node-webrtc · markandrus · Nov 3, 2020 · Dec 25, 2020 · Dec 25, 2020
diff --git a/.circleci/config.yml b/.circleci/config.yml
diff --git a/CHANGELOG.md b/CHANGELOG.md
@@ -1,3 +1,13 @@
+0.4.7
+=====
+
+New Features
+------------
+
+- Updated to WebRTC M87. Check the WebRTC release notes for changes included in
+  M82—M87.
+- Added support for RTCRtpTransceiverDirection "stopped".
+
 0.4.6
 =====


diff --git a/README.md b/README.md
@@ -41,30 +41,14 @@ The following platforms are confirmed to work with node-webrtc and have prebuilt
  </thead>
  <tbody>
    <tr>
-      <th rowspan="6">Node</th>
-      <th>8</th>
-        <td align="center">✓</td>
-        <td align="center">✓</td>
-        <td align="center">✓</td>
-      <td align="center">✓</td>
-      <td align="center">✓</td>
-    </tr>
-    <tr>
+      <th rowspan="4">Node</th>
      <th>10</th>
        <td align="center">✓</td>
        <td align="center">✓</td>
        <td align="center">✓</td>
      <td align="center">✓</td>
      <td align="center">✓</td>
    </tr>
-    <tr>
-      <th>11</th>
-        <td align="center">✓</td>
-        <td align="center">✓</td>
-        <td align="center">✓</td>
-      <td align="center">✓</td>
-      <td align="center">✓</td>
-    </tr>
    <tr>
      <th>12</th>
        <td align="center">✓</td>
@@ -74,15 +58,15 @@ The following platforms are confirmed to work with node-webrtc and have prebuilt
      <td align="center">✓</td>
    </tr>
    <tr>
-      <th>13</th>
+      <th>14</th>
        <td align="center">✓</td>
        <td align="center">✓</td>
        <td align="center">✓</td>
      <td align="center">✓</td>
      <td align="center">✓</td>
    </tr>
    <tr>
-      <th>14</th>
+      <th>15</th>
        <td align="center">✓</td>
        <td align="center">✓</td>
        <td align="center">✓</td>

diff --git a/appveyor.yml b/appveyor.yml
@@ -4,7 +4,7 @@ environment:
  node_pre_gyp_secretAccessKey:
    secure: NVvJWst8RJXhz/G0Gu+dpljZz5iNT0ko4sPybD313JuIn2wAf9xGDnKEipmY1mow
  matrix:
-    - nodejs_version: "8"
+    - nodejs_version: "10"
      platform: x64
      msvs_toolset: 15
      CHECK_REFLEXIVE: true

diff --git a/lib/index.js b/lib/index.js
@@ -70,6 +70,7 @@ module.exports = {
  RTCIceTransport,
  RTCPeerConnection: require('./peerconnection'),
  RTCPeerConnectionIceEvent: require('./rtcpeerconnectioniceevent'),
+  RTCPeerConnectionIceErrorEvent: require('./rtcpeerconnectioniceerrorevent'),
  RTCRtpReceiver,
  RTCRtpSender,
  RTCRtpTransceiver,

diff --git a/lib/peerconnection.js b/lib/peerconnection.js
@@ -42,9 +42,6 @@ function RTCPeerConnection() {
  };

  pc.onicecandidateerror = function onicecandidateerror(eventInitDict) {
-    var pair = eventInitDict.hostCandidate.split(':');
-    eventInitDict.address = pair[0];
-    eventInitDict.port = pair[1];
    eventInitDict.target = self;
    var icecandidateerror = new RTCPeerConnectionIceErrorEvent('icecandidateerror', eventInitDict);
    self.dispatchEvent(icecandidateerror);

diff --git a/scripts/build-appveyor.bat b/scripts/build-appveyor.bat
@@ -14,33 +14,23 @@ ECHO npm run lint
 CALL npm run lint
 IF %ERRORLEVEL% NEQ 0 GOTO ERROR

