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webrtcsrc: Fix caps used when creating transceiver

Browse source 
We used to pass all media keys and attributes to the caps which
incorrect. Instead we should be using only the keys from the map
and remove all information related to rtcp which is irrelevant
to create the transceiver.

This also simplifies the code.

New caps look like:

```
Caps(
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 96,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP8",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 102,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 104,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 106,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 108,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 127,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 39,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 98,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "0",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 100,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "2",
    },
)
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1216>
This commit is contained in:
Thibault Saunier 2023-05-18 10:40:50 -04:00 committed by Sebastian Dröge
parent cc59ff9052
commit 482ff879a4
1 changed files with 27 additions and 35 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

64
net/webrtc/src/webrtcsrc/imp.rs
View file

@ -9,6 +9,7 @@ use core::ops::Deref;
use gst::glib;
use gst::subclass::prelude::*;
use once_cell::sync::Lazy;
use std::collections::HashSet;
use std::str::FromStr;
use std::sync::atomic::AtomicBool;
use std::sync::atomic::AtomicU16;
@ -696,48 +697,39 @@ impl WebRTCSrc {
let direction = gst_webrtc::WebRTCRTPTransceiverDirection::Recvonly;
let webrtcbin = self.webrtcbin();
for (i, media) in sdp.medias().enumerate() {
let all_caps_for_media = media
.formats()
.filter_map(|format| {
format.parse::<i32>().ok().and_then(|pt| {
let mut tmpcaps = media.caps_from_media(pt)?;
{
let tmpcaps = tmpcaps.get_mut().unwrap();
tmpcaps
.structure_mut(0)
.unwrap()
.set_name("application/x-rtp");
if let Err(err) = media.attributes_to_caps(tmpcaps) {
gst::error!(CAT, "Couldn't copy media attributes to caps: {err:?}")
}
}
Some(tmpcaps)
})
})
.collect::<Vec<gst::Caps>>();
let mut caps = gst::Caps::new_empty();
let codec_names = {
let settings = self.settings.lock().unwrap();
for codec in settings
settings
.video_codecs
.iter()
.chain(settings.audio_codecs.iter())
{
for media_caps in &all_caps_for_media {
let encoding_name = media_caps
.structure(0)
.unwrap()
.get::<&str>("encoding-name")
.unwrap();
if encoding_name == codec.name {
caps.get_mut().unwrap().append(media_caps.clone());
.map(|codec| codec.name.clone())
.collect::<HashSet<String>>()
};
let caps = media
.formats()
.filter_map(|format| {
format.parse::<i32>().ok().and_then(|pt| {
let mediacaps = media.caps_from_media(pt)?;
let s = mediacaps.structure(0).unwrap();
if !codec_names.contains(s.get::<&str>("encoding-name").ok()?) {
return None;
}
let mut filtered_s = gst::Structure::new_empty("application/x-rtp");
filtered_s.extend(s.iter().filter_map(|(key, value)| {
if key.starts_with("rtcp-") {
None
} else {
Some((key, value.to_owned()))
}
}
drop(settings);
}));
Some(filtered_s)
})
})
.collect::<gst::Caps>();
if !caps.is_empty() {
let stream_id = self.get_stream_id(None, Some(i as u32)).unwrap();