@jsonstudio/zterm-daemon-darwin-arm64 0.0.0-stage → 0.1.3

package/runtime/node_modules/@roamhq/wrtc/CHANGELOG.md

@@ -0,0 +1,1094 @@
+# 0.10.0
+
+## New Features
+
+- Update libwebrtc to M106!
+
+## Bug Fixes
+
+- Fix segfault/abort during program cleanup; some destructors that relied on
+  the V8 environment were being called after said environment had shut down.
+  Forcing them to run earlier (on `pc.close()`) seems to fix things.
+
+# 0.9.1
+
+## Bug Fixes
+
+- Fix I420 helpers for images with odd resolutions
+  (<https://github.com/WonderInventions/node-webrtc/pull/38>)
+
+# 0.9.0
+
+## New Features
+
+- Added support for RTCRtpTransceiverDirection "stopped".
+  (<https://github.com/node-webrtc/node-webrtc/pull/672>)
+- More standards-compliance for `addIceCandidate`; now, all the different
+  "empty candidate" options are accepted as an "end-of-candidates" signal.
+
+## Bug Fixes
+
+- Fixed too-early GCs (<https://github.com/node-webrtc/node-webrtc/pull/667>)
+
+# 0.8.0
+
+## New Features
+
+- Update libwebrtc to M98!
+- Compile linux-x64 with GCC 10 in order to support stabler Linux distributions
+  like Amazon Linux 2023.
+
+## Bug Fixes
+
+- Changed binary path resolution to make things more consistent on environments
+  that don't automatically resolve `node_modules`, like Amazon Linux 2023.
+
+# 0.7.3
+
+## Bug Fixes
+
+- `ondata` audio problems should be fixed
+  (<https://github.com/WonderInventions/node-webrtc/issues/2>)
+
+# 0.7.2
+
+Added `linux-arm64` prebuilt binary, built & unit-tested all x64 binaries on
+Node 20.
+
+# 0.7.0
+
+## New Features
+
+- Update to libwebrtc M94! This brings some quality improvements over M87's
+  encoders and other network capabilities.
+
+## Breaking Changes
+
+- Remove `legacyGetStats` API from `RTCPeerConnection`.
+
+## Bug Fixes
+
+- `RTCIceTransport.getGatheringState()` no longer always returns `"complete"`.
+
+# 0.6.0
+
+## New Features
+
+- `RTCPeerConnection.getStats()` can now take in a `MediaStreamTrack` as a
+  selector, updating that API to the latest WebRTC standard.
+- Typescript definition files provided to let this library (somewhat
+  inaccurately, unfortunately) polyfill for WebRTC types included with the
+  standard, and also define the `nonstandard` types.
+
+# 0.5.1
+
+## Bug Fixes
+
+- Take care of case where `RTCPeerConnection` is closed while `OnTrack` events
+  are still pending.
+
+  # 0.5.0
+
+## New Features
+
+- Support Node 18, with updated build system
+- Newer prebuilt binary system not dependent on node-pre-gyp
+
+# 0.4.6
+
+## New Features
+
+- Added `target` property to RTCPeerConnection events (thanks, @CharlesRA).
+- Support for additional APIs in `lib/browser.js` (thanks, @piranna).
+- Added a naïve version of `getDeviceMedia` that delegates to `getUserMedia`
+  (thanks, @piranna).
+
+## Bug Fixes
+
+- Although Node 14 support was confirmed in v0.4.5, it was not included in the
+  "engines" property of package.json (thanks, @farnabaz).
+- Potential fix for a crash mentioned in #637 (thanks, @thedracle).
+
+# 0.4.5
+
+## New Features
+
+- Updated to WebRTC M81.
+- Added support for Node 14.
+- Added rollback support.
+
+## Bug Fixes
+
+- RTCPeerConnection no longer raises "icegatheringstatechange" when the
+  RTCPeerConnection is closed. Thanks, @arlolra. (#625)
+
+## Breaking Changes
+
+- With the update from M79 to M81, the `dtx`, `ptime`, and `codecPayloadType`
+  parameters to RTCRtpEncodingParameters no longer take affect. They've also
+  been removed from the WebRTC 1.0 specification (see [here](https://github.com/w3c/webrtc-pc/issues/2350)).
+  Although this is technically a SemVer-breaking change, few users of this
+  library are depending on the removed functionality, and I prefer not to
+  increment the version number at this time.