-@REM ECHO Test using Node 8
-@REM SET nodejs_version=8
-@REM CALL scripts\run-tests.bat
-@REM IF %ERRORLEVEL% NEQ 0 GOTO ERROR
-
-@REM ECHO Test using Node 10
-@REM SET nodejs_version=10
-@REM CALL scripts\run-tests.bat
-@REM IF %ERRORLEVEL% NEQ 0 GOTO ERROR
-
-@REM ECHO Test using Node 11
-@REM SET nodejs_version=11
-@REM CALL scripts\run-tests.bat
-@REM IF %ERRORLEVEL% NEQ 0 GOTO ERROR
+ECHO Test using Node 10
+SET nodejs_version=10
+CALL scripts\run-tests.bat
+IF %ERRORLEVEL% NEQ 0 GOTO ERROR

 ECHO Test using Node 12
 SET nodejs_version=12
 CALL scripts\run-tests.bat
 IF %ERRORLEVEL% NEQ 0 GOTO ERROR

-ECHO Test using Node 13
-SET nodejs_version=13
+ECHO Test using Node 14
+SET nodejs_version=14
 CALL scripts\run-tests.bat
 IF %ERRORLEVEL% NEQ 0 GOTO ERROR

-ECHO Test using Node 14
-SET nodejs_version=14
+ECHO Test using Node 15
+SET nodejs_version=15
 CALL scripts\run-tests.bat
 IF %ERRORLEVEL% NEQ 0 GOTO ERROR


diff --git a/src/dictionaries/rtc_peer_connection_on_ice_error_event.h b/src/dictionaries/rtc_peer_connection_on_ice_error_event.h
@@ -6,7 +6,8 @@

 #define RTC_PEER_CONNECTION_ICE_ERROR_EVENT RTCPeerConnectionIceErrorEvent
 #define RTC_PEER_CONNECTION_ICE_ERROR_EVENT_LIST \
-  DICT_REQUIRED(std::string, hostCandidate, "hostCandidate") \
+  DICT_REQUIRED(std::string, address, "address") \
+  DICT_REQUIRED(int, port, "port") \
  DICT_REQUIRED(std::string, url, "url") \
  DICT_REQUIRED(uint8_t, errorCode, "errorCoode") \
  DICT_REQUIRED(std::string, errorText, "errorText")

diff --git a/src/dictionaries/webrtc/data_channel_init.cc b/src/dictionaries/webrtc/data_channel_init.cc
@@ -37,8 +37,8 @@ static Validation<webrtc::DataChannelInit> DATA_CHANNEL_INIT_FN(
  }
  webrtc::DataChannelInit init;
  init.ordered = ordered;
-  init.maxRetransmitTime = maxPacketLifeTime.Map([](auto i) { return static_cast<int>(i); }).FromMaybe(-1);
-  init.maxRetransmits = maxRetransmits.Map([](auto i) { return static_cast<int>(i); }).FromMaybe(-1);
+  init.maxRetransmitTime = maxPacketLifeTime.Map([](auto i) { return absl::make_optional(static_cast<int>(i)); }).FromMaybe(absl::optional<int>());
+  init.maxRetransmits = maxRetransmits.Map([](auto i) { return absl::make_optional(static_cast<int>(i)); }).FromMaybe(absl::optional<int>());
  init.protocol = protocol;
  init.negotiated = negotiated;
  init.id = id.Map([](auto i) { return static_cast<int>(i); }).FromMaybe(-1);

diff --git a/src/dictionaries/webrtc/rtc_error.cc b/src/dictionaries/webrtc/rtc_error.cc
@@ -24,6 +24,7 @@ TO_NAPI_IMPL(const webrtc::RTCError*, pair) {
    case webrtc::RTCErrorType::NONE:
      return Validation<Napi::Value>::Invalid("No error? Please file a bug on https://github.com/node-webrtc/node-webrtc");
    case webrtc::RTCErrorType::UNSUPPORTED_PARAMETER:
+      return Pure(scope.Escape(ErrorFactory::CreateInvalidModificationError(env, error->message())));
    case webrtc::RTCErrorType::INVALID_PARAMETER:
      return Pure(scope.Escape(ErrorFactory::CreateInvalidAccessError(env, error->message())));
    case webrtc::RTCErrorType::INVALID_RANGE:

