Sean DuBois
dd0c3da122
Revert jsfiddle links with getUserMedia
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AudioCapture and VideoCapture is blocked by jsfiddle when using this
feature. Need to debug+fix
2018-07-07 11:41:56 -07:00
Sean DuBois
76a07068c9
Use HTTPS for jsfiddle examples
2018-07-07 11:32:58 -07:00
Sean DuBois
7aa47c7d99
Add empty css files to jsfiddle demos
2018-07-07 11:32:58 -07:00
Sean DuBois
074e3391bf
Move JSfiddle snippets to git
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jsfiddle provides a way to create snippets from Github via a URL. This
way we can still provide easy demos, but get all the nice things from
having them in Git
Closes #32
2018-07-07 11:32:58 -07:00
Raphael Randschau
98ea0b791e
fix warnings in examples
2018-07-06 15:23:40 -07:00
Raphael Randschau
4f6983307f
update examples with new constructor
2018-07-06 15:23:40 -07:00
John Bradley
cfba14cfea
Add H264 send/receive and packetization support
2018-07-05 17:16:51 -05:00
Sean DuBois
c6d8334cf7
Update gstreamer-send jsfiddle
2018-07-04 00:49:23 -07:00
Sean DuBois
24a312c34d
Fix poorly named function in RTCPeerConnection
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CreateOffer -> CreateAnswer currently `pion-WebRTC` can only generate
offers not answers.
2018-07-04 00:46:01 -07:00
Sean DuBois
3b3ed9a544
Fix lint, vet and fmt errors
2018-07-03 21:11:25 -07:00
Sean DuBois
366f9ec268
Implement sample count generation in gstreamer-send
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Audio+Video now works in gstreamer-send
2018-07-03 20:55:10 -07:00
Sean DuBois
a5cf1702e8
Use unique names for static globals in GStreamer cgo
2018-07-03 17:34:16 -07:00
Sean DuBois
b1da546d24
Implement multi-pipeline gstreamer-send example
2018-07-03 17:11:56 -07:00
John Bradley
912a8e18f8
Add opus sending support
2018-07-03 18:00:45 -05:00
Sean DuBois
f7ae8e3d0a
Copy @backkem doc fixes to gstreamer-receive and save-to-disk
2018-07-03 11:42:40 -07:00
backkem
8df477e38d
gstreamer-send: document running in Windows.
2018-07-03 11:18:46 -07:00
Sean DuBois
6eb22ad669
Add 'gstreamer-send' to README.md
2018-07-03 00:10:38 -07:00
Sean DuBois
093a4efac4
Gofmt
2018-07-02 21:58:08 -07:00
John R. Bradley
799e02d8f8
Add VP8 payloading
2018-07-02 23:28:53 -05:00
Sean DuBois
176efccff2
Update docs to mention audio support
2018-07-01 02:08:34 -07:00
Sean DuBois
de2fb09778
Add support for receiving audio
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Update gstreamer-receive to create pipelines for each input.
Currently we don't allow the user to pass in what codecs they support and we don't
take into account the offer. The API will need to be updated to catch
both these signaling errors. The user will pass a slice of support
codecs.
2018-07-01 02:04:47 -07:00
Sean DuBois
5bf9d5af34
Add ICE connection state change notification and timeouts
2018-06-30 02:57:47 -07:00
Sean DuBois
a623369bca
Fix lint errors
2018-06-24 23:11:14 -07:00
Sean DuBois
5235a4f78c
Prepare for send-peer
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Only thing left is creating the RTP packets, using packets directly
works
2018-06-24 19:48:45 -07:00
Sean DuBois
51136804ac
Significant send progress
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* GStreamer sends RTP packets to Go
* pion-WebRTC generates proper SDP, and has certificate ready
* Just need to implement SRTP functionality and rough MVP should be done
2018-06-22 01:04:07 -07:00
John Bradley
2041b2e6d0
Fix windows build instructions
2018-06-20 01:31:23 -05:00
John Bradley
d26bf8c80a
Add build instructions for windows
2018-06-20 01:30:01 -05:00
Sean DuBois
2ac78a4500
Fix gstreamer-receive import
2018-06-18 22:35:33 -07:00
Sean DuBois
882619f02c
Doc updates
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* Finish README.md
* Create README.md for every example
* Finish DESIGN.md
2018-06-18 22:22:19 -07:00
Sean DuBois
d46382e382
Call RTCPeerConnection.Ontrack with a new goroutine
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Every implementation should do this anyway. Also
new users might not understand and block all events for
RTCPeerConnections
2018-06-13 00:03:08 -07:00
Sean DuBois
c7ca757fa8
Change from Media -> Track
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We expect single tracks, so use proper terminology
2018-06-12 23:59:56 -07:00
Sean DuBois
a325be40a1
Add <stdlib.h> to GStreamer code
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Run clang-format on DTLS+SRTP
2018-06-12 22:31:55 -07:00
Sean DuBois
ddbb6c8ba8
Fix all golint errors
2018-06-12 22:24:52 -07:00
Sean DuBois
c21bc319d2
Set directions on channels
2018-06-12 21:25:49 -07:00
Sean DuBois
7e7f155e33
Small GStreamer pipeline changes
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* Don't specify caps that are already fixated
* Use autovideosink
2018-06-12 20:50:10 -07:00
John Bradley
cb36b4a0e5
Add gstreamer example that renders VP8 peer
2018-06-12 17:39:41 -05:00
John Bradley
06fc5b846c
Refactor VP8 RTP Packet parsing
2018-06-12 17:39:33 -05:00
Sean DuBois
2f3fa046e6
Fix all megacheck warnings
2018-06-12 01:23:15 -07:00
Sean DuBois
9e72c2913e
Fix errcheck warnings
2018-06-12 01:13:09 -07:00
Sean DuBois
d1608eda58
Add support for RTP header extensions, and IVF videos now fully work!
2018-06-11 01:42:41 -07:00
John Bradley
882b87b346
Add VP8 aware RTP decoding in IVF writer
2018-06-11 01:15:32 -05:00
Sean DuBois
c7a6caa462
Finish IVF writer
2018-06-10 19:01:53 -07:00
Sean DuBois
fd96da48ea
Add save-to-disk example
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Currently the constructed IVF don't work, but everything saves properly.
Hopefully off-by-one somewhere
2018-06-10 17:01:14 -07:00
Sean DuBois
279a786207
Refactor to match WebRTC Native API
2018-06-10 01:18:02 -07:00