+
+# 0.4.4
+
+## New Features
+
+- `addTrack` now supports multiple MediaStream arguments (#548). Additionally,
+  MediaStreams can now be constructed with arbitrary IDs. For more information,
+  see below. Thanks, @csheely and @sgodin.
+- `setStreams` now supports multiple MediaStream arguments.
+
+## MediaStream
+
+MediaStreams in node-webrtc can be constructed with arbitrary IDs. For example,
+the following MediaStream, `stream`, has its ID set to "foo".
+
+```js
+const stream = new MediaStream({ id: "foo" });
+stream.id === "foo"; // true
+```
+
+# 0.4.3
+
+## New Features
+
+- Updated to WebRTC M79.
+- Added support for Node 13.
+- Added support for a number of new standard APIs (see below).
+
+### RTCPeerConnection
+
+- Added support for `restartIce`.
+- Added support for "icecandidateerror" events.
+
+### RTCDtlsTransport
+
+- Added support for `getRemoteCertificates`.
+
+### RTCRtpSender, RTCRtpReceiver & RTCRtpTransceiver
+
+- Added support for `getCapabilities` to RTCRtpSender and RTCRtpReceiver.
+- Added support for `setParameters` to RTCRtpSender.
+- Added support for `setStreams` to RTCRtpSender (at this time, up to one
+  MediaStream argument is supported).
+- Added support for `sendEncodings` to RTCRtpTransceiverInit.
+- Added support for `setCodecPreferences` to RTCRtpTransceiver.
+
+## Bug Fixes
+
+- Fixed a bug where VideoFrame timestamps reported via RTCP were incorrect
+  (#566). Thanks, @lonocvb.
+- Fixed a bug where, in some cases, Ninja builds failed on macOS (#582). Thanks,
+  @taylorhoward92.
+- Fixed bugs related to N-API usage in recent version of Node 12 and 13.
+
+# 0.4.2
+
+## Bug Fixes
+
+- Fix image stride issue at certain resolutions. (#536)
+
+# 0.4.1
+
+## Bug Fixes
+
+- Fix memory leak when receiving strings over RTCDataChannel. (#528)
+
+# 0.4.0
+
+node-webrtc is now implemented using N-API, which is ABI stable across Node
+releases. This means we can now ship fewer binaries while supporting a
+potentially greater number of Node releases. As of 0.4.0, node-webrtc targets
+N-API version 3.
+
+## New Features
+
+- Updated to WebRTC M74.
+- Added support for Node 12.
+- Added support for Electron 4 and 5.
+- Added initial RTCIceTransport support (see below).
+- Added initial RTCSctpTransport support (see below).
+- Expanded RTCIceCandidate support (see below).
+
+## Bug Fixes
+
+- Avoid crashing when `createDataChannel` fails. (#508)
+
+## Breaking Changes
+
+- Dropped support for Node 6.
+- Installing from NPM only downloads pre-built binaries. If you wish to build
+  from source, install from the source repository. (#200)
+- Unified Plan is now the default. For Plan B behavior, set the `sdpSemantics`
+  RTCConfiguration property or the `SDP_SEMANTICS` environment variable to
+  "plan-b".
+
+## RTCIceTransport
+
+RTCDtlsTransport now exposes RTCIceTransport under the `iceTransport` property.
+
+The following attributes are supported:
+
+- `role`
+- `component`
+- `state`
+- `gatheringState`
+
+The following events are supported:
+
+- "statechange"
+- "gatheringstatechange"
+
+## RTCSctpTransport
+
+RTCPeerConnection now exposes RTCSctpTransport under the `sctp` property.
+
+The following attributes are supported:
+
+- `transport`
+- `state`
+
+The following attributes are partially supported:
+
+- `maxMessageSize` (always `null`)
+- `maxChannels` (always `null`)
+
+The "statechange" event is also supported.
+
+## RTCIceCandidate
+
+RTCIceCandidates now include the following attributes:
+
+- `foundation`
+- `component`
+- `priority`
+- `address`
+- `protocol`
+- `port`
+- `type`
+- `tcpType`
+- `relatedAddress`
+- `relatedPort`
+- `usernameFragment`
+
+# 0.3.7
+
+## New Features
+
+### RTCDtlsTransport
+
+RTCRtpSender and RTCRtpReceiver now provide access to RTCDtlsTransport via the
+`transport` property (initially `null`). Currently, RTCDtlsTransport only
+supports the `state` property, the "statechange" event, and the "error" event.