diff --git a/src/dictionaries/webrtc/rtp_encoding_parameters.cc b/src/dictionaries/webrtc/rtp_encoding_parameters.cc
@@ -41,33 +41,28 @@ static Validation<webrtc::RtpEncodingParameters> RTP_ENCODING_PARAMETERS_FN(
    const node_webrtc::Maybe<uint64_t> maxBitrate,
    const node_webrtc::Maybe<double> maxFramerate,
    const node_webrtc::Maybe<double> scaleResolutionDownBy) {
+  // NOTE(mroberts): No longer supported in WebRTC 1.0.
+  (void) codecPayloadType;
+  (void) dtx;
+  (void) priority;
+  (void) ptime;
+
  webrtc::RtpEncodingParameters parameters;
  if (rid.IsJust()) {
    parameters.rid = rid.UnsafeFromJust();
  }
  if (ssrc.IsJust()) {
-    parameters.ssrc = absl::optional<uint32_t>(ssrc.UnsafeFromJust());
-  }
-  if (codecPayloadType.IsJust()) {
-    // NOTE(mroberts): No longer supported in WebRTC 1.0.
-  }
-  if (dtx.IsJust()) {
-    // NOTE(mroberts): No longer supported in WebRTC 1.0.
+    parameters.ssrc = ssrc.UnsafeFromJust();
  }
  parameters.active = active;
-  // TODO(mroberts): Do something with this.
-  (void) priority;
-  if (ptime.IsJust()) {
-    // NOTE(mroberts): No longer supported in WebRTC 1.0.
-  }
  if (maxBitrate.IsJust()) {
-    parameters.max_bitrate_bps = absl::optional<int>(maxBitrate.UnsafeFromJust());
+    parameters.max_bitrate_bps = maxBitrate.UnsafeFromJust();
  }
  if (maxFramerate.IsJust()) {
-    parameters.max_framerate = absl::optional<int>(maxFramerate.UnsafeFromJust());
+    parameters.max_framerate = maxFramerate.UnsafeFromJust();
  }
  if (scaleResolutionDownBy.IsJust()) {
-    parameters.scale_resolution_down_by = absl::optional<double>(scaleResolutionDownBy.UnsafeFromJust());
+    parameters.scale_resolution_down_by = scaleResolutionDownBy.UnsafeFromJust();
  }
  return Pure(parameters);
 }

diff --git a/src/dictionaries/webrtc/rtp_parameters.cc b/src/dictionaries/webrtc/rtp_parameters.cc
@@ -2,8 +2,10 @@

 #include <webrtc/api/rtp_parameters.h>

+#include "src/converters/absl.h"
 #include "src/converters/object.h"
 #include "src/enums/webrtc/degradation_preference.h"
+#include "src/functional/maybe.h"
 #include "src/dictionaries/macros/napi.h"
 #include "src/dictionaries/webrtc/rtcp_parameters.h"
 #include "src/dictionaries/webrtc/rtp_codec_parameters.h"
@@ -27,10 +29,9 @@ TO_NAPI_IMPL(webrtc::RtpParameters, pair) {
  NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "codecs", parameters.codecs)
  NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "rtcp", parameters.rtcp)
  NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "encodings", parameters.encodings)
-  // TODO: fix this
-  // if (parameters.degradation_preference != webrtc::DegradationPreference::BALANCED) {
-  //   NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "degradationPreference", parameters.degradation_preference)
-  // }
+  if (parameters.degradation_preference && parameters.degradation_preference != webrtc::DegradationPreference::BALANCED) {
+    NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "degradationPreference", parameters.degradation_preference)
+  }
  return Pure(scope.Escape(object));
 }

@@ -40,14 +41,16 @@ static webrtc::RtpParameters NapiToRtpParameters(
    webrtc::RtcpParameters rtcp,
    std::vector<webrtc::RtpCodecParameters> codecs,
    std::vector<webrtc::RtpEncodingParameters> encodings,
-    webrtc::DegradationPreference degradationPreference) {
+    node_webrtc::Maybe<webrtc::DegradationPreference> degradationPreference) {
  webrtc::RtpParameters parameters;
  parameters.transaction_id = transactionId;
  parameters.header_extensions = headerExtensions;
  parameters.rtcp = rtcp;
  parameters.codecs = codecs;
  parameters.encodings = encodings;
-  parameters.degradation_preference = degradationPreference;
+  parameters.degradation_preference = degradationPreference
+  .Map([](auto degradationPreference) { return absl::make_optional(static_cast<webrtc::DegradationPreference>(degradationPreference)); })
+  .FromMaybe(absl::optional<webrtc::DegradationPreference>());
  return parameters;
 }

