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ZLMediaKit
Commit a0de7577 by custompal
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Merge branch 'master' into cus_pr 

同步代码
parents 82f6bfc7 d4ad44c8
正在显示 包含 594 行增加 和 266 行删除
.clang-format
......@@ -71,6 +71,7 @@ SpaceBeforeInheritanceColon: true
SpaceBeforeParens: ControlStatements
# 空 {} 中不加空格
SpaceInEmptyBlock: false
Standard: C++11
# Tab 占 4 位
TabWidth: 4
# 不使用 TAB
......
.github/workflows/android.yml
name: Android CI
name: Android
on: [push, pull_request]
jobs:
build:
......
.github/workflows/ccpp.yml → .github/workflows/linux.yml
name: linux C/C++ CI
name: Linux
on: [push, pull_request]
......
.github/workflows/macos.yml
name: macos C/C++ CI
name: macOS
on: [push, pull_request]
......
.github/workflows/style.yml 0 → 100644
name: style check
on: [pull_request]
jobs:
check:
runs-on: ubuntu-latest
steps:
- uses: actions/checkout@v2
with:
# with all history
fetch-depth: 0
- name: Validate BOM
run: |
ret=0
for i in $(git diff --name-only origin/${GITHUB_BASE_REF}...${GITHUB_SHA}); do
if [ -f ${i} ]; then
case ${i} in
*.c|*.cc|*.cpp|*.h)
if file ${i} | grep -qv BOM; then
echo "Missing BOM in ${i}" && ret=1;
fi
;;
esac
fi
done
exit ${ret}
.github/workflows/msvc-ci.yml → .github/workflows/windows.yml
name: MSVC C/C++ CI
name: Windows
on: [push, pull_request]
......
ZLToolKit @ 61f2c6c8
Subproject commit 136b6b2f28193da218f577423db217aeb0f7aa6a
Subproject commit 61f2c6c8d4288c2c60299a84473d9cfec113891c
CMakeLists.txt
......@@ -22,6 +22,10 @@ endif()
set(CMAKE_CXX_STANDARD 11)
#加载自定义模块
set(CMAKE_MODULE_PATH ${CMAKE_MODULE_PATH} "${CMAKE_CURRENT_SOURCE_DIR}/cmake")
set(DEP_ROOT_DIR ${CMAKE_SOURCE_DIR}/3rdpart/external-${CMAKE_SYSTEM_NAME})
if(NOT EXISTS ${DEP_ROOT_DIR})
file(MAKE_DIRECTORY ${DEP_ROOT_DIR})
endif()
find_program(CCACHE_FOUND ccache)
if(CCACHE_FOUND)
......@@ -108,6 +112,7 @@ option(ENABLE_MSVC_MT "Enable MSVC Mt/Mtd lib" true)
option(ENABLE_API_STATIC_LIB "Enable mk_api static lib" false)
option(USE_SOLUTION_FOLDERS "Enable solution dir supported" ON)
option(ENABLE_SRT "Enable SRT" true)
option(ENABLE_JEMALLOC_STATIC "Enable static linking to the jemalloc library" false)
# ----------------------------------------------------------------------------
# Solution folders:
# ----------------------------------------------------------------------------
......@@ -299,6 +304,13 @@ if (ENABLE_FFMPEG)
endif ()
if(ENABLE_JEMALLOC_STATIC)
include(cmake/Jemalloc.cmake)
include_directories(${DEP_ROOT_DIR}/${JEMALLOC_NAME}/include/jemalloc)
link_directories(${DEP_ROOT_DIR}/${JEMALLOC_NAME}/lib)
set(JEMALLOC_ROOT_DIR "${DEP_ROOT_DIR}/${JEMALLOC_NAME}")
endif ()
#默认链接jemalloc库避免内存碎片
find_package(JEMALLOC QUIET)
if (JEMALLOC_FOUND)
......
README.md
![logo](https://raw.githubusercontent.com/xia-chu/ZLMediaKit/master/www/logo.png)
![logo](https://raw.githubusercontent.com/ZLMediaKit/ZLMediaKit/master/www/logo.png)
# 一个基于C++11的高性能运营级流媒体服务框架
[![license](http://img.shields.io/badge/license-MIT-green.svg)](https://github.com/xia-chu/ZLMediaKit/blob/master/LICENSE)
[![C++](https://img.shields.io/badge/language-c++-red.svg)](https://en.cppreference.com/)
[![platform](https://img.shields.io/badge/platform-linux%20|%20macos%20|%20windows-blue.svg)](https://github.com/xia-chu/ZLMediaKit)
[![PRs Welcome](https://img.shields.io/badge/PRs-welcome-yellow.svg)](https://github.com/xia-chu/ZLMediaKit/pulls)
[![Build Status](https://travis-ci.org/xia-chu/ZLMediaKit.svg?branch=master)](https://travis-ci.org/xia-chu/ZLMediaKit)
[![Docker](https://img.shields.io/docker/pulls/zlmediakit/zlmediakit)](https://hub.docker.com/r/zlmediakit/zlmediakit/tags)
[![](https://img.shields.io/badge/license-MIT-green.svg)](https://github.com/ZLMediaKit/ZLMediaKit/blob/master/LICENSE)
[![](https://img.shields.io/badge/language-c++-red.svg)](https://en.cppreference.com/)
[![](https://img.shields.io/badge/platform-linux%20|%20macos%20|%20windows-blue.svg)](https://github.com/ZLMediaKit/ZLMediaKit)
[![](https://img.shields.io/badge/PRs-welcome-yellow.svg)](https://github.com/ZLMediaKit/ZLMediaKit/pulls)
[![](https://github.com/ZLMediaKit/ZLMediaKit/actions/workflows/android.yml/badge.svg)](https://github.com/ZLMediaKit/ZLMediaKit)
[![](https://github.com/ZLMediaKit/ZLMediaKit/actions/workflows/linux.yml/badge.svg)](https://github.com/ZLMediaKit/ZLMediaKit)
[![](https://github.com/ZLMediaKit/ZLMediaKit/actions/workflows/macos.yml/badge.svg)](https://github.com/ZLMediaKit/ZLMediaKit)
[![](https://github.com/ZLMediaKit/ZLMediaKit/actions/workflows/windows.yml/badge.svg)](https://github.com/ZLMediaKit/ZLMediaKit)
[![](https://github.com/ZLMediaKit/ZLMediaKit/actions/workflows/docker.yml/badge.svg)](https://hub.docker.com/r/zlmediakit/zlmediakit/tags)
[![](https://img.shields.io/docker/pulls/zlmediakit/zlmediakit)](https://hub.docker.com/r/zlmediakit/zlmediakit/tags)
## 项目特点
......@@ -16,10 +22,10 @@
- 使用多路复用/多线程/异步网络IO模式开发，并发性能优越，支持海量客户端连接。
- 代码经过长期大量的稳定性、性能测试，已经在线上商用验证已久。
- 支持linux、macos、ios、android、windows全平台。
- 支持画面秒开、极低延时([500毫秒内，最低可达100毫秒](https://github.com/xia-chu/ZLMediaKit/wiki/%E5%BB%B6%E6%97%B6%E6%B5%8B%E8%AF%95))。
- 提供完善的标准[C API](https://github.com/xia-chu/ZLMediaKit/tree/master/api/include),可以作SDK用，或供其他语言调用。
- 提供完整的[MediaServer](https://github.com/xia-chu/ZLMediaKit/tree/master/server)服务器，可以免开发直接部署为商用服务器。