+
+### Miscellaneous
+
+- Updated to WebRTC M73.
+- Added `maxPacketLifeTime` getter to RTCDataChannel (#492).
+- Added `negotiated` getter to RTCDataChannel.
+
+## Bug Fixes
+
+- Fixed `addIceCandidate` queueing behavior (#498).
+
+# 0.3.6
+
+## New Features
+
+### Programmatic Audio
+
+This release of node-webrtc adds non-standard, programmatic audio APIs in the
+form of RTCAudioSource and RTCAudioSink. These APIs are similar to the
+previously added RTCVideoSource and RTCVideoSink APIs. With these APIs, you can
+
+- Pass audio samples to RTCAudioSource via the `onData` method. Then use the
+  RTCAudioSource's `createTrack` method to create a local audio
+  MediaStreamTrack.
+- Construct an RTCAudioSink from a local or remote audio MediaStreamTrack. The
+  RTCAudioSink will emit a "data" event every time audio samples are received.
+  When you're finished, stop the RTCAudioSink by calling `stop`.
+
+Because these APIs are non-standard, they are exposed via a `nonstandard`
+property on node-webrtc's exports object. For example,
+
+```js
+const { RTCAudioSource, RTCAudioSink } = require("wrtc").nonstandard;
+
+const source = new RTCAudioSource();
+const track = source.createTrack();
+const sink = new RTCAudioSink(track);
+
+const sampleRate = 8000;
+const samples = new Int16Array(sampleRate / 100); // 10 ms of 16-bit mono audio
+
+const data = {
+  samples,
+  sampleRate,
+};
+
+const interval = setInterval(() => {
+  // Update audioData in some way before sending.
+  source.onData(data);
+});
+
+sink.ondata = (data) => {
+  // Do something with the received audio samples.
+};
+
+setTimeout(() => {
+  clearInterval(interval);
+  track.stop();
+  sink.stop();
+}, 10000);
+```
+
+#### RTCAudioSource
+
+```webidl
+[constructor]
+interface RTCAudioSource {
+  MediaStreamTrack createTrack();
+  void onData(RTCAudioData data);
+};
+
+dictionary RTCAudioData {
+  required Int16Array samples;
+  required unsigned short sampleRate;
+  octet bitsPerSample = 16;
+  octet channelCount = 1;
+  unsigned short numberOfFrames;
+};
+```
+
+- Calling `createTrack` will return a local audio MediaStreamTrack whose source
+  is the RTCAudioSource.
+- Calling `onData` with RTCAudioData pushes a new audio samples to every
+  non-stopped local audio MediaStreamTrack created with `createTrack`.
+- RTCAudioData should represent 10 ms worth of 16-bit audio samples.
+
+#### RTCAudioSink
+
+```webidl
+[constructor(MediaStreamTrack track)]
+interface RTCAudioSink {
+  void stop();
+  readonly attribute boolean stopped;
+  attribute EventHandler ondata;
+};
+```
+
+- RTCAudioSink's constructor accepts a local or remote audio MediaStreamTrack.
+- As long as neither the RTCAudioSink nor the RTCAudioSink's MediaStreamTrack
+  are stopped, the RTCAudioSink will raise a "data" event any time RTCAudioData
+  is received.
+- The "data" event has all the properties of RTCAudioData.
+- RTCAudioSink must be stopped by calling `stop`.
+
+### RTCVideoFrame `rotation`
+
+The RTCVideoFrame raised in RTCVideoSink's "frame" event now includes a
+property, `rotation`, which indicates rotation of the RTCVideoFrame. Possible
+values are 0, 90, 180, and 270.
+
+### EventListener `handleEvent`
+
+EventListener instances now support [`handleEvent`](https://developer.mozilla.org/en-US/docs/Web/API/EventListener/handleEvent).
+
+# 0.3.5
+
+## New Features
+
+### Programmatic Video
+
+This release of node-webrtc adds non-standard, programmatic video APIs in the
+form of RTCVideoSource and RTCVideoSink. With these APIs, you can
+
+- Pass [I420](https://wiki.videolan.org/YUV/#I420) frames to RTCVideoSource via
+  the `onFrame` method. Then use RTCVideoSource's `createTrack` method to create
+  a local video MediaStreamTrack.
