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Author SHA1 Message Date
679ecfc066 Bumped version to 0.10.5 2021-01-21 14:40:21 +01:00
2518 changed files with 498 additions and 7415 deletions

3
.gitignore vendored
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@ -5,7 +5,6 @@ node_modules/
*.a
*.so
compile_commands.json
/tests
tests
.DS_Store
.idea

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@ -1,6 +1,6 @@
cmake_minimum_required(VERSION 3.7)
project(libdatachannel
VERSION 0.11.7
VERSION 0.10.5
LANGUAGES CXX)
set(PROJECT_DESCRIPTION "WebRTC Data Channels Library")
@ -58,39 +58,13 @@ set(LIBDATACHANNEL_SOURCES
${CMAKE_CURRENT_SOURCE_DIR}/src/log.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/message.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/peerconnection.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/logcounter.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/rtcpreceivingsession.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/rtcp.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/sctptransport.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/threadpool.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/tls.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/track.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/processor.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/capi.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/rtppacketizationconfig.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/rtcpsrreporter.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/rtppacketizer.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/opusrtppacketizer.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/opuspacketizationhandler.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/h264rtppacketizer.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/nalunit.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/h264packetizationhandler.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/mediachainablehandler.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/mediahandlerelement.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/mediahandlerrootelement.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/rtcpnackresponder.cpp
)
set(LIBDATACHANNEL_PRIVATE_HEADERS
${CMAKE_CURRENT_SOURCE_DIR}/src/certificate.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/dtlssrtptransport.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/dtlstransport.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/icetransport.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/logcounter.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/sctptransport.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/threadpool.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/tls.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/processor.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/transport.hpp
)
set(LIBDATACHANNEL_WEBSOCKET_SOURCES
@ -102,14 +76,6 @@ set(LIBDATACHANNEL_WEBSOCKET_SOURCES
${CMAKE_CURRENT_SOURCE_DIR}/src/wstransport.cpp
)
set(LIBDATACHANNEL_WEBSOCKET_PRIVATE_HEADERS
${CMAKE_CURRENT_SOURCE_DIR}/src/base64.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/tcptransport.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/tlstransport.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/verifiedtlstransport.hpp
${CMAKE_CURRENT_SOURCE_DIR}/src/wstransport.hpp
)
set(LIBDATACHANNEL_HEADERS
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/candidate.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/channel.hpp
@ -117,8 +83,7 @@ set(LIBDATACHANNEL_HEADERS
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/configuration.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/datachannel.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/description.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/mediahandler.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtcpreceivingsession.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtcp.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/include.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/init.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/log.hpp
@ -131,24 +96,11 @@ set(LIBDATACHANNEL_HEADERS
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtp.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/track.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/websocket.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtppacketizationconfig.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtcpsrreporter.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtppacketizer.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/opusrtppacketizer.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/opuspacketizationhandler.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/h264rtppacketizer.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/nalunit.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/h264packetizationhandler.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/mediachainablehandler.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/mediahandlerelement.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/mediahandlerrootelement.hpp
${CMAKE_CURRENT_SOURCE_DIR}/include/rtc/rtcpnackresponder.hpp
)
set(TESTS_SOURCES
${CMAKE_CURRENT_SOURCE_DIR}/test/main.cpp
${CMAKE_CURRENT_SOURCE_DIR}/test/connectivity.cpp
${CMAKE_CURRENT_SOURCE_DIR}/test/turn_connectivity.cpp
${CMAKE_CURRENT_SOURCE_DIR}/test/track.cpp
${CMAKE_CURRENT_SOURCE_DIR}/test/capi_connectivity.cpp
${CMAKE_CURRENT_SOURCE_DIR}/test/capi_track.cpp
@ -156,24 +108,6 @@ set(TESTS_SOURCES
${CMAKE_CURRENT_SOURCE_DIR}/test/benchmark.cpp
)
set(TESTS_UWP_RESOURCES
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/tests/Logo.png
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/tests/package.appxManifest
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/tests/SmallLogo.png
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/tests/SmallLogo44x44.png
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/tests/SplashScreen.png
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/tests/StoreLogo.png
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/tests/Windows_TemporaryKey.pfx)
set(BENCHMARK_UWP_RESOURCES
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/benchmark/Logo.png
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/benchmark/package.appxManifest
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/benchmark/SmallLogo.png
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/benchmark/SmallLogo44x44.png
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/benchmark/SplashScreen.png
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/benchmark/StoreLogo.png
${CMAKE_CURRENT_SOURCE_DIR}/test/uwp/benchmark/Windows_TemporaryKey.pfx)
set(CMAKE_THREAD_PREFER_PTHREAD TRUE)
set(THREADS_PREFER_PTHREAD_FLAG TRUE)
find_package(Threads REQUIRED)
@ -194,28 +128,18 @@ add_library(Usrsctp::Usrsctp ALIAS usrsctp)
if (NO_WEBSOCKET)
add_library(datachannel SHARED
${LIBDATACHANNEL_SOURCES}
${LIBDATACHANNEL_PRIVATE_HEADERS}
${LIBDATACHANNEL_HEADERS})
${LIBDATACHANNEL_SOURCES})
add_library(datachannel-static STATIC EXCLUDE_FROM_ALL
${LIBDATACHANNEL_SOURCES}
${LIBDATACHANNEL_PRIVATE_HEADERS}
${LIBDATACHANNEL_HEADERS})
${LIBDATACHANNEL_SOURCES})
target_compile_definitions(datachannel PUBLIC RTC_ENABLE_WEBSOCKET=0)
target_compile_definitions(datachannel-static PUBLIC RTC_ENABLE_WEBSOCKET=0)
else()
add_library(datachannel SHARED
${LIBDATACHANNEL_SOURCES}
${LIBDATACHANNEL_PRIVATE_HEADERS}
${LIBDATACHANNEL_WEBSOCKET_SOURCES}
${LIBDATACHANNEL_WEBSOCKET_PRIVATE_HEADERS}
${LIBDATACHANNEL_HEADERS})
${LIBDATACHANNEL_WEBSOCKET_SOURCES})
add_library(datachannel-static STATIC EXCLUDE_FROM_ALL
${LIBDATACHANNEL_SOURCES}
${LIBDATACHANNEL_PRIVATE_HEADERS}
${LIBDATACHANNEL_WEBSOCKET_SOURCES}
${LIBDATACHANNEL_WEBSOCKET_PRIVATE_HEADERS}
${LIBDATACHANNEL_HEADERS})
${LIBDATACHANNEL_WEBSOCKET_SOURCES})
target_compile_definitions(datachannel PUBLIC RTC_ENABLE_WEBSOCKET=1)
target_compile_definitions(datachannel-static PUBLIC RTC_ENABLE_WEBSOCKET=1)
endif()
@ -341,30 +265,24 @@ endif()
# Tests
if(NOT NO_TESTS)
if(CMAKE_SYSTEM_NAME STREQUAL "WindowsStore")
# Add resource files needed for UWP apps.
add_executable(datachannel-tests ${TESTS_SOURCES} ${TESTS_UWP_RESOURCES})
else()
add_executable(datachannel-tests ${TESTS_SOURCES})
endif()
add_executable(datachannel-tests ${TESTS_SOURCES})
set_target_properties(datachannel-tests PROPERTIES
VERSION ${PROJECT_VERSION}
CXX_STANDARD 17)
set_target_properties(datachannel-tests PROPERTIES OUTPUT_NAME tests)
if(NOT CMAKE_SYSTEM_NAME STREQUAL "WindowsStore") # Prevent a bug in manifest generation for UWP
set_target_properties(datachannel-tests PROPERTIES OUTPUT_NAME tests)
endif()
target_include_directories(datachannel-tests PRIVATE ${CMAKE_CURRENT_SOURCE_DIR}/src)
target_link_libraries(datachannel-tests datachannel)
# Benchmark
if(CMAKE_SYSTEM_NAME STREQUAL "WindowsStore")