- 提供完善的[restful api](https://github.com/xia-chu/ZLMediaKit/wiki/MediaServer%E6%94%AF%E6%8C%81%E7%9A%84HTTP-API)以及[web hook](https://github.com/xia-chu/ZLMediaKit/wiki/MediaServer%E6%94%AF%E6%8C%81%E7%9A%84HTTP-HOOK-API)，支持丰富的业务逻辑。
- 支持画面秒开、极低延时([500毫秒内，最低可达100毫秒](https://github.com/ZLMediaKit/ZLMediaKit/wiki/%E5%BB%B6%E6%97%B6%E6%B5%8B%E8%AF%95))。
- 提供完善的标准[C API](https://github.com/ZLMediaKit/ZLMediaKit/tree/master/api/include),可以作SDK用，或供其他语言调用。
- 提供完整的[MediaServer](https://github.com/ZLMediaKit/ZLMediaKit/tree/master/server)服务器，可以免开发直接部署为商用服务器。
- 提供完善的[restful api](https://github.com/ZLMediaKit/ZLMediaKit/wiki/MediaServer%E6%94%AF%E6%8C%81%E7%9A%84HTTP-API)以及[web hook](https://github.com/ZLMediaKit/ZLMediaKit/wiki/MediaServer%E6%94%AF%E6%8C%81%E7%9A%84HTTP-HOOK-API)，支持丰富的业务逻辑。
- 打通了视频监控协议栈与直播协议栈，对RTSP/RTMP支持都很完善。
- 全面支持H265/H264/AAC/G711/OPUS。
- 功能完善，支持集群、按需转协议、按需推拉流、先播后推、断连续推等功能。
......@@ -58,7 +64,7 @@
- 支持websocket-flv直播
- 支持H264/H265/AAC/G711/OPUS编码，其他编码能转发但不能转协议
- 支持[RTMP-H265](https://github.com/ksvc/FFmpeg/wiki)
- 支持[RTMP-OPUS](https://github.com/xia-chu/ZLMediaKit/wiki/RTMP%E5%AF%B9H265%E5%92%8COPUS%E7%9A%84%E6%94%AF%E6%8C%81)
- 支持[RTMP-OPUS](https://github.com/ZLMediaKit/ZLMediaKit/wiki/RTMP%E5%AF%B9H265%E5%92%8COPUS%E7%9A%84%E6%94%AF%E6%8C%81)
- HLS
- 支持HLS文件生成，自带HTTP文件服务器
......@@ -126,15 +132,15 @@
## 编译以及测试
**编译前务必仔细参考wiki:[快速开始](https://github.com/xia-chu/ZLMediaKit/wiki/%E5%BF%AB%E9%80%9F%E5%BC%80%E5%A7%8B)操作!!!**
**编译前务必仔细参考wiki:[快速开始](https://github.com/ZLMediaKit/ZLMediaKit/wiki/%E5%BF%AB%E9%80%9F%E5%BC%80%E5%A7%8B)操作!!!**
## 怎么使用
你有三种方法使用ZLMediaKit，分别是：
- 1、使用c api，作为sdk使用，请参考[这里](https://github.com/xia-chu/ZLMediaKit/tree/master/api/include).
- 2、作为独立的流媒体服务器使用，不想做c/c++开发的，可以参考 [restful api](https://github.com/xia-chu/ZLMediaKit/wiki/MediaServer支持的HTTP-API) 和 [web hook](https://github.com/xia-chu/ZLMediaKit/wiki/MediaServer支持的HTTP-HOOK-API ).
- 3、如果想做c/c++开发，添加业务逻辑增加功能，可以参考这里的[测试程序](https://github.com/xia-chu/ZLMediaKit/tree/master/tests).
- 1、使用c api，作为sdk使用，请参考[这里](https://github.com/ZLMediaKit/ZLMediaKit/tree/master/api/include).
- 2、作为独立的流媒体服务器使用，不想做c/c++开发的，可以参考 [restful api](https://github.com/ZLMediaKit/ZLMediaKit/wiki/MediaServer支持的HTTP-API) 和 [web hook](https://github.com/ZLMediaKit/ZLMediaKit/wiki/MediaServer支持的HTTP-HOOK-API ).
- 3、如果想做c/c++开发，添加业务逻辑增加功能，可以参考这里的[测试程序](https://github.com/ZLMediaKit/ZLMediaKit/tree/master/tests).
## Docker 镜像
......@@ -167,6 +173,7 @@ bash build_docker_images.sh
- [Go实现的海康ehome服务器](https://github.com/tsingeye/FreeEhome)
- 客户端
- [c sdk完整c#包装库](https://github.com/malegend/ZLMediaKit.Autogen)
- [基于C SDK实现的推流客户端](https://github.com/hctym1995/ZLM_ApiDemo)
- [C#版本的Http API与Hook](https://github.com/chengxiaosheng/ZLMediaKit.HttpApi)
- [DotNetCore的RESTful客户端](https://github.com/MingZhuLiu/ZLMediaKit.DotNetCore.Sdk)
......@@ -195,7 +202,7 @@ bash build_docker_images.sh
- 1、仔细看下readme、wiki，如果有必要可以查看下issue.
- 2、如果您的问题还没解决，可以提issue.
- 3、有些问题，如果不具备参考性的，无需在issue提的，可以在qq群提.
- 4、QQ私聊一般不接受无偿技术咨询和支持([为什么不提倡QQ私聊](https://github.com/xia-chu/ZLMediaKit/wiki/%E4%B8%BA%E4%BB%80%E4%B9%88%E4%B8%8D%E5%BB%BA%E8%AE%AEQQ%E7%A7%81%E8%81%8A%E5%92%A8%E8%AF%A2%E9%97%AE%E9%A2%98%EF%BC%9F)).
- 4、QQ私聊一般不接受无偿技术咨询和支持([为什么不提倡QQ私聊](https://github.com/ZLMediaKit/ZLMediaKit/wiki/%E4%B8%BA%E4%BB%80%E4%B9%88%E4%B8%8D%E5%BB%BA%E8%AE%AEQQ%E7%A7%81%E8%81%8A%E5%92%A8%E8%AF%A2%E9%97%AE%E9%A2%98%EF%BC%9F)).
## 特别感谢
......@@ -278,4 +285,4 @@ bash build_docker_images.sh
本项目已经得到不少公司和个人开发者的认可，据作者不完全统计，
使用本项目的公司包括知名的互联网巨头、国内排名前列的云服务公司、多家知名的AI独角兽公司，
以及一系列中小型公司。使用者可以通过在 [issue](https://github.com/xia-chu/ZLMediaKit/issues/511) 上粘贴公司的大名和相关项目介绍为本项目背书，感谢支持！
以及一系列中小型公司。使用者可以通过在 [issue](https://github.com/ZLMediaKit/ZLMediaKit/issues/511) 上粘贴公司的大名和相关项目介绍为本项目背书，感谢支持！
api/source/mk_media.cpp
......@@ -187,7 +187,6 @@ API_EXPORT int API_CALL mk_media_init_video(mk_media ctx, int codec_id, int widt
info.iWidth = width;
info.iHeight = height;
info.iBitRate = bit_rate;
(*obj)->getChannel()->initVideo(info);
return (*obj)->getChannel()->initVideo(info);
}
......
api/source/mk_track.cpp
......@@ -10,12 +10,13 @@
#include "mk_track.h"
#include "Extension/Track.h"
#include "Extension/Factory.h"
using namespace std;
using namespace toolkit;
using namespace mediakit;
class VideoTrackForC : public VideoTrack {
class VideoTrackForC : public VideoTrack, public std::enable_shared_from_this<VideoTrackForC> {
public:
VideoTrackForC(int codec_id, codec_args *args) {
_codec_id = (CodecId) codec_id;
......@@ -49,7 +50,8 @@ public:
}
Track::Ptr clone() override {
return std::make_shared<std::remove_reference<decltype(*this)>::type>(*this);
auto track_in = std::shared_ptr<Track>(shared_from_this());
return Factory::getTrackByAbstractTrack(track_in);
}
Sdp::Ptr getSdp() override {
......@@ -61,15 +63,16 @@ private:
codec_args _args;
};
class AudioTrackForC : public AudioTrackImp {
class AudioTrackForC : public AudioTrackImp, public std::enable_shared_from_this<AudioTrackForC> {
public:
~AudioTrackForC() override = default;
AudioTrackForC(int codec_id, codec_args *args) :
AudioTrackImp((CodecId) codec_id, args->audio.sample_rate, args->audio.channels, 16) {}
AudioTrackImp((CodecId) codec_id, args->audio.sample_rate, args->audio.channels, 16) {}
Track::Ptr clone() override {
return std::make_shared<std::remove_reference<decltype(*this)>::type>(*this);
auto track_in = std::shared_ptr<Track>(shared_from_this());
return Factory::getTrackByAbstractTrack(track_in);
}
Sdp::Ptr getSdp() override {
......