+- Construct an RTCVideoSink from a local or remote video MediaStreamTrack. The
+  RTCVideoSink will emit a "frame" event every time an I420 frame is received.
+  When you're finished, stop the RTCVideoSink by calling `stop`.
+
+Because these APIs are non-standard, they are exposed via a `nonstandard`
+property on node-webrtc's exports object. For example,
+
+```js
+const { RTCVideoSource, RTCVideoSink } = require("wrtc").nonstandard;
+
+const source = new RTCVideoSource();
+const track = source.createTrack();
+const sink = new RTCVideoSink(track);
+
+const width = 320;
+const height = 240;
+const data = new Uint8ClampedArray(width * height * 1.5);
+const frame = { width, height, data };
+
+const interval = setInterval(() => {
+  // Update the frame in some way before sending.
+  source.onFrame(frame);
+});
+
+sink.onframe = ({ frame }) => {
+  // Do something with the received frame.
+};
+
+setTimeout(() => {
+  clearInterval(interval);
+  track.stop();
+  sink.stop();
+}, 10000);
+```
+
+This release also adds bindings to some [libyuv](https://chromium.googlesource.com/libyuv/libyuv/)
+functions for handling I420 frames. These can be useful when converting to and
+from RGBA.
+
+#### RTCVideoSource
+
+```webidl
+[constructor(optional RTCVideoSourceInit init)]
+interface RTCVideoSource {
+  readonly attribute boolean isScreencast;
+  readonly attribute boolean? needsDenoising;
+  MediaStreamTrack createTrack();
+  void onFrame(RTCVideoFrame frame);
+};
+
+dictionary RTCVideoSourceInit {
+  boolean isScreencast = false;
+  boolean needsDenoising;
+};
+
+dictionary RTCVideoFrame {
+  required unsigned long width;
+  required unsigned long height;
+  required Uint8ClampedArray data;
+};
+```
+
+- Calling `createTrack` will return a local video MediaStreamTrack whose source
+  is the RTCVideoSource.
+- Calling `onFrame` with an RTCVideoFrame pushes a new video frame to every
+  non-stopped local video MediaStreamTrack created with `createTrack`.
+- An RTCVideoFrame represents an I420 frame.
+
+#### RTCVideoSink
+
+```webidl
+[constructor(MediaStreamTrack track)]
+interface RTCVideoSink {
+  void stop();
+  readonly attribute boolean stopped;
+  attribute EventHandler onframe;
+};
+```
+
+- RTCVideoSink's constructor accepts a local or remote video MediaStreamTrack.
+- As long as neither the RTCVideoSink nor the RTCVideoSink's MediaStreamTrack
+  are stopped, the RTCVideoSink will raise a "frame" event any time an
+  RTCVideoFrame is received.
+- The "frame" event has a property, `frame`, of type RTCVideoFrame.
+- RTCVideoSink must be stopped by calling `stop`.
+
+#### `i420ToRgba` and `rgbaToI420`
+
+These two functions are bindings to libyuv that provide conversions between I420
+and RGBA frames. WebRTC expects I420, whereas APIs like the
+[Canvas API](https://developer.mozilla.org/en-US/docs/Web/API/Canvas_API) expect
+RGBA, so these functions are useful for converting between. For example,
+
+```js
+const { i420ToRgba, rgbaToI420 } = require("wrtc").nonstandard;
+
+const width = 640;
+const height = 480;
+const i420Data = new Uint8ClampedArray(width * height * 1.5);
+const rgbaData = new Uint8ClampedArray(width * height * 4);
+const i420Frame = { width, height, data: i420Data };
+const rgbaFrame = { width, height, data: rgbaData };
+
+i420ToRgba(i420Frame, rgbaFrame);
+rgbaToI420(rgbaFrame, i420Frame);
+```
+
+### MediaStreamTrack
+
+- Added support for setting MediaStreamTrack's `enabled` property (#475).
+
+# 0.3.4
+
+## New Features
+
+- Updated to WebRTC M71.
+- Relay remote audio MediaStreamTracks on Windows (0.1.5 initially introduced
+  this feature for Linux and macOS; now, Windows supports it, too).
+- Added support for [pkg](https://github.com/zeit/pkg) (#404).
