RtpTransceiver bindings & requirements for publishing tracks (#39)

- RtpSender
- RtpReceiver
- RtpTransceiver
- RtpParameters
- VideoFrameBuilder
- Interior mutability on c++ side
- Cleanup bindings ( Use #pragma once instead of include guards )
     - webrtc-sys/src/* headers are now used in only one place 
     - Avoid confusion between generated headers and our headers
- AdaptedVideoTrackSource
- Cleanup the workaround with unsupported Vec<Shared<T>>
    - Put everything inside one file

webrtc-sys/include/livekit/rtp_parameters.h

@@ -0,0 +1,45 @@
+#pragma once
+
+#include <memory>
+
+#include "api/media_types.h"
+#include "api/priority.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_transceiver_direction.h"
+#include "webrtc-sys/src/rtp_parameters.rs.h"
+
+namespace livekit {
+
+webrtc::RtcpFeedback to_native_rtcp_feedback(RtcpFeedback feedback);
+webrtc::RtpCodecCapability to_native_rtp_codec_capability(
+    RtpCodecCapability capability);
+webrtc::RtpHeaderExtensionCapability to_native_rtp_header_extension_capability(
+    RtpHeaderExtensionCapability header);
+webrtc::RtpExtension to_native_rtp_extension(RtpExtension ext);
+webrtc::RtpFecParameters to_rtp_fec_parameters(RtpFecParameters fec);
+webrtc::RtpRtxParameters to_rtp_rtx_parameters(RtpRtxParameters rtx);
+webrtc::RtpEncodingParameters to_native_rtp_encoding_paramters(
+    RtpEncodingParameters parameters);
+webrtc::RtpCodecParameters to_native_rtp_codec_parameters(
+    RtpCodecParameters params);
+webrtc::RtpCapabilities to_rtp_capabilities(RtpCapabilities capabilities);
+webrtc::RtcpParameters to_native_rtcp_paramaters(RtcpParameters params);
+webrtc::RtpParameters to_native_rtp_parameters(RtpParameters params);
+
+RtcpFeedback to_rust_rtcp_feedback(webrtc::RtcpFeedback feedback);
+RtpCodecCapability to_rust_rtp_codec_capability(
+    webrtc::RtpCodecCapability capability);
+RtpHeaderExtensionCapability to_rust_rtp_header_extension_capability(
+    webrtc::RtpHeaderExtensionCapability header);
+RtpExtension to_rust_rtp_extension(webrtc::RtpExtension ext);
+RtpFecParameters to_rust_rtp_fec_parameters(webrtc::RtpFecParameters fec);
+RtpRtxParameters to_rust_rtp_rtx_parameters(webrtc::RtpRtxParameters param);
+RtpEncodingParameters to_rust_rtp_encoding_parameters(
+    webrtc::RtpEncodingParameters params);
+RtpCodecParameters to_rust_rtp_codec_parameters(
+    webrtc::RtpCodecParameters params);
+RtpCapabilities to_rust_rtp_capabilities(webrtc::RtpCapabilities capabilities);
+RtcpParameters to_rust_rtcp_parameters(webrtc::RtcpParameters params);
+RtpParameters to_rust_rtp_parameters(webrtc::RtpParameters params);
+
+}  // namespace livekit