@@ -59,7 +62,7 @@ FROM_NAPI_IMPL(webrtc::RtpParameters, value) {
        * GetRequired<webrtc::RtcpParameters>(object, "rtcp")
        * GetRequired<std::vector<webrtc::RtpCodecParameters>>(object, "codecs")
        * GetRequired<std::vector<webrtc::RtpEncodingParameters>>(object, "encodings")
-        * GetOptional<webrtc::DegradationPreference>(object, "degradationPreference", webrtc::DegradationPreference::BALANCED);
+        * GetOptional<webrtc::DegradationPreference>(object, "degradationPreference");
  });
 }


diff --git a/src/enums/webrtc/rtp_transceiver_direction.h b/src/enums/webrtc/rtp_transceiver_direction.h
@@ -10,7 +10,8 @@
  ENUM_SUPPORTED(RTP_TRANSCEIVER_DIRECTION::kSendRecv, "sendrecv") \
  ENUM_SUPPORTED(RTP_TRANSCEIVER_DIRECTION::kSendOnly, "sendonly") \
  ENUM_SUPPORTED(RTP_TRANSCEIVER_DIRECTION::kRecvOnly, "recvonly") \
-  ENUM_SUPPORTED(RTP_TRANSCEIVER_DIRECTION::kInactive, "inactive")
+  ENUM_SUPPORTED(RTP_TRANSCEIVER_DIRECTION::kInactive, "inactive") \
+  ENUM_SUPPORTED(RTP_TRANSCEIVER_DIRECTION::kStopped, "stopped")

 #define ENUM(X) RTP_TRANSCEIVER_DIRECTION ## X
 #include "src/enums/macros/decls.h"

diff --git a/src/enums/webrtc/video_frame_buffer_type.h b/src/enums/webrtc/video_frame_buffer_type.h
@@ -11,7 +11,8 @@
  ENUM_SUPPORTED(VIDEO_FRAME_BUFFER_TYPE::kI420, "I420") \
  ENUM_SUPPORTED(VIDEO_FRAME_BUFFER_TYPE::kI420A, "I420A") \
  ENUM_SUPPORTED(VIDEO_FRAME_BUFFER_TYPE::kI444, "I444") \
-  ENUM_SUPPORTED(VIDEO_FRAME_BUFFER_TYPE::kI010, "I010")
+  ENUM_SUPPORTED(VIDEO_FRAME_BUFFER_TYPE::kI010, "I010") \
+  ENUM_SUPPORTED(VIDEO_FRAME_BUFFER_TYPE::kNV12, "NV12")

 #define ENUM(X) VIDEO_FRAME_BUFFER_TYPE ## X
 #include "src/enums/macros/decls.h"

diff --git a/src/interfaces/media_stream.cc b/src/interfaces/media_stream.cc
@@ -27,6 +27,10 @@ Napi::FunctionReference& MediaStream::constructor() {
  return constructor;
 }

+MediaStream::~MediaStream() {
+  wrap()->Release(this);
+}
+
 MediaStream::Impl::Impl(PeerConnectionFactory* factory)
  : _factory(factory ? factory : PeerConnectionFactory::GetOrCreateDefault())
  , _stream(_factory->factory()->CreateLocalMediaStream(rtc::CreateRandomUuid()))

diff --git a/src/interfaces/media_stream.h b/src/interfaces/media_stream.h
@@ -27,7 +27,9 @@ struct RTCMediaStreamInit;
 class MediaStream
  : public Napi::ObjectWrap<MediaStream> {
 public:
-  MediaStream(const Napi::CallbackInfo&);
+  explicit MediaStream(const Napi::CallbackInfo&);
+
+  ~MediaStream();

  static void Init(Napi::Env, Napi::Object);


diff --git a/src/interfaces/rtc_dtls_transport.cc b/src/interfaces/rtc_dtls_transport.cc
@@ -84,6 +84,7 @@ RTCDtlsTransport::~RTCDtlsTransport() {
  Napi::HandleScope scope(PeerConnectionFactory::constructor().Env());
  _factory->Unref();
  _factory = nullptr;
+  wrap()->Release(this);
 }  // NOLINT

 void RTCDtlsTransport::Stop() {

diff --git a/src/interfaces/rtc_peer_connection.cc b/src/interfaces/rtc_peer_connection.cc
@@ -17,6 +17,7 @@
 #include <webrtc/p2p/client/basic_port_allocator.h>

