Implements extended API for creating RtcRtpTransceivers

CHANGELOG.md

@@ -11,6 +11,10 @@ All user visible changes to this project will be documented in this file. This p
 
 [Diff](https://github.com/instrumentisto/medea-flutter-webrtc/compare/0.8.2...0.8.3)
 
+### Added
+
+- `RtpTransceiverInit.sendEncodings` field with `SendEncodingParameters`s. ([#125])
+
 ### Changed
 
 - Refactor Audio Device Module to use [OpenAL] library for playout. ([#117])
@@ -21,6 +25,7 @@ All user visible changes to this project will be documented in this file. This p
 [#119]: https://github.com/instrumentisto/medea-flutter-webrtc/pull/119
 [#120]: https://github.com/instrumentisto/medea-flutter-webrtc/pull/120
 [#123]: https://github.com/instrumentisto/medea-flutter-webrtc/pull/123
+[#125]: https://github.com/instrumentisto/medea-flutter-webrtc/pull/125
 [116.0.5845.110]: https://github.com/instrumentisto/libwebrtc-bin/releases/tag/116.0.5845.110
 
 

Makefile

@@ -134,7 +134,7 @@ flutter.run:
 # Run Flutter plugin integration tests on the current host as desktop.
 #
 # Usage:
-#	make flutter.test [debug=(no|yes)]
+#	make flutter.test.desktop [debug=(no|yes)]
 
 flutter.test.desktop:
 	cd example/ && \
@@ -302,7 +302,8 @@ endif
 		--dart-output=lib/src/api/bridge.g.dart \
 		--skip-add-mod-to-lib \
 		--no-build-runner \
-		--dart-format-line-length=80
+		--dart-enums-style \
+		--inline-rust
 	flutter pub run build_runner build --delete-conflicting-outputs
 
 
@@ -392,6 +393,7 @@ endif
 swift.fmt:
 ifeq ($(dockerized),yes)
 	docker run --rm -v "$(PWD)":/app -w /app \
+		-u $(shell id -u):$(shell id -g) \
 		ghcr.io/nicklockwood/swiftformat:latest \
 			$(if $(call eq,$(check),yes),--lint,) ios/Classes/
 else

android/gradle/wrapper/gradle-wrapper.properties

@@ -1 +1,2 @@
-distributionUrl=https\://services.gradle.org/distributions/gradle-7.2-bin.zip
+#Thu Sep 28 10:50:19 CEST 2023
+distributionUrl=https\://services.gradle.org/distributions/gradle-7.4.2-bin.zip

android/src/main/kotlin/com/instrumentisto/medea_flutter_webrtc/model/Encoding.kt

@@ -0,0 +1,48 @@
+package com.instrumentisto.medea_flutter_webrtc.model
+
+import org.webrtc.RtpParameters.Encoding as WEncoding
+
+/**
+ * Representation of an [org.webrtc.RtpParameters.Encoding].
+ *
+ * @property rid 'rid' of this encoding parameters, created from this config.
+ * @property active Indicates whether this parameters are active.
+ * @property maxBitrate Maximum bitrate for this parameters.
+ * @property maxFramerate Maximum framerate for this parameters.
+ * @property scaleResolutionDownBy Resolution will be scaled down for this parameters.
+ */
+data class Encoding(
+    var rid: String,
+    var active: Boolean,
+    var maxBitrate: Int?,
+    var maxFramerate: Double?,
+    var scaleResolutionDownBy: Double?,
+) {
+  companion object {
+    /**
+     * Creates a new [Encoding] object based on the method call received from the Flutter side.
+     *
+     * @return [Encoding] created from the provided [Map].
+     */
+    fun fromMap(map: Map<String, Any>): Encoding {
+      return Encoding(
+          map["rid"] as String,
+          map["active"] as Boolean,
+          map["maxBitrate"] as Int?,
+          map["maxFramerate"] as Double?,
+          map["scaleResolutionDownBy"] as Double?)
+    }
+  }
+
+  /**
+   * Converts this [Encoding] into an [org.webrtc.RtpParameters.Encoding].
+   *
+   * @return [org.webrtc.RtpParameters.Encoding] created based on this [Encoding].
+   */
+  fun intoWebRtc(): WEncoding {
+    var encoding: WEncoding = WEncoding(rid, active, scaleResolutionDownBy)
+    encoding.maxBitrateBps = maxBitrate
+    encoding.maxFramerate = maxFramerate?.toInt()
+    return encoding
+  }
+}

android/src/main/kotlin/com/instrumentisto/medea_flutter_webrtc/model/RtpTransceiverInit.kt