# Add resource files needed for UWP apps.
add_executable(datachannel-benchmark test/benchmark.cpp ${BENCHMARK_UWP_RESOURCES})
else()
add_executable(datachannel-benchmark test/benchmark.cpp)
endif()
add_executable(datachannel-benchmark test/benchmark.cpp)
set_target_properties(datachannel-benchmark PROPERTIES
VERSION ${PROJECT_VERSION}
CXX_STANDARD 17)
set_target_properties(datachannel-benchmark PROPERTIES OUTPUT_NAME benchmark)
if(NOT CMAKE_SYSTEM_NAME STREQUAL "WindowsStore") # Prevent a bug in manifest generation for UWP
set_target_properties(datachannel-benchmark PROPERTIES OUTPUT_NAME benchmark)
endif()
target_compile_definitions(datachannel-benchmark PRIVATE BENCHMARK_MAIN=1)
target_include_directories(datachannel-benchmark PRIVATE ${CMAKE_CURRENT_SOURCE_DIR}/src)
target_link_libraries(datachannel-benchmark datachannel)
@ -375,11 +293,8 @@ if(NOT NO_EXAMPLES AND NOT CMAKE_SYSTEM_NAME STREQUAL "WindowsStore")
set(JSON_BuildTests OFF CACHE INTERNAL "")
add_subdirectory(deps/json)
add_subdirectory(examples/client)
if(NOT NO_MEDIA)
add_subdirectory(examples/media)
add_subdirectory(examples/sfu-media)
add_subdirectory(examples/streamer)
endif()
add_subdirectory(examples/copy-paste)
add_subdirectory(examples/copy-paste-capi)
endif()

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@ -10,7 +10,6 @@ LDFLAGS=-pthread
LIBS=
LOCALLIBS=libusrsctp.a
USRSCTP_DIR=deps/usrsctp
SRTP_DIR=deps/libsrtp
JUICE_DIR=deps/libjuice
PLOG_DIR=deps/plog
@ -39,22 +38,15 @@ ifneq ($(USE_GNUTLS), 0)
endif
endif
NO_MEDIA ?= 0
USE_SYSTEM_SRTP ?= 0
ifeq ($(NO_MEDIA), 0)
USE_SRTP ?= 0
ifneq ($(USE_SRTP), 0)
CPPFLAGS+=-DRTC_ENABLE_MEDIA=1
ifneq ($(USE_SYSTEM_SRTP), 0)
CPPFLAGS+=-DRTC_SYSTEM_SRTP=1
LIBS+=srtp
else
CPPFLAGS+=-DRTC_SYSTEM_SRTP=0
INCLUDES+=-I$(SRTP_DIR)/include
LOCALLIBS+=libsrtp2.a
endif
else
CPPFLAGS+=-DRTC_ENABLE_MEDIA=0
endif
NO_WEBSOCKET ?= 0
ifeq ($(NO_WEBSOCKET), 0)
CPPFLAGS+=-DRTC_ENABLE_WEBSOCKET=1
@ -62,8 +54,8 @@ else
CPPFLAGS+=-DRTC_ENABLE_WEBSOCKET=0
endif
INCLUDES+=$(if $(LIBS),$(shell pkg-config --cflags $(LIBS)),)
LDLIBS+=$(LOCALLIBS) $(if $(LIBS),$(shell pkg-config --libs $(LIBS)),)
INCLUDES+=$(shell pkg-config --cflags $(LIBS))
LDLIBS+=$(LOCALLIBS) $(shell pkg-config --libs $(LIBS))
SRCS=$(shell printf "%s " src/*.cpp)
OBJS=$(subst .cpp,.o,$(SRCS))
@ -81,7 +73,7 @@ test/%.o: test/%.cpp
-include $(subst .cpp,.d,$(SRCS))
$(NAME).a: $(LOCALLIBS) $(OBJS)
$(NAME).a: $(OBJS)
$(AR) crf $@ $(OBJS)
$(NAME).so: $(LOCALLIBS) $(OBJS)
@ -105,22 +97,15 @@ dist-clean: clean
-$(RM) src/*~
-$(RM) test/*~
-cd $(USRSCTP_DIR) && make clean
-cd $(SRTP_DIR) && make clean
-cd $(JUICE_DIR) && make clean
libusrsctp.a:
cd $(USRSCTP_DIR) && \
./bootstrap && \
./configure --enable-static --disable-debug CFLAGS="-fPIC" && \
./configure --enable-static --disable-debug CFLAGS="$(CPPFLAGS) -Wno-error=format-truncation" && \
make
cp $(USRSCTP_DIR)/usrsctplib/.libs/libusrsctp.a .
libsrtp2.a:
cd $(SRTP_DIR) && \
./configure && \
make
cp $(SRTP_DIR)/libsrtp2.a .
libjuice.a:
ifneq ($(USE_GNUTLS), 0)
cd $(JUICE_DIR) && make USE_NETTLE=1

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@ -21,7 +21,7 @@ Protocol stack:
- SCTP-based Data Channels ([draft-ietf-rtcweb-data-channel-13](https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13))
- SRTP-based Media Transport ([draft-ietf-rtcweb-rtp-usage-26](https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26))
- DTLS/UDP ([RFC7350](https://tools.ietf.org/html/rfc7350) and [RFC8261](https://tools.ietf.org/html/rfc8261))
- ICE ([RFC8445](https://tools.ietf.org/html/rfc8445)) with STUN ([RFC8489](https://tools.ietf.org/html/rfc8489)) and its extension TURN ([RFC8656](https://tools.ietf.org/html/rfc8656))
- ICE ([RFC8445](https://tools.ietf.org/html/rfc8445)) with STUN ([RFC5389](https://tools.ietf.org/html/rfc5389))
Features:
- Full IPv6 support
@ -30,6 +30,7 @@ Features:
- Multicast DNS candidates ([draft-ietf-rtcweb-mdns-ice-candidates-04](https://tools.ietf.org/html/draft-ietf-rtcweb-mdns-ice-candidates-04))
- SRTP and SRTCP key derivation from DTLS ([RFC5764](https://tools.ietf.org/html/rfc5764))
- Differentiated Services QoS ([draft-ietf-tsvwg-rtcweb-qos-18](https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18))
- TURN relaying ([RFC5766](https://tools.ietf.org/html/rfc5766)) with [libnice](https://github.com/libnice/libnice) as ICE backend
Note only SDP BUNDLE mode is supported for media multiplexing ([draft-ietf-mmusic-sdp-bundle-negotiation-54](https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-54)). The behavior is equivalent to the JSEP bundle-only policy: the library always negociates one unique network component, where SRTP media streams are multiplexed with SRTCP control packets ([RFC5761](https://tools.ietf.org/html/rfc5761)) and SCTP/DTLS data traffic ([RFC5764](https://tools.ietf.org/html/rfc5764)).
@ -73,7 +74,7 @@ $ git submodule update --init --recursive
The CMake library targets `libdatachannel` and `libdatachannel-static` respectively correspond to the shared and static libraries. The default target will build tests and examples. The option `USE_GNUTLS` allows to switch between OpenSSL (default) and GnuTLS, and the option `USE_NICE` allows to switch between libjuice as submodule (default) and libnice.
If you only need Data Channels, the option `NO_MEDIA` allows to make the library lighter by removing media support. Similarly, `NO_WEBSOCKET` removes WebSocket support.
On Windows, the DLL resulting from the shared library build only exposes the C API, use the static library for the C++ API.
#### POSIX-compliant operating systems (including Linux and Apple macOS)
```bash
@ -123,8 +124,6 @@ $ nmake
The option `USE_GNUTLS` allows to switch between OpenSSL (default) and GnuTLS, and the option `USE_NICE` allows to switch between libjuice as submodule (default) and libnice.
If you only need Data Channels, the option `NO_MEDIA` removes media support. Similarly, `NO_WEBSOCKET` removes WebSocket support.
```bash
$ make USE_GNUTLS=1 USE_NICE=0
```

2
deps/libjuice vendored

2
deps/usrsctp vendored

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@ -8,7 +8,6 @@ This directory contains different WebRTC clients and compatible WebSocket + JSON
- [signaling-server-rust](signaling-server-rust) contains a similar signaling server in Rust (see [lerouxrgd/datachannel-rs](https://github.com/lerouxrgd/datachannel-rs) for Rust wrappers)
- [media](media) is a copy/paste demo to send the webcam from your browser into gstreamer.
- [streamer](streamer) streams h264 and opus samples to web browsers (signaling-server-python is required).