api/tests/h264_media_server.c
......@@ -64,8 +64,12 @@ int main(int argc, char *argv[]) {
mk_media media = mk_media_create("__defaultVhost__", "live", "test", 0, 0, 0);
//h264的codec
mk_media_init_video(media, 0, 0, 0, 0, 2 * 104 * 1024);
//mk_media_init_video(media, 0, 0, 0, 0, 2 * 104 * 1024);
codec_args v_args={0};
mk_track v_track = mk_track_create(MKCodecH264,&v_args);
mk_media_init_track(media,v_track);
mk_media_init_complete(media);
mk_track_unref(v_track);
//创建h264分帧器
mk_h264_splitter splitter = mk_h264_splitter_create(on_h264_frame, media);
......
cmake/FindJEMALLOC.cmake
find_path(JEMALLOC_INCLUDE_DIR
NAMES jemalloc/jemalloc.h
)
# Tries to find Jemalloc headers and libraries.
#
# Usage of this module as follows:
#
# find_package(jemalloc)
#
# Variables used by this module, they can change the default behaviour and need
# to be set before calling find_package:
#
# JEMALLOC_ROOT_DIR Set this variable to the root installation of
# Jemalloc if the module has problems finding
# the proper installation path.
#
# Variables defined by this module:
#
# JEMALLOC_FOUND System has Jemalloc libs/headers
# JEMALLOC_LIBRARIES The Jemalloc libraries
# JEMALLOC_INCLUDE_DIR The location of Jemalloc headers
if (ENABLE_JEMALLOC_STATIC)
find_path(JEMALLOC_INCLUDE_DIR
NAMES jemalloc.h
HINTS ${JEMALLOC_ROOT_DIR}/include/jemalloc
NO_DEFAULT_PATH)
find_library(JEMALLOC_LIBRARY
NAMES jemalloc
)
find_library(JEMALLOC_LIBRARIES
NAMES jemalloc
HINTS ${JEMALLOC_ROOT_DIR}/lib
NO_DEFAULT_PATH)
else ()
find_path(JEMALLOC_INCLUDE_DIR
NAMES jemalloc/jemalloc.h
)
set(JEMALLOC_INCLUDE_DIRS ${JEMALLOC_INCLUDE_DIR})
set(JEMALLOC_LIBRARIES ${JEMALLOC_LIBRARY})
find_library(JEMALLOC_LIBRARIES
NAMES jemalloc
)
endif ()
include(FindPackageHandleStandardArgs)
find_package_handle_standard_args(JEMALLOC DEFAULT_MSG
JEMALLOC_LIBRARIES
JEMALLOC_INCLUDE_DIR)
find_package_handle_standard_args(JEMALLOC DEFAULT_MSG JEMALLOC_LIBRARY JEMALLOC_INCLUDE_DIR)
mark_as_advanced(
JEMALLOC_ROOT_DIR
JEMALLOC_LIBRARIES
JEMALLOC_INCLUDE_DIR)
cmake/Jemalloc.cmake 0 → 100644
# Download and build Jemalloc
set(JEMALLOC_VERSION 5.2.1)
set(JEMALLOC_NAME jemalloc-${JEMALLOC_VERSION})
set(JEMALLOC_TAR_PATH ${DEP_ROOT_DIR}/${JEMALLOC_NAME}.tar.bz2)
list(APPEND jemalloc_CONFIG_ARGS --disable-initial-exec-tls)
list(APPEND jemalloc_CONFIG_ARGS --without-export)
list(APPEND jemalloc_CONFIG_ARGS --disable-stats)
list(APPEND jemalloc_CONFIG_ARGS --disable-libdl)
#list(APPEND jemalloc_CONFIG_ARGS --disable-cxx)
#list(APPEND jemalloc_CONFIG_ARGS --with-jemalloc-prefix=je_)
#list(APPEND jemalloc_CONFIG_ARGS --enable-debug)
if(NOT EXISTS ${JEMALLOC_TAR_PATH})
message(STATUS "Downloading ${JEMALLOC_NAME}...")
file(DOWNLOAD https://github.com/jemalloc/jemalloc/releases/download/${JEMALLOC_VERSION}/${JEMALLOC_NAME}.tar.bz2
${JEMALLOC_TAR_PATH})
endif()
SET( DIR_CONTAINING_JEMALLOC ${DEP_ROOT_DIR}/${JEMALLOC_NAME} )
if(NOT EXISTS ${DIR_CONTAINING_JEMALLOC})
message(STATUS "Extracting jemalloc...")
execute_process(COMMAND ${CMAKE_COMMAND} -E tar xzf ${JEMALLOC_TAR_PATH} WORKING_DIRECTORY ${DEP_ROOT_DIR})
endif()
if(NOT EXISTS ${DIR_CONTAINING_JEMALLOC}/Makefile)
message("Configuring jemalloc locally...")
# Builds with "--with-jemalloc-prefix=je_" on OSX
# SET( BASH_COMMAND_TO_RUN bash -l -c "cd ${DIR_CONTAINING_JEMALLOC} && ./configure ${jemalloc_CONFIG_ARGS}" )
#
# EXECUTE_PROCESS( COMMAND ${BASH_COMMAND_TO_RUN}
# WORKING_DIRECTORY ${DIR_CONTAINING_JEMALLOC} RESULT_VARIABLE JEMALLOC_CONFIGURE )
execute_process(COMMAND ./configure ${jemalloc_CONFIG_ARGS} WORKING_DIRECTORY ${DIR_CONTAINING_JEMALLOC} RESULT_VARIABLE JEMALLOC_CONFIGURE)
if(NOT JEMALLOC_CONFIGURE EQUAL 0)
message(FATAL_ERROR "${JEMALLOC_NAME} configure failed!")
message("${JEMALLOC_CONFIGURE}")
endif()
endif()
if(NOT EXISTS ${DIR_CONTAINING_JEMALLOC}/lib/libjemalloc.a)
message("Building jemalloc locally...")
execute_process(COMMAND make "build_lib_static" WORKING_DIRECTORY ${DIR_CONTAINING_JEMALLOC})
if(NOT EXISTS ${DIR_CONTAINING_JEMALLOC}/lib/libjemalloc.a)
message(FATAL_ERROR "${JEMALLOC_NAME} build failed!")
endif()
endif()
\ No newline at end of file
conf/config.ini
......@@ -298,6 +298,7 @@ g711a_pt=8
#rtc播放推流、播放超时时间
timeoutSec=15
#本机对rtc客户端的可见ip，作为服务器时一般为公网ip，可有多个，用','分开，当置空时，会自动获取网卡ip
#同时支持环境变量，以$开头，如"$EXTERN_IP"; 请参考：https://github.com/ZLMediaKit/ZLMediaKit/pull/1786
externIP=
#rtc udp服务器监听端口号，所有rtc客户端将通过该端口传输stun/dtls/srtp/srtcp数据，
#该端口是多线程的，同时支持客户端网络切换导致的连接迁移
......