+
+## Bug Fixes
+
+- Calling certain methods, like `addTrack`, `removeTrack`, etc., with objects
+  that were not instances of MediaStreamTrack, RTCRtpSender, etc., could lead to
+  segfaults. This was because we did not properly validate objects before
+  attempting to unwrap them. (#448)
+
+# 0.3.3
+
+## New Features
+
+- Experimental support for armv7l and arm64. Binaries built for these
+  architectures have been tested with QEMU but not on real devices. Please test
+  them out. If you install node-webrtc directly on an ARM device, node-pre-gyp
+  should pull the correct binaries automatically. Otherwise, you may need to set
+  the `TARGET_ARCH` environment variable to "arm" (armv7l) or "arm64". For
+  example,
+
+  ```
+  TARGET_ARCH=arm64 npm install
+  ```
+
+- Set `DEBUG=true` to install debug binaries (Linux- and macOS-only). For
+  example,
+
+  ```
+  DEBUG=true npm install
+  ```
+
+# 0.3.2
+
+## New Features
+
+- Support for Node 11 on Windows.
+
+# 0.3.1
+
+This release adds a number of new features and brings us closer to
+spec-compliance, thanks to the tests at
+[web-platform-tests/wpt](http://github.com/web-platform-tests/wpt).
+
+## New Features
+
+### `getUserMedia`
+
+This release adds limited `getUserMedia` support. You can create audio and video
+MediaStreamTracks; however, the resulting MediaStreamTracks do not capture any
+media. You can add these MediaStreamTracks to an RTCPeerConnection; however, no
+media will be transmitted. You can confirm by checking `bytesSent` and
+`bytesReceived` in `getStats`.
+
+```js
+const { getUserMedia } = require("wrtc");
+
+getUserMedia({
+  audio: true,
+  video: true,
+}).then((stream) => {
+  stream.getTracks().forEach((track) => stop());
+});
+```
+
+Although we will parse and validate some members of the MediaStreamConstraints
+and related dictionaries, we do not use their values at this time.
+
+### `getStats`
+
+This release adds limited standards-compliant `getStats` support. Previous
+node-webrtc releases exposed the legacy, callback-based `getStats` API. This
+release preserves that API but adds the Promise-based API. Neither the
+MediaStreamTrack selector argument nor the RTCRtpSender- and
+RTCRtpReceiver-level `getStats` APIs are implemented at this time.
+
+```js
+// Legacy API
+pc.getStats((response) => {
+  /* ... */
+}, console.error);
+
+// Standards-compliant API
+pc.getStats().then((report) => {
+  /* ... */
+}, console.error);
+```
+
+### Unified Plan and `sdpSemantics`
+
+This release adds support for RTCRtpTransceivers and Unified Plan SDP via
+
+- A non-standard RTCConfiguration option, `sdpSemantics`, and
+- An environment variable, `SDP_SEMANTICS`.
+
+Construct an RTCPeerConnection with `sdpSemantics` set to "unified-plan" or
+launch your application with `SDP_SEMANTICS=unified-plan` to enable
+RTCRtpTransceiver support; otherwise, "plan-b" is the default.
+
+```js
+const { RTCPeerConnection } = require("wrtc");
+
+const pc = new RTCPeerConnection({
+  sdpSemantics: "unified-plan", // default is "plan-b"
+});
+```
+
+```
+SDP_SEMANTICS=unified-plan node app.js
+```
+
+### RTCRtpTransceiver
+
+You can use RTCRtpTransceivers and related APIs when Unified Plan is enabled.
+This includes the following RTCPeerConnection methods
+
+- `addTransceiver`
+- `getTransceivers`
+
+and the following RTCTrackEvent properties
+
+- `transceiver`
+
+The following RTCRtpTransceiver methods are supported
+
+- `stop`
+
+as well as the following RTCRtpTransceiver properties
+
+- `mid`
+- `sender`
+- `receiver`
+- `stopped`
+- `direction`
+- `currentDirection`
+
+`setCodecPreferences` is not yet implemented. When calling `addTransceiver`,
+only the following RTCRtpTransceiverInit dictionary members are supported
+
+- `direction`
+- `streams`
+
+```js
+const assert = require("assert");
+const { MediaStream, RTCPeerConnection, RTCRtpTransceiver } = require("wrtc");
+
+const pc = new RTCPeerConnection({
+  sdpSemantics: "unified-plan",
+});
+
+const t1 = pc.addTransceiver("audio", {
+  direction: "recvonly",
+});
+
+const t2 = pc.addTransceiver(t1.receiver.track, {
+  direction: "sendonly",
+  streams: [new MediaStream()],
+});
+```
+
+### MediaStreamTrack
+
+Added limited support for the `muted` property (it always returns `false`).