Additionally, it contains two debugging tools for libdatachannel with copy-pasting as signaling:
- [copy-paste](copy-paste) using the C++ API

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@ -4,10 +4,10 @@ if(POLICY CMP0079)
endif()
if(WIN32)
add_executable(datachannel-client main.cpp parse_cl.cpp parse_cl.h getopt.cpp getopt.h)
target_compile_definitions(datachannel-client PUBLIC STATIC_GETOPT)
add_executable(datachannel-client main.cpp parse_cl.cpp parse_cl.h getopt.cpp getopt.h)
target_compile_definitions(datachannel-client PUBLIC STATIC_GETOPT)
else()
add_executable(datachannel-client main.cpp parse_cl.cpp parse_cl.h)
add_executable(datachannel-client main.cpp parse_cl.cpp parse_cl.h)
endif()
set_target_properties(datachannel-client PROPERTIES
@ -15,10 +15,3 @@ set_target_properties(datachannel-client PROPERTIES
OUTPUT_NAME client)
target_link_libraries(datachannel-client datachannel nlohmann_json)
if(WIN32)
add_custom_command(TARGET datachannel-client POST_BUILD
COMMAND ${CMAKE_COMMAND} -E copy_if_different
"$<TARGET_FILE_DIR:datachannel>/datachannel.dll"
$<TARGET_FILE_DIR:datachannel-client>
)
endif()

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@ -13,15 +13,3 @@ set_target_properties(datachannel-copy-paste-capi-answerer PROPERTIES
OUTPUT_NAME answerer)
target_link_libraries(datachannel-copy-paste-capi-answerer datachannel)
if(WIN32)
add_custom_command(TARGET datachannel-copy-paste-capi-offerer POST_BUILD
COMMAND ${CMAKE_COMMAND} -E copy_if_different
"$<TARGET_FILE_DIR:datachannel>/datachannel.dll"
$<TARGET_FILE_DIR:datachannel-copy-paste-capi-offerer>
)
add_custom_command(TARGET datachannel-copy-paste-capi-answerer POST_BUILD
COMMAND ${CMAKE_COMMAND} -E copy_if_different
"$<TARGET_FILE_DIR:datachannel>/datachannel.dll"
$<TARGET_FILE_DIR:datachannel-copy-paste-capi-answerer>
)
endif()

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@ -12,15 +12,3 @@ set_target_properties(datachannel-copy-paste-answerer PROPERTIES
OUTPUT_NAME answerer)
target_link_libraries(datachannel-copy-paste-answerer datachannel)
if(WIN32)
add_custom_command(TARGET datachannel-copy-paste-offerer POST_BUILD
COMMAND ${CMAKE_COMMAND} -E copy_if_different
"$<TARGET_FILE_DIR:datachannel>/datachannel.dll"
$<TARGET_FILE_DIR:datachannel-copy-paste-offerer>
)
add_custom_command(TARGET datachannel-copy-paste-answerer POST_BUILD
COMMAND ${CMAKE_COMMAND} -E copy_if_different
"$<TARGET_FILE_DIR:datachannel>/datachannel.dll"
$<TARGET_FILE_DIR:datachannel-copy-paste-answerer>
)
endif()

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@ -6,10 +6,3 @@ set_target_properties(datachannel-media PROPERTIES
OUTPUT_NAME media)
target_link_libraries(datachannel-media datachannel nlohmann_json)
if(WIN32)
add_custom_command(TARGET datachannel-media POST_BUILD
COMMAND ${CMAKE_COMMAND} -E copy_if_different
"$<TARGET_FILE_DIR:datachannel>/datachannel.dll"
$<TARGET_FILE_DIR:datachannel-media>
)
endif()

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@ -6,10 +6,3 @@ set_target_properties(datachannel-sfu-media PROPERTIES
OUTPUT_NAME sfu-media)
target_link_libraries(datachannel-sfu-media datachannel nlohmann_json)
if(WIN32)
add_custom_command(TARGET datachannel-sfu-media POST_BUILD
COMMAND ${CMAKE_COMMAND} -E copy_if_different
"$<TARGET_FILE_DIR:datachannel>/datachannel.dll"
$<TARGET_FILE_DIR:datachannel-sfu-media>
)
endif()

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@ -1,75 +0,0 @@
/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#include "ArgParser.hpp"
#include <iostream>
ArgParser::ArgParser(std::vector<std::pair<std::string, std::string>> options, std::vector<std::pair<std::string, std::string>> flags) {
for(auto option: options) {
this->options.insert(option.first);
this->options.insert(option.second);
shortToLongMap.emplace(option.first, option.second);
shortToLongMap.emplace(option.second, option.second);
}
for(auto flag: flags) {
this->flags.insert(flag.first);
this->flags.insert(flag.second);
shortToLongMap.emplace(flag.first, flag.second);
shortToLongMap.emplace(flag.second, flag.second);
}
}
std::optional<std::string> ArgParser::toKey(std::string prefixedKey) {
if (prefixedKey.find("--") == 0) {
return prefixedKey.substr(2, prefixedKey.length());
} else if (prefixedKey.find("-") == 0) {
return prefixedKey.substr(1, prefixedKey.length());
} else {
return std::nullopt;
}
}
bool ArgParser::parse(int argc, char **argv, std::function<bool (std::string, std::string)> onOption, std::function<bool (std::string)> onFlag) {
std::optional<std::string> currentOption = std::nullopt;
for(int i = 1; i < argc; i++) {
std::string current = argv[i];
auto optKey = toKey(current);
if (!currentOption.has_value() && optKey.has_value() && flags.find(optKey.value()) != flags.end()) {
auto check = onFlag(shortToLongMap.at(optKey.value()));
if (!check) {
return false;
}
} else if (!currentOption.has_value() && optKey.has_value() && options.find(optKey.value()) != options.end()) {
currentOption = optKey.value();
} else if (currentOption.has_value()) {
auto check = onOption(shortToLongMap.at(currentOption.value()), current);
if (!check) {
return false;
}
currentOption = std::nullopt;
} else {
std::cerr << "Unrecognized option " << current << std::endl;
return false;
}
}
if (currentOption.has_value()) {
std::cerr << "Missing value for " << currentOption.value() << std::endl;
return false;
}
return true;
}

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@ -1,41 +0,0 @@
/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef ArgParser_hpp
#define ArgParser_hpp
#include <functional>
#include <vector>
#include <utility>
#include <string>
#include <set>
#include <unordered_map>
#include <optional>
struct ArgParser {
private:
std::set<std::string> options{};
std::set<std::string> flags{};
std::unordered_map<std::string, std::string> shortToLongMap{};
public:
ArgParser(std::vector<std::pair<std::string, std::string>> options, std::vector<std::pair<std::string, std::string>> flags);
std::optional<std::string> toKey(std::string prefixedKey);
bool parse(int argc, char **argv, std::function<bool (std::string, std::string)> onOption, std::function<bool (std::string)> onFlag);
};
#endif /* ArgParser_hpp */

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@ -1,40 +0,0 @@
cmake_minimum_required(VERSION 3.7)
if(POLICY CMP0079)
cmake_policy(SET CMP0079 NEW)
endif()
add_executable(streamer
main.cpp
dispatchqueue.cpp
dispatchqueue.hpp
h264fileparser.cpp
h264fileparser.hpp
helpers.cpp
helpers.hpp
opusfileparser.cpp
opusfileparser.hpp
fileparser.cpp
fileparser.hpp
stream.cpp
stream.hpp
ArgParser.cpp
ArgParser.hpp
)
if(WIN32)
target_compile_definitions(streamer PUBLIC STATIC_GETOPT)
endif()
set_target_properties(streamer PROPERTIES
CXX_STANDARD 17
OUTPUT_NAME streamer)
target_link_libraries(streamer datachannel nlohmann_json)
if(WIN32)
add_custom_command(TARGET streamer POST_BUILD
COMMAND ${CMAKE_COMMAND} -E copy_if_different
"$<TARGET_FILE_DIR:datachannel>/datachannel.dll"
$<TARGET_FILE_DIR:streamer>
)
endif()

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# Streaming H264 and opus
This example streams H264 and opus<sup id="a1">[1](#f1)</sup> samples to the connected browser client.
## Starting signaling server
```sh
$ python3 ../signaling-server-python/signaling-server.py
```
## Starting php
```sh
$ php -S 127.0.0.1:8080
```
Now you can open demo at [127.0.0.1:8080](127.0.0.1:8080).
## Arguments
- `-a` Directory with OPUS samples (default: *../../../../examples/streamer/samples/opus/*).
- `-b` Directory with H264 samples (default: *../../../../examples/streamer/samples/h264/*).
- `-d` Signaling server IP address (default: 127.0.0.1).
- `-p` Signaling server port (default: 8000).
- `-v` Enable debug logs.
- `-h` Print this help and exit.
## Generating H264 and Opus samples
You can generate H264 and Opus sample with *samples/generate_h264.py* and *samples/generate_opus.py* respectively. This require ffmpeg, python3 and kaitaistruct library to be installed. Use `-h`/`--help` to learn more about arguments.
