This is not needed. We don't perform any operations on inbound RTCP
packets. Receiver Reports and TWCC are generated by Reading RTP packets.
This reverts commit 080d7b8427.
The play-from-disk examples sees the average bitrate using
Chromium 90.0.4412.3 when enabled on loopback for a 3 minute
session.
Before: 744.443
After: 3927.669
Introduced with pion/interceptor@v0.0.10
* Correctly calculate page CRC
Ogg uses slightly non-standard version of CRC, we are unable to
use the Go's version. You can see the details here[0], in summary
"direct algorithm, initial val and final XOR = 0,
generator polynomial=0x04c11db7"
[0] https://xiph.org/vorbis/doc/framing.html
* Properly set EOS value
Before we created a new page with an EOS. Instead seek backwards
and update the last page with valid Opus to have EOS header and
re-generate the CRC
* Only use 0 timestamp/granule for headers
Audio itself should start from 0
* Rename OpusWriter -> OggWriter
Ogg supports more then just Opus, renaming gives us the flexibility to
expand in the future.
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
- Implemented a common "Writer" interface
- Made IVFWriter compliant to the new interface
- Implemented an OGG Opus writer
- Updated the save-file example
- Updated janus-gateway/streaming example
- Wrote a few unit tests for opus-writer
- Wrote a few unit tests for ivf-writer
- Fixed filenames
- Improved API to allow tests using an io.Writer
Let's pull off the bandaid!
* Reduces studdering: webrtc.RTCTrack -> webrtc.Track
* Makes it easier to find types by editor autocomplete
* Makes code read more fluently (less repetition)
Since we're breaking the API in 2.0, our only chance to
do this is now.
Relates to #408
This deviates from the WebRTC spec, so we need to fix it. This is a
massively breaking change, so we need to figure out the best way to help
users with this.
I also renamed our RTCPeerConnection constructor. The hope is that
people will refer to the examples/backlog and see what changed.
Resolves#309
Methods that were marked as deprecated weren't properly handled. There
was a mix of old+new ones supported which caused broken behavior.
SDP creation didn't add SCTP to Offer
Resolves#156