src/Codec/Transcode.cpp
......@@ -47,13 +47,13 @@ static void on_ffmpeg_log(void *ctx, int level, const char *fmt, va_list args) {
}
LogLevel lev;
switch (level) {
case AV_LOG_FATAL: lev = LError; break;
case AV_LOG_FATAL:
case AV_LOG_ERROR: lev = LError; break;
case AV_LOG_WARNING: lev = LWarn; break;
case AV_LOG_INFO: lev = LInfo; break;
case AV_LOG_VERBOSE: lev = LDebug; break;
case AV_LOG_VERBOSE:
case AV_LOG_DEBUG: lev = LDebug; break;
case AV_LOG_TRACE: lev = LTrace; break;
case AV_LOG_TRACE:
default: lev = LTrace; break;
}
LoggerWrapper::printLogV(::toolkit::getLogger(), lev, __FILE__, ctx ? av_default_item_name(ctx) : "NULL", level, fmt, args);
......@@ -63,7 +63,9 @@ static bool setupFFmpeg_l() {
av_log_set_level(AV_LOG_TRACE);
av_log_set_flags(AV_LOG_PRINT_LEVEL);
av_log_set_callback(on_ffmpeg_log);
#if (LIBAVCODEC_VERSION_MAJOR < 58)
avcodec_register_all();
#endif
return true;
}
......@@ -243,14 +245,14 @@ AVFrame *FFmpegFrame::get() const {
void FFmpegFrame::fillPicture(AVPixelFormat target_format, int target_width, int target_height) {
assert(_data == nullptr);
_data = new char[avpicture_get_size(target_format, target_width, target_height)];
avpicture_fill((AVPicture *) _frame.get(), (uint8_t *) _data, target_format, target_width, target_height);
_data = new char[av_image_get_buffer_size(target_format, target_width, target_height, 1)];
av_image_fill_arrays(_frame->data, _frame->linesize, (uint8_t *) _data, target_format, target_width, target_height,1);
}
///////////////////////////////////////////////////////////////////////////
template<bool decoder = true>
static inline AVCodec *getCodec_l(const char *name) {
static inline const AVCodec *getCodec_l(const char *name) {
auto codec = decoder ? avcodec_find_decoder_by_name(name) : avcodec_find_encoder_by_name(name);
if (codec) {
InfoL << (decoder ? "got decoder:" : "got encoder:") << name;
......@@ -261,7 +263,7 @@ static inline AVCodec *getCodec_l(const char *name) {
}
template<bool decoder = true>
static inline AVCodec *getCodec_l(enum AVCodecID id) {
static inline const AVCodec *getCodec_l(enum AVCodecID id) {
auto codec = decoder ? avcodec_find_decoder(id) : avcodec_find_encoder(id);
if (codec) {
InfoL << (decoder ? "got decoder:" : "got encoder:") << avcodec_get_name(id);
......@@ -277,7 +279,7 @@ public:
CodecName(enum AVCodecID id) : _id(id) {}
template <bool decoder>
AVCodec *getCodec() const {
const AVCodec *getCodec() const {
if (!_codec_name.empty()) {
return getCodec_l<decoder>(_codec_name.data());
}
......@@ -290,8 +292,8 @@ private:
};
template <bool decoder = true>
static inline AVCodec *getCodec(const std::initializer_list<CodecName> &codec_list) {
AVCodec *ret = nullptr;
static inline const AVCodec *getCodec(const std::initializer_list<CodecName> &codec_list) {
const AVCodec *ret = nullptr;
for (int i = codec_list.size(); i >= 1; --i) {
ret = codec_list.begin()[i - 1].getCodec<decoder>();
if (ret) {
......@@ -303,8 +305,8 @@ static inline AVCodec *getCodec(const std::initializer_list<CodecName> &codec_li
FFmpegDecoder::FFmpegDecoder(const Track::Ptr &track, int thread_num) {
setupFFmpeg();
AVCodec *codec = nullptr;
AVCodec *codec_default = nullptr;
const AVCodec *codec = nullptr;
const AVCodec *codec_default = nullptr;
switch (track->getCodecId()) {
case CodecH264:
codec_default = getCodec({AV_CODEC_ID_H264});
......@@ -358,7 +360,9 @@ FFmpegDecoder::FFmpegDecoder(const Track::Ptr &track, int thread_num) {
}
//保存AVFrame的引用
#ifdef FF_API_OLD_ENCDEC
_context->refcounted_frames = 1;
#endif
_context->flags |= AV_CODEC_FLAG_LOW_DELAY;
_context->flags2 |= AV_CODEC_FLAG2_FAST;
if (track->getTrackType() == TrackVideo) {
......@@ -539,7 +543,7 @@ FFmpegSwr::~FFmpegSwr() {
FFmpegFrame::Ptr FFmpegSwr::inputFrame(const FFmpegFrame::Ptr &frame) {
if (frame->get()->format == _target_format &&
frame->get()->channels == _target_channels &&
frame->get()->channel_layout == _target_channel_layout &&
frame->get()->channel_layout == (uint64_t)_target_channel_layout &&
frame->get()->sample_rate == _target_samplerate) {
//不转格式
return frame;
......@@ -596,7 +600,8 @@ int FFmpegSws::inputFrame(const FFmpegFrame::Ptr &frame, uint8_t *data) {
}
AVFrame dst;
memset(&dst, 0, sizeof(dst));
avpicture_fill((AVPicture *) &dst, data, _target_format, _target_width, _target_height);
av_image_fill_arrays(dst.data, dst.linesize, data, _target_format, _target_width, _target_height,1);
if (!_ctx) {
_ctx = sws_getContext(frame->get()->width, frame->get()->height, (enum AVPixelFormat) frame->get()->format,
_target_width, _target_height, _target_format, SWS_FAST_BILINEAR, NULL, NULL, NULL);
......
src/Codec/Transcode.h
......@@ -25,6 +25,7 @@ extern "C" {
#include "libavcodec/avcodec.h"
#include "libswresample/swresample.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/imgutils.h"
#ifdef __cplusplus
}
#endif
......
src/Common/MultiMediaSourceMuxer.cpp
......@@ -354,64 +354,12 @@ void MultiMediaSourceMuxer::resetTracks() {
}
}
//该类实现frame级别的时间戳覆盖
class FrameModifyStamp : public Frame{
public:
typedef std::shared_ptr<FrameModifyStamp> Ptr;
FrameModifyStamp(const Frame::Ptr &frame, Stamp &stamp){
_frame = frame;
//覆盖时间戳
stamp.revise(frame->dts(), frame->pts(), _dts, _pts, true);
}
~FrameModifyStamp() override {}
uint32_t dts() const override{
return (uint32_t)_dts;
}
uint32_t pts() const override{
return (uint32_t)_pts;
}
size_t prefixSize() const override {
return _frame->prefixSize();
}
bool keyFrame() const override {
return _frame->keyFrame();
}
bool configFrame() const override {
return _frame->configFrame();
}
bool cacheAble() const override {
return _frame->cacheAble();
}
char *data() const override {
return _frame->data();
}
size_t size() const override {
return _frame->size();
}
CodecId getCodecId() const override {
return _frame->getCodecId();
}
private:
int64_t _dts;
int64_t _pts;
Frame::Ptr _frame;
};
bool MultiMediaSourceMuxer::onTrackFrame(const Frame::Ptr &frame_in) {
GET_CONFIG(bool, modify_stamp, General::kModifyStamp);
auto frame = frame_in;
if (modify_stamp) {
//开启了时间戳覆盖
frame = std::make_shared<FrameModifyStamp>(frame, _stamp[frame->getTrackType()]);
frame = std::make_shared<FrameStamp>(frame, _stamp[frame->getTrackType()],true);
}
bool ret = false;
......