<b id="f1">1</b> Opus samples are generated from music downloaded at [bensound](https://www.bensound.com). [](#a1)

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/** @type {RTCPeerConnection} */
let rtc;
const iceConnectionLog = document.getElementById('ice-connection-state'),
iceGatheringLog = document.getElementById('ice-gathering-state'),
signalingLog = document.getElementById('signaling-state'),
dataChannelLog = document.getElementById('data-channel');
function randomString(len) {
const charSet = 'ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789';
let randomString = '';
for (let i = 0; i < len; i++) {
const randomPoz = Math.floor(Math.random() * charSet.length);
randomString += charSet.substring(randomPoz, randomPoz + 1);
}
return randomString;
}
const receiveID = randomString(10);
const websocket = new WebSocket('ws://127.0.0.1:8000/' + receiveID);
websocket.onopen = function () {
document.getElementById('start').disabled = false;
}
// data channel
let dc = null, dcTimeout = null;
function createPeerConnection() {
const config = {
sdpSemantics: 'unified-plan',
bundlePolicy: "max-bundle",
};
if (document.getElementById('use-stun').checked) {
config.iceServers = [{urls: ['stun:stun.l.google.com:19302']}];
}
let pc = new RTCPeerConnection(config);
// register some listeners to help debugging
pc.addEventListener('icegatheringstatechange', function () {
iceGatheringLog.textContent += ' -> ' + pc.iceGatheringState;
}, false);
iceGatheringLog.textContent = pc.iceGatheringState;
pc.addEventListener('iceconnectionstatechange', function () {
iceConnectionLog.textContent += ' -> ' + pc.iceConnectionState;
}, false);
iceConnectionLog.textContent = pc.iceConnectionState;
pc.addEventListener('signalingstatechange', function () {
signalingLog.textContent += ' -> ' + pc.signalingState;
}, false);
signalingLog.textContent = pc.signalingState;
// connect audio / video
pc.addEventListener('track', function (evt) {
document.getElementById('media').style.display = 'block';
const videoTag = document.getElementById('video');
videoTag.srcObject = evt.streams[0];
videoTag.play();
});
let time_start = null;
function current_stamp() {
if (time_start === null) {
time_start = new Date().getTime();
return 0;
} else {
return new Date().getTime() - time_start;
}
}
pc.ondatachannel = function (event) {
dc = event.channel;
dc.onopen = function () {
dataChannelLog.textContent += '- open\n';
dataChannelLog.scrollTop = dataChannelLog.scrollHeight;
};
dc.onmessage = function (evt) {
dataChannelLog.textContent += '< ' + evt.data + '\n';
dataChannelLog.scrollTop = dataChannelLog.scrollHeight;
dcTimeout = setTimeout(function () {
if (dc == null && dcTimeout != null) {
dcTimeout = null;
return
}
const message = 'Pong ' + current_stamp();
dataChannelLog.textContent += '> ' + message + '\n';
dataChannelLog.scrollTop = dataChannelLog.scrollHeight;
dc.send(message);
}, 1000);
}
dc.onclose = function () {
clearTimeout(dcTimeout);
dcTimeout = null;
dataChannelLog.textContent += '- close\n';
dataChannelLog.scrollTop = dataChannelLog.scrollHeight;
};
}
return pc;
}
function sendAnswer(pc) {
return pc.createAnswer()
.then((answer) => rtc.setLocalDescription(answer))
.then(function () {
// wait for ICE gathering to complete
return new Promise(function (resolve) {
if (pc.iceGatheringState === 'complete') {
resolve();
} else {
function checkState() {
if (pc.iceGatheringState === 'complete') {
pc.removeEventListener('icegatheringstatechange', checkState);
resolve();
}
}
pc.addEventListener('icegatheringstatechange', checkState);
}
});
}).then(function () {
const answer = pc.localDescription;
document.getElementById('answer-sdp').textContent = answer.sdp;
return websocket.send(JSON.stringify(
{
id: "server",
type: answer.type,
sdp: answer.sdp,
}));
}).catch(function (e) {
alert(e);
});
}
function handleOffer(offer) {
rtc = createPeerConnection();
return rtc.setRemoteDescription(offer)
.then(() => sendAnswer(rtc));
}
function sendStreamRequest() {
websocket.send(JSON.stringify(
{
id: "server",
type: "streamRequest",
receiver: receiveID,
}));
}
async function start() {
document.getElementById('start').style.display = 'none';
document.getElementById('stop').style.display = 'inline-block';
document.getElementById('media').style.display = 'block';
sendStreamRequest();
}
function stop() {
document.getElementById('stop').style.display = 'none';
document.getElementById('media').style.display = 'none';
document.getElementById('start').style.display = 'inline-block';
// close data channel
if (dc) {
dc.close();
dc = null;
}
// close transceivers
if (rtc.getTransceivers) {
rtc.getTransceivers().forEach(function (transceiver) {
if (transceiver.stop) {
transceiver.stop();
}
});
}
// close local audio / video
rtc.getSenders().forEach(function (sender) {
const track = sender.track;
if (track !== null) {
sender.track.stop();
}
});
// close peer connection
setTimeout(function () {
rtc.close();
rtc = null;
}, 500);
}
websocket.onmessage = async function (evt) {
const received_msg = evt.data;
const object = JSON.parse(received_msg);
if (object.type == "offer") {
document.getElementById('offer-sdp').textContent = object.sdp;
await handleOffer(object)
}
}

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/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#include "dispatchqueue.hpp"
DispatchQueue::DispatchQueue(std::string name, size_t threadCount) :
name{std::move(name)}, threads(threadCount) {
for(size_t i = 0; i < threads.size(); i++)
{
threads[i] = std::thread(&DispatchQueue::dispatchThreadHandler, this);
}
}
DispatchQueue::~DispatchQueue() {
// Signal to dispatch threads that it's time to wrap up
std::unique_lock<std::mutex> lock(lockMutex);
quit = true;
lock.unlock();
condition.notify_all();
// Wait for threads to finish before we exit
for(size_t i = 0; i < threads.size(); i++)
{
if(threads[i].joinable())
{
threads[i].join();
}
}
}
void DispatchQueue::removePending() {
std::unique_lock<std::mutex> lock(lockMutex);
queue = {};
}
void DispatchQueue::dispatch(const fp_t& op) {
std::unique_lock<std::mutex> lock(lockMutex);
queue.push(op);
// Manual unlocking is done before notifying, to avoid waking up
// the waiting thread only to block again (see notify_one for details)
lock.unlock();
condition.notify_one();
}
void DispatchQueue::dispatch(fp_t&& op) {
std::unique_lock<std::mutex> lock(lockMutex);
queue.push(std::move(op));
// Manual unlocking is done before notifying, to avoid waking up
// the waiting thread only to block again (see notify_one for details)
lock.unlock();
condition.notify_one();
}
void DispatchQueue::dispatchThreadHandler(void) {
std::unique_lock<std::mutex> lock(lockMutex);
do {
//Wait until we have data or a quit signal
condition.wait(lock, [this]{
return (queue.size() || quit);
});
//after wait, we own the lock
if(!quit && queue.size())
{
auto op = std::move(queue.front());
queue.pop();
//unlock now that we're done messing with the queue
lock.unlock();
op();
lock.lock();
}
} while (!quit);
}

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/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef dispatchqueue_hpp
#define dispatchqueue_hpp
#include <thread>
#include <mutex>
#include <condition_variable>
#include <queue>
#include <functional>
class DispatchQueue {
typedef std::function<void(void)> fp_t;
public:
DispatchQueue(std::string name, size_t threadCount = 1);
~DispatchQueue();
// dispatch and copy
void dispatch(const fp_t& op);
// dispatch and move
void dispatch(fp_t&& op);
void removePending();
// Deleted operations
DispatchQueue(const DispatchQueue& rhs) = delete;
DispatchQueue& operator=(const DispatchQueue& rhs) = delete;
DispatchQueue(DispatchQueue&& rhs) = delete;
DispatchQueue& operator=(DispatchQueue&& rhs) = delete;
private:
std::string name;
std::mutex lockMutex;
std::vector<std::thread> threads;
std::queue<fp_t> queue;
std::condition_variable condition;
bool quit = false;
void dispatchThreadHandler(void);
};
#endif /* dispatchqueue_hpp */

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/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#include "fileparser.hpp"
#include <fstream>
using namespace std;
FileParser::FileParser(string directory, string extension, uint32_t samplesPerSecond, bool loop): sampleDuration_us(1000 * 1000 / samplesPerSecond), StreamSource() {
