Merge pull request webrtc-rs#287 from webrtc-rs/chore/prepare-0.5.0-release

Prepare 0.5.0 release

Author: lookback-hugotunius

examples/Cargo.toml

@@ -1,6 +1,6 @@
 [package]
 name = "examples"
-version = "0.4.0"
+version = "0.5.0"
 authors = ["Rain Liu <[email protected]>"]
 edition = "2018"
 description = "Examples of WebRTC.rs stack"

webrtc/CHANGELOG.md

@@ -2,6 +2,109 @@
 
 ## Unreleased
 
-## Prior to 0.4.0
+## 0.5.0
 
-Before 0.4.0 there was no changelog, previous changes are sometimes, but not always, available in the [GitHub Releases](https://github.com/webrtc-rs/webrtc/releases).
+### Changes
+
+#### Breaking changes
+
+* The serialized format for `RTCIceCandidateInit` has changed to match what the specification i.e. keys are camelCase. [#153 Make RTCIceCandidateInit conform to WebRTC spec](https://github.com/webrtc-rs/webrtc/pull/153) contributed by [jmatss](https://github.com/jmatss).
+* Improved robustness when proposing RTP extension IDs and handling of collisions in these. This change is only breaking if you have assumed anything about the nature of these extension IDs. [#154 Fix RTP extension id collision](https://github.com/webrtc-rs/webrtc/pull/154) contributed by [k0nserv](https://github.com/k0nserv)
+* Transceivers will now not stop when either or both directions are disabled. That is, applying and SDP with `a=inactive` will not stop the transceiver, instead attached senders and receivers will pause. A transceiver can be resurrected by setting direction back to e.g. `a=sendrecv`. The desired direction can be controlled with the newly introduced public method `set_direction` on `RTCRtpTransceiver`.  
+  * [#201 Handle inactive transceivers more correctly](https://github.com/webrtc-rs/webrtc/pull/201) contributed by [k0nserv](https://github.com/k0nserv)
+  * [#210 Rework transceiver direction support further](https://github.com/webrtc-rs/webrtc/pull/210) contributed by [k0nserv](https://github.com/k0nserv) 
+  * [#214 set_direction add missing Send + Sync bound](https://github.com/webrtc-rs/webrtc/pull/214) contributed by [algesten](https://github.com/algesten)
+  * [#213 set_direction add missing Sync bound](https://github.com/webrtc-rs/webrtc/pull/213) contributed by [algesten](https://github.com/algesten)
+  * [#212 Public RTCRtpTransceiver::set_direction](https://github.com/webrtc-rs/webrtc/pull/212) contributed by [algesten](https://github.com/algesten) 
+  * [#268 Fix current direction update when applying answer](https://github.com/webrtc-rs/webrtc/pull/268) contributed by [k0nserv](https://github.com/k0nserv)
+  * [#236 Pause RTP writing if direction indicates it](https://github.com/webrtc-rs/webrtc/pull/236) contributed by [algesten](https://github.com/algesten)
+* Generated the `a=msid` line for `m=` line sections according to the specification. This might be break remote peers that relied on the previous, incorrect, behaviour. This also fixes a bug where an endless negotiation loop could happen. [#217 Correct msid handling for RtpSender](https://github.com/webrtc-rs/webrtc/pull/217) contributed by [k0nserv](https://github.com/k0nserv)
+* Improve data channel id negotiation. We've slightly adjust the public interface for creating pre-negotiated data channels. Instead of a separate `negotiated: Option<bool>` and `id: Option<u16>` in `RTCDataChannelInit` there's now a more idiomatic `negotiated: Option<u16>`. If you have a pre-negotiated data channel simply set `negotiated: Some(id)` when creating the data channel.