+
+### Miscellaneous
+
+- APIs that should throw DOMExceptions, such as `addTrack`, will use
+  [domexception](https://github.com/jsdom/domexception) to construct those
+  DOMExceptions, if installed.
+
+## Bug Fixes
+
+- Calling `addTrack` twice with the same MediaStreamTrack should throw an
+  InvalidAccessError (#442).
+- MediaStream's `getTrackById` did not work for video MediaStreamTracks.
+- MediaStream's `clone` method did not `clone` MediaStreamTracks.
+- MediaStreamTrack's `readyState` was not updated when `stop` was called.
+
+  # 0.3.0
+
+## New Features
+
+- Support for Node 11. Binaries are available for Linux and macOS. Windows
+  binaries will become available in a subsequent release once AppVeyor gains
+  support for Node 11.
+- Updated to WebRTC M70. This release no longer uses
+  [mayeut/libwebrtc](https://github.com/mayeut/libwebrtc); instead, WebRTC is
+  built from source.
+
+## Breaking Changes
+
+- Dropped support for Node 9
+- Minimum CMake version bumped to 3.12
+- Minimum GCC version bumped to 5.4
+- Minimum Microsoft Visual Studio version bumped to 2017
+
+## Bug Fixes
+
+- Updating to WebRTC M70 fixes an RTCDataChannel-related interop bug with recent
+  Firefox releases (#444).
+
+# 0.2.2
+
+## Bug Fixes
+
+- Destroy AudioDeviceModule on the worker thread.
+
+# 0.2.1
+
+## Bug Fixes
+
+- Fixed an AudioDeviceModule memory and thread leak (#429).
+- Fixed an issue where closing an RTCPeerConnection would raise "open" events on
+  any RTCDataChannels whose `readyState` was "connecting" (#436).
+
+# 0.2.0
+
+## Breaking Changes
+
+- Dropped support for Node 4, 5 and 7 (#408).
+
+## Bug Fixes
+
+- Fixed a race when closing an RTCDataChannel (#358).
+- Fixed memory leaks in `createOffer`, `createAnswer`, `addIceCandidate`, and
+  `getStats` (#425).
+
+# 0.1.6
+
+## Bug Fixes
+
+- Fixed an issue with receiving multiple ArrayBuffers over an RTCDataChannel
+  that could cause invalid memory accesses (#406).
+
+# 0.1.5
+
+## New Features
+
+- This release allows relaying remote MediaStreamTracks. This can be useful for
+  test applications. **Note:** currently, Windows cannot relay audio
+  MediaStreamTracks, only video.
+
+### MediaStream
+
+This release adds support for MediaStream. Most MediaStream APIs are supported,
+excluding the "addtrack" and "removetrack" events. You can construct
+MediaStreams as follows:
+
+```js
+const { MediaStream } = require("wrtc");
+
+const stream1 = new MediaStream();
+const stream2 = new MediaStream(stream1);
+```
+
+Assuming you already have some Array of MediaStreamTracks, `tracks`, you can
+also construct a MediaStream with
+
+```js
+const stream3 = new MediaStream(tracks);
+```
+
+Or, if you have an existing MediaStream, you can `clone` it.
+
+```js
+const stream4 = stream3.clone();
+```
+
+This release also adds support for the following methods
+
+- `getTracks`
+- `getAudioTracks`
+- `getVideoTracks`
+- `getTrackById`
+- `addTrack`
+- `removeTrack`
+
+and the following attributes
+
+- `id`
+- `active`
+
+### RTCTrackEvent
+
+This release adds support for the `streams` property.
+
+### RTCPeerConnection
+
+This release adds support for `addTrack`, `removeTrack`, and `getSenders`.
+Although we don't yet provide a way to construct local MediaStreamTracks, you
+can relay remote MediaStreamTracks as follows:
+
+```js
+pc.ontrack = ({ track, streams }) => {
+  pc.addTrack(track, ...streams);
+};
+```
+
+### RTCRtpSender
+
+This release adds support for the following methods
+
+- `getCapabilities` (always throws for now)
+- `getParameters`
+- `setParameters` (always returns a rejected Promise for now)
+- `getStats` (always returns a rejected Promise for now)
+- `replaceTrack`
+
+and the following attributes
+
+- `track`
+- `transport` (always returns `null` for now)
+- `rtcpTransport` (always returns `null` for now)
+
+# 0.1.4
+
+## New Features
+
+- Added support for Node 10 (#402)
+
+### RTCPeerConnection
+
+- Added support for `getReceivers`.