this->directory = directory;
this->extension = extension;
this->loop = loop;
}
void FileParser::start() {
sampleTime_us = -sampleDuration_us;
loadNextSample();
}
void FileParser::stop() {
StreamSource::stop();
counter = -1;
}
void FileParser::loadNextSample() {
string frame_id = to_string(++counter);
string url = directory + "/sample-" + frame_id + extension;
ifstream source(url, ios_base::binary);
if (!source) {
if (loop && counter > 0) {
loopTimestampOffset = sampleTime_us;
counter = -1;
loadNextSample();
return;
}
sample = {};
return;
}
vector<uint8_t> fileContents((std::istreambuf_iterator<char>(source)), std::istreambuf_iterator<char>());
sample = *reinterpret_cast<vector<byte> *>(&fileContents);
sampleTime_us += sampleDuration_us;
}

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/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef fileparser_hpp
#define fileparser_hpp
#include <string>
#include <vector>
#include "stream.hpp"
class FileParser: public StreamSource {
std::string directory;
std::string extension;
uint32_t counter = -1;
bool loop;
uint64_t loopTimestampOffset = 0;
public:
const uint64_t sampleDuration_us;
virtual void start();
virtual void stop();
FileParser(std::string directory, std::string extension, uint32_t samplesPerSecond, bool loop);
virtual void loadNextSample();
};
#endif /* fileparser_hpp */

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/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#include "h264fileparser.hpp"
#include <fstream>
#include "rtc/rtc.hpp"
using namespace std;
H264FileParser::H264FileParser(string directory, uint32_t fps, bool loop): FileParser(directory, ".h264", fps, loop) { }
void H264FileParser::loadNextSample() {
FileParser::loadNextSample();
unsigned long long i = 0;
while (i < sample.size()) {
assert(i + 4 < sample.size());
auto lengthPtr = (uint32_t *) (sample.data() + i);
uint32_t length = ntohl(*lengthPtr);
auto naluStartIndex = i + 4;
auto naluEndIndex = naluStartIndex + length;
assert(naluEndIndex <= sample.size());
auto header = reinterpret_cast<rtc::NalUnitHeader *>(sample.data() + naluStartIndex);
auto type = header->unitType();
switch (type) {
case 7:
previousUnitType7 = {sample.begin() + i, sample.begin() + naluEndIndex};
break;
case 8:
previousUnitType8 = {sample.begin() + i, sample.begin() + naluEndIndex};;
break;
case 5:
previousUnitType5 = {sample.begin() + i, sample.begin() + naluEndIndex};;
break;
}
i = naluEndIndex;
}
}
vector<byte> H264FileParser::initialNALUS() {
vector<byte> units{};
if (previousUnitType7.has_value()) {
auto nalu = previousUnitType7.value();
units.insert(units.end(), nalu.begin(), nalu.end());
}
if (previousUnitType8.has_value()) {
auto nalu = previousUnitType8.value();
units.insert(units.end(), nalu.begin(), nalu.end());
}
if (previousUnitType5.has_value()) {
auto nalu = previousUnitType5.value();
units.insert(units.end(), nalu.begin(), nalu.end());
}
return units;
}

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/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef h264fileparser_hpp
#define h264fileparser_hpp
#include "fileparser.hpp"
#include <optional>
class H264FileParser: public FileParser {
std::optional<std::vector<std::byte>> previousUnitType5 = std::nullopt;
std::optional<std::vector<std::byte>> previousUnitType7 = std::nullopt;
std::optional<std::vector<std::byte>> previousUnitType8 = std::nullopt;
public:
H264FileParser(std::string directory, uint32_t fps, bool loop);
void loadNextSample() override;
std::vector<std::byte> initialNALUS();
};
#endif /* h264fileparser_hpp */

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/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#include "helpers.hpp"
#include <ctime>
#if _WIN32
// taken from https://stackoverflow.com/questions/10905892/equivalent-of-gettimeday-for-windows
#include <Windows.h>
struct timezone {
int tz_minuteswest;
int tz_dsttime;
};
int gettimeofday(struct timeval *tv, struct timezone *tz)
{
if (tv) {
FILETIME filetime; /* 64-bit value representing the number of 100-nanosecond intervals since January 1, 1601 00:00 UTC */
ULARGE_INTEGER x;
ULONGLONG usec;
static const ULONGLONG epoch_offset_us = 11644473600000000ULL; /* microseconds betweeen Jan 1,1601 and Jan 1,1970 */
#if _WIN32_WINNT >= _WIN32_WINNT_WIN8
GetSystemTimePreciseAsFileTime(&filetime);
#else
GetSystemTimeAsFileTime(&filetime);
#endif
x.LowPart = filetime.dwLowDateTime;
x.HighPart = filetime.dwHighDateTime;
usec = x.QuadPart / 10 - epoch_offset_us;
tv->tv_sec = (time_t)(usec / 1000000ULL);
tv->tv_usec = (long)(usec % 1000000ULL);
}
if (tz) {
TIME_ZONE_INFORMATION timezone;
GetTimeZoneInformation(&timezone);
tz->tz_minuteswest = timezone.Bias;
tz->tz_dsttime = 0;
}
return 0;
}
#endif
using namespace std;
using namespace rtc;
ClientTrackData::ClientTrackData(shared_ptr<Track> track, shared_ptr<RtcpSrReporter> sender) {
this->track = track;
this->sender = sender;
}
void Client::setState(State state) {
std::unique_lock lock(_mutex);
this->state = state;
}
Client::State Client::getState() {
std::shared_lock lock(_mutex);
return state;
}
ClientTrack::ClientTrack(string id, shared_ptr<ClientTrackData> trackData) {
this->id = id;
this->trackData = trackData;
}
uint64_t currentTimeInMicroSeconds() {
struct timeval time;
gettimeofday(&time, NULL);
return uint64_t(time.tv_sec) * 1000 * 1000 + time.tv_usec;
}

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/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef helpers_hpp
#define helpers_hpp
#include "rtc/rtc.hpp"
struct ClientTrackData {
std::shared_ptr<rtc::Track> track;
std::shared_ptr<rtc::RtcpSrReporter> sender;
ClientTrackData(std::shared_ptr<rtc::Track> track, std::shared_ptr<rtc::RtcpSrReporter> sender);
};
struct Client {
enum class State {
Waiting,
WaitingForVideo,
WaitingForAudio,
Ready
};
const std::shared_ptr<rtc::PeerConnection> & peerConnection = _peerConnection;
Client(std::shared_ptr<rtc::PeerConnection> pc) {
_peerConnection = pc;
}
std::optional<std::shared_ptr<ClientTrackData>> video;
std::optional<std::shared_ptr<ClientTrackData>> audio;
std::optional<std::shared_ptr<rtc::DataChannel>> dataChannel{};
void setState(State state);
State getState();
private:
std::shared_mutex _mutex;
State state = State::Waiting;
std::string id;
std::shared_ptr<rtc::PeerConnection> _peerConnection;
};
struct ClientTrack {
std::string id;
std::shared_ptr<ClientTrackData> trackData;
ClientTrack(std::string id, std::shared_ptr<ClientTrackData> trackData);
};
uint64_t currentTimeInMicroSeconds();
#endif /* helpers_hpp */

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<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<title>libdatachannel media example</title>
<style>
button {
padding: 8px 16px;
}
pre {
overflow-x: hidden;
overflow-y: auto;
}
video {
width: 100%;
}
.option {
margin-bottom: 8px;
}
#media {
max-width: 1280px;
}
</style>
</head>
<body>
<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>
<h2>Options</h2>
<div class="option">
<input id="use-stun" type="checkbox"/>
<label for="use-stun">Use STUN server</label>
</div>
<button id="start" onclick="start()" disabled>Start</button>
<button id="stop" style="display: none" onclick="stop()">Stop</button>
<h2>State</h2>
<p>
ICE gathering state: <span id="ice-gathering-state"></span>
</p>
<p>
ICE connection state: <span id="ice-connection-state"></span>
</p>
<p>
Signaling state: <span id="signaling-state"></span>
</p>
<div id="media" style="display: none">
<h2>Media</h2>
<video id="video" autoplay playsinline></video>
</div>
<h2>Data channel</h2>
<pre id="data-channel" style="height: 200px;"></pre>
<h2>SDP</h2>
<h3>Offer</h3>
<pre id="offer-sdp"></pre>
<h3>Answer</h3>
<pre id="answer-sdp"></pre>
<script src="client.js"></script>
</body>
</html>

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/*
* libdatachannel client example
* Copyright (c) 2019-2020 Paul-Louis Ageneau
* Copyright (c) 2019 Murat Dogan
* Copyright (c) 2020 Will Munn
* Copyright (c) 2020 Nico Chatzi
* Copyright (c) 2020 Lara Mackey
* Copyright (c) 2020 Erik Cota-Robles
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#include "nlohmann/json.hpp"
#include "h264fileparser.hpp"
#include "opusfileparser.hpp"
#include "helpers.hpp"
#include "ArgParser.hpp"
using namespace rtc;
using namespace std;
