codec has apt fmtp can't negotiation because it's
apt payloadtype can't be found in negotiation codecs.
rtx codec must be partial match if apt codec is partial.
Was reverted in eeb67e1Fixes#1785
Exact match if fmtp parameters are not inconsistent.
e.g. default OPUS fmtp of the browsers are:
Chrome: minptime=10;useinbandfec=1
Firefox: maxplaybackrate=48000;stereo=1;useinbandfec=1
They should be treated as matched.
If an API is shared between PeerConnections they would use the same
MediaEngine. A MediaEngine contains negotiated PayloadTypes so if the
PeerConnections were answering you would end up in invalid states.
Add DisableMediaEngineCopy to SettingEngine in case user needs old
behavior.
Resolves#1662
RegisterHeaderExtension now allows users to enable headers depending on
the type of transceiver that was created.
Also expose GetParameters on RTPSender and RTPReceiver
Co-authored-by: OrlandoCo <luisorlando.co@gmail.com>
Resolves#1554
Instead of having logic in pion/webrtc and pion/sdp move all logic to
pion/ice. Users of pion/ice don't want to import these packages to use
basic pion/ice functionality.
Formatting parameters are not only allowed on video
codecs but also on most audio codecs. Without this change,
custom parameters of audio codecs would be dropped when
parsing SDPs (e.g. opus parameters)
This causes users who use PopulateFromSDP to declare
they support transport-cc. Since they don't send receiver
reports. They will get a degraded experience unexpectedly.
Partially revert ce8de088e7.
Resolves#1026
This fixes the echo program so it works properly on Safari
and Firefox, where the preferred offered dynamic media type
is not 96/VP8. It loads MediaEngine with codecs found in the
offer and then uses the payload type of the offer's preferred
video codec in the answer.
This commit has breaking changes. This API change means we
can no longer support an arbitrary number of receivers. For every track
you want to receive you MUST call PeerConnection.AddTransceiver
We do now support sending an multiple audio/video feeds. You can see
this behavior via gstreamer-receive and gstreamer-send currently.
Resolves#54
Fixes#515
This includes a few small and closely related changes:
1. All occurrences of the build tag `+build js` have been changed to the
more precise `+build js,wasm`. This will exclude the files from being
included by third-party compilers like GopherJS, with which they are
incompatible.
2. Some files which are incompatible with JavaScript/Wasm now have the
correct build tag (`+build -js`) so they will be excluded from Wasm
builds.
3. Some configuration options which are incompatible with
JavaScript/Wasm (or at least the current bindings) will now no longer
appear in Wasm builds. This meant creating new files with new struct
definitions and the appropriate build tags.
Syntax could be better but works in a standards-compliant way.
```
codec := webrtc.NewRTPH264Codec(webrtc.DefaultPayloadTypeH264, 90000)
codec.RTPCodecCapability.RTCPFeedback = []webrtc.RTCPFeedback{{
Type: "nack",
}, {
Type: "nack",
Parameter: "pli",
}}
```