+- Added partial support for the RTCTrackEvent. RTCPeerConnection will emit the
+  RTCTrackEvent with two attributes: `track` and `receiver`. In the future, we
+  will add support for the `streams` and `transceiver` attributes.
+
+### RTCRtpReceiver
+
+This release adds partial support for RTCRtpReceiver, including methods
+
+- `getParameters`
+- `getContributingSources` (always returns an empty array for now)
+- `getSynchronizationSources` (always returns an empty array for now)
+- `getStats` (always returns a rejected Promise for now)
+
+and attributes
+
+- `track`
+- `transport` (always returns `null` for now)
+- `rtcpTransport` (always returns `null` for now)
+
+RTCRtpReceiver also includes partial support for the static method
+`getCapabilities`; however, it always returns a rejected Promise.
+
+### MediaStreamTrack
+
+This release adds partial support for remote MediaStreamTracks, including
+attributes
+
+- `enabled` (read-only for now)
+- `id`
+- `kind`
+- `readyState`
+
+# 0.1.3
+
+## Bug Fixes
+
+- Fixed memory leaks related to RTCPeerConnection events.
+
+# 0.1.2
+
+## Bug Fixes
+
+- Fixed memory leaks related to sending and receiving messages over
+  RTCDataChannels (#205, #304, #319). There are some less severe leaks related
+  to RTCPeerConnection events that remain. These will be addressed in a future
+  release.
+
+# 0.1.1
+
+## Bug Fixes
+
+- Calling `createDataChannel` on a closed RTCPeerConnection no longer returns
+  `undefined`; instead, it raises an InvalidStateError (#314, #382).
+- Worked around WebRTC [Issue 7585](https://bugs.chromium.org/p/webrtc/issues/detail?id=7585)
+  on Linux by backporting the `epoll`-based PhysicalSocketServer from WebRTC
+  M61 into node-webrtc. This allows many more concurrent RTCPeerConnections on
+  Linux (for example, up to 3000 in my tests, not exceeding thread limits).
+  (#362)
+
+# 0.1.0
+
+This project will begin to follow [SemVer](https://semver.org) in preparation
+for a 1.0.0 release.
+
+## New Features
+
+Besides updating to WebRTC M60 (using
+[mayeut/libwebrtc](https://github.com/mayeut/libwebrtc)), this release adds a
+number of features that bring node-webrtc closer to standards-compliance. We
+still have a ways to go, but we're now testing against
+[w3c/web-platform-tests](https://github.com/w3c/web-platform-tests).
+
+### RTCConfiguration
+
+RTCPeerConnection's constructor now accepts the following standard properties:
+
+- `bundlePolicy`
+- `iceCandidatePoolSize`
+- `iceServers` (no support for OAuth yet)
+- `iceTransportPolicy`
+- `rtcpMuxPolicy`
+
+RTCConfiguration also accepts a non-standard property, `portRange`. This
+property constrains the port range used by the RTCPeerConnection's ICE
+transports. For example,
+
+```js
+const { RTCPeerConnection } = require("wrtc");
+
+const pc = new RTCPeerConnection({
+  portRange: {
+    min: 10000, // defaults to 0
+    max: 20000, // defaults to 65535
+  },
+});
+```
+
+### RTCPeerConnection
+
+RTCPeerConnection now supports two new methods:
+
+- `getConfiguration`
+- `setConfiguration`
+
+RTCPeerConnection now supports the following properties:
+
+- `canTrickleIceCandidates` (always returns `null` for now)
+- `connectionState` (derived from `iceConnectionState`)
+- `currentLocalDescription`
+- `currentRemoteDescription`
+- `pendingLocalDescription`
+- `pendingRemoteDescription`
+
+RTCPeerConnection now supports the following events:
+
+- "connectionstatechange"
+- "negotiationneeded"
+
+### RTCOfferOptions and RTCAnswerOptions
+
+RTCPeerConnection's `createOffer` method now accepts RTCOfferOptions, and
+RTCPeerConnection's `createAnswer` method now accepts RTCAnswerOptions.