using namespace std::chrono_literals;
using json = nlohmann::json;
template <class T> weak_ptr<T> make_weak_ptr(shared_ptr<T> ptr) { return ptr; }
/// all connected clients
unordered_map<string, shared_ptr<Client>> clients{};
/// Creates peer connection and client representation
/// @param config Configuration
/// @param wws Websocket for signaling
/// @param id Client ID
/// @returns Client
shared_ptr<Client> createPeerConnection(const Configuration &config,
weak_ptr<WebSocket> wws,
string id);
/// Creates stream
/// @param h264Samples Directory with H264 samples
/// @param fps Video FPS
/// @param opusSamples Directory with opus samples
/// @returns Stream object
shared_ptr<Stream> createStream(const string h264Samples, const unsigned fps, const string opusSamples);
/// Add client to stream
/// @param client Client
/// @param adding_video True if adding video
void addToStream(shared_ptr<Client> client, bool isAddingVideo);
/// Start stream
void startStream();
/// Main dispatch queue
DispatchQueue MainThread("Main");
/// Audio and video stream
optional<shared_ptr<Stream>> avStream = nullopt;
const string defaultRootDirectory = "../../../../examples/streamer/samples/";
const string defaultH264SamplesDirectory = defaultRootDirectory + "h264/";
string h264SamplesDirectory = defaultH264SamplesDirectory;
const string defaultOpusSamplesDirectory = defaultRootDirectory + "opus/";
string opusSamplesDirectory = defaultOpusSamplesDirectory;
const string defaultIPAddress = "127.0.0.1";
const uint16_t defaultPort = 8000;
string ip_address = defaultIPAddress;
uint16_t port = defaultPort;
/// Incomming message handler for websocket
/// @param message Incommint message
/// @param config Configuration
/// @param ws Websocket
void wsOnMessage(json message, Configuration config, shared_ptr<WebSocket> ws) {
auto it = message.find("id");
if (it == message.end())
return;
string id = it->get<string>();
it = message.find("type");
if (it == message.end())
return;
string type = it->get<string>();
if (type == "streamRequest") {
shared_ptr<Client> c = createPeerConnection(config, make_weak_ptr(ws), id);
clients.emplace(id, c);
} else if (type == "answer") {
shared_ptr<Client> c;
if (auto jt = clients.find(id); jt != clients.end()) {
auto pc = clients.at(id)->peerConnection;
auto sdp = message["sdp"].get<string>();
auto description = Description(sdp, type);
pc->setRemoteDescription(description);
}
}
}
int main(int argc, char **argv) try {
bool enableDebugLogs = false;
bool printHelp = false;
int c = 0;
auto parser = ArgParser({{"a", "audio"}, {"b", "video"}, {"d", "ip"}, {"p","port"}}, {{"h", "help"}, {"v", "verbose"}});
auto parsingResult = parser.parse(argc, argv, [](string key, string value) {
if (key == "audio") {
opusSamplesDirectory = value + "/";
} else if (key == "video") {
h264SamplesDirectory = value + "/";
} else if (key == "ip") {
ip_address = value;
} else if (key == "port") {
port = atoi(value.data());
} else {
cerr << "Invalid option --" << key << " with value " << value << endl;
return false;
}
return true;
}, [&enableDebugLogs, &printHelp](string flag){
if (flag == "verbose") {
enableDebugLogs = true;
} else if (flag == "help") {
printHelp = true;
} else {
cerr << "Invalid flag --" << flag << endl;
return false;
}
return true;
});
if (!parsingResult) {
return 1;
}
if (printHelp) {
cout << "usage: stream-h264 [-a opus_samples_folder] [-b h264_samples_folder] [-d ip_address] [-p port] [-v] [-h]" << endl
<< "Arguments:" << endl
<< "\t -a " << "Directory with opus samples (default: " << defaultOpusSamplesDirectory << ")." << endl
<< "\t -b " << "Directory with H264 samples (default: " << defaultH264SamplesDirectory << ")." << endl
<< "\t -d " << "Signaling server IP address (default: " << defaultIPAddress << ")." << endl
<< "\t -p " << "Signaling server port (default: " << defaultPort << ")." << endl
<< "\t -v " << "Enable debug logs." << endl
<< "\t -h " << "Print this help and exit." << endl;
return 0;
}
if (enableDebugLogs) {
InitLogger(LogLevel::Debug);
}
Configuration config;
string stunServer = "stun:stun.l.google.com:19302";
cout << "Stun server is " << stunServer << endl;
config.iceServers.emplace_back(stunServer);
string localId = "server";
cout << "The local ID is: " << localId << endl;
auto ws = make_shared<WebSocket>();
ws->onOpen([]() { cout << "WebSocket connected, signaling ready" << endl; });
ws->onClosed([]() { cout << "WebSocket closed" << endl; });
ws->onError([](const string &error) { cout << "WebSocket failed: " << error << endl; });
ws->onMessage([&](variant<binary, string> data) {
if (!holds_alternative<string>(data))
return;
json message = json::parse(get<string>(data));
MainThread.dispatch([message, config, ws]() {
wsOnMessage(message, config, ws);
});
});
const string url = "ws://" + ip_address + ":" + to_string(port) + "/" + localId;
cout << "Url is " << url << endl;
ws->open(url);
cout << "Waiting for signaling to be connected..." << endl;
while (!ws->isOpen()) {
if (ws->isClosed())
return 1;
this_thread::sleep_for(100ms);
}
while (true) {
string id;
cout << "Enter to exit" << endl;
cin >> id;
cin.ignore();
cout << "exiting" << endl;
break;
}
cout << "Cleaning up..." << endl;
return 0;
} catch (const std::exception &e) {
std::cout << "Error: " << e.what() << std::endl;
return -1;
}
shared_ptr<ClientTrackData> addVideo(const shared_ptr<PeerConnection> pc, const uint8_t payloadType, const uint32_t ssrc, const string cname, const string msid, const function<void (void)> onOpen) {
auto video = Description::Video(cname);
video.addH264Codec(payloadType);
video.addSSRC(ssrc, cname, msid, cname);
auto track = pc->addTrack(video);
// create RTP configuration
auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, H264RtpPacketizer::defaultClockRate));
// create packetizer
auto packetizer = shared_ptr<H264RtpPacketizer>(new H264RtpPacketizer(H264RtpPacketizer::Separator::Length, rtpConfig));
// create H264 handler
shared_ptr<H264PacketizationHandler> h264Handler(new H264PacketizationHandler(packetizer));
// add RTCP SR handler
auto srReporter = make_shared<RtcpSrReporter>(rtpConfig);
h264Handler->addToChain(srReporter);
// add RTCP NACK handler
auto nackResponder = make_shared<RtcpNackResponder>();
h264Handler->addToChain(nackResponder);
// set handler
track->setRtcpHandler(h264Handler);
track->onOpen(onOpen);
auto trackData = make_shared<ClientTrackData>(track, srReporter);
return trackData;
}
shared_ptr<ClientTrackData> addAudio(const shared_ptr<PeerConnection> pc, const uint8_t payloadType, const uint32_t ssrc, const string cname, const string msid, const function<void (void)> onOpen) {
auto audio = Description::Audio(cname);
audio.addOpusCodec(payloadType);
audio.addSSRC(ssrc, cname, msid, cname);
auto track = pc->addTrack(audio);
// create RTP configuration
auto rtpConfig = shared_ptr<RtpPacketizationConfig>(new RtpPacketizationConfig(ssrc, cname, payloadType, OpusRtpPacketizer::defaultClockRate));
// create packetizer
auto packetizer = make_shared<OpusRtpPacketizer>(rtpConfig);
// create opus handler
auto opusHandler = make_shared<OpusPacketizationHandler>(packetizer);
// add RTCP SR handler
auto srReporter = make_shared<RtcpSrReporter>(rtpConfig);
opusHandler->addToChain(srReporter);
// add RTCP NACK handler
auto nackResponder = make_shared<RtcpNackResponder>();
opusHandler->addToChain(nackResponder);
// set handler
track->setRtcpHandler(opusHandler);
track->onOpen(onOpen);
auto trackData = make_shared<ClientTrackData>(track, srReporter);
return trackData;
}
// Create and setup a PeerConnection
shared_ptr<Client> createPeerConnection(const Configuration &config,
weak_ptr<WebSocket> wws,
string id) {
auto pc = make_shared<PeerConnection>(config);
shared_ptr<Client> client(new Client(pc));
pc->onStateChange([id](PeerConnection::State state) {
cout << "State: " << state << endl;
if (state == PeerConnection::State::Disconnected ||
state == PeerConnection::State::Failed ||
state == PeerConnection::State::Closed) {
// remove disconnected client
MainThread.dispatch([id]() {
clients.erase(id);
});
}
});
pc->onGatheringStateChange(
[wpc = make_weak_ptr(pc), id, wws](PeerConnection::GatheringState state) {
cout << "Gathering State: " << state << endl;
if (state == PeerConnection::GatheringState::Complete) {
if(auto pc = wpc.lock()) {
auto description = pc->localDescription();
json message = {
{"id", id},
{"type", description->typeString()},
{"sdp", string(description.value())}
};
// Gathering complete, send answer
if (auto ws = wws.lock()) {
ws->send(message.dump());
}
}
}
});
client->video = addVideo(pc, 102, 1, "video-stream", "stream1", [id, wc = make_weak_ptr(client)]() {
MainThread.dispatch([wc]() {
if (auto c = wc.lock()) {
addToStream(c, true);
}
});
cout << "Video from " << id << " opened" << endl;
});
client->audio = addAudio(pc, 111, 2, "audio-stream", "stream1", [id, wc = make_weak_ptr(client)]() {
MainThread.dispatch([wc]() {
if (auto c = wc.lock()) {
addToStream(c, false);
}
});
cout << "Audio from " << id << " opened" << endl;
});
auto dc = pc->addDataChannel("ping-pong");
dc->onOpen([id, wdc = make_weak_ptr(dc)]() {
if (auto dc = wdc.lock()) {
dc->send("Ping");
}
});
dc->onMessage(nullptr, [id, wdc = make_weak_ptr(dc)](string msg) {
cout << "Message from " << id << " received: " << msg << endl;
if (auto dc = wdc.lock()) {
dc->send("Ping");
}
});
client->dataChannel = dc;
pc->setLocalDescription();
return client;
};
/// Create stream
shared_ptr<Stream> createStream(const string h264Samples, const unsigned fps, const string opusSamples) {
// video source
auto video = make_shared<H264FileParser>(h264Samples, fps, true);
// audio source
auto audio = make_shared<OPUSFileParser>(opusSamples, true);
auto stream = make_shared<Stream>(video, audio);
// set callback responsible for sample sending
stream->onSample([ws = make_weak_ptr(stream)](Stream::StreamSourceType type, uint64_t sampleTime, rtc::binary sample) {
vector<ClientTrack> tracks{};
string streamType = type == Stream::StreamSourceType::Video ? "video" : "audio";
// get track for given type
function<optional<shared_ptr<ClientTrackData>> (shared_ptr<Client>)> getTrackData = [type](shared_ptr<Client> client) {
return type == Stream::StreamSourceType::Video ? client->video : client->audio;
};
// get all clients with Ready state
for(auto id_client: clients) {
auto id = id_client.first;
auto client = id_client.second;
auto optTrackData = getTrackData(client);
if (client->getState() == Client::State::Ready && optTrackData.has_value()) {
auto trackData = optTrackData.value();
tracks.push_back(ClientTrack(id, trackData));
}
}
if (!tracks.empty()) {
auto message = make_message(move(sample));
for (auto clientTrack: tracks) {
auto client = clientTrack.id;
auto trackData = clientTrack.trackData;
// sample time is in us, we need to convert it to seconds
auto elapsedSeconds = double(sampleTime) / (1000 * 1000);
auto rtpConfig = trackData->sender->rtpConfig;
// get elapsed time in clock rate
uint32_t elapsedTimestamp = rtpConfig->secondsToTimestamp(elapsedSeconds);
// set new timestamp
rtpConfig->timestamp = rtpConfig->startTimestamp + elapsedTimestamp;
// get elapsed time in clock rate from last RTCP sender report
auto reportElapsedTimestamp = rtpConfig->timestamp - trackData->sender->previousReportedTimestamp;
// check if last report was at least 1 second ago
if (rtpConfig->timestampToSeconds(reportElapsedTimestamp) > 1) {
trackData->sender->setNeedsToReport();
}
cout << "Sending " << streamType << " sample with size: " << to_string(message->size()) << " to " << client << endl;
bool send = false;
try {
// send sample
send = trackData->track->send(*message);
} catch (...) {
send = false;
}
if (!send) {
cerr << "Unable to send "<< streamType << " packet" << endl;
break;
}
}
}
MainThread.dispatch([ws]() {
if (clients.empty()) {
// we have no clients, stop the stream
if (auto stream = ws.lock()) {
stream->stop();
}
}
});
});
return stream;
}
/// Start stream
void startStream() {
shared_ptr<Stream> stream;
if (avStream.has_value()) {
stream = avStream.value();
if (stream->isRunning) {
// stream is already running
return;
}
} else {
stream = createStream(h264SamplesDirectory, 30, opusSamplesDirectory);
avStream = stream;
}
stream->start();
}
/// Send previous key frame so browser can show something to user
/// @param stream Stream
/// @param video Video track data
void sendInitialNalus(shared_ptr<Stream> stream, shared_ptr<ClientTrackData> video) {
auto h264 = dynamic_cast<H264FileParser *>(stream->video.get());
auto initialNalus = h264->initialNALUS();
// send previous NALU key frame so users don't have to wait to see stream works
if (!initialNalus.empty()) {
const double frameDuration_s = double(h264->sampleDuration_us) / (1000 * 1000);
const uint32_t frameTimestampDuration = video->sender->rtpConfig->secondsToTimestamp(frameDuration_s);
video->sender->rtpConfig->timestamp = video->sender->rtpConfig->startTimestamp - frameTimestampDuration * 2;
video->track->send(initialNalus);
video->sender->rtpConfig->timestamp += frameTimestampDuration;
// Send initial NAL units again to start stream in firefox browser
video->track->send(initialNalus);
}
}
/// Add client to stream
/// @param client Client
/// @param adding_video True if adding video
void addToStream(shared_ptr<Client> client, bool isAddingVideo) {
if (client->getState() == Client::State::Waiting) {
client->setState(isAddingVideo ? Client::State::WaitingForAudio : Client::State::WaitingForVideo);
} else if ((client->getState() == Client::State::WaitingForAudio && !isAddingVideo)
|| (client->getState() == Client::State::WaitingForVideo && isAddingVideo)) {
// Audio and video tracks are collected now
assert(client->video.has_value() && client->audio.has_value());
auto video = client->video.value();
auto audio = client->audio.value();
auto currentTime_us = double(currentTimeInMicroSeconds());
auto currentTime_s = currentTime_us / (1000 * 1000);
// set start time of stream
video->sender->rtpConfig->setStartTime(currentTime_s, RtpPacketizationConfig::EpochStart::T1970);
audio->sender->rtpConfig->setStartTime(currentTime_s, RtpPacketizationConfig::EpochStart::T1970);
// start stat recording of RTCP SR
video->sender->startRecording();
audio->sender->startRecording();
if (avStream.has_value()) {
sendInitialNalus(avStream.value(), video);
}
client->setState(Client::State::Ready);
}
if (client->getState() == Client::State::Ready) {
startStream();
}
}

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@ -1,23 +0,0 @@
/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#include "opusfileparser.hpp"
using namespace std;
OPUSFileParser::OPUSFileParser(string directory, bool loop, uint32_t samplesPerSecond): FileParser(directory, ".opus", samplesPerSecond, loop) { }

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@ -1,32 +0,0 @@
/*
* libdatachannel streamer example
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef opusfileparser_hpp
#define opusfileparser_hpp
#include "fileparser.hpp"
class OPUSFileParser: public FileParser {
static const uint32_t defaultSamplesPerSecond = 50;
public:
OPUSFileParser(std::string directory, bool loop, uint32_t samplesPerSecond = OPUSFileParser::defaultSamplesPerSecond);
};
#endif /* opusfileparser_hpp */

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#!/usr/bin/env python3
import os
import getopt
import sys
import glob
from functools import reduce
from typing import Optional, List
class H264ByteStream:
@staticmethod
def nalu_type(nalu: bytes) -> int:
return nalu[0] & 0x1F
@staticmethod
def merge_sample(sample: List[bytes]) -> bytes:
result = bytes()
for nalu in sample:
result += len(nalu).to_bytes(4, byteorder='big') + nalu
return result
@staticmethod
def reduce_nalus_to_samples(samples: List[List[bytes]], current: bytes) -> List[List[bytes]]:
last_nalus = samples[-1]
samples[-1] = last_nalus + [current]
if H264ByteStream.nalu_type(current) in [1, 5]:
samples.append([])
return samples
def __init__(self, file_name: str):
with open(file_name, "rb") as file:
byte_stream = file.read()
long_split = byte_stream.split(b"\x00\x00\x00\x01")
splits = reduce(lambda acc, x: acc + x.split(b"\x00\x00\x01"), long_split, [])
nalus = filter(lambda x: len(x) > 0, splits)
self.samples = list(
filter(lambda x: len(x) > 0, reduce(H264ByteStream.reduce_nalus_to_samples, nalus, [[]])))
def generate(input_file: str, output_dir: str, max_samples: Optional[int], fps: Optional[int]):
if output_dir[-1] != "/":
output_dir += "/"
if os.path.isdir(output_dir):
files_to_delete = glob.glob(output_dir + "*.h264")
if len(files_to_delete) > 0:
print("Remove following files?")
for file in files_to_delete:
print(file)
response = input("Remove files? [y/n] ").lower()
if response != "y" and response != "yes":
print("Cancelling...")
return
print("Removing files")
for file in files_to_delete:
os.remove(file)
else:
os.makedirs(output_dir, exist_ok=True)
video_stream_file = "_video_stream.h264"
if os.path.isfile(video_stream_file):
os.remove(video_stream_file)
fps_line = "" if fps is None else "-filter:v fps=fps={} ".format(fps)
command = 'ffmpeg -i {} -an -vcodec libx264 -preset slow -profile baseline {}{}'.format(input_file, fps_line,
video_stream_file)
os.system(command)
data = H264ByteStream(video_stream_file)
index = 0
for sample in data.samples[:max_samples]:
name = "{}sample-{}.h264".format(output_dir, index)
index += 1
with open(name, 'wb') as file:
merged_sample = H264ByteStream.merge_sample(sample)
file.write(merged_sample)
os.remove(video_stream_file)
def main(argv):
input_file = None
default_output_dir = "h264/"
output_dir = default_output_dir
max_samples = None
fps = None
try:
opts, args = getopt.getopt(argv, "hi:o:m:f:", ["help", "ifile=", "odir=", "max=", "fps"])
except getopt.GetoptError:
print('generate_h264.py -i <input_files> [-o <output_files>] [-m <max_samples>] [-f <fps>] [-h]')
sys.exit(2)
for opt, arg in opts:
if opt in ("-h", "--help"):
print("Usage: generate_h264.py -i <input_files> [-o <output_files>] [-m <max_samples>] [-f <fps>] [-h]")
print("Arguments:")
print("\t-i,--ifile: Input file")
print("\t-o,--odir: Output directory (default: " + default_output_dir + ")")
print("\t-m,--max: Maximum generated samples")
print("\t-f,--fps: Output fps")
print("\t-h,--help: Print this help and exit")
sys.exit()
elif opt in ("-i", "--ifile"):
input_file = arg
elif opt in ("-o", "--ofile"):
output_dir = arg
elif opt in ("-m", "--max"):
max_samples = int(arg)
elif opt in ("-f", "--fps"):
fps = int(arg)
if input_file is None:
print("Missing argument -i")
sys.exit(2)
generate(input_file, output_dir, max_samples, fps)
if __name__ == "__main__":
main(sys.argv[1:])

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@ -1,142 +0,0 @@
#!/usr/bin/env python3
from kaitaistruct import KaitaiStruct, ValidationNotEqualError
import os
import getopt
import sys
import glob
from functools import reduce
class Ogg(KaitaiStruct):
"""Ogg is a popular media container format, which provides basic
streaming / buffering mechanisms and is content-agnostic. Most
popular codecs that are used within Ogg streams are Vorbis (thus
making Ogg/Vorbis streams) and Theora (Ogg/Theora).
Ogg stream is a sequence Ogg pages. They can be read sequentially,
or one can jump into arbitrary stream location and scan for "OggS"
sync code to find the beginning of a new Ogg page and continue
decoding the stream contents from that one.
"""
def __init__(self, _io, _parent=None, _root=None):
KaitaiStruct.__init__(self, _io)
self._parent = _parent
self._root = _root if _root else self
self._read()
def _read(self):
self.pages = []
i = 0
while not self._io.is_eof():
self.pages.append(Ogg.Page(self._io, self, self._root))
i += 1
class Page(KaitaiStruct):
"""Ogg page is a basic unit of data in an Ogg bitstream, usually
it's around 4-8 KB, with a maximum size of 65307 bytes.
"""
def __init__(self, _io, _parent=None, _root=None):
KaitaiStruct.__init__(self, _io)
self._parent = _parent
self._root = _root if _root else self
self._read()
def _read(self):
self.sync_code = self._io.read_bytes(4)
if not self.sync_code == b"\x4F\x67\x67\x53":
raise ValidationNotEqualError(b"\x4F\x67\x67\x53", self.sync_code, self._io,
u"/types/page/seq/0")
self.version = self._io.read_bytes(1)
if not self.version == b"\x00":
raise ValidationNotEqualError(b"\x00", self.version, self._io, u"/types/page/seq/1")
self.reserved1 = self._io.read_bits_int_be(5)
self.is_end_of_stream = self._io.read_bits_int_be(1) != 0
self.is_beginning_of_stream = self._io.read_bits_int_be(1) != 0
self.is_continuation = self._io.read_bits_int_be(1) != 0
self._io.align_to_byte()
self.granule_pos = self._io.read_u8le()
self.bitstream_serial = self._io.read_u4le()
self.page_seq_num = self._io.read_u4le()
self.crc32 = self._io.read_u4le()
self.num_segments = self._io.read_u1()
self.len_segments = [None] * self.num_segments
for i in range(self.num_segments):
self.len_segments[i] = self._io.read_u1()
self.segments = [None] * self.num_segments
for i in range(self.num_segments):
self.segments[i] = self._io.read_bytes(self.len_segments[i])
def generate(input_file: str, output_dir: str, max_samples: int):
if output_dir[-1] != "/":
output_dir += "/"
if os.path.isdir(output_dir):
files_to_delete = glob.glob(output_dir + "*.opus")
if len(files_to_delete) > 0:
print("Remove following files?")
for file in files_to_delete:
print(file)
response = input("Remove files? [y/n] ").lower()
if response != "y" and response != "yes":
print("Cancelling...")
return
print("Removing files")
for file in files_to_delete:
os.remove(file)
else:
os.makedirs(output_dir, exist_ok=True)
audio_stream_file = "_audio_stream.ogg"
if os.path.isfile(audio_stream_file):
os.remove(audio_stream_file)
os.system('ffmpeg -i {} -vn -ar 48000 -ac 2 -vbr off -acodec libopus -ab 64k {}'.format(input_file, audio_stream_file))
data = Ogg.from_file(audio_stream_file)
index = 0
valid_pages = data.pages[2:]
segments = list(reduce(lambda x, y: x + y.segments, valid_pages, []))[:max_samples]
for segment in segments:
name = "{}sample-{}.opus".format(output_dir, index)
index += 1
with open(name, 'wb') as file:
assert len(list(segment)) == 160
file.write(segment)
os.remove(audio_stream_file)
def main(argv):
input_file = None
default_output_dir = "opus/"
output_dir = default_output_dir
max_samples = None
try:
opts, args = getopt.getopt(argv, "hi:o:m:", ["help", "ifile=", "odir=", "max="])
except getopt.GetoptError:
print('generate_opus.py -i <input_files> [-o <output_files>] [-m <max_samples>] [-h]')
sys.exit(2)
for opt, arg in opts:
if opt in ("-h", "--help"):
print("Usage: generate_opus.py -i <input_files> [-o <output_files>] [-m <max_samples>] [-h]")
print("Arguments:")
print("\t-i,--ifile: Input file")
print("\t-o,--odir: Output directory (default: " + default_output_dir + ")")
print("\t-m,--max: Maximum generated samples")
print("\t-h,--help: Print this help and exit")
sys.exit()
elif opt in ("-i", "--ifile"):
input_file = arg
elif opt in ("-o", "--ofile"):
output_dir = arg
elif opt in ("-m", "--max"):
max_samples = int(arg)
if input_file is None:
print("Missing argument -i")
sys.exit(2)
generate(input_file, output_dir, max_samples)
if __name__ == "__main__":
main(sys.argv[1:])

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