@@ -6,8 +6,12 @@ import org.webrtc.RtpTransceiver.RtpTransceiverInit as WRtpTransceiverInit
  * Representation of an [org.webrtc.RtpTransceiver.RtpTransceiverInit].
  *
  * @property direction Direction of the transceiver, created from this config.
+ * @property encodings [List] of the [Encoding]s, created from this config.
  */
-data class RtpTransceiverInit(val direction: RtpTransceiverDirection) {
+data class RtpTransceiverInit(
+    val direction: RtpTransceiverDirection,
+    var encodings: List<Encoding>
+) {
   companion object {
     /**
      * Creates a new [RtpTransceiverInit] object based on the method call received from the Flutter
@@ -16,7 +20,11 @@ data class RtpTransceiverInit(val direction: RtpTransceiverDirection) {
      * @return [RtpTransceiverInit] created from the provided [Map].
      */
     fun fromMap(map: Map<String, Any>): RtpTransceiverInit {
-      return RtpTransceiverInit(RtpTransceiverDirection.fromInt(map["direction"] as Int))
+      return RtpTransceiverInit(
+          RtpTransceiverDirection.fromInt(map["direction"] as Int),
+          (map["sendEncodings"] as List<Map<String, Map<String, Any>>>).map { encoding ->
+            Encoding.fromMap(encoding)
+          })
     }
   }
 
@@ -27,6 +35,7 @@ data class RtpTransceiverInit(val direction: RtpTransceiverDirection) {
    * [RtpTransceiverInit].
    */
   fun intoWebRtc(): WRtpTransceiverInit {
-    return WRtpTransceiverInit(direction.intoWebRtc())
+    return WRtpTransceiverInit(
+        direction.intoWebRtc(), listOf(), encodings.map { e -> e.intoWebRtc() })
   }
 }

crates/libwebrtc-sys/include/bridge.h

@@ -10,11 +10,11 @@
 #include "api/video_codecs/builtin_video_encoder_factory.h"
 #include "api/video_track_source_proxy_factory.h"
 #if __APPLE__
-  #include "libwebrtc-sys/include/device_info_mac.h"
-  #include "mac_capturer.h"
-  #include "device_info_mac.h"
+#include "device_info_mac.h"
+#include "libwebrtc-sys/include/device_info_mac.h"
+#include "mac_capturer.h"
 #else
-  #include "device_video_capturer.h"
+#include "device_video_capturer.h"
 #endif
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/video_capture/video_capture_factory.h"
@@ -30,8 +30,8 @@
 #include "adm_proxy.h"
 
 #include "media/base/fake_frame_source.h"
-#include "pc/test/fake_video_track_source.h"
 #include "modules/audio_device/include/test_audio_device.h"
+#include "pc/test/fake_video_track_source.h"
 
 namespace bridge {
 
@@ -172,8 +172,9 @@ int32_t set_audio_recording_device(const AudioDeviceModule& audio_device_module,
 int32_t stop_playout(const AudioDeviceModule& audio_device_module);
 
 // Sets stereo availability of the specified playout device.
-int32_t stereo_playout_is_available(const AudioDeviceModule& audio_device_module,
-                                    bool available);
+int32_t stereo_playout_is_available(
+    const AudioDeviceModule& audio_device_module,
+    bool available);
 
 // Initializes the specified audio playout device.
 int32_t init_playout(const AudioDeviceModule& audio_device_module);
@@ -383,6 +384,21 @@ std::unique_ptr<RTCOfferAnswerOptions> create_rtc_offer_answer_options(
     bool ice_restart,
     bool use_rtp_mux);
 
+// Creates a new default `RtpTransceiverInit`.
+std::unique_ptr<RtpTransceiverInit> create_default_rtp_transceiver_init();
+
+// Sets a `RtpTransceiverDirection` for the provided `RtpTransceiverInit`.
+void set_rtp_transceiver_init_direction(RtpTransceiverInit& init,
+                                        RtpTransceiverDirection direction);
+
+// Adds a `RtpEncodingParameters` into the provided `RtpTransceiverInit`.
+void add_rtp_transceiver_init_send_encoding(
+    RtpTransceiverInit& init,
+    const RtpEncodingParametersContainer& params);
+
+// Creates a new default `RtpEncodingParameters`.
+RtpEncodingParametersContainer create_rtp_encoding_parameters();
+
 // Creates a new `CreateSessionDescriptionObserver` from the provided
 // `bridge::DynCreateSdpCallback`.
 std::unique_ptr<CreateSessionDescriptionObserver>

