stuff

Makefile

@@ -7,13 +7,13 @@ WEBRTC_GN_ARGS ?= \
 	use_sysroot=false \
 	rtc_include_tests=false \
 	rtc_use_h264=true \
+	rtc_use_x11=false \
 	treat_warnings_as_errors=false \
 	is_clang=false \
 	use_custom_libcxx=false \
 	use_rtti=true \
 	$(WEBRTC_GN_ARGS_EXTRA)
 WEBRTC_TARGETS ?= default $(WEBRTC_TARGETS_EXTRA)
-WEBRTC_LIBS ?= libwebrtc.a $(WEBRTC_LIBS_EXTRA)
 WEBRTC_INCLUDES = \
 	api audio base call common_audio common_video logging media modules p2p pc \
 	system_wrappers rtc_base rtc_tools video third_party common_types.h \
@@ -22,8 +22,9 @@ WEBRTC_INCLUDES = \
 WEBRTC_PLATFORM ?= POSIX
 
 WEBRTC_FLAGS ?= \
-	-I$(WEBRTC_BASE)/include -I$(WEBRTC_BASE)/include/webrtc \
-	-I$(WEBRTC_BASE)/include/webrtc/third_party/abseil-cpp \
+	-I$(WEBRTC_BASE)/include \
+	-I$(WEBRTC_DIR)/include -I$(WEBRTC_DIR)/include/webrtc \
+	-I$(WEBRTC_DIR)/include/webrtc/third_party/abseil-cpp \
 	-DWEBRTC_$(WEBRTC_PLATFORM)
 
 WEBRTC_SETPATH = PATH=$(WEBRTC_DIR)/depot_tools:"$$PATH"
@@ -31,13 +32,12 @@ RTC_STAMP_DEPOT = $(WEBRTC_DIR)/depot_tools/.stamp
 RTC_STAMP_REPO = $(WEBRTC_DIR)/webrtc-checkout/.stamp
 RTC_STAMP_VERSION = $(WEBRTC_DIR)/webrtc-checkout/.stamps/stamp-$(WEBRTC_VERSION)
 RTC_STAMP_COMPILE = $(WEBRTC_DIR)/webrtc-checkout/src/out/$(WEBRTC_OUT)/.stamp
-RTC_STAMP_INCLUDES = $(WEBRTC_BASE)/include/webrtc/.stamp
+RTC_STAMP_INCLUDES = $(WEBRTC_DIR)/include/webrtc/.stamp
 
 .PHONY: rtcall
 rtc-all: \
-		$(WEBRTC_BASE)/include/webrtc/.stamp \
-		$(patsubst %,$(WEBRTC_BASE)/libs/%,$(WEBRTC_LIBS)) \
-		$(WEBRTC_BASE)/libs/libutil.a
+		$(WEBRTC_DIR)/include/webrtc/.stamp \
+		$(WEBRTC_DIR)/lib/libwebrtc.a
 
 # Get depot_tools, which contains all the tools required to work with Google software
 $(RTC_STAMP_DEPOT):
@@ -70,30 +70,23 @@ $(RTC_STAMP_COMPILE):
 
 # Includes
 $(RTC_STAMP_INCLUDES): $(RTC_STAMP_VERSION)
-	rm -rf include/webrtc && mkdir -p include/webrtc
+	rm -rf $(WEBRTC_DIR)/include/webrtc && mkdir -p $(WEBRTC_DIR)/include/webrtc
 	rsync -m --recursive --include='*/' --include='*.h' --exclude='*' \
 		$(patsubst %,$(WEBRTC_DIR)/webrtc-checkout/src/%,$(WEBRTC_INCLUDES)) \
-		include/webrtc
+		$(WEBRTC_DIR)/include/webrtc
 	touch $@
 
 # Libraries
+.PRECIOUS: $(WEBRTC_DIR)/webrtc-checkout/src/out/$(WEBRTC_OUT)/obj/%.a
 $(WEBRTC_DIR)/webrtc-checkout/src/out/$(WEBRTC_OUT)/obj/%.a: $(RTC_STAMP_COMPILE)
-	touch $@ # Make doesn't believe us unless there's a recipe body
-$(WEBRTC_BASE)/libs/%.a: $(WEBRTC_DIR)/webrtc-checkout/src/out/$(WEBRTC_OUT)/obj/%.a
+	touch $@
+$(WEBRTC_DIR)/lib/%.a: $(WEBRTC_DIR)/webrtc-checkout/src/out/$(WEBRTC_OUT)/obj/%.a
 	@mkdir -p $(@D)
 	cp $< $@
 