+  * [#237 Fix datachannel id setting for 0.5.0 release](https://github.com/webrtc-rs/webrtc/pull/237) contributed by [stuqdog](https://github.com/stuqdog)
+  * [#229 Revert "base id updating on whether it's been negotiated, not on its …](https://github.com/webrtc-rs/webrtc/pull/229) contributed by [melekes](https://github.com/melekes)
+
+  * [#226 base id updating on whether it's been finalized, not on its value](https://github.com/webrtc-rs/webrtc/pull/226) contributed by [stuqdog](https://github.com/stuqdog)
+
+
+#### Other improvememnts
+
+We made various improvements and fixes since 0.4.0, including merging all subcrates into a single git repo. The old crate repos are archived and all development will now happen in https://github.com/webrtc-rs/webrtc/.
+
+* We now provide stats reporting via the standardized `RTCPeerConnection::get_stats` method.
+  * [#277 Implement Remote Inbound Stats](https://github.com/webrtc-rs/webrtc/pull/277) contributed by [k0nserv](https://github.com/k0nserv)
+  * [#220 Make stats types pub so they can be used directly](https://github.com/webrtc-rs/webrtc/pull/220) contributed by [k0nserv](https://github.com/k0nserv)
+  * [#225 Add RTP Stats to stats report](https://github.com/webrtc-rs/webrtc/pull/225) contributed by [k0nserv](https://github.com/k0nserv)
+  * [#189 Serialize stats](https://github.com/webrtc-rs/webrtc/pull/189) contributed by [sax](https://github.com/sax)
+  * [#180 Get stats from peer connection](https://github.com/webrtc-rs/webrtc/pull/180) contributed by [sax](https://github.com/sax)
+
+* [#278 Fix async-global-executor](https://github.com/webrtc-rs/webrtc/pull/278) contributed by [k0nserv](https://github.com/k0nserv)
+* [#276 relax regex version requirement](https://github.com/webrtc-rs/webrtc/pull/276) contributed by [melekes](https://github.com/melekes)
+* [#244 Update README.md instructions after monorepo merge](https://github.com/webrtc-rs/webrtc/pull/244) contributed by [k0nserv](https://github.com/k0nserv)
+* [#241 move profile to workspace](https://github.com/webrtc-rs/webrtc/pull/241) contributed by [xnorpx](https://github.com/xnorpx)
+* [#240 Increase timeout to "fix" test breaking](https://github.com/webrtc-rs/webrtc/pull/240) contributed by [algesten](https://github.com/algesten)
+* [#239 One repo (again)](https://github.com/webrtc-rs/webrtc/pull/239) contributed by [algesten](https://github.com/algesten)
+* [#234 Fix recent clippy lints](https://github.com/webrtc-rs/webrtc/pull/234) contributed by [k0nserv](https://github.com/k0nserv)
+* [#224 update call to DataChannel::accept as per data pr #14](https://github.com/webrtc-rs/webrtc/pull/224) contributed by [melekes](https://github.com/melekes)
+* [#223 dtls_transport: always set remote certificate](https://github.com/webrtc-rs/webrtc/pull/223) contributed by [melekes](https://github.com/melekes)
+* [#216 Lower case mime types for comparison in fmpt lines](https://github.com/webrtc-rs/webrtc/pull/216) contributed by [k0nserv](https://github.com/k0nserv)
+* [#211 Helper to trigger negotiation_needed](https://github.com/webrtc-rs/webrtc/pull/211) contributed by [algesten](https://github.com/algesten)
+* [#209 MID generator feature](https://github.com/webrtc-rs/webrtc/pull/209) contributed by [algesten](https://github.com/algesten)
+* [#208 update deps + loosen some requirements](https://github.com/webrtc-rs/webrtc/pull/208) contributed by [melekes](https://github.com/melekes)
+* [#205 data_channel: handle stream EOF](https://github.com/webrtc-rs/webrtc/pull/205) contributed by [melekes](https://github.com/melekes)
+* [#204 [peer_connection] allow persistent certificates](https://github.com/webrtc-rs/webrtc/pull/204) contributed by [melekes](https://github.com/melekes)
+* [#202 bugfix-Udp connection not close (reopen #174) #195](https://github.com/webrtc-rs/webrtc/pull/202) contributed by [shiqifeng2000](https://github.com/shiqifeng2000)
+* [#199 Upgrade ICE to 0.7.0](https://github.com/webrtc-rs/webrtc/pull/199) contributed by [k0nserv](https://github.com/k0nserv)
+* [#194 Add AV1 MimeType and RtpCodecParameters](https://github.com/webrtc-rs/webrtc/pull/194) contributed by [billylindeman](https://github.com/billylindeman)
+* [#188 Improve operations debuggability](https://github.com/webrtc-rs/webrtc/pull/188) contributed by [k0nserv](https://github.com/k0nserv)
+* [#187 Fix SDP for rejected tracks to conform to RFC](https://github.com/webrtc-rs/webrtc/pull/187) contributed by [k0nserv](https://github.com/k0nserv)
+* [#185 Adding some debug and display traits](https://github.com/webrtc-rs/webrtc/pull/185) contributed by [sevensidedmarble](https://github.com/sevensidedmarble)
+* [#179 Fix example names in README](https://github.com/webrtc-rs/webrtc/pull/179) contributed by [ethagnawl](https://github.com/ethagnawl)
+* [#176 Time overflow armv7 workaround](https://github.com/webrtc-rs/webrtc/pull/176) contributed by [frjol](https://github.com/frjol)
+* [#171 close DTLS conn upon err](https://github.com/webrtc-rs/webrtc/pull/171) contributed by [melekes](https://github.com/melekes)
+* [#170 always start sctp](https://github.com/webrtc-rs/webrtc/pull/170) contributed by [melekes](https://github.com/melekes)
+* [#167 Add offer/answer/pranswer constructors for RTCSessionDescription](https://github.com/webrtc-rs/webrtc/pull/167) contributed by [sax](https://github.com/sax)
+
+#### Subcrate updates
+
+The various sub-crates have been updated as follows:
+
+* util: 0.5.3 => 0.6.0
+* sdp:  0.5.1 => 0.5.2
+* mdns: 0.4.2 => 0.5.0
+* stun: 0.4.2 => 0.4.3
+* turn: 0.5.3 => 0.6.0
+* ice: 0.6.4 => 0.8.0
+* dtls: 0.5.2 => 0.6.0
+* rtcp: 0.6.5 => 0.7.0
+* rtp: 0.6.5 => 0.6.7
+* srtp: 0.8.9 => 0.9.0
+* scpt: 0.4.3 => 0.6.1
+* data: 0.3.3 => 0.5.0
+* interceptor: 0.7.6 => 0.8.0
+* media: 0.4.5 => 0.4.7
+
+Their respective change logs are found in the old, now archived, repositories and within their respective `CHANGELOG.md` files in the monorepo.
+
+### Contributors 
+
+A big thanks to all the contributors that have made this release happen:
+
+* [morajabi](https://github.com/morajabi)
+* [sax](https://github.com/sax)
+* [ethagnawl](https://github.com/ethagnawl)
+* [xnorpx](https://github.com/xnorpx)
+* [frjol](https://github.com/frjol)
+* [algesten](https://github.com/algesten)
+* [shiqifeng2000](https://github.com/shiqifeng2000)
+* [billylindeman](https://github.com/billylindeman)
+* [sevensidedmarble](https://github.com/sevensidedmarble)
+* [k0nserv](https://github.com/k0nserv)
+* [stuqdog](https://github.com/stuqdog)
+* [neonphog](https://github.com/neonphog)
+* [melekes](https://github.com/melekes)
+* [jmatss](https://github.com/jmatss)
+
+
+## Prior to 0.5.0
+
+Before 0.5.0 there was no changelog, previous changes are sometimes, but not always, available in the [GitHub Releases](https://github.com/webrtc-rs/webrtc/releases).

webrtc/Cargo.toml

@@ -1,6 +1,6 @@
 [package]
 name = "webrtc"
-version = "0.4.0"
+version = "0.5.0"
 authors = ["Rain Liu <[email protected]>"]
 edition = "2018"
 description = "A pure Rust implementation of WebRTC API"