crates/libwebrtc-sys/include/peer_connection.h

@@ -17,6 +17,7 @@ using SessionDescriptionInterface = webrtc::SessionDescriptionInterface;
 using RtpTransceiverInterface =
     rtc::scoped_refptr<webrtc::RtpTransceiverInterface>;
 using RtpTransceiverDirection = webrtc::RtpTransceiverDirection;
+using RtpTransceiverInit = webrtc::RtpTransceiverInit;
 
 struct TransceiverContainer;
 struct DynPeerConnectionEventsHandler;
@@ -214,7 +215,7 @@ void set_remote_description(PeerConnectionInterface& peer,
 std::unique_ptr<RtpTransceiverInterface> add_transceiver(
     PeerConnectionInterface& peer,
     cricket::MediaType media_type,
-    RtpTransceiverDirection direction);
+    const RtpTransceiverInit& init);
 
 // Returns a list of `RtpTransceiverInterface`s attached to the provided
 // `PeerConnectionInterface`.

crates/libwebrtc-sys/include/rtp_encoding_parameters.h

@@ -1,25 +1,44 @@
 #pragma once
 
 #include "bridge.h"
+#include "rust/cxx.h"
 
 namespace bridge {
 
+// Sets the `RtpEncodingParameters.rid` field value.
+void set_rtp_encoding_parameters_rid(webrtc::RtpEncodingParameters& encoding,
+                                     rust::String rid);
+
 // Returns the `RtpEncodingParameters.active` field value.
 bool rtp_encoding_parameters_active(
     const webrtc::RtpEncodingParameters& encoding);
 
+// Sets the `RtpEncodingParameters.active` field value.
+void set_rtp_encoding_parameters_active(webrtc::RtpEncodingParameters& encoding,
+                                        bool active);
+
 // Returns the `RtpEncodingParameters.maxBitrate` field value.
-int32_t rtp_encoding_parameters_maxBitrate(
+int32_t rtp_encoding_parameters_max_bitrate(
     const webrtc::RtpEncodingParameters& encoding);
 
+// Returns the `RtpEncodingParameters.maxBitrate` field value.
+void set_rtp_encoding_parameters_max_bitrate(
+    webrtc::RtpEncodingParameters& encoding,
+    int32_t max_bitrate);
+
 // Returns the `RtpEncodingParameters.minBitrate` field value.
-int32_t rtp_encoding_parameters_minBitrate(
+int32_t rtp_encoding_parameters_min_bitrate(
     const webrtc::RtpEncodingParameters& encoding);
 
 // Returns the `RtpEncodingParameters.maxFramerate` field value.
-double rtp_encoding_parameters_maxFramerate(
+double rtp_encoding_parameters_max_framerate(
     const webrtc::RtpEncodingParameters& encoding);
 
+// Sets the `RtpEncodingParameters.maxFramerate` field value.
+void set_rtp_encoding_parameters_max_framerate(
+    webrtc::RtpEncodingParameters& encoding,
+    double max_framrate);
+
 // Returns the `RtpEncodingParameters.ssrc` field value.
 int64_t rtp_encoding_parameters_ssrc(
     const webrtc::RtpEncodingParameters& encoding);
@@ -28,4 +47,14 @@ int64_t rtp_encoding_parameters_ssrc(
 double rtp_encoding_parameters_scale_resolution_down_by(
     const webrtc::RtpEncodingParameters& encoding);
 
+// Sets the `RtpEncodingParameters.scale_resolution_down_by` field value.
+void set_rtp_encoding_parameters_scale_resolution_down_by(
+    webrtc::RtpEncodingParameters& encoding,
+    double scale_resolution_down_by);
+
+// Sets the `RtpEncodingParameters.scalability_mode` field value.
+void set_rtp_encoding_parameters_scalability_mode(
+    webrtc::RtpEncodingParameters& encoding,
+    rust::String scalability_mode);
+
 }  // namespace bridge

