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Configurable JB size, send PLIs, send loss events, sep RTP/RTCP logging

This commit is contained in:
Jasper Hugo 2022-03-09 10:56:21 +07:00 committed by Jasper
parent 6ef974719b
commit d7e2320fe5
4 changed files with 109 additions and 52 deletions
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41
gst-meet/src/main.rs
View File

@ -25,80 +25,113 @@ use tracing::{error, info, trace, warn};
struct Opt {
#[structopt(long)]
web_socket_url: String,
#[structopt(
long,
help = "If not specified, assumed to be the host part of <web-socket-url>",
)]
xmpp_domain: Option<String>,
#[structopt(long)]
room_name: String,
#[structopt(
long,
help = "If not specified, assumed to be conference.<xmpp-domain>",
)]
muc_domain: Option<String>,
#[structopt(
long,
help = "If not specified, assumed to be focus@auth.<xmpp-domain>/focus",
)]
focus_jid: Option<String>,
#[structopt(
long,
default_value = "vp8",
help = "The video codec to negotiate support for. One of: vp8, vp9, h264",
)]
video_codec: String,
#[structopt(long, default_value = "gst-meet")]
nick: String,
#[structopt(long)]
region: Option<String>,
#[structopt(long)]
send_pipeline: Option<String>,
#[structopt(long)]
recv_pipeline_participant_template: Option<String>,
#[structopt(
long,
help = "Comma-separated endpoint IDs to select (prioritise receiving of)",
)]
select_endpoints: Option<String>,
#[structopt(
long,
help = "The maximum number of video streams we would like to receive",
)]
last_n: Option<u16>,
#[structopt(
long,
help = "The maximum height to receive video at."
)]
recv_video_height: Option<u16>,
#[structopt(
long,
help = "The maximum height we plan to send video at (used for stats only)."
)]
send_video_height: Option<u16>,
#[structopt(
long,
help = "The video type to signal that we are sending. One of: camera, desktop"
)]
video_type: Option<String>,
#[structopt(
long,
default_value = "200",
help = "The size of the jitter buffers in milliseconds. Larger values are more resilient to packet loss and jitter, smaller values give lower latency.",
)]
buffer_size: u32,
#[structopt(long)]
start_bitrate: Option<u32>,
#[structopt(long)]
stereo: Option<bool>,
#[structopt(short, long, parse(from_occurrences))]
verbose: u8,
#[cfg(feature = "tls-insecure")]
#[structopt(
long,
help = "Disable TLS certificate verification (use with extreme caution)"
)]
tls_insecure: bool,
#[cfg(feature = "log-rtp")]
#[structopt(
long,
help = "Log all RTP/RTCP packets at DEBUG level"
help = "Log all RTP packets at DEBUG level (extremely verbose)"
)]
log_rtp: bool,
#[cfg(feature = "log-rtp")]
#[structopt(
long,
help = "Log all RTCP packets at DEBUG level"
)]
log_rtcp: bool,
}
#[cfg(not(target_os = "macos"))]
@ -204,10 +237,13 @@ async fn main_inner() -> Result<()> {
video_codec,
recv_pipeline_participant_template,
send_video_height,
buffer_size,
start_bitrate,
stereo,
#[cfg(feature = "log-rtp")]
log_rtp,
#[cfg(feature = "log-rtp")]
log_rtcp,
..
} = opt;
@ -220,8 +256,11 @@ async fn main_inner() -> Result<()> {
extra_muc_features: vec![],
start_bitrate: start_bitrate.unwrap_or(800),
stereo: stereo.unwrap_or_default(),
buffer_size,
#[cfg(feature = "log-rtp")]
log_rtp,
#[cfg(feature = "log-rtp")]
log_rtcp,
};
let main_loop = glib::MainLoop::new(None, false);

4
lib-gst-meet-c/src/lib.rs
View File

@ -160,9 +160,11 @@ pub unsafe extern "C" fn gstmeet_connection_join_conference(
// TODO
start_bitrate: 800,
stereo: false,
buffer_size: 200,
#[cfg(feature = "log-rtp")]
log_rtp: false,
#[cfg(feature = "log-rtp")]
log_rtcp: false,
};
(*context)
.runtime

3
lib-gst-meet/src/conference.rs
View File

@ -80,8 +80,11 @@ pub struct JitsiConferenceConfig {
pub extra_muc_features: Vec<String>,
pub start_bitrate: u32,
pub stereo: bool,
pub buffer_size: u32,
#[cfg(feature = "log-rtp")]
pub log_rtp: bool,
#[cfg(feature = "log-rtp")]
pub log_rtcp: bool,
}
#[derive(Clone)]

