From 2a914030201d3d0ae031b84074f1cd5991addfdf Mon Sep 17 00:00:00 2001 From: Hugo Tunius Date: Wed, 7 Sep 2022 16:55:13 +0100 Subject: [PATCH 1/2] Prepare 0.5.0 release --- webrtc/CHANGELOG.md | 107 +++++++++++++++++++++++++++++++++++++++++++- webrtc/Cargo.toml | 2 +- 2 files changed, 106 insertions(+), 3 deletions(-) diff --git a/webrtc/CHANGELOG.md b/webrtc/CHANGELOG.md index 0480a1057..5bd2d7510 100644 --- a/webrtc/CHANGELOG.md +++ b/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` and `id: Option` in `RTCDataChannelInit` there's now a more idiomatic `negotiated: Option`. 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). diff --git a/webrtc/Cargo.toml b/webrtc/Cargo.toml index 664f8ae96..5d4c42cd0 100644 --- a/webrtc/Cargo.toml +++ b/webrtc/Cargo.toml @@ -1,6 +1,6 @@ [package] name = "webrtc" -version = "0.4.0" +version = "0.5.0" authors = ["Rain Liu "] edition = "2018" description = "A pure Rust implementation of WebRTC API" From f109936340dbc919102f8c8cd3f0d62d87588572 Mon Sep 17 00:00:00 2001 From: Hugo Tunius Date: Wed, 7 Sep 2022 16:57:37 +0100 Subject: [PATCH 2/2] Bump examples version to 0.5.0 --- examples/Cargo.toml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/examples/Cargo.toml b/examples/Cargo.toml index a12ace73f..d95b25cd7 100644 --- a/examples/Cargo.toml +++ b/examples/Cargo.toml @@ -1,6 +1,6 @@ [package] name = "examples" -version = "0.4.0" +version = "0.5.0" authors = ["Rain Liu "] edition = "2018" description = "Examples of WebRTC.rs stack"