crates/libwebrtc-sys/src/bridge.rs

@@ -1519,41 +1519,90 @@ pub(crate) mod webrtc {
         ) -> bool;
     }
 
+    #[rustfmt::skip]
+    unsafe extern "C++" {
+        pub type RtpTransceiverInit;
+
+        /// Creates a new default [`RtpTransceiverInit`].
+        #[must_use]
+        pub fn create_default_rtp_transceiver_init() -> UniquePtr<RtpTransceiverInit>;
+
+        /// Sets a [`RtpTransceiverDirection`] for the provided [`RtpTransceiverInit`].
+        pub fn set_rtp_transceiver_init_direction(
+            init: Pin<&mut RtpTransceiverInit>,
+            direction: RtpTransceiverDirection
+        );
+
+        /// Adds a [`RtpEncodingParameters`] into the provided [`RtpTransceiverInit`].
+        pub fn add_rtp_transceiver_init_send_encoding(
+            init: Pin<&mut RtpTransceiverInit>,
+            encoding: &RtpEncodingParametersContainer
+        );
+
+        /// Creates a new default [`RtpEncodingParameters`].
+        #[must_use]
+        pub fn create_rtp_encoding_parameters() -> RtpEncodingParametersContainer;
+    }
+
     #[rustfmt::skip]
     unsafe extern "C++" {
         include!("libwebrtc-sys/include/rtp_encoding_parameters.h");
 
         #[namespace = "webrtc"]
         pub type RtpEncodingParameters;
 
+        /// Sets the `rid` of the provided [`RtpEncodingParameters`].
+        pub fn set_rtp_encoding_parameters_rid(
+            encoding: Pin<&mut RtpEncodingParameters>,
+            rid: String
+        );
+
         /// Returns the `active` of the provided [`RtpEncodingParameters`].
         #[must_use]
         pub fn rtp_encoding_parameters_active(
             encoding: &RtpEncodingParameters,
         ) -> bool;
 
+        /// Sets the `active` of the provided [`RtpEncodingParameters`].
+        pub fn set_rtp_encoding_parameters_active(
+            encoding: Pin<&mut RtpEncodingParameters>,
+            active: bool
+        );
+
         /// Returns the `maxBitrate` of the provided [`RtpEncodingParameters`].
         ///
         /// [`Result::Err`] means [`None`].
-        pub fn rtp_encoding_parameters_maxBitrate(
+        pub fn rtp_encoding_parameters_max_bitrate(
             encoding: &RtpEncodingParameters,
         ) -> Result<i32>;
 
+        /// Sets the `maxBitrate` of the provided [`RtpEncodingParameters`].
+        pub fn set_rtp_encoding_parameters_max_bitrate(
+            encoding: Pin<&mut RtpEncodingParameters>,
+            max_bitrate: i32
+        );
+
         /// Returns the `minBitrate` of the provided [`RtpEncodingParameters`].
         ///
         /// [`Result::Err`] means [`None`].
-        pub fn rtp_encoding_parameters_minBitrate(
+        pub fn rtp_encoding_parameters_min_bitrate(
             encoding: &RtpEncodingParameters,
         ) -> Result<i32>;
 
         /// Returns the `maxFramerate` of the provided
         /// [`RtpEncodingParameters`].
         ///
         /// [`Result::Err`] means [`None`].
-        pub fn rtp_encoding_parameters_maxFramerate(
+        pub fn rtp_encoding_parameters_max_framerate(
             encoding: &RtpEncodingParameters,
         ) -> Result<f64>;
 
+        /// Sets the `maxFramerate` of the provided [`RtpEncodingParameters`].
+        pub fn set_rtp_encoding_parameters_max_framerate(
+            encoding: Pin<&mut RtpEncodingParameters>,
+            max_framrate: f64
+        );
+
         /// Returns the `ssrc` of the provided [`RtpEncodingParameters`].
         ///
         /// [`Result::Err`] means [`None`].
@@ -1568,6 +1617,20 @@ pub(crate) mod webrtc {
         pub fn rtp_encoding_parameters_scale_resolution_down_by(
             encoding: &RtpEncodingParameters,
         ) -> Result<f64>;
+
+        /// Sets the `scale_resolution_down_by` of the provided
+        /// [`RtpEncodingParameters`].
+        pub fn set_rtp_encoding_parameters_scale_resolution_down_by(
+            encoding: Pin<&mut RtpEncodingParameters>,
+            scale_resolution_down_by: f64
+        );
+
+        /// Sets the `scalability_mode` of the provided
+        /// [`RtpEncodingParameters`].
+        pub fn set_rtp_encoding_parameters_scalability_mode(
+            encoding: Pin<&mut RtpEncodingParameters>,
+            scalability_mode: String
+        );
     }
 
     #[rustfmt::skip]
@@ -2031,7 +2094,7 @@ pub(crate) mod webrtc {
         pub fn add_transceiver(
             peer_connection_interface: Pin<&mut PeerConnectionInterface>,
             media_type: MediaType,
-            direction: RtpTransceiverDirection
+            init: &RtpTransceiverInit
         ) -> UniquePtr<RtpTransceiverInterface>;
 
         /// Returns a sequence of [`RtpTransceiverInterface`] objects