 #include "src/converters.h"
+#include "src/converters/absl.h"
 #include "src/converters/arguments.h"
 #include "src/converters/interfaces.h"
 #include "src/converters/napi.h"
@@ -163,25 +164,28 @@ void RTCPeerConnection::OnIceCandidate(const webrtc::IceCandidateInterface* ice_
 }

 static Validation<Napi::Value> CreateRTCPeerConnectionIceErrorEvent(
-    const Napi::Value hostCandidate,
+    const Napi::Value address,
+    const Napi::Value port,
    const Napi::Value url,
    const Napi::Value errorCode,
    const Napi::Value errorText) {
-  auto env = hostCandidate.Env();
+  auto env = address.Env();
  Napi::EscapableHandleScope scope(env);
  NODE_WEBRTC_CREATE_OBJECT_OR_RETURN(env, object)
-  NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "hostCandidate", hostCandidate)
+  NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "address", address)
+  NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "port", port)
  NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "url", url)
  NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "errorCode", errorCode)
  NODE_WEBRTC_CONVERT_AND_SET_OR_RETURN(env, object, "errorText", errorText)
  return Pure(scope.Escape(object));
 }

-void RTCPeerConnection::OnIceCandidateError(const std::string& host_candidate, const std::string& url, int error_code, const std::string& error_text) {
-  Dispatch(CreateCallback<RTCPeerConnection>([this, host_candidate, url, error_code, error_text]() {
+void RTCPeerConnection::OnIceCandidateError(const std::string& address, int port, const std::string& url, int error_code, const std::string& error_text) {
+  Dispatch(CreateCallback<RTCPeerConnection>([this, address, port, url, error_code, error_text]() {
    auto env = Env();
    auto maybeEvent = Validation<Napi::Value>::Join(curry(CreateRTCPeerConnectionIceErrorEvent)
-            % From<Napi::Value>(std::make_pair(env, host_candidate))
+            % From<Napi::Value>(std::make_pair(env, address))
+            * From<Napi::Value>(std::make_pair(env, port))
            * From<Napi::Value>(std::make_pair(env, url))
            * From<Napi::Value>(std::make_pair(env, error_code))
            * From<Napi::Value>(std::make_pair(env, error_text)));
@@ -315,9 +319,10 @@ Napi::Value RTCPeerConnection::RemoveTrack(const Napi::CallbackInfo& info) {
    Napi::Error(env, ErrorFactory::CreateInvalidAccessError(env, "Cannot removeTrack")).ThrowAsJavaScriptException();
    return env.Undefined();
  }
-  if (!_jinglePeerConnection->RemoveTrack(sender->sender())) {
-    Napi::Error(env, ErrorFactory::CreateInvalidAccessError(env, "Cannot removeTrack")).ThrowAsJavaScriptException();
-    return env.Undefined();
+  auto error = _jinglePeerConnection->RemoveTrackNew(sender->sender());
+  if (!error.ok()) {
+    CONVERT_OR_THROW_AND_RETURN_NAPI(info.Env(), &error, result, Napi::Value)
+    Napi::Error(info.Env(), result).ThrowAsJavaScriptException();
  }
  return env.Undefined();
 }
@@ -618,7 +623,13 @@ Napi::Value RTCPeerConnection::RestartIce(const Napi::CallbackInfo& info) {
 }

 Napi::Value RTCPeerConnection::GetCanTrickleIceCandidates(const Napi::CallbackInfo& info) {
-  return info.Env().Null();
+  auto env = info.Env();
+  if (!_jinglePeerConnection) {
+    return env.Null();
+  }
+  auto canTrickleIceCandidates = _jinglePeerConnection->can_trickle_ice_candidates();
+  CONVERT_OR_THROW_AND_RETURN_NAPI(env, canTrickleIceCandidates, result, Napi::Value)
+  return result;
 }

 Napi::Value RTCPeerConnection::GetConnectionState(const Napi::CallbackInfo& info) {

diff --git a/src/interfaces/rtc_peer_connection.h b/src/interfaces/rtc_peer_connection.h
@@ -47,7 +47,7 @@ class RTCPeerConnection
  void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
  void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
-  void OnIceCandidateError(const std::string& host_candidate, const std::string& url, int error_code, const std::string& error_text) override;
+  void OnIceCandidateError(const std::string& address, int port, const std::string& url, int error_code, const std::string& error_text) override;
  void OnRenegotiationNeeded() override;

  void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;