloader_caps_mut = payloader_caps.make_mut();
payloader_caps_mut.set_simple(&[("ssrc", &ssrc)]);
}
gst::info!(
CAT,
obj: element,
"Requesting WebRTC pad for consumer {} with caps {}",
self.peer_id,
payloader_caps
);
let pad = self
.webrtcbin
.request_pad_simple(&format!("sink_{}", media_idx))
.unwrap();
let transceiver = pad.property::<gst_webrtc::WebRTCRTPTransceiver>("transceiver");
transceiver.set_property(
"direction",
gst_webrtc::WebRTCRTPTransceiverDirection::Sendonly,
);
transceiver.set_property("codec-preferences", &payloader_caps);
if stream.sink_pad.name().starts_with("video_") {
if settings.do_fec {
transceiver.set_property("fec-type", gst_webrtc::WebRTCFECType::UlpRed);
}
transceiver.set_property("do-nack", settings.do_retransmission);
}
self.webrtc_pads.insert(
ssrc,
WebRTCPad {
pad,
in_caps: stream.in_caps.as_ref().unwrap().clone(),
media_idx: media_idx as u32,
ssrc,
stream_name: stream.sink_pad.name().to_string(),
payload: None,
},
);
}
/// Called when we have received an answer, connects an InputStream
/// to a given WebRTCPad
fn connect_input_stream(
&mut self,
element: &super::WebRTCSink,
producer: &StreamProducer,
webrtc_pad: &WebRTCPad,
codecs: &BTreeMap<i32, Codec>,
) -> Result<(), Error> {
gst::info!(
CAT,
obj: element,
"Connecting input stream {} for consumer {}",
webrtc_pad.stream_name,
self.peer_id
);
let payload = webrtc_pad.payload.unwrap();
let codec = codecs
.get(&payload)
.ok_or_else(|| anyhow!("No codec for payload {}", payload))?;
let appsrc = make_element("appsrc", Some(&webrtc_pad.stream_name))?;
self.pipeline.add(&appsrc).unwrap();
let pay_filter = make_element("capsfilter", None)?;
self.pipeline.add(&pay_filter).unwrap();
let (enc, raw_filter, pay) = setup_encoding(
&self.pipeline,
&appsrc,
&webrtc_pad.in_caps,
codec,
Some(webrtc_pad.ssrc),
false,
)?;
element.emit_by_name::<bool>(
"encoder-setup",
&[&self.peer_id, &webrtc_pad.stream_name, &enc],
);
// At this point, the peer has provided its answer, and we want to
// let the payloader / encoder perform negotiation according to that.
//
// This means we need to unset our codec preferences, as they would now
// conflict with what the peer actually requested (see webrtcbin's
// caps query implementation), and instead install a capsfilter downstream
// of the payloader with caps constructed from the relevant SDP media.
let transceiver = webrtc_pad
.pad
.property::<gst_webrtc::WebRTCRTPTransceiver>("transceiver");
transceiver.set_property("codec-preferences", None::<gst::Caps>);
let mut global_caps = gst::Caps::new_simple("application/x-unknown", &[]);
let sdp = self.sdp.as_ref().unwrap();
let sdp_media = sdp.media(webrtc_pad.media_idx).unwrap();
sdp.attributes_to_caps(global_caps.get_mut().unwrap())
.unwrap();
sdp_media
.attributes_to_caps(global_caps.get_mut().unwrap())
.unwrap();
let caps = sdp_media
.caps_from_media(payload)
.unwrap()
.intersect(&global_caps);
let s = caps.structure(0).unwrap();
let mut filtered_s = gst::Structure::new_empty("application/x-rtp");
filtered_s.extend(s.iter().filter_map(|(key, value)| {
if key.starts_with("a-") {
None
} else {
Some((key, value.to_owned()))
}
}));
filtered_s.set("ssrc", webrtc_pad.ssrc);
let caps = gst::Caps::builder_full().structure(filtered_s).build();
pay_filter.set_property("caps", caps);
if codec.is_video() {
let video_info = gst_video::VideoInfo::from_caps(&webrtc_pad.in_caps)?;
let mut enc = VideoEncoder::new(
enc,
raw_filter,
video_info,
&self.peer_id,
codec.caps.structure(0).unwrap().name(),
transceiver,
);
match self.cc_info.heuristic {
WebRTCSinkCongestionControl::Disabled => {
// If congestion control is disabled, we simply use the highest
// known "safe" value for the bitrate.
enc.set_bitrate(element, self.cc_info.max_bitrate as i32);
enc.transceiver.set_property("fec-percentage", 50u32);
}
WebRTCSinkCongestionControl::Homegrown => {
if let Some(congestion_controller) = self.congestion_controller.as_mut() {
congestion_controller.target_bitrate_on_delay += enc.bitrate();
congestion_controller.target_bitrate_on_loss =
congestion_controller.target_bitrate_on_delay;
enc.transceiver.set_property("fec-percentage", 0u32);
} else {
/* If congestion control is disabled, we simply use the highest
* known "safe" value for the bitrate. */
enc.set_bitrate(element, self.cc_info.max_bitrate as i32);
enc.transceiver.set_property("fec-percentage", 50u32);
}
}
_ => enc.transceiver.set_property("fec-percentage", 0u32),
}
self.encoders.push(enc);
if let Some(rtpgccbwe) = self.rtpgccbwe.as_ref() {
let max_bitrate = self.cc_info.max_bitrate * (self.encoders.len() as u32);
rtpgccbwe.set_property("max-bitrate", max_bitrate);
}
}
let appsrc = appsrc.downcast::<gst_app::AppSrc>().unwrap();
gst_utils::StreamProducer::configure_consumer(&appsrc);
self.pipeline
.sync_children_states()
.with_context(|| format!("Connecting input stream for {}", self.peer_id))?;
pay.link(&pay_filter)?;
let srcpad = pay_filter.static_pad("src").unwrap();
srcpad
.link(&webrtc_pad.pad)
.with_context(|| format!("Connecting input stream for {}", self.peer_id))?;
match producer.add_consumer(&appsrc) {
Ok(link) => {
self.links.insert(webrtc_pad.ssrc, link);
Ok(())
}
Err(err) => Err(anyhow!("Could not link producer: {:?}", err)),
}
}
}
impl Drop for PipelineWrapper {
fn drop(&mut self) {
let _ = self.0.set_state(gst::State::Null);
}
}
impl InputStream {
/// Called when transitioning state up to Paused
fn prepare(&mut self, element: &super::WebRTCSink) -> Result<(), Error> {
let clocksync = make_element("clocksync", None)?;
let appsink = make_element("appsink", None)?
