haskal
/
gst-meet
1
0
Fork 0
Code Issues Pull Requests Projects Releases Wiki Activity

split RTP description parsing into its own function 

This keeps the ugly Options used after in the code, in the future these
could be replaced with the ParsedRtpDescription struct instead.
This commit is contained in:
Emmanuel Gil Peyrot 2021-10-23 16:33:10 +02:00 committed by Jasper
parent 0040d20151
commit c920fdf0a5
1 changed files with 220 additions and 148 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

110
lib-gst-meet/src/jingle.rs
View File

@ -19,6 +19,7 @@ use xmpp_parsers::{
jingle::{Action, Content, Creator, Description, Jingle, Senders, Transport},
jingle_dtls_srtp::{Fingerprint, Setup},
jingle_ice_udp::{self, Transport as IceUdpTransport},
jingle_rtcp_fb::RtcpFb,
jingle_rtp::{self, Description as RtpDescription, PayloadType, RtcpMux},
jingle_rtp_hdrext::RtpHdrext,
jingle_ssma::{self, Parameter},
@ -40,6 +41,24 @@ const RTP_HDREXT_TRANSPORT_CC: &str =
const DEFAULT_STUN_PORT: u16 = 3478;
const DEFAULT_TURNS_PORT: u16 = 5349;
struct ParsedRtpDescription {
opus_payload_type: Option<u8>,
opus_rtcp_fbs: Option<Vec<RtcpFb>>,
h264_payload_type: Option<u8>,
h264_rtx_payload_type: Option<u8>,
h264_rtcp_fbs: Option<Vec<RtcpFb>>,
vp8_payload_type: Option<u8>,
vp8_rtx_payload_type: Option<u8>,
vp8_rtcp_fbs: Option<Vec<RtcpFb>>,
vp9_payload_type: Option<u8>,
vp9_rtx_payload_type: Option<u8>,
vp9_rtcp_fbs: Option<Vec<RtcpFb>>,
audio_hdrext_ssrc_audio_level: Option<u8>,
audio_hdrext_transport_cc: Option<u8>,
video_hdrext_abs_send_time: Option<u8>,
video_hdrext_transport_cc: Option<u8>,
}
pub(crate) struct JingleSession {
pipeline: gstreamer::Pipeline,
audio_sink_element: gstreamer::Element,
@ -91,19 +110,7 @@ impl JingleSession {
Ok(self.pipeline_state_null_rx.await?)
}
pub(crate) async fn initiate(conference: &JitsiConference, jingle: Jingle) -> Result<Self> {
let initiator = jingle
.initiator
.as_ref()
.ok_or_else(|| anyhow!("session-initiate with no initiator"))?
.clone();
debug!("Received Jingle session-initiate from {}", initiator);
let mut ice_remote_candidates = None;
let mut ice_remote_ufrag = None;
let mut ice_remote_pwd = None;
let mut dtls_fingerprint = None;
fn parse_rtp_description(description: &RtpDescription, remote_ssrc_map: &mut HashMap<u32, Source>) -> Result<Option<ParsedRtpDescription>> {
let mut opus_payload_type = None;
let mut opus_rtcp_fbs = None;
let mut h264_payload_type = None;
@ -119,12 +126,7 @@ impl JingleSession {
let mut audio_hdrext_transport_cc = None;
let mut video_hdrext_abs_send_time = None;
let mut video_hdrext_transport_cc = None;
let mut colibri_url = None;
let mut remote_ssrc_map = HashMap::new();
for content in &jingle.contents {
if let Some(Description::Rtp(description)) = &content.description {
if description.media == "audio" {
opus_payload_type = description
.payload_types
@ -237,7 +239,7 @@ impl JingleSession {
}
else {
debug!("skipping media: {}", description.media);
continue;
return Ok(None);
}
for ssrc in &description.ssrcs {
@ -274,6 +276,76 @@ impl JingleSession {
},
);
}
Ok(Some(ParsedRtpDescription {
opus_payload_type,
opus_rtcp_fbs,
h264_payload_type,
h264_rtx_payload_type,
h264_rtcp_fbs,
vp8_payload_type,
vp8_rtx_payload_type,
vp8_rtcp_fbs,
vp9_payload_type,
vp9_rtx_payload_type,
vp9_rtcp_fbs,
audio_hdrext_ssrc_audio_level,
audio_hdrext_transport_cc,
video_hdrext_abs_send_time,
video_hdrext_transport_cc,
}))
}
pub(crate) async fn initiate(conference: &JitsiConference, jingle: Jingle) -> Result<Self> {
let initiator = jingle
.initiator
.as_ref()
.ok_or_else(|| anyhow!("session-initiate with no initiator"))?
.clone();
debug!("Received Jingle session-initiate from {}", initiator);
let mut ice_remote_candidates = None;
let mut ice_remote_ufrag = None;
let mut ice_remote_pwd = None;
let mut dtls_fingerprint = None;
let mut opus_payload_type = None;
let mut opus_rtcp_fbs = None;
let mut h264_payload_type = None;
let mut h264_rtx_payload_type = None;
let mut h264_rtcp_fbs = None;
let mut vp8_payload_type = None;
let mut vp8_rtx_payload_type = None;
let mut vp8_rtcp_fbs = None;
let mut vp9_payload_type = None;
let mut vp9_rtx_payload_type = None;
let mut vp9_rtcp_fbs = None;
let mut audio_hdrext_ssrc_audio_level = None;
let mut audio_hdrext_transport_cc = None;
let mut video_hdrext_abs_send_time = None;
let mut video_hdrext_transport_cc = None;
let mut colibri_url = None;
let mut remote_ssrc_map = HashMap::new();
for content in &jingle.contents {
if let Some(Description::Rtp(description)) = &content.description {
if let Some(description) = JingleSession::parse_rtp_description(description, &mut remote_ssrc_map)? {
opus_payload_type = opus_payload_type.or(description.opus_payload_type);
opus_rtcp_fbs = opus_rtcp_fbs.or(description.opus_rtcp_fbs);
h264_payload_type = h264_payload_type.or(description.h264_payload_type);
h264_rtx_payload_type = h264_rtx_payload_type.or(description.h264_rtx_payload_type);
h264_rtcp_fbs = h264_rtcp_fbs.or(description.h264_rtcp_fbs);
vp8_payload_type = vp8_payload_type.or(description.vp8_payload_type);
vp8_rtx_payload_type = vp8_rtx_payload_type.or(description.vp8_rtx_payload_type);
vp8_rtcp_fbs = vp8_rtcp_fbs.or(description.vp8_rtcp_fbs);
vp9_payload_type = vp9_payload_type.or(description.vp9_payload_type);
vp9_rtx_payload_type = vp9_rtx_payload_type.or(description.vp9_rtx_payload_type);
vp9_rtcp_fbs = vp9_rtcp_fbs.or(description.vp9_rtcp_fbs);
audio_hdrext_ssrc_audio_level = audio_hdrext_ssrc_audio_level.or(description.audio_hdrext_ssrc_audio_level);
audio_hdrext_transport_cc = audio_hdrext_transport_cc.or(description.audio_hdrext_transport_cc);
video_hdrext_abs_send_time = video_hdrext_abs_send_time.or(description.video_hdrext_abs_send_time);
video_hdrext_transport_cc = video_hdrext_transport_cc.or(description.video_hdrext_transport_cc);
}
}
if let Some(Transport::IceUdp(transport)) = &content.transport {