Drop unused audio processing and video render smoothing

python-webrtc/cpp/src/interfaces/peer_connection_factory.cpp

@@ -17,7 +17,6 @@
 #include "../utils/port_allocator.h"
 #include "../utils/task_queue_factory.h"
 
-#include <api/audio/builtin_audio_processing_builder.h>
 #include <api/audio_codecs/builtin_audio_decoder_factory.h>
 #include <api/audio_codecs/builtin_audio_encoder_factory.h>
 #include <api/create_modular_peer_connection_factory.h>
@@ -84,7 +83,7 @@ namespace python_webrtc {
     }
   } // namespace
 
-  // webrtc::CreatePeerConnectionFactory without the RTC event log, which keeps every packet's buffer in memory
+  // webrtc::CreatePeerConnectionFactory without the RTC event log (holds every packet) or audio processing (no capture)
   webrtc::PeerConnectionFactoryDependencies PeerConnectionFactory::Dependencies() {
     webrtc::PeerConnectionFactoryDependencies dependencies;
     dependencies.network_thread = _workerThread.get();
@@ -95,7 +94,6 @@ namespace python_webrtc {
     dependencies.adm = _audioDeviceModule;
     dependencies.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
     dependencies.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
-    dependencies.audio_processing_builder = std::make_unique<webrtc::BuiltinAudioProcessingBuilder>();
     dependencies.video_encoder_factory = std::make_unique<VideoEncoderFactory>();
     dependencies.video_decoder_factory = std::make_unique<VideoDecoderFactory>();
     webrtc::EnableMedia(dependencies);
@@ -117,9 +115,8 @@ namespace python_webrtc {
 
   PeerConnectionFactory::PeerConnectionFactory()
       : _fieldTrials(_trials), _networkIgnoreMask(_loopbackAllowed ? 0 : webrtc::kDefaultNetworkIgnoreMask) {
-    _workerThread = Started(std::make_unique<webrtc::Thread>(std::make_unique<NetworkSocketServer>()),
-                            "PeerConnectionFactory:workerThread");
-    _signalingThread = Started(webrtc::Thread::Create(), "PeerConnectionFactory:signalingThread");
+    _workerThread = Started(std::make_unique<webrtc::Thread>(std::make_unique<NetworkSocketServer>()), "wrtc-network");
+    _signalingThread = Started(webrtc::Thread::Create(), "wrtc-signaling");
     // sanitizer builds check that these threads never enter Python
     _workerThread->PostTask(&TagNativeThread);
     _signalingThread->PostTask(&TagNativeThread);

python-webrtc/cpp/src/interfaces/rtc_data_channel.cpp

@@ -204,7 +204,7 @@ namespace python_webrtc {
   }
 
   webrtc::Priority RTCDataChannel::GetPriority() {
-    // the ranges Chromium maps priority values to
+    // splits at the midpoints between the DCEP priority values of W3C webrtc-priority (128, 256, 512, 1024)
     constexpr int veryLowMax = 192;
     constexpr int lowMax = 384;
     constexpr int mediumMax = 768;

python-webrtc/cpp/src/interfaces/rtc_peer_connection/rtc_peer_connection.cpp

@@ -32,6 +32,8 @@ namespace python_webrtc {
     auto configuration = webrtc::PeerConnectionInterface::RTCConfiguration();
     configuration.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
     configuration = _configuration.Apply(configuration);
+    // renders each decoded frame at once, without a smoothing queue per video receiver
+    configuration.set_prerenderer_smoothing(false);
 
     webrtc::PeerConnectionDependencies dependencies(this);
     dependencies.allocator = _factory->CreatePortAllocator(configuration);

python-webrtc/cpp/src/interfaces/rtc_rtp_sender.cpp

@@ -22,7 +22,7 @@ namespace python_webrtc {
 
   namespace {
 
-    // only what setParameters may change, as in Chromium
+    // only the encoding members setParameters may change, the rest keep their current values
     void applySettable(const webrtc::RtpEncodingParameters &requested, webrtc::RtpEncodingParameters &current) {
       current.active = requested.active;
       current.request_key_frame = requested.request_key_frame;

python-webrtc/python/webrtc/exceptions.py

@@ -212,7 +212,7 @@ def __init__(self, init: RTCErrorInit, message: str = '') -> None:
 
 
 _BY_RTC_ERROR_TYPE = {
-    # the same mapping as Chromium's
+    # the W3C error names for libwebrtc's RTCErrorType values
     'UNSUPPORTED_OPERATION': OperationError,
     'UNSUPPORTED_PARAMETER': InvalidAccessError,
     'INVALID_PARAMETER': InvalidAccessError,

python-webrtc/python/webrtc/models/rtp_parameters.py

@@ -31,7 +31,7 @@
 
 _NativeCodecT = TypeVar('_NativeCodecT', bound='wrtc.RtpCodec')
 
-# the bitrate priorities of libwebrtc for RTCRtpEncodingParameters.priority, as Chromium maps them
+# libwebrtc's bitrate priorities for RTCRtpEncodingParameters.priority, doubling per level from low (the default) at 1.0
 _BITRATE_PRIORITY = {
     RTCPriorityType.very_low: 0.5,
     RTCPriorityType.low: 1.0,