.downcast::<gst_app::AppSink>()
.unwrap();
element.add(&clocksync).unwrap();
element.add(&appsink).unwrap();
clocksync
.link(&appsink)
.with_context(|| format!("Linking input stream {}", self.sink_pad.name()))?;
element
.sync_children_states()
.with_context(|| format!("Linking input stream {}", self.sink_pad.name()))?;
self.sink_pad
.set_target(Some(&clocksync.static_pad("sink").unwrap()))
.unwrap();
let producer = StreamProducer::from(&appsink);
producer.forward();
self.producer = Some(producer);
Ok(())
}
/// Called when transitioning state back down to Ready
fn unprepare(&mut self, element: &super::WebRTCSink) {
self.sink_pad.set_target(None::<&gst::Pad>).unwrap();
if let Some(clocksync) = self.clocksync.take() {
element.remove(&clocksync).unwrap();
clocksync.set_state(gst::State::Null).unwrap();
}
if let Some(producer) = self.producer.take() {
let appsink = producer.appsink().upcast_ref::<gst::Element>();
element.remove(appsink).unwrap();
appsink.set_state(gst::State::Null).unwrap();
}
}
}
impl NavigationEventHandler {
pub fn new(element: &super::WebRTCSink, webrtcbin: &gst::Element) -> Self {
gst::info!(CAT, "Creating navigation data channel");
let channel = webrtcbin.emit_by_name::<WebRTCDataChannel>(
"create-data-channel",
&[
&"input",
&gst::Structure::new(
"config",
&[("priority", &gst_webrtc::WebRTCPriorityType::High)],
),
],
);
let weak_element = element.downgrade();
Self((
channel.connect("on-message-string", false, move |values| {
if let Some(element) = weak_element.upgrade() {
let _channel = values[0].get::<WebRTCDataChannel>().unwrap();
let msg = values[1].get::<&str>().unwrap();
create_navigation_event(&element, msg);
}
None
}),
channel,
))
}
}
impl WebRTCSink {
/// Build an ordered map of Codecs, given user-provided audio / video caps */
fn lookup_codecs(&self) -> BTreeMap<i32, Codec> {
/* First gather all encoder and payloader factories */
let encoders = gst::ElementFactory::factories_with_type(
gst::ElementFactoryType::ENCODER,
gst::Rank::Marginal,
);
let payloaders = gst::ElementFactory::factories_with_type(
gst::ElementFactoryType::PAYLOADER,
gst::Rank::Marginal,
);
/* Now iterate user-provided codec preferences and determine
* whether we can fulfill these preferences */
let settings = self.settings.lock().unwrap();
let mut payload = 96..128;
settings
.video_caps
.iter()
.chain(settings.audio_caps.iter())
.filter_map(move |s| {
let caps = gst::Caps::builder_full().structure(s.to_owned()).build();
Option::zip(
encoders
.iter()
.find(|factory| factory.can_src_any_caps(&caps)),
payloaders
.iter()
.find(|factory| factory.can_sink_any_caps(&caps)),
)
.and_then(|(encoder, payloader)| {
/* Assign a payload type to the codec */
if let Some(pt) = payload.next() {
Some(Codec {
encoder: encoder.clone(),
payloader: payloader.clone(),
caps,
payload: pt,
})
} else {
gst::warning!(
CAT,
imp: self,
"Too many formats for available payload type range, ignoring {}",
s
);
None
}
})
})
.map(|codec| (codec.payload, codec))
.collect()
}
/// Prepare for accepting consumers, by setting
/// up StreamProducers for each of our sink pads
fn prepare(&self, element: &super::WebRTCSink) -> Result<(), Error> {
gst::debug!(CAT, obj: element, "preparing");
self.state
.lock()
.unwrap()
.streams
.iter_mut()
.try_for_each(|(_, stream)| stream.prepare(element))?;
Ok(())
}
/// Unprepare by stopping consumers, then the signaller object.
/// Might abort codec discovery
fn unprepare(&self, element: &super::WebRTCSink) -> Result<(), Error> {
gst::info!(CAT, obj: element, "unpreparing");
let mut state = self.state.lock().unwrap();
let session_ids: Vec<_> = state.sessions.keys().map(|k| k.to_owned()).collect();
for id in session_ids {
state.end_session(element, &id, true);
}
state
.streams
.iter_mut()
.for_each(|(_, stream)| stream.unprepare(element));
if let Some(handle) = state.codecs_abort_handle.take() {
handle.abort();
}
if let Some(receiver) = state.codecs_done_receiver.take() {
task::block_on(async {
let _ = receiver.await;
});
}
state.maybe_stop_signaller(element);
state.codec_discovery_done = false;
state.codecs = BTreeMap::new();
Ok(())
}
/// When using a custom signaller
pub fn set_signaller(&self, signaller: Box<dyn super::SignallableObject>) -> Result<(), Error> {
let mut state = self.state.lock().unwrap();
state.signaller = signaller;
Ok(())
}
/// Called by the signaller when it has encountered an error
pub fn handle_signalling_error(&self, element: &super::WebRTCSink, error: anyhow::Error) {
gst::error!(CAT, obj: element, "Signalling error: {:?}", error);
gst::element_error!(
element,
gst::StreamError::Failed,
["Signalling error: {:?}", error]
);
}
fn on_offer_created(
&self,
element: &super::WebRTCSink,
offer: gst_webrtc::WebRTCSessionDescription,
session_id: &str,
) {
let mut state = self.state.lock().unwrap();
if let Some(session) = state.sessions.get(session_id) {
session
.webrtcbin
.emit_by_name::<()>("set-local-description", &[&offer, &None::<gst::Promise>]);
if let Err(err) = state.signaller.handle_sdp(element, session_id, &offer) {
gst::warning!(
CAT,
"Failed to handle SDP for session {}: {}",
session_id,
err
);
state.end_session(element, session_id, true);
}
}
}
fn negotiate(&self, element: &super::WebRTCSink, session_id: &str) {
let state = self.state.lock().unwrap();
gst::debug!(CAT, obj: element, "Negotiating for session {}", session_id);
if let Some(session) = state.sessions.get(session_id) {
let element = element.downgrade();
gst::debug!(CAT, "Creating offer for session {}", session_id);
let session_id = session_id.to_string();
let promise = gst::Promise::with_change_func(move |reply| {
gst::debug!(CAT, "Created offer for session {}", session_id);
if let Some(element) = element.upgrade() {
let this = element.imp();
let reply = match reply {
Ok(Some(reply)) => reply,
Ok(None) => {
gst::warning!(
CAT,
obj: element,
"Promise returned without a reply for {}",
session_id
);
let _ = this.remove_session(&element, &session_id, true);
return;
}
Err(err) => {
gst::warning!(
CAT,
obj: element,
"Promise returned with an error for {}: {:?}",
session_id,
err
);
let _ = this.remove_session(&element, &session_id, true);
return;
}
};
if let Ok(offer) = reply
.value("offer")
.map(|offer| offer.get::<gst_webrtc::WebRTCSessionDescription>().unwrap())
{
this.on_offer_created(&element, offer, &session_id);
} else {
gst::warning!(
CAT,
"Reply without an offer for session {}: {:?}",
session_id,
reply
);
let _ = this.remove_session(&element, &session_id, true);
}
}
});
session
.webrtcbin
.emit_by_name::<()>("create-offer", &[&None::<gst::Structure>, &promise]);