src/Extension/AAC.cpp
......@@ -274,6 +274,9 @@ bool AACTrack::inputFrame(const Frame::Ptr &frame) {
bool ret = false;
//有adts头，尝试分帧
int64_t dts = frame->dts();
int64_t pts = frame->pts();
auto ptr = frame->data();
auto end = frame->data() + frame->size();
while (ptr < end) {
......@@ -284,7 +287,7 @@ bool AACTrack::inputFrame(const Frame::Ptr &frame) {
if (frame_len == frame->size()) {
return inputFrame_l(frame);
}
auto sub_frame = std::make_shared<FrameInternal<FrameFromPtr> >(frame, (char *) ptr, frame_len, ADTS_HEADER_LEN);
auto sub_frame = std::make_shared<FrameTSInternal<FrameFromPtr> >(frame, (char *) ptr, frame_len, ADTS_HEADER_LEN,dts,pts);
ptr += frame_len;
if (ptr > end) {
WarnL << "invalid aac length in adts header: " << frame_len
......@@ -295,6 +298,8 @@ bool AACTrack::inputFrame(const Frame::Ptr &frame) {
if (inputFrame_l(sub_frame)) {
ret = true;
}
dts += 1024*1000/getAudioSampleRate();
pts += 1024*1000/getAudioSampleRate();
}
return ret;
}
......
src/Extension/CommonRtp.cpp
......@@ -76,7 +76,7 @@ bool CommonRtpEncoder::inputFrame(const Frame::Ptr &frame){
auto len = frame->size() - frame->prefixSize();
auto remain_size = len;
auto max_size = getMaxSize();
bool is_key = frame->keyFrame();
bool mark = false;
while (remain_size > 0) {
size_t rtp_size;
......@@ -86,9 +86,10 @@ bool CommonRtpEncoder::inputFrame(const Frame::Ptr &frame){
rtp_size = remain_size;
mark = true;
}
RtpCodec::inputRtp(makeRtp(getTrackType(), ptr, rtp_size, mark, stamp), false);
RtpCodec::inputRtp(makeRtp(getTrackType(), ptr, rtp_size, mark, stamp), is_key);
ptr += rtp_size;
remain_size -= rtp_size;
is_key = false;
}
return len > 0;
}
\ No newline at end of file
src/Extension/Factory.cpp
......@@ -95,6 +95,32 @@ Track::Ptr Factory::getTrackBySdp(const SdpTrack::Ptr &track) {
}
}
Track::Ptr Factory::getTrackByAbstractTrack(const Track::Ptr& track) {
auto codec = track->getCodecId();
switch (codec) {
case CodecG711A:
case CodecG711U: {
auto audio_track = dynamic_pointer_cast<AudioTrackImp>(track);
return std::make_shared<G711Track>(codec, audio_track->getAudioSampleRate(), audio_track->getAudioChannel(), 16);
}
case CodecL16: {
auto audio_track = dynamic_pointer_cast<AudioTrackImp>(track);
return std::make_shared<L16Track>(audio_track->getAudioSampleRate(), audio_track->getAudioChannel());
}
case CodecAAC : return std::make_shared<AACTrack>();
case CodecOpus : return std::make_shared<OpusTrack>();
case CodecH265 : return std::make_shared<H265Track>();
case CodecH264 : return std::make_shared<H264Track>();
default: {
//其他codec不支持
WarnL << "暂不支持该该编码类型创建Track:" << track->getCodecName();
return nullptr;
}
}
}
RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
GET_CONFIG(uint32_t,audio_mtu,Rtp::kAudioMtuSize);
GET_CONFIG(uint32_t,video_mtu,Rtp::kVideoMtuSize);
......
src/Extension/Factory.h
......@@ -28,6 +28,11 @@ public:
static Track::Ptr getTrackBySdp(const SdpTrack::Ptr &track);
/**
* 根据c api 抽象的Track生成具体Track对象
*/
static Track::Ptr getTrackByAbstractTrack(const Track::Ptr& track);
/**
* 根据sdp生成rtp编码器
* @param sdp sdp对象
*/
......
src/Extension/Frame.h
......@@ -15,6 +15,7 @@
#include <functional>
#include "Util/RingBuffer.h"
#include "Network/Socket.h"
#include "Common/Stamp.h"
namespace mediakit{
......@@ -263,6 +264,27 @@ private:
};
/**
* 一个Frame类中可以有多个帧(AAC)，时间戳会变化
* ZLMediaKit会先把这种复合帧split成单个帧然后再处理
* 一个复合帧可以通过无内存拷贝的方式切割成多个子Frame
* 提供该类的目的是切割复合帧时防止内存拷贝，提高性能
*/
template<typename Parent>
class FrameTSInternal : public Parent{
public:
typedef std::shared_ptr<FrameTSInternal> Ptr;
FrameTSInternal(const Frame::Ptr &parent_frame, char *ptr, size_t size, size_t prefix_size,uint32_t dts,uint32_t pts)
: Parent(ptr, size, dts, pts, prefix_size) {
_parent_frame = parent_frame;
}
bool cacheAble() const override {
return _parent_frame->cacheAble();
}
private:
Frame::Ptr _parent_frame;
};
/**
* 写帧接口的抽象接口类
*/
class FrameWriterInterface {
......@@ -369,17 +391,18 @@ class FrameFromPtr : public Frame{
public:
typedef std::shared_ptr<FrameFromPtr> Ptr;
FrameFromPtr(CodecId codec_id, char *ptr, size_t size, uint32_t dts, uint32_t pts = 0, size_t prefix_size = 0)
: FrameFromPtr(ptr, size, dts, pts, prefix_size) {
FrameFromPtr(CodecId codec_id, char *ptr, size_t size, uint32_t dts, uint32_t pts = 0, size_t prefix_size = 0,bool is_key = false )
: FrameFromPtr(ptr, size, dts, pts, prefix_size,is_key) {
_codec_id = codec_id;
}
FrameFromPtr(char *ptr, size_t size, uint32_t dts, uint32_t pts = 0, size_t prefix_size = 0){
FrameFromPtr(char *ptr, size_t size, uint32_t dts, uint32_t pts = 0, size_t prefix_size = 0,bool is_key = false){
_ptr = ptr;
_size = size;
_dts = dts;
_pts = pts;
_prefix_size = prefix_size;
_is_key = is_key;
}
char *data() const override{
......@@ -418,7 +441,7 @@ public:
}
bool keyFrame() const override {
return false;
return _is_key;
}
bool configFrame() const override{
......@@ -435,6 +458,7 @@ protected:
size_t _size;
size_t _prefix_size;
CodecId _codec_id = CodecInvalid;
bool _is_key;
};
/**
......@@ -498,6 +522,58 @@ private:
FrameImp::Ptr _buffer;
};
//该类实现frame级别的时间戳覆盖
class FrameStamp : public Frame{
public:
typedef std::shared_ptr<FrameStamp> Ptr;
FrameStamp(const Frame::Ptr &frame, Stamp &stamp,bool modify_stamp){
_frame = frame;
//覆盖时间戳
stamp.revise(frame->dts(), frame->pts(), _dts, _pts, modify_stamp);
}
~FrameStamp() override {}
uint32_t dts() const override{
return (uint32_t)_dts;
}
uint32_t pts() const override{
return (uint32_t)_pts;
}
size_t prefixSize() const override {
return _frame->prefixSize();
}
bool keyFrame() const override {
return _frame->keyFrame();
}
bool configFrame() const override {
return _frame->configFrame();
}
bool cacheAble() const override {
return _frame->cacheAble();
}
char *data() const override {
return _frame->data();
}
size_t size() const override {
return _frame->size();
}
CodecId getCodecId() const override {
return _frame->getCodecId();
}
private:
int64_t _dts;
int64_t _pts;
Frame::Ptr _frame;
};
/**
* 该对象可以把Buffer对象转换成可缓存的Frame对象
*/
......