-# Util
-$(WEBRTC_BASE)/libs/libutil.a: $(WEBRTC_BASE)/build/util/Conductor.cc.o
-	@mkdir -p $(@D)
-	ar cr $@ $^
-$(WEBRTC_BASE)/build/%.cc.o: $(WEBRTC_BASE)/%.cc $(RTC_STAMP_INCLUDES) $(wildcard $(WEBRTC_BASE)/util/*.h)
-	@mkdir -p $(@D)
-	$(CXX) $(CXXFLAGS) -I$(WEBRTC_BASE)/include/rtcutil $(WEBRTC_FLAGS) -c $< -o $@
-
 .PHONY: rtc-clean
 rtc-clean:
-	rm -rf include/webrtc libs build
+	rm -rf $(WEBRTC_DIR)/include/webrtc $(WEBRTC_DIR)/lib
 
 .PHONY: rtc-gn-clean
 rtc-gn-clean: rtc-clean
@@ -104,6 +97,10 @@ rtc-gn-clean: rtc-clean
 rtc-clean-all: rtc-clean
 	rm -rf $(WEBRTC_DIR)/depot_tools $(WEBRTC_DIR)/webrtc-checkout
 
+.PHONY: rtc-redo
+rtc-redo:
+	rm -f $(RTC_STAMP_COMPILE)
+
 .PHONY: rtc-mock-stamps
 rtc-mock-stamps:
 	mkdir -p $(dir $(RTC_STAMP_DEPOT)) && touch $(RTC_STAMP_DEPOT)

include/rtcutil/Conductor.h

@@ -16,8 +16,8 @@ public:
 
 	virtual void onIceCandidate(std::string mid, int index, std::string sdp) = 0;
 	virtual void onAnswer(std::string sdp) = 0;
-	virtual void onRemoteTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track);
-	virtual void removeRemoteTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track);
+	virtual void onRemoteTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track) = 0;
+	virtual void removeRemoteTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track) = 0;
 };
 
 class Conductor:

util/Conductor.cc

@@ -1,4 +1,4 @@
-#include "Conductor.h"
+#include <rtcutil/Conductor.h>
 
 #include <webrtc/api/create_peerconnection_factory.h>
 #include <webrtc/api/audio_codecs/builtin_audio_encoder_factory.h>
@@ -39,6 +39,7 @@ private:
 };
 
 void Conductor::receiveOffer(const std::string &sdp) {
+	printf("receive offer 1\n");
 	webrtc::SdpParseError error;
 	std::unique_ptr<webrtc::SessionDescriptionInterface> desc =
 		webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, sdp, &error);
@@ -52,6 +53,7 @@ void Conductor::receiveOffer(const std::string &sdp) {
 }
 
 void Conductor::receiveOffer(std::unique_ptr<webrtc::SessionDescriptionInterface> offer) {
+	printf("receive offer 2\n");
 	ensurePeerConnection();
 
 	peer_conn_->SetRemoteDescription(
@@ -63,6 +65,7 @@ void Conductor::receiveOffer(std::unique_ptr<webrtc::SessionDescriptionInterface
 }
 
 void Conductor::receiveIceCandidate(std::string mid, int index, std::string sdp) {
+	printf("receive ice candidate\n");
 	webrtc::SdpParseError error;
 	std::unique_ptr<webrtc::IceCandidateInterface> candidate(
 			webrtc::CreateIceCandidate(mid, index, sdp, &error));
@@ -148,6 +151,7 @@ Conductor::doCreatePeerConnection(webrtc::PeerConnectionFactoryInterface &factor
 }
 
 void Conductor::OnSuccess(webrtc::SessionDescriptionInterface *desc) {
+	fprintf(stderr, "on success; got answer\n");
 	peer_conn_->SetLocalDescription(
 			SetSessionDescriptionObserver::Create(observer_), desc);