} else {
gst::debug!(
CAT,
obj: element,
"consumer for session {} no longer exists (sessions: {:?}",
session_id,
state.sessions.keys()
);
}
}
fn on_ice_candidate(
&self,
element: &super::WebRTCSink,
session_id: String,
sdp_m_line_index: u32,
candidate: String,
) {
let mut state = self.state.lock().unwrap();
if let Err(err) = state.signaller.handle_ice(
element,
&session_id,
&candidate,
Some(sdp_m_line_index),
None,
) {
gst::warning!(
CAT,
"Failed to handle ICE in session {}: {}",
session_id,
err
);
state.end_session(element, &session_id, true);
}
}
/// Called by the signaller to add a new session
pub fn start_session(
&self,
element: &super::WebRTCSink,
session_id: &str,
peer_id: &str,
) -> Result<(), WebRTCSinkError> {
let settings = self.settings.lock().unwrap();
let mut state = self.state.lock().unwrap();
let peer_id = peer_id.to_string();
let session_id = session_id.to_string();
if state.sessions.contains_key(&session_id) {
return Err(WebRTCSinkError::DuplicateSessionId(session_id));
}
gst::info!(
CAT,
obj: element,
"Adding session: {} for peer: {}",
peer_id,
session_id
);
let pipeline = gst::Pipeline::builder()
.name(&format!("session-pipeline-{}", session_id))
.build();
let webrtcbin = make_element("webrtcbin", Some(&format!("webrtcbin-{}", session_id)))
.map_err(|err| WebRTCSinkError::SessionPipelineError {
session_id: session_id.clone(),
peer_id: peer_id.clone(),
details: err.to_string(),
})?;
webrtcbin.set_property_from_str("bundle-policy", "max-bundle");
if let Some(stun_server) = settings.stun_server.as_ref() {
webrtcbin.set_property("stun-server", stun_server);
}
for turn_server in settings.turn_servers.iter() {
webrtcbin.emit_by_name::<bool>("add-turn-server", &[&turn_server]);
}
let rtpgccbwe = match settings.cc_info.heuristic {
WebRTCSinkCongestionControl::GoogleCongestionControl => {
let rtpgccbwe = match gst::ElementFactory::make("rtpgccbwe").build() {
Err(err) => {
glib::g_warning!(
"webrtcsink",
"The `rtpgccbwe` element is not available \
not doing any congestion control: {err:?}"
);
None
}
Ok(cc) => {
webrtcbin.connect_closure(
"request-aux-sender",
false,
glib::closure!(@watch element, @strong session_id, @weak-allow-none cc
=> move |_webrtcbin: gst::Element, _transport: gst::Object| {
let cc = cc.unwrap();
let settings = element.imp().settings.lock().unwrap();
// TODO: Bind properties with @element's
cc.set_properties(&[
("min-bitrate", &settings.cc_info.min_bitrate),
("estimated-bitrate", &settings.cc_info.start_bitrate),
("max-bitrate", &settings.cc_info.max_bitrate),
]);
cc.connect_notify(Some("estimated-bitrate"),
glib::clone!(@weak element, @strong session_id
=> move |bwe, pspec| {
element.imp().set_bitrate(&element, &session_id,
bwe.property::<u32>(pspec.name()));
}
));
Some(cc)
}),
);
Some(cc)
}
};
webrtcbin.connect_closure(
"deep-element-added",
false,
glib::closure!(@watch element, @strong session_id
=> move |_webrtcbin: gst::Element, _bin: gst::Bin, e: gst::Element| {
if e.factory().map_or(false, |f| f.name() == "rtprtxsend") {
if e.has_property("stuffing-kbps", Some(i32::static_type())) {
element.imp().set_rtptrxsend(element, &session_id, e);
} else {
gst::warning!(CAT, "rtprtxsend doesn't have a `stuffing-kbps` \
property, stuffing disabled");
}
}
}),
);
rtpgccbwe
}
_ => None,
};
pipeline.add(&webrtcbin).unwrap();
let element_clone = element.downgrade();
let session_id_clone = session_id.clone();
webrtcbin.connect("on-ice-candidate", false, move |values| {
if let Some(element) = element_clone.upgrade() {
let this = element.imp();
let sdp_m_line_index = values[1].get::<u32>().expect("Invalid argument");
let candidate = values[2].get::<String>().expect("Invalid argument");
this.on_ice_candidate(
&element,
session_id_clone.to_string(),
sdp_m_line_index,
candidate,
);
}
None
});
let element_clone = element.downgrade();
let peer_id_clone = peer_id.clone();
let session_id_clone = session_id.clone();
webrtcbin.connect_notify(Some("connection-state"), move |webrtcbin, _pspec| {
if let Some(element) = element_clone.upgrade() {
let state =
webrtcbin.property::<gst_webrtc::WebRTCPeerConnectionState>("connection-state");
match state {
gst_webrtc::WebRTCPeerConnectionState::Failed => {
let this = element.imp();
gst::warning!(
CAT,
obj: element,
"Connection state for in session {} (peer {}) failed",
session_id_clone,
peer_id_clone
);
let _ = this.remove_session(&element, &session_id_clone, true);
}
_ => {
gst::log!(
CAT,
obj: element,
"Connection state in session {} (peer {}) changed: {:?}",
session_id_clone,
peer_id_clone,
state
);
}
}
}
});
let element_clone = element.downgrade();
let peer_id_clone = peer_id.clone();
let session_id_clone = session_id.clone();
webrtcbin.connect_notify(Some("ice-connection-state"), move |webrtcbin, _pspec| {
if let Some(element) = element_clone.upgrade() {
let state = webrtcbin
.property::<gst_webrtc::WebRTCICEConnectionState>("ice-connection-state");
let this = element.imp();
match state {
gst_webrtc::WebRTCICEConnectionState::Failed => {
gst::warning!(
CAT,
obj: element,
"Ice connection state in session {} (peer {}) failed",
session_id_clone,
peer_id_clone,
);
let _ = this.remove_session(&element, &session_id_clone, true);
}
_ => {
gst::log!(
CAT,
obj: element,
"Ice connection state in session {} (peer {}) changed: {:?}",
session_id_clone,
peer_id_clone,
state
);
}
}
if state == gst_webrtc::WebRTCICEConnectionState::Completed {
let state = this.state.lock().unwrap();
if let Some(session) = state.sessions.get(&session_id_clone) {
for webrtc_pad in session.webrtc_pads.values() {
if let Some(srcpad) = webrtc_pad.pad.peer() {
srcpad.send_event(
gst_video::UpstreamForceKeyUnitEvent::builder()
.all_headers(true)
.build(),
);
}
}
}
}
}
});
let element_clone = element.downgrade();
let peer_id_clone = peer_id.clone();
let session_id_clone = session_id.clone();
webrtcbin.connect_notify(Some("ice-gathering-state"), move |webrtcbin, _pspec| {
let state =
webrtcbin.property::<gst_webrtc::WebRTCICEGatheringState>("ice-gathering-state");
if let Some(element) = element_clone.upgrade() {
gst::log!(
CAT,
obj: element,
"Ice gathering state in session {} (peer {}) changed: {:?}",
session_id_clone,
peer_id_clone,
state
);
}
});
let mut session = Session::new(
session_id.clone(),
pipeline.clone(),
webrtcbin.clone(),
peer_id.clone(),
match settings.cc_info.heuristic {
WebRTCSinkCongestionControl::Homegrown => Some(CongestionController::new(
&peer_id,
settings.cc_info.min_bitrate,
settings.cc_info.max_bitrate,
)),
_ => None,
},
rtpgccbwe,