src/Rtcp/RtcpContext.h
......@@ -11,9 +11,9 @@
#ifndef ZLMEDIAKIT_RTCPCONTEXT_H
#define ZLMEDIAKIT_RTCPCONTEXT_H
#include <stdint.h>
#include <stddef.h>
#include "Rtcp.h"
#include <stddef.h>
#include <stdint.h>
namespace mediakit {
......@@ -56,6 +56,13 @@ public:
virtual toolkit::Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc);
/**
* @brief 创建xr的dlrr包，用于接收者估算rtt
*
* @return toolkit::Buffer::Ptr
*/
virtual toolkit::Buffer::Ptr createRtcpXRDLRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc);
/**
* 创建RR rtcp包
* @param rtcp_ssrc rtcp的ssrc
* @param rtp_ssrc rtp的ssrc
......@@ -74,11 +81,11 @@ public:
virtual size_t geLostInterval();
protected:
//收到或发送的rtp的字节数
// 收到或发送的rtp的字节数
size_t _bytes = 0;
//收到或发送的rtp的个数
// 收到或发送的rtp的个数
size_t _packets = 0;
//上次的rtp时间戳,毫秒
// 上次的rtp时间戳,毫秒
uint32_t _last_rtp_stamp = 0;
uint64_t _last_ntp_stamp_ms = 0;
};
......@@ -86,8 +93,11 @@ protected:
class RtcpContextForSend : public RtcpContext {
public:
toolkit::Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc) override;
void onRtcp(RtcpHeader *rtcp) override;
toolkit::Buffer::Ptr createRtcpXRDLRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) override;
/**
* 获取rtt
* @param ssrc rtp ssrc
......@@ -96,8 +106,11 @@ public:
uint32_t getRtt(uint32_t ssrc) const;
private:
std::map<uint32_t/*ssrc*/, uint32_t/*rtt*/> _rtt;
std::map<uint32_t/*last_sr_lsr*/, uint64_t/*ntp stamp*/> _sender_report_ntp;
std::map<uint32_t /*ssrc*/, uint32_t /*rtt*/> _rtt;
std::map<uint32_t /*last_sr_lsr*/, uint64_t /*ntp stamp*/> _sender_report_ntp;
std::map<uint32_t /*ssrc*/, uint64_t /*xr rrtr sys stamp*/> _xr_rrtr_recv_sys_stamp;
std::map<uint32_t /*ssrc*/, uint32_t /*last rr */> _xr_xrrtr_recv_last_rr;
};
class RtcpContextForRecv : public RtcpContext {
......@@ -111,29 +124,29 @@ public:
void onRtcp(RtcpHeader *rtcp) override;
private:
//时间戳抖动值
// 时间戳抖动值
double _jitter = 0;
//第一个seq的值
// 第一个seq的值
uint16_t _seq_base = 0;
//rtp最大seq
// rtp最大seq
uint16_t _seq_max = 0;
//rtp回环次数
// rtp回环次数
uint16_t _seq_cycles = 0;
//上次回环发生时，记录的rtp包数
// 上次回环发生时，记录的rtp包数
size_t _last_cycle_packets = 0;
//上次的seq
// 上次的seq
uint16_t _last_rtp_seq = 0;
//上次的rtp的系统时间戳(毫秒)用于统计抖动
// 上次的rtp的系统时间戳(毫秒)用于统计抖动
uint64_t _last_rtp_sys_stamp = 0;
//上次统计的丢包总数
// 上次统计的丢包总数
size_t _last_lost = 0;
//上次统计应收rtp包总数
// 上次统计应收rtp包总数
size_t _last_expected = 0;
//上次收到sr包时计算出的Last SR timestamp
// 上次收到sr包时计算出的Last SR timestamp
uint32_t _last_sr_lsr = 0;
//上次收到sr时的系统时间戳,单位毫秒
// 上次收到sr时的系统时间戳,单位毫秒
uint64_t _last_sr_ntp_sys = 0;
};
}//namespace mediakit
#endif //ZLMEDIAKIT_RTCPCONTEXT_H
} // namespace mediakit
#endif // ZLMEDIAKIT_RTCPCONTEXT_H
src/Rtmp/RtmpPlayerImp.h
......@@ -99,7 +99,11 @@ private:
if (_rtmp_src) {
_rtmp_src->setMetaData(val);
}
if(_demuxer){
return;
}
_demuxer = std::make_shared<RtmpDemuxer>();
//TraceL<<" _wait_track_ready "<<_wait_track_ready;
_demuxer->setTrackListener(this, _wait_track_ready);
_demuxer->loadMetaData(val);
}
......
src/Rtmp/RtmpSession.cpp
......@@ -28,7 +28,7 @@ RtmpSession::~RtmpSession() {
}
void RtmpSession::onError(const SockException& err) {
bool is_player = !_push_src;
bool is_player = !_push_src_ownership;
uint64_t duration = _ticker.createdTime() / 1000;
WarnP(this) << (is_player ? "RTMP播放器(" : "RTMP推流器(")
<< _media_info._vhost << "/"
......@@ -219,10 +219,11 @@ void RtmpSession::onCmd_publish(AMFDecoder &dec) {
}
void RtmpSession::onCmd_deleteStream(AMFDecoder &dec) {
_push_src = nullptr;
//此时回复可能触发broken pipe事件，从而直接触发onError回调；所以需要先把_push_src置空，防止触发断流续推功能
sendStatus({ "level", "status",
"code", "NetStream.Unpublish.Success",
"description", "Stop publishing." });
//_push_src = nullptr;
throw std::runtime_error(StrPrinter << "Stop publishing" << endl);
}
......
src/Rtp/Decoder.cpp
......@@ -24,6 +24,15 @@
using namespace toolkit;
namespace mediakit {
void Decoder::setOnDecode(Decoder::onDecode cb) {
_on_decode = std::move(cb);
}
void Decoder::setOnStream(Decoder::onStream cb) {
_on_stream = std::move(cb);
}
static Decoder::Ptr createDecoder_l(DecoderImp::Type type) {
switch (type){
case DecoderImp::decoder_ps:
......
src/Rtp/Decoder.h
......@@ -25,12 +25,16 @@ public:
typedef std::function<void(int stream, int codecid, const void *extra, size_t bytes, int finish)> onStream;
virtual ssize_t input(const uint8_t *data, size_t bytes) = 0;
virtual void setOnDecode(onDecode cb) = 0;
virtual void setOnStream(onStream cb) = 0;
void setOnDecode(onDecode cb);
void setOnStream(onStream cb);
protected:
Decoder() = default;
virtual ~Decoder() = default;
protected:
onDecode _on_decode;
onStream _on_stream;
};
class DecoderImp{
......
src/Rtp/PSDecoder.cpp
......@@ -53,14 +53,6 @@ ssize_t PSDecoder::input(const uint8_t *data, size_t bytes) {
return bytes;
}
void PSDecoder::setOnDecode(Decoder::onDecode cb) {
_on_decode = std::move(cb);
}
void PSDecoder::setOnStream(Decoder::onStream cb) {
_on_stream = std::move(cb);
}
const char *PSDecoder::onSearchPacketTail(const char *data, size_t len) {
try {
auto ret = ps_demuxer_input(static_cast<struct ps_demuxer_t *>(_ps_demuxer), reinterpret_cast<const uint8_t *>(data), len);
......@@ -81,4 +73,4 @@ const char *PSDecoder::onSearchPacketTail(const char *data, size_t len) {
}
}//namespace mediakit
#endif//#if defined(ENABLE_RTPPROXY)
\ No newline at end of file
#endif//#if defined(ENABLE_RTPPROXY)
src/Rtp/PSDecoder.h
......@@ -25,8 +25,6 @@ public:
~PSDecoder();
ssize_t input(const uint8_t* data, size_t bytes) override;
void setOnDecode(onDecode cb) override;
void setOnStream(onStream cb) override;
// HttpRequestSplitter interface
private:
......@@ -36,8 +34,6 @@ private:
private:
void *_ps_demuxer = nullptr;
onDecode _on_decode;
onStream _on_stream;
};
}//namespace mediakit
......