113
lib-gst-meet/src/jingle.rs
View File

@ -548,6 +548,8 @@ impl JingleSession {
let rtpbin = gstreamer::ElementFactory::make("rtpbin", Some("rtpbin"))?;
rtpbin.set_property_from_str("rtp-profile", "savpf");
rtpbin.set_property("autoremove", true);
rtpbin.set_property("do-lost", true);
rtpbin.set_property("do-sync-event", true);
pipeline.add(&rtpbin)?;
let nicesrc = gstreamer::ElementFactory::make("nicesrc", None)?;
@ -604,7 +606,8 @@ impl JingleSession {
caps = caps
.field("media", "video")
.field("clock-rate", 90000)
.field("encoding-name", codec.encoding_name());
.field("encoding-name", codec.encoding_name())
.field("rtcp-fb-nack-pli", true);
if let Some(hdrext) = video_hdrext_transport_cc {
caps = caps.field(&format!("extmap-{}", hdrext), RTP_HDREXT_TRANSPORT_CC);
}
@ -640,6 +643,7 @@ impl JingleSession {
let handle = Handle::current();
let jingle_session = conference.jingle_session.clone();
let buffer_size = conference.config.buffer_size;
rtpbin.connect("new-jitterbuffer", false, move |values| {
let handle = handle.clone();
let jingle_session = jingle_session.clone();
@ -669,6 +673,8 @@ impl JingleSession {
if source.media_type == MediaType::Video && source.participant_id.is_some() {
debug!("enabling RTX for ssrc {}", ssrc);
rtpjitterbuffer.set_property("do-retransmission", true);
rtpjitterbuffer.set_property("drop-on-latency", true);
rtpjitterbuffer.set_property("latency", buffer_size);
}
Ok::<_, anyhow::Error>(())
};
@ -1023,60 +1029,67 @@ impl JingleSession {
pipeline.add(&rtcp_send_identity)?;
#[cfg(feature = "log-rtp")]
if conference.config.log_rtp {
debug!("setting up RTP/RTCP packet logging");
{
if conference.config.log_rtp {
debug!("setting up RTP packet logging");
let make_rtp_logger = |direction: &'static str| {
move |values: &[glib::Value]| -> Option<glib::Value> {
let f = || {
let buffer: gstreamer::Buffer = values[1].get()?;
let rtp_buffer = RTPBuffer::from_buffer_readable(&buffer)?;
debug!(
ssrc=rtp_buffer.ssrc(),
pt=rtp_buffer.payload_type(),
seq=rtp_buffer.seq(),
ts=rtp_buffer.timestamp(),
marker=rtp_buffer.is_marker(),
extension=rtp_buffer.is_extension(),
payload_size=rtp_buffer.payload_size(),
"RTP {}",
direction,
);
Ok::<_, anyhow::Error>(())
};
if let Err(e) = f() {
warn!("RTP {}: {:?}", direction, e);
let make_rtp_logger = |direction: &'static str| {
move |values: &[glib::Value]| -> Option<glib::Value> {
let f = || {
let buffer: gstreamer::Buffer = values[1].get()?;
let rtp_buffer = RTPBuffer::from_buffer_readable(&buffer)?;
debug!(
ssrc=rtp_buffer.ssrc(),
pt=rtp_buffer.payload_type(),
seq=rtp_buffer.seq(),
ts=rtp_buffer.timestamp(),
marker=rtp_buffer.is_marker(),
extension=rtp_buffer.is_extension(),
payload_size=rtp_buffer.payload_size(),
"RTP {}",
direction,
);
Ok::<_, anyhow::Error>(())
};
if let Err(e) = f() {
warn!("RTP {}: {:?}", direction, e);
}
None
}
None
}
};
};
let make_rtcp_logger = |direction: &'static str| {
move |values: &[glib::Value]| -> Option<glib::Value> {
let f = || {
let buffer: gstreamer::Buffer = values[1].get()?;
let mut buf = [0u8; 1500];
buffer.copy_to_slice(0, &mut buf[..buffer.size()]).map_err(|_| anyhow!("invalid RTCP packet size"))?;
let decoded = rtcp::packet::unmarshal(&mut &buf[..buffer.size()])?;
debug!(
"RTCP {} size={}\n{:#?}",
direction,
buffer.size(),
decoded,
);
Ok::<_, anyhow::Error>(())
};
if let Err(e) = f() {
warn!("RTCP {}: {:?}", direction, e);
rtp_recv_identity.connect("handoff", false, make_rtp_logger("RECV"));
rtp_send_identity.connect("handoff", false, make_rtp_logger("SEND"));
}
if conference.config.log_rtcp {
debug!("setting up RTCP packet logging");
let make_rtcp_logger = |direction: &'static str| {
move |values: &[glib::Value]| -> Option<glib::Value> {
let f = || {
let buffer: gstreamer::Buffer = values[1].get()?;
let mut buf = [0u8; 1500];
buffer.copy_to_slice(0, &mut buf[..buffer.size()]).map_err(|_| anyhow!("invalid RTCP packet size"))?;
let decoded = rtcp::packet::unmarshal(&mut &buf[..buffer.size()])?;
debug!(
"RTCP {} size={}\n{:#?}",
direction,
buffer.size(),
decoded,
);
Ok::<_, anyhow::Error>(())
};
if let Err(e) = f() {
warn!("RTCP {}: {:?}", direction, e);
}
None
}
None
}
};
};
rtp_recv_identity.connect("handoff", false, make_rtp_logger("RECV"));
rtp_send_identity.connect("handoff", false, make_rtp_logger("SEND"));
rtcp_recv_identity.connect("handoff", false, make_rtcp_logger("RECV"));
rtcp_send_identity.connect("handoff", false, make_rtcp_logger("SEND"));
rtcp_recv_identity.connect("handoff", false, make_rtcp_logger("RECV"));
rtcp_send_identity.connect("handoff", false, make_rtcp_logger("SEND"));
}
}
debug!("link dtlssrtpdec -> rtpbin");