settings.cc_info,
);
let rtpbin = webrtcbin
.dynamic_cast_ref::<gst::ChildProxy>()
.unwrap()
.child_by_name("rtpbin")
.unwrap();
if session.congestion_controller.is_some() {
let session_id_str = session_id.to_string();
if session.stats_sigid.is_none() {
session.stats_sigid = Some(rtpbin.connect_closure("on-new-ssrc", true,
glib::closure!(@weak-allow-none element, @weak-allow-none webrtcbin
=> move |rtpbin: gst::Object, session_id: u32, _src: u32| {
let rtp_session = rtpbin.emit_by_name::<gst::Element>("get-session", &[&session_id]);
let element = element.expect("on-new-ssrc emited when webrtcsink has been disposed?");
let webrtcbin = webrtcbin.unwrap();
let mut state = element.imp().state.lock().unwrap();
if let Some(mut session) = state.sessions.get_mut(&session_id_str) {
session.stats_sigid = Some(rtp_session.connect_notify(Some("twcc-stats"),
glib::clone!(@strong session_id_str, @weak webrtcbin, @weak element => @default-panic, move |sess, pspec| {
// Run the Loss-based control algorithm on new peer TWCC feedbacks
element.imp().process_loss_stats(&element, &session_id_str, &sess.property::<gst::Structure>(pspec.name()));
})
));
}
})
));
}
}
state
.streams
.iter()
.for_each(|(_, stream)| session.request_webrtcbin_pad(element, &settings, stream));
let clock = element.clock();
pipeline.use_clock(clock.as_ref());
pipeline.set_start_time(gst::ClockTime::NONE);
pipeline.set_base_time(element.base_time().unwrap());
let mut bus_stream = pipeline.bus().unwrap().stream();
let element_clone = element.downgrade();
let pipeline_clone = pipeline.downgrade();
let session_id_clone = session_id.to_owned();
task::spawn(async move {
while let Some(msg) = bus_stream.next().await {
if let Some(element) = element_clone.upgrade() {
let this = element.imp();
match msg.view() {
gst::MessageView::Error(err) => {
gst::error!(
CAT,
"session {} error: {}, details: {:?}",
session_id_clone,
err.error(),
err.debug()
);
let _ = this.remove_session(&element, &session_id_clone, true);
}
gst::MessageView::StateChanged(state_changed) => {
if let Some(pipeline) = pipeline_clone.upgrade() {
if Some(pipeline.clone().upcast()) == state_changed.src() {
pipeline.debug_to_dot_file_with_ts(
gst::DebugGraphDetails::all(),
format!(
"webrtcsink-session-{}-{:?}-to-{:?}",
session_id_clone,
state_changed.old(),
state_changed.current()
),
);
}
}
}
gst::MessageView::Latency(..) => {
if let Some(pipeline) = pipeline_clone.upgrade() {
gst::info!(CAT, obj: pipeline, "Recalculating latency");
let _ = pipeline.recalculate_latency();
}
}
gst::MessageView::Eos(..) => {
gst::error!(
CAT,
"Unexpected end of stream in session {}",
session_id_clone,
);
let _ = this.remove_session(&element, &session_id_clone, true);
}
_ => (),
}
}
}
});
pipeline.set_state(gst::State::Ready).map_err(|err| {
WebRTCSinkError::SessionPipelineError {
session_id: session_id.to_string(),
peer_id: peer_id.to_string(),
details: err.to_string(),
}
})?;
if settings.enable_data_channel_navigation {
state.navigation_handler = Some(NavigationEventHandler::new(element, &webrtcbin));
}
state.sessions.insert(session_id.to_string(), session);
drop(state);
drop(settings);
// This is intentionally emitted with the pipeline in the Ready state,
// so that application code can create data channels at the correct
// moment.
element.emit_by_name::<()>("consumer-added", &[&peer_id, &webrtcbin]);
// We don't connect to on-negotiation-needed, this in order to call the above
// signal without holding the state lock:
//
// Going to Ready triggers synchronous emission of the on-negotiation-needed
// signal, during which time the application may add a data channel, causing
// renegotiation, which we do not support at this time.
//
// This is completely safe, as we know that by now all conditions are gathered:
// webrtcbin is in the Ready state, and all its transceivers have codec_preferences.
self.negotiate(element, &session_id);
pipeline.set_state(gst::State::Playing).map_err(|err| {
WebRTCSinkError::SessionPipelineError {
session_id: session_id.to_string(),
peer_id: peer_id.to_string(),
details: err.to_string(),
}
})?;
Ok(())
}
/// Called by the signaller to remove a consumer
pub fn remove_session(
&self,
element: &super::WebRTCSink,
session_id: &str,
signal: bool,
) -> Result<(), WebRTCSinkError> {
let mut state = self.state.lock().unwrap();
if !state.sessions.contains_key(session_id) {
return Err(WebRTCSinkError::NoSessionWithId(session_id.to_string()));
}
if let Some(session) = state.end_session(element, session_id, signal) {
drop(state);
element.emit_by_name::<()>("consumer-removed", &[&session.peer_id, &session.webrtcbin]);
}
Ok(())
}
fn process_loss_stats(
&self,
element: &super::WebRTCSink,
session_id: &str,
stats: &gst::Structure,
) {
let mut state = element.imp().state.lock().unwrap();
if let Some(mut session) = state.sessions.get_mut(session_id) {
if let Some(congestion_controller) = session.congestion_controller.as_mut() {
congestion_controller.loss_control(element, stats, &mut session.encoders);
}
session.stats = stats.to_owned();
}
}
fn process_stats(
&self,
element: &super::WebRTCSink,
webrtcbin: gst::Element,
session_id: &str,
) {
let session_id = session_id.to_string();
let promise = gst::Promise::with_change_func(
glib::clone!(@strong session_id, @weak element => move |reply| {
if let Ok(Some(stats)) = reply {
let mut state = element.imp().state.lock().unwrap();
if let Some(mut session) = state.sessions.get_mut(&session_id) {
if let Some(congestion_controller) = session.congestion_controller.as_mut() {
congestion_controller.delay_control(&element, stats, &mut session.encoders,);
}
session.stats = stats.to_owned();
}
}
}),
);
webrtcbin.emit_by_name::<()>("get-stats", &[&None::<gst::Pad>, &promise]);
}
fn set_rtptrxsend(&self, element: &super::WebRTCSink, peer_id: &str, rtprtxsend: gst::Element) {
let mut state = element.imp().state.lock().unwrap();
if let Some(session) = state.sessions.get_mut(peer_id) {
session.rtprtxsend = Some(rtprtxsend);
}
}
fn set_bitrate(&self, element: &super::WebRTCSink, peer_id: &str, bitrate: u32) {
let settings = element.imp().settings.lock().unwrap();
let mut state = element.imp().state.lock().unwrap();
if let Some(session) = state.sessions.get_mut(peer_id) {
let fec_ratio = {
if settings.do_fec && bitrate > DO_FEC_THRESHOLD {
(bitrate as f64 - DO_FEC_THRESHOLD as f64)
/ (session.cc_info.max_bitrate as f64 - DO_FEC_THRESHOLD as f64)
} else {
0f64
}
};
let fec_percentage = fec_ratio * 50f64;
let encoders_bitrate = ((bitrate as f64)
/ (1. + (fec_percentage / 100.))