+RTCOfferOptions supports
+
+- `iceRestart`
+- `offerToReceiveAudio`
+- `offerToReceiveVideo`
+
+Both RTCOfferOptions and RTCAnswerOptions support `voiceActivityDetection`.
+
+### RTCDataChannelInit
+
+RTCPeerConnection's `createDataChannel` method now accepts
+
+- `id`
+- `maxPacketLifeTime`
+- `maxRetransmits`
+- `negotiated`
+- `ordered`
+- `protocol`
+
+### RTCDataChannel
+
+RTCDataChannel supports the following properties:
+
+- `id`
+- `maxRetransmits`
+- `ordered`
+- `priority` (always returns "high")
+- `protocol`
+
+RTCDataChannel's `send` method now supports sending Blobs provided by
+[jsdom](https://github.com/jsdom/jsdom); however, there is no support for
+receiving Blobs.
+
+### Top-level Exports
+
+Added top-level exports for
+
+- RTCDataChannel
+- RTCDataChannelEvent
+- RTCPeerConnectionIceEvent
+
+## Bug Fixes
+
+- Fixed a failed assertion when closing RTCPeerConnection's or RTCDataChannel's
+  event loop (#376).
+- Moved AudioDeviceModule construction to the worker thread. This fixes a thread
+  checker assertion raised by debug builds of libwebrtc.
+- Copy StatsReports on the signaling thread. This fixes a thread checker
+  assertion raised by debug builds of libwebrtc.
+
+## Breaking Changes
+
+- Dropped support for "moz"- and "webkit"-prefixed WebRTC APIs in
+  `lib/browser.js`. This means that, when bundling JavaScript that depends on
+  node-webrtc for the browser, the resulting bundle will no longer depend on
+  these APIs (#369).
+- Dropped support for the now non-standard `url` attribute in RTCIceServer.
+- Dropped support for the `RTCIceConnectionStates`, `RTCIceGatheringStates`, and
+  `RTCSignalingStates` properties on the RTCPeerConnection prototype. These were
+  an implementation detail that some libraries used for detecting node-webrtc.
+- Dropped support for the `RTCDataStates` and `BinaryTypes` properties on the
+  RTCDataChannel prototype. This, too, was an implementation detail.
+
+# 0.0.67
+
+## Bug Fixes
+
+- ObjectWrap instances accessed in an event loop (like PeerConnection and
+  DataChannel) were getting freed before the event loop completed, which caused
+  segfaults. Now, we call `Ref` in the ObjectWrap instances' constructors and
+  `Unref` in their event loops.
+- Fixed another source of segfaults, where, if a DataChannel's
+  PeerConnectionFactory was freed, accessing the underlying DataChannelInterface
+  would try to use Threads which had been freed.
+
+# 0.0.66
+
+## Bug Fixes
+
+- Fixed a CPU regression introduced in 0.0.63. We now share a single
+  PeerConnectionFactoryInterface across PeerConnectionInterfaces, and we now use
+  a "dummy" AudioDeviceModule instead of FakeAudioDeviceModule.
+
+# 0.0.65
+
+## New Features
+
+- Added support for sending Buffers (#103)
+
+## Bug Fixes
+
+- Sending an ArrayBufferView over an RTCDataChannel did not take into account
+  the ArrayBufferView's `offset` or `length` properties. This resulted in
+  sending the entire backing ArrayBuffer instead of just the data in the
+  ArrayBufferView.
+- unzip-stream 0.2.2 breaks compatibility with Node 4 and 5. This release pins
+  to unzip-stream 0.2.1.
+
+## Breaking Changes
+
+- Building from source requires CMake 3.1 or newer
+
+# 0.0.64
+
+## Bug Fixes
+
+- We no longer `Externalize` ArrayBuffers. This fixes an error when sending
+  ArrayBuffers mutliple times (#262 and #264) and a memory leak (#304).
+- Fixed RTCDataChannel-related segfaults by checking for `nullptr` (#236 and
+  #325)
+
+# 0.0.63
+
+## New Features
+
+- Support for Node 9
+- Updated to WebRTC M57 (using [libwebrtc](https://github.com/aisouard/libwebrtc))
+
+## Breaking Changes
+
+- Minimum Mac OS X version bumped to 10.9
+- Minimum Microsoft Visual Studio version bumped to 2015