src/Rtp/PSEncoder.cpp
......@@ -23,7 +23,7 @@ PSEncoderImp::PSEncoderImp(uint32_t ssrc, uint8_t payload_type) : MpegMuxer(true
_rtp_encoder = std::make_shared<CommonRtpEncoder>(CodecInvalid, ssrc, video_mtu, 90000, payload_type, 0);
_rtp_encoder->setRtpRing(std::make_shared<RtpRing::RingType>());
_rtp_encoder->getRtpRing()->setDelegate(std::make_shared<RingDelegateHelper>([this](RtpPacket::Ptr rtp, bool is_key){
onRTP(std::move(rtp));
onRTP(std::move(rtp),is_key);
}));
InfoL << this << " " << printSSRC(_rtp_encoder->getSsrc());
}
......@@ -36,7 +36,7 @@ void PSEncoderImp::onWrite(std::shared_ptr<Buffer> buffer, uint32_t stamp, bool
if (!buffer) {
return;
}
_rtp_encoder->inputFrame(std::make_shared<FrameFromPtr>(buffer->data(), buffer->size(), stamp, stamp));
_rtp_encoder->inputFrame(std::make_shared<FrameFromPtr>(buffer->data(), buffer->size(), stamp, stamp,0,key_pos));
}
}//namespace mediakit
......
src/Rtp/PSEncoder.h
......@@ -27,7 +27,7 @@ public:
protected:
//rtp打包后回调
virtual void onRTP(toolkit::Buffer::Ptr rtp) = 0;
virtual void onRTP(toolkit::Buffer::Ptr rtp,bool is_key = false) = 0;
protected:
void onWrite(std::shared_ptr<toolkit::Buffer> buffer, uint32_t stamp, bool key_pos) override;
......
src/Rtp/RawEncoder.cpp
......@@ -35,7 +35,7 @@ bool RawEncoderImp::addTrack(const Track::Ptr &track){
_rtp_encoder = createRtpEncoder(track);
_rtp_encoder->setRtpRing(std::make_shared<RtpRing::RingType>());
_rtp_encoder->getRtpRing()->setDelegate(std::make_shared<RingDelegateHelper>([this](RtpPacket::Ptr rtp, bool is_key){
onRTP(std::move(rtp));
onRTP(std::move(rtp),true);
}));
return true;
}
......@@ -44,7 +44,7 @@ bool RawEncoderImp::addTrack(const Track::Ptr &track){
_rtp_encoder = createRtpEncoder(track);
_rtp_encoder->setRtpRing(std::make_shared<RtpRing::RingType>());
_rtp_encoder->getRtpRing()->setDelegate(std::make_shared<RingDelegateHelper>([this](RtpPacket::Ptr rtp, bool is_key){
onRTP(std::move(rtp));
onRTP(std::move(rtp),is_key);
}));
return true;
}
......
src/Rtp/RawEncoder.h
......@@ -43,10 +43,9 @@ public:
protected:
//rtp打包后回调
virtual void onRTP(toolkit::Buffer::Ptr rtp) = 0;
virtual void onRTP(toolkit::Buffer::Ptr rtp, bool is_key = false) = 0;
private:
RtpCodec::Ptr createRtpEncoder(const Track::Ptr &track);
uint32_t _ssrc;
uint8_t _payload_type;
bool _sendAudio;
......
src/Rtp/RtpCache.cpp
......@@ -19,25 +19,37 @@ namespace mediakit{
RtpCache::RtpCache(onFlushed cb) {
_cb = std::move(cb);
}
bool RtpCache::firstKeyReady(bool in) {
if(_first_key){
return _first_key;
}
_first_key = in;
return _first_key;
}
void RtpCache::onFlush(std::shared_ptr<List<Buffer::Ptr> > rtp_list, bool) {
_cb(std::move(rtp_list));
}
void RtpCache::input(uint64_t stamp, Buffer::Ptr buffer) {
inputPacket(stamp, true, std::move(buffer), false);
void RtpCache::input(uint64_t stamp, Buffer::Ptr buffer,bool is_key ) {
inputPacket(stamp, true, std::move(buffer), is_key);
}
void RtpCachePS::onRTP(Buffer::Ptr buffer) {
void RtpCachePS::onRTP(Buffer::Ptr buffer,bool is_key) {
if(!firstKeyReady(is_key)){
return;
}
auto rtp = std::static_pointer_cast<RtpPacket>(buffer);
auto stamp = rtp->getStampMS();
input(stamp, std::move(buffer));
input(stamp, std::move(buffer),is_key);
}
void RtpCacheRaw::onRTP(Buffer::Ptr buffer) {
void RtpCacheRaw::onRTP(Buffer::Ptr buffer,bool is_key) {
if(!firstKeyReady(is_key)){
return;
}
auto rtp = std::static_pointer_cast<RtpPacket>(buffer);
auto stamp = rtp->getStampMS();
input(stamp, std::move(buffer));
input(stamp, std::move(buffer),is_key);
}
}//namespace mediakit
......
src/Rtp/RtpCache.h
......@@ -30,13 +30,15 @@ protected:
* 输入rtp(目的是为了合并写)
* @param buffer rtp数据
*/
void input(uint64_t stamp, toolkit::Buffer::Ptr buffer);
void input(uint64_t stamp, toolkit::Buffer::Ptr buffer,bool is_key = false);
bool firstKeyReady(bool in);
protected:
void onFlush(std::shared_ptr<toolkit::List<toolkit::Buffer::Ptr> > rtp_list, bool) override;
private:
onFlushed _cb;
bool _first_key = false;
};
class RtpCachePS : public RtpCache, public PSEncoderImp{
......@@ -45,7 +47,7 @@ public:
~RtpCachePS() override = default;
protected:
void onRTP(toolkit::Buffer::Ptr rtp) override;
void onRTP(toolkit::Buffer::Ptr rtp,bool is_key = false) override;
};
......@@ -55,7 +57,7 @@ public:
~RtpCacheRaw() override = default;
protected:
void onRTP(toolkit::Buffer::Ptr rtp) override;
void onRTP(toolkit::Buffer::Ptr rtp,bool is_key = false) override;
};
}//namespace mediakit
......
src/Rtp/TSDecoder.cpp
......@@ -101,14 +101,6 @@ ssize_t TSDecoder::input(const uint8_t *data, size_t bytes) {
return bytes;
}
void TSDecoder::setOnDecode(Decoder::onDecode cb) {
_on_decode = std::move(cb);
}
void TSDecoder::setOnStream(Decoder::onStream cb) {
_on_stream = std::move(cb);
}
#endif//defined(ENABLE_HLS)
}//namespace mediakit
src/Rtp/TSDecoder.h
......@@ -45,14 +45,10 @@ public:
TSDecoder();
~TSDecoder();
ssize_t input(const uint8_t* data, size_t bytes) override ;
void setOnDecode(onDecode cb) override;
void setOnStream(onStream cb) override;
private:
TSSegment _ts_segment;
struct ts_demuxer_t* _demuxer_ctx = nullptr;
onDecode _on_decode;
onStream _on_stream;
};
#endif//defined(ENABLE_HLS)
......
src/Rtsp/RtspSession.cpp
......@@ -60,7 +60,7 @@ RtspSession::~RtspSession() {
}
void RtspSession::onError(const SockException &err) {
bool is_player = !_push_src;
bool is_player = !_push_src_ownership;
uint64_t duration = _alive_ticker.createdTime() / 1000;
WarnP(this) << (is_player ? "RTSP播放器(" : "RTSP推流器(")
<< _media_info._vhost << "/"
......@@ -867,8 +867,9 @@ void RtspSession::handleReq_Pause(const Parser &parser) {
}
void RtspSession::handleReq_Teardown(const Parser &parser) {
sendRtspResponse("200 OK");
_push_src = nullptr;
//此时回复可能触发broken pipe事件，从而直接触发onError回调；所以需要先把_push_src置空，防止触发断流续推功能
sendRtspResponse("200 OK");
throw SockException(Err_shutdown,"recv teardown request");
}
......