/ (session.encoders.len() as f64)) as i32;
if let Some(rtpxsend) = session.rtprtxsend.as_ref() {
rtpxsend.set_property("stuffing-kbps", (bitrate as f64 / 1000.) as i32);
}
for encoder in session.encoders.iter_mut() {
encoder.set_bitrate(element, encoders_bitrate);
encoder
.transceiver
.set_property("fec-percentage", fec_percentage as u32);
}
}
}
fn on_remote_description_set(&self, element: &super::WebRTCSink, session_id: String) {
let mut state = self.state.lock().unwrap();
let mut remove = false;
let codecs = state.codecs.clone();
if let Some(mut session) = state.sessions.remove(&session_id) {
for webrtc_pad in session.webrtc_pads.clone().values() {
let transceiver = webrtc_pad
.pad
.property::<gst_webrtc::WebRTCRTPTransceiver>("transceiver");
if let Some(mid) = transceiver.mid() {
state
.mids
.insert(mid.to_string(), webrtc_pad.stream_name.clone());
}
if let Some(producer) = state
.streams
.get(&webrtc_pad.stream_name)
.and_then(|stream| stream.producer.clone())
{
drop(state);
if let Err(err) =
session.connect_input_stream(element, &producer, webrtc_pad, &codecs)
{
gst::error!(
CAT,
obj: element,
"Failed to connect input stream {} for session {}: {}",
webrtc_pad.stream_name,
session_id,
err
);
remove = true;
state = self.state.lock().unwrap();
break;
}
state = self.state.lock().unwrap();
} else {
gst::error!(
CAT,
obj: element,
"No producer to connect session {} to",
session_id,
);
remove = true;
break;
}
}
session.pipeline.debug_to_dot_file_with_ts(
gst::DebugGraphDetails::all(),
format!("webrtcsink-peer-{}-remote-description-set", session_id,),
);
let element_clone = element.downgrade();
let webrtcbin = session.webrtcbin.downgrade();
task::spawn(async move {
let mut interval =
async_std::stream::interval(std::time::Duration::from_millis(100));
while interval.next().await.is_some() {
let element_clone = element_clone.clone();
if let (Some(webrtcbin), Some(element)) =
(webrtcbin.upgrade(), element_clone.upgrade())
{
element
.imp()
.process_stats(&element, webrtcbin, &session_id);
} else {
break;
}
}
});
if remove {
state.finalize_session(element, &mut session, true);
} else {
state.sessions.insert(session.id.clone(), session);
}
}
}
/// Called by the signaller with an ice candidate
pub fn handle_ice(
&self,
_element: &super::WebRTCSink,
session_id: &str,
sdp_m_line_index: Option<u32>,
_sdp_mid: Option<String>,
candidate: &str,
) -> Result<(), WebRTCSinkError> {
let state = self.state.lock().unwrap();
let sdp_m_line_index = sdp_m_line_index.ok_or(WebRTCSinkError::MandatorySdpMlineIndex)?;
if let Some(session) = state.sessions.get(session_id) {
gst::trace!(CAT, "adding ice candidate for session {}", session_id);
session
.webrtcbin
.emit_by_name::<()>("add-ice-candidate", &[&sdp_m_line_index, &candidate]);
Ok(())
} else {
Err(WebRTCSinkError::NoSessionWithId(session_id.to_string()))
}
}
/// Called by the signaller with an answer to our offer
pub fn handle_sdp(
&self,
element: &super::WebRTCSink,
session_id: &str,
desc: &gst_webrtc::WebRTCSessionDescription,
) -> Result<(), WebRTCSinkError> {
let mut state = self.state.lock().unwrap();
if let Some(session) = state.sessions.get_mut(session_id) {
let sdp = desc.sdp();
session.sdp = Some(sdp.to_owned());
for webrtc_pad in session.webrtc_pads.values_mut() {
let media_idx = webrtc_pad.media_idx;
/* TODO: support partial answer, webrtcbin doesn't seem
* very well equipped to deal with this at the moment */
if let Some(media) = sdp.media(media_idx) {
if media.attribute_val("inactive").is_some() {
let media_str = sdp
.media(webrtc_pad.media_idx)
.and_then(|media| media.as_text().ok());
gst::warning!(
CAT,
"consumer from session {} refused media {}: {:?}",
session_id,
media_idx,
media_str
);
state.end_session(element, session_id, true);
return Err(WebRTCSinkError::ConsumerRefusedMedia {
session_id: session_id.to_string(),
media_idx,
});
}
}
if let Some(payload) = sdp
.media(webrtc_pad.media_idx)
.and_then(|media| media.format(0))
.and_then(|format| format.parse::<i32>().ok())
{
webrtc_pad.payload = Some(payload);
} else {
gst::warning!(
CAT,
"consumer from session {} did not provide valid payload for media index {} for session {}",
session_id,
media_idx,
session_id,
);
state.end_session(element, session_id, true);
return Err(WebRTCSinkError::ConsumerNoValidPayload {
session_id: session_id.to_string(),
media_idx,
});
}
}
let element = element.downgrade();
let session_id = session_id.to_string();
let promise = gst::Promise::with_change_func(move |reply| {
gst::debug!(CAT, "received reply {:?}", reply);
if let Some(element) = element.upgrade() {
let this = element.imp();
this.on_remote_description_set(&element, session_id);
}
});
session
.webrtcbin
.emit_by_name::<()>("set-remote-description", &[desc, &promise]);
Ok(())
} else {
Err(WebRTCSinkError::NoSessionWithId(session_id.to_string()))
}
}
async fn run_discovery_pipeline(
element: &super::WebRTCSink,
codec: &Codec,
caps: &gst::Caps,
) -> Result<gst::Structure, Error> {
let pipe = PipelineWrapper(gst::Pipeline::default());
let src = if codec.is_video() {
make_element("videotestsrc", None)?
} else {
make_element("audiotestsrc", None)?
};
let mut elements = vec![src.clone()];
if codec.is_video() {
elements.push(make_converter_for_video_caps(caps)?);
}
gst::debug!(
CAT,
obj: element,
"Running discovery pipeline for caps {caps} with codec {codec:?}"
);
let capsfilter = make_element("capsfilter", None)?;
elements.push(capsfilter.clone());
let elements_slice = &elements.iter().collect::<Vec<_>>();
pipe.0.add_many(elements_slice).unwrap();
gst::Element::link_many(elements_slice)
.with_context(|| format!("Running discovery pipeline for caps {}", caps))?;
let (_, _, pay) = setup_encoding(&pipe.0, &capsfilter, caps, codec, None, true)?;
let sink = make_element("fakesink", None)?;
pipe.0.add(&sink).unwrap();
pay.link(&sink)
.with_context(|| format!("Running discovery pipeline for caps {}", caps))?;
capsfilter.set_property("caps", caps);
src.set_property("num-buffers", 1);
let mut stream = pipe.0.bus().unwrap().stream();
pipe.0
.set_state(gst::State::Playing)
.with_context(|| format!("Running discovery pipeline for caps {}", caps))?;
let in_caps = caps;
while let Some(msg) = stream.next().await {
match msg.view() {
gst::MessageView::Error(err) => {
pipe.0.debug_to_dot_file_with_ts(
gst::DebugGraphDetails::all(),
"webrtcsink-discovery-error",
);
return Err(err.error().into());
}
gst::MessageView::Eos(_) => {
let caps = pay.static_pad("src").unwrap().current_caps().unwrap();
pipe.0.debug_to_dot_file_with_ts(
gst::DebugGraphDetails::all(),
"webrtcsink-discovery-done",
);
if let Some(s) = caps.structure(0) {
let mut s = s.to_owned();
s.remove_fields(&[
"timestamp-offset",
"seqnum-offset",
"ssrc",
"sprop-parameter-sets",
"a-framerate",
]);
s.set("payload", codec.payload);
gst::debug!(
CAT,
obj: element,
"Codec discovery pipeline for caps {in_caps} with codec {codec:?} succeeded: {s}"
);
return Ok(s);
} else {
return Err(anyhow!("Discovered empty caps"));
}
}
_ => {
continue;
}
}
}
unreachable!()
}
async fn lookup_caps(
element: &super::WebRTCSink,
name: String,
in_caps: gst::Caps,
codecs: &BTreeMap<i32, Codec>,
) -> (String, gst::Caps) {
let sink_caps = in_caps.as_ref().to_owned();
let is_video = match sink_caps.structure(0).unwrap().name() {
"video/x-raw" => true,
"audio/x-raw" => false,
_ => unreachable!(),
};
let mut payloader_caps = gst::Caps::new_empty();
let payloader_caps_mut = payloader_caps.make_mut();
let futs = codecs
.iter()
.filter(|(_, codec)| codec.is_video() == is_video)
.map(|(_, codec)| WebRTCSink::run_discovery_pipeline(element, codec, &sink_caps));
for ret in futures::future::join_all(futs).await {
match ret {
Ok(s) => {
payloader_caps_mut.append_structure(s);
}
Err(err) => {
/* We don't consider this fatal, as long as we end up with one
* potential codec for each input stream
*/
gst::warning!(
CAT,
obj: element,
"Codec discovery pipeline failed: {}",
err
);
}
}
}
(name, payloader_caps)
}
async fn lookup_streams_caps(&self, element: &super::WebRTCSink) -> Result<(), Error> {
let codecs = self.lookup_codecs();
gst::debug!(CAT, obj: element, "Looked up codecs {codecs:?}");
let futs: Vec<_> = self
.state
.lock()
.unwrap()
.streams
.iter()
.map(|(name, stream)| {
WebRTCSink::lookup_caps(
element,
name.to_owned(),
stream.in_caps.as_ref().unwrap().to_owned(),
&codecs,
)
})
.collect();
let caps: Vec<(String, gst::Caps)> = futures::future::join_all(futs).await;
let mut state = self.state.lock().unwrap();
for (name, caps) in caps {
if caps.is_empty() {
return Err(anyhow!("No caps found for stream {}", name));
}
if let Some(mut stream) = state.streams.get_mut(&name) {
stream.out_caps = Some(caps);
}
}
state.codecs = codecs;
Ok(())
}
fn gather_stats(&self) -> gst::Structure {
gst::Structure::from_iter(
"application/x-webrtcsink-stats",
self.state
.lock()
.unwrap()
.sessions
.iter()
.map(|(name, consumer)| (name.as_str(), consumer.gather_stats().to_send_value())),
)
}
fn sink_event(&self, pad: &gst::Pad, element: &super::WebRTCSink, event: gst::Event) -> bool {
use gst::EventView;
match event.view() {
EventView::Caps(e) => {
if let Some(caps) = pad.current_caps() {
if caps.is_strictly_equal(e.caps()) {
// Nothing changed
true
} else {
gst::error!(CAT, obj: pad, "Renegotiation is not supported");
false
}
} else {
gst::info!(CAT, obj: pad, "Received caps event {:?}", e);
let mut all_pads_have_caps = true;
self.state
.lock()
.unwrap()
.streams
.iter_mut()
.for_each(|(_, mut stream)| {
if stream.sink_pad.upcast_ref::<gst::Pad>() == pad {
stream.in_caps = Some(e.caps().to_owned());
} else if stream.in_caps.is_none() {
all_pads_have_caps = false;
}
});
if all_pads_have_caps {
let element_clone = element.downgrade();
task::spawn(async move {
if let Some(element) = element_clone.upgrade() {
let this = element.imp();
let (fut, handle) =
futures::future::abortable(this.lookup_streams_caps(&element));
let (codecs_done_sender, codecs_done_receiver) =
futures::channel::oneshot::channel();
// Compiler isn't budged by dropping state before await,
// so let's make a new scope instead.
{
let mut state = this.state.lock().unwrap();
state.codecs_abort_handle = Some(handle);
state.codecs_done_receiver = Some(codecs_done_receiver);
}
match fut.await {
Ok(Err(err)) => {
gst::error!(CAT, obj: element, "error: {}", err);
gst::element_error!(
element,
gst::StreamError::CodecNotFound,
["Failed to look up output caps: {}", err]
);
}
Ok(Ok(_)) => {
let mut state = this.state.lock().unwrap();
state.codec_discovery_done = true;
state.maybe_start_signaller(&element);
}
_ => (),
}
let _ = codecs_done_sender.send(());
}
});
}
gst::Pad::event_default(pad, Some(element), event)
}
}
_ => gst::Pad::event_default(pad, Some(element), event),
}
}
}
#[glib::object_subclass]
impl ObjectSubclass for WebRTCSink {
const NAME: &'static str = "GstWebRTCSink";
type Type = super::WebRTCSink;
type ParentType = gst::Bin;
type Interfaces = (gst::ChildProxy, gst_video::Navigation);
}
impl ObjectImpl for WebRTCSink {
fn properties() -> &'static [glib::ParamSpec] {
static PROPERTIES: Lazy<Vec<glib::ParamSpec>> = Lazy::new(|| {
vec![
glib::ParamSpecBoxed::new(
"video-caps",
"Video encoder caps",
"Governs what video codecs will be proposed",
gst::Caps::static_type(),
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
),
glib::ParamSpecBoxed::new(
"audio-caps",
"Audio encoder caps",
"Governs what audio codecs will be proposed",
gst::Caps::static_type(),
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
),
glib::ParamSpecString::new(
"stun-server",
"STUN Server",
"The STUN server of the form stun://hostname:port",
DEFAULT_STUN_SERVER,
glib::ParamFlags::READWRITE,
),
gst::ParamSpecArray::new(
"turn-servers",
"List of TURN Servers to user",
"The TURN servers of the form <\"turn(s)://username:password@host:port\", \"turn(s)://username1:password1@host1:port1\">",
Some(&glib::ParamSpecString::new(
"turn-server",
"TURN Server",
"The TURN server of the form turn(s)://username:password@host:port.",
None,
glib::ParamFlags::READWRITE,
)),
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
),
glib::ParamSpecEnum::new(
"congestion-control",
"Congestion control",
"Defines how congestion is controlled, if at all",
WebRTCSinkCongestionControl::static_type(),
DEFAULT_CONGESTION_CONTROL as i32,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
),
glib::ParamSpecUInt::new(
"min-bitrate",
"Minimal Bitrate",
"Minimal bitrate to use (in bit/sec) when computing it through the congestion control algorithm",
1,
u32::MAX as u32,
DEFAULT_MIN_BITRATE,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
),
glib::ParamSpecUInt::new(
"max-bitrate",
"Minimal Bitrate",
"Minimal bitrate to use (in bit/sec) when computing it through the congestion control algorithm",
1,
u32::MAX as u32,
DEFAULT_MAX_BITRATE,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
),
glib::ParamSpecUInt::new(
"start-bitrate",
"Start Bitrate",
"Start bitrate to use (in bit/sec)",
1,
u32::MAX as u32,
DEFAULT_START_BITRATE,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
),
glib::ParamSpecBoxed::new(
"stats",
"Consumer statistics",
"Statistics for the current consumers",
gst::Structure::static_type(),
glib::ParamFlags::READABLE,
),
glib::ParamSpecBoolean::new(
"do-fec",
"Do Forward Error Correction",
"Whether the element should negotiate and send FEC data",
DEFAULT_DO_FEC,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY
),
glib::ParamSpecBoolean::new(
"do-retransmission",
"Do retransmission",
"Whether the element should offer to honor retransmission requests",
DEFAULT_DO_RETRANSMISSION,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY
),
glib::ParamSpecBoolean::new(
"enable-data-channel-navigation",
"Enable data channel navigation",
"Enable navigation events through a dedicated WebRTCDataChannel",
DEFAULT_ENABLE_DATA_CHANNEL_NAVIGATION,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY
),
glib::ParamSpecBoxed::new(
"meta",
"Meta",
"Free form metadata about the producer",
gst::Structure::static_type(),
glib::ParamFlags::READWRITE,
),
]
});
PROPERTIES.as_ref()
}
fn set_property(&self, _id: usize, value: &glib::Value, pspec: &glib::ParamSpec) {
match pspec.name() {
"video-caps" => {
let mut settings = self.settings.lock().unwrap();
settings.video_caps = value
.get::<Option<gst::Caps>>()
.expect("type checked upstream")
.unwrap_or_else(gst::Caps::new_empty);
}
"audio-caps" => {
let mut settings = self.settings.lock().unwrap();
settings.audio_caps = value
.get::<Option<gst::Caps>>()
.expect("type checked upstream")
.unwrap_or_else(gst::Caps::new_empty);
}
"stun-server" => {
let mut settings = self.settings.lock().unwrap();
settings.stun_server = value
.get::<Option<String>>()
.expect("type checked upstream")
}
"turn-servers" => {
let mut settings = self.settings.lock().unwrap();
settings.turn_servers = value.get::<gst::Array>().expect("type checked upstream")
}
"congestion-control" => {
let mut settings = self.settings.lock().unwrap();
settings.cc_info.heuristic = value
.get::<WebRTCSinkCongestionControl>()
.expect("type checked upstream");
}
"min-bitrate" => {
let mut settings = self.settings.lock().unwrap();
settings.cc_info.min_bitrate = value.get::<u32>().expect("type checked upstream");
}
"max-bitrate" => {
let mut settings = self.settings.lock().unwrap();
settings.cc_info.max_bitrate = value.get::<u32>().expect("type checked upstream");
}
"start-bitrate" => {
let mut settings = self.settings.lock().unwrap();
settings.cc_info.start_bitrate = value.get::<u32>().expect("type checked upstream");
}
"do-fec" => {
let mut settings = self.settings.lock().unwrap();
settings.do_fec = value.get::<bool>().expect("type checked upstream");
}
"do-retransmission" => {
let mut settings = self.settings.lock().unwrap();
settings.do_retransmission = value.get::<bool>().expect("type checked upstream");
}
"enable-data-channel-navigation" => {
let mut settings = self.settings.lock().unwrap();
settings.enable_data_channel_navigation =
value.get::<bool>().expect("type checked upstream");
}
"meta" => {
let mut settings = self.settings.lock().unwrap();
settings.meta = value
.get::<Option<gst::Structure>>()
.expect("type checked upstream")
}
_ => unimplemented!(),
}
}
fn property(&self, _id: usize, pspec: &glib::ParamSpec) -> glib::Value {
match pspec.name() {
"video-caps" => {
let settings = self.settings.lock().unwrap();
settings.video_caps.to_value()
}
"audio-caps" => {
let settings = self.settings.lock().unwrap();
settings.audio_caps.to_value()
}
"congestion-control" => {
let settings = self.settings.lock().unwrap();
settings.cc_info.heuristic.to_value()
}
"stun-server" => {
let settings = self.settings.lock().unwrap();
settings.stun_server.to_value()
}
"turn-servers" => {
let settings = self.settings.lock().unwrap();
settings.turn_servers.to_value()
}
"min-bitrate" => {
let settings = self.settings.lock().unwrap();
settings.cc_info.min_bitrate.to_value()
}
"max-bitrate" => {
let settings = self.settings.lock().unwrap();
settings.cc_info.max_bitrate.to_value()
}
"start-bitrate" => {
let settings = self.settings.lock().unwrap();
settings.cc_info.start_bitrate.to_value()
}
"do-fec" => {
let settings = self.settings.lock().unwrap();
settings.do_fec.to_value()
}
"do-retransmission" => {
let settings = self.settings.lock().unwrap();
settings.do_retransmission.to_value()
}
"enable-data-channel-navigation" => {
let settings = self.settings.lock().unwrap();
settings.enable_data_channel_navigation.to_value()
}
"stats" => self.gather_stats().to_value(),
"meta" => {
let settings = self.settings.lock().unwrap();
settings.meta.to_value()
}
_ => unimplemented!(),
}
}
fn signals() -> &'static [glib::subclass::Signal] {
static SIGNALS: Lazy<Vec<glib::subclass::Signal>> = Lazy::new(|| {
vec![
/**
* RsWebRTCSink::consumer-added:
* @consumer_id: Identifier of the consumer added
* @webrtcbin: The new webrtcbin
*
* This signal can be used to tweak @webrtcbin, creating a data
* channel for example.
*/
glib::subclass::Signal::builder("consumer-added")
.param_types([String::static_type(), gst::Element::static_type()])
.build(),
/**
* RsWebRTCSink::consumer_removed:
* @consumer_id: Identifier of the consumer that was removed
* @webrtcbin: The webrtcbin connected to the newly removed consumer
*
* This signal is emitted right after the connection with a consumer
* has been dropped.
*/
glib::subclass::Signal::builder("consumer-removed")
.param_types([String::static_type(), gst::Element::static_type()])
.build(),
/**
* RsWebRTCSink::get_sessions:
*
* List all sessions (by ID).
*/
glib::subclass::Signal::builder("get-sessions")
.action()
.class_handler(|_, args| {
let element = args[0].get::<super::WebRTCSink>().expect("signal arg");
let this = element.imp();
let res = Some(
this.state
.lock()
.unwrap()
.sessions
.keys()
.cloned()
.collect::<Vec<String>>()
.to_value(),
);
res
})
.return_type::<Vec<String>>()
.build(),
/**
* RsWebRTCSink::encoder-setup:
* @consumer_id: Identifier of the consumer
* @pad_name: The name of the corresponding input pad
* @encoder: The constructed encoder
*
* This signal can be used to tweak @encoder properties.
*
* Returns: True if the encoder is entirely configured,
* False to let other handlers run
*/
glib::subclass::Signal::builder("encoder-setup")
.param_types([
String::static_type(),
String::static_type(),
gst::Element::static_type(),
])
.return_type::<bool>()
.accumulator(|_hint, _ret, value| !value.get::<bool>().unwrap())
.class_handler(|_, args| {
let element = args[0].get::<super::WebRTCSink>().expect("signal arg");
let enc = args[3].get::<gst::Element>().unwrap();
gst::debug!(
CAT,
obj: element,
"applying default configuration on encoder {:?}",
enc
);
let this = element.imp();
let settings = this.settings.lock().unwrap();
configure_encoder(&enc, settings.cc_info.start_bitrate);
// Return false here so that latter handlers get called
Some(false.to_value())
})
.build(),
]
});
SIGNALS.as_ref()
}
fn constructed(&self) {
self.parent_constructed();
let obj = self.obj();
obj.set_suppressed_flags(gst::ElementFlags::SINK | gst::ElementFlags::SOURCE);
obj.set_element_flags(gst::ElementFlags::SINK);
}
}
impl GstObjectImpl for WebRTCSink {}
impl ElementImpl for WebRTCSink {
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
gst::subclass::ElementMetadata::new(
"WebRTCSink",
"Sink/Network/WebRTC",
"WebRTC sink",
"Mathieu Duponchelle <mathieu@centricular.com>",
)
});
Some(&*ELEMENT_METADATA)
}
fn pad_templates() -> &'static [gst::PadTemplate] {
static PAD_TEMPLATES: Lazy<Vec<gst::PadTemplate>> = Lazy::new(|| {
let caps = gst::Caps::builder_full()
.structure(gst::Structure::builder("video/x-raw").build())
.structure_with_features(
gst::Structure::builder("video/x-raw").build(),
gst::CapsFeatures::new(&[CUDA_MEMORY_FEATURE]),
)
.structure_with_features(
gst::Structure::builder("video/x-raw").build(),
gst::CapsFeatures::new(&[GL_MEMORY_FEATURE]),
)
.structure_with_features(
gst::Structure::builder("video/x-raw").build(),
gst::CapsFeatures::new(&[NVMM_MEMORY_FEATURE]),
)
.build();
let video_pad_template = gst::PadTemplate::new(
"video_%u",
gst::PadDirection::Sink,
gst::PadPresence::Request,
&caps,
)
.unwrap();
let caps = gst::Caps::builder("audio/x-raw").build();
let audio_pad_template = gst::PadTemplate::new(
"audio_%u",
gst::PadDirection::Sink,
gst::PadPresence::Request,
&caps,
)
.unwrap();
vec![video_pad_template, audio_pad_template]
});
PAD_TEMPLATES.as_ref()
}
fn request_new_pad(
&self,
templ: &gst::PadTemplate,
_name: Option<&str>,
_caps: Option<&gst::Caps>,
) -> Option<gst::Pad> {
let element = self.obj();
if element.current_state() > gst::State::Ready {
gst::error!(CAT, "element pads can only be requested before starting");
return None;
}
let mut state = self.state.lock().unwrap();
let name = if templ.name().starts_with("video_") {
let name = format!("video_{}", state.video_serial);
state.video_serial += 1;
name
} else {
let name = format!("audio_{}", state.audio_serial);
state.audio_serial += 1;
name
};
let sink_pad = gst::GhostPad::builder_with_template(templ, Some(name.as_str()))
.event_function(|pad, parent, event| {
WebRTCSink::catch_panic_pad_function(
parent,
|| false,
|this| this.sink_event(pad.upcast_ref(), &this.obj(), event),
)
})
.build();
sink_pad.set_active(true).unwrap();
sink_pad.use_fixed_caps();
element.add_pad(&sink_pad).unwrap();
state.streams.insert(
name,
InputStream {
sink_pad: sink_pad.clone(),
producer: None,
in_caps: None,
out_caps: None,
clocksync: None,
},
);
Some(sink_pad.upcast())
}
fn change_state(
&self,
transition: gst::StateChange,
) -> Result<gst::StateChangeSuccess, gst::StateChangeError> {
let element = self.obj();
if let gst::StateChange::ReadyToPaused = transition {
if let Err(err) = self.prepare(&element) {
gst::element_error!(
element,
gst::StreamError::Failed,
["Failed to prepare: {}", err]
);
return Err(gst::StateChangeError);
}
}
let mut ret = self.parent_change_state(transition);
match transition {
gst::StateChange::PausedToReady => {
if let Err(err) = self.unprepare(&element) {
gst::element_error!(
element,
gst::StreamError::Failed,
["Failed to unprepare: {}", err]
);
return Err(gst::StateChangeError);
}
}
gst::StateChange::ReadyToPaused => {
ret = Ok(gst::StateChangeSuccess::NoPreroll);
}
gst::StateChange::PausedToPlaying => {
let mut state = self.state.lock().unwrap();
state.maybe_start_signaller(&element);
}
_ => (),
}
ret
}
}
impl BinImpl for WebRTCSink {}
impl ChildProxyImpl for WebRTCSink {
fn child_by_index(&self, _index: u32) -> Option<glib::Object> {
None
}
fn children_count(&self) -> u32 {
0
}
fn child_by_name(&self, name: &str) -> Option<glib::Object> {
match name {
"signaller" => Some(
self.state
.lock()
.unwrap()
.signaller
.as_ref()
.as_ref()
.clone(),
),
_ => None,
}
}
}
impl NavigationImpl for WebRTCSink {
fn send_event(&self, event_def: gst::Structure) {
let mut state = self.state.lock().unwrap();
let event = gst::event::Navigation::new(event_def);
state.streams.iter_mut().for_each(|(_, stream)| {
if stream.sink_pad.name().starts_with("video_") {
gst::log!(CAT, "Navigating to: {:?}", event);
// FIXME: Handle multi tracks.
if !stream.sink_pad.push_event(event.clone()) {
gst::info!(CAT, "Could not send event: {:?}", event);
}
}
});
}
}
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