srt/Packet.cpp
#include "Util/MD5.h"
#include <atomic>
#include "Util/MD5.h"
#include "Util/logger.h"
#include <atomic>
#include "Packet.hpp"
......@@ -225,7 +225,35 @@ size_t ControlPacket::size() const {
uint32_t ControlPacket::getSocketID(uint8_t *buf, size_t len) {
return loadUint32(buf + 12);
}
std::string HandshakePacket::dump(){
_StrPrinter printer;
printer <<"flag:"<< (int)f<<"\r\n";
printer <<"control_type:"<< (int)control_type<<"\r\n";
printer <<"sub_type:"<< (int)sub_type<<"\r\n";
printer <<"type_specific_info:"<< (int)type_specific_info[0]<<":"<<(int)type_specific_info[1]<<":"<<(int)type_specific_info[2]<<":"<<(int)type_specific_info[3]<<"\r\n";
printer <<"timestamp:"<< timestamp<<"\r\n";
printer <<"dst_socket_id:"<< dst_socket_id<<"\r\n";
printer <<"version:"<< version<<"\r\n";
printer <<"encryption_field:"<< encryption_field<<"\r\n";
printer <<"extension_field:"<< extension_field<<"\r\n";
printer <<"initial_packet_sequence_number:"<< initial_packet_sequence_number<<"\r\n";
printer <<"mtu:"<< mtu<<"\r\n";
printer <<"max_flow_window_size:"<< max_flow_window_size<<"\r\n";
printer <<"handshake_type:"<< handshake_type<<"\r\n";
printer <<"srt_socket_id:"<< srt_socket_id<<"\r\n";
printer <<"syn_cookie:"<< syn_cookie<<"\r\n";
printer <<"peer_ip_addr:";
for(size_t i=0;i<sizeof(peer_ip_addr);++i){
printer<<(int)peer_ip_addr[i]<<":";
}
printer<<"\r\n";
for(size_t i=0;i<ext_list.size();++i){
printer<<ext_list[i]->dump()<<"\r\n";
}
return std::move(printer);
}
bool HandshakePacket::loadFromData(uint8_t *buf, size_t len) {
if (HEADER_SIZE + HS_CONTENT_MIN_SIZE > len) {
ErrorL << "size too smalle " << encryption_field;
......@@ -435,15 +463,11 @@ uint32_t HandshakePacket::generateSynCookie(
while (true) {
// SYN cookie
char clienthost[NI_MAXHOST];
char clientport[NI_MAXSERV];
getnameinfo(
(struct sockaddr *)addr, sizeof(struct sockaddr_storage), clienthost, sizeof(clienthost), clientport,
sizeof(clientport), NI_NUMERICHOST | NI_NUMERICSERV);
int64_t timestamp = (DurationCountMicroseconds(SteadyClock::now() - ts) / 60000000) + distractor.load()
+ correction; // secret changes every one minute
std::stringstream cookiestr;
cookiestr << clienthost << ":" << clientport << ":" << timestamp;
cookiestr << SockUtil::inet_ntoa((struct sockaddr *)addr) << ":" << SockUtil::inet_port((struct sockaddr *)addr)
<< ":" << timestamp;
union {
unsigned char cookie[16];
uint32_t cookie_val;
......
srt/Packet.hpp
......@@ -118,9 +118,9 @@ public:
USERDEFINEDTYPE = 0x7FFF
};
uint32_t sub_type : 16;
uint32_t control_type : 15;
uint32_t f : 1;
uint16_t sub_type;
uint16_t control_type;
uint8_t f;
uint8_t type_specific_info[4];
uint32_t timestamp;
uint32_t dst_socket_id;
......@@ -189,7 +189,7 @@ public:
static uint32_t getSynCookie(uint8_t *buf, size_t len);
static uint32_t
generateSynCookie(struct sockaddr_storage *addr, TimePoint ts, uint32_t current_cookie = 0, int correction = 0);
std::string dump();
void assignPeerIP(struct sockaddr_storage *addr);
///////ControlPacket override///////
bool loadFromData(uint8_t *buf, size_t len) override;
......
srt/SrtTransport.cpp
......@@ -86,6 +86,7 @@ void SrtTransport::inputSockData(uint8_t *buf, int len, struct sockaddr_storage
handleDataPacket(buf, len, addr);
} else {
WarnL<<"DataPacket switch to other transport: "<<socketId;
switchToOtherTransport(buf, len, socketId, addr);
}
} else {
......@@ -94,6 +95,7 @@ void SrtTransport::inputSockData(uint8_t *buf, int len, struct sockaddr_storage
uint16_t type = ControlPacket::getControlType(buf, len);
if (type != ControlPacket::HANDSHAKE && socketId != _socket_id && _socket_id != 0) {
// socket id not same
WarnL<<"ControlPacket: "<< (int)type <<" switch to other transport: "<<socketId;
switchToOtherTransport(buf, len, socketId, addr);
return;
}
......@@ -168,7 +170,7 @@ void SrtTransport::handleHandshakeConclusion(HandshakePacket &pkt, struct sockad
if (delay <= 120) {
delay = 120;
}
for (auto ext : pkt.ext_list) {
for (auto& ext : pkt.ext_list) {
// TraceL << getIdentifier() << " ext " << ext->dump();
if (!req) {
req = std::dynamic_pointer_cast<HSExtMessage>(ext);
......@@ -228,7 +230,10 @@ void SrtTransport::handleHandshakeConclusion(HandshakePacket &pkt, struct sockad
void SrtTransport::handleHandshake(uint8_t *buf, int len, struct sockaddr_storage *addr) {
HandshakePacket pkt;
assert(pkt.loadFromData(buf, len));
if(!pkt.loadFromData(buf, len)){
WarnL<<"is not vaild HandshakePacket";
return;
}
if (pkt.handshake_type == HandshakePacket::HS_TYPE_INDUCTION) {
handleHandshakeInduction(pkt, addr);
......@@ -236,6 +241,7 @@ void SrtTransport::handleHandshake(uint8_t *buf, int len, struct sockaddr_storag
handleHandshakeConclusion(pkt, addr);
} else {
WarnL << " not support handshake type = " << pkt.handshake_type;
WarnL <<pkt.dump();
}
_ack_ticker.resetTime(_now);
_nak_ticker.resetTime(_now);
......@@ -288,13 +294,13 @@ void SrtTransport::handleNAK(uint8_t *buf, int len, struct sockaddr_storage *add
bool empty = false;
bool flush = false;
for (auto it : pkt.lost_list) {
for (auto& it : pkt.lost_list) {
if (pkt.lost_list.back() == it) {
flush = true;
}
empty = true;
auto re_list = _send_buf->findPacketBySeq(it.first, it.second - 1);
for (auto pkt : re_list) {
for (auto& pkt : re_list) {
pkt->R = 1;
pkt->storeToHeader();
sendPacket(pkt, flush);
......@@ -325,7 +331,7 @@ void SrtTransport::handleDropReq(uint8_t *buf, int len, struct sockaddr_storage
return;
}
uint32_t max_seq = 0;
for (auto data : list) {
for (auto& data : list) {
max_seq = data->packet_seq_number;
if (_last_pkt_seq + 1 != data->packet_seq_number) {
TraceL << "pkt lost " << _last_pkt_seq + 1 << "->" << data->packet_seq_number;
......@@ -495,7 +501,7 @@ void SrtTransport::handleDataPacket(uint8_t *buf, int len, struct sockaddr_stora
// when no data ok send nack to sender immediately
} else {
uint32_t last_seq;
for (auto data : list) {
for (auto& data : list) {
last_seq = data->packet_seq_number;
if (_last_pkt_seq + 1 != data->packet_seq_number) {
TraceL << "pkt lost " << _last_pkt_seq + 1 << "->" << data->packet_seq_number;
......
srt/SrtTransportImp.cpp
......@@ -17,7 +17,7 @@ SrtTransportImp::~SrtTransportImp() {
GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
if (_total_bytes >= iFlowThreshold * 1024) {
NoticeCenter::Instance().emitEvent(
Broadcast::kBroadcastFlowReport, _media_info, _total_bytes, duration, false,
Broadcast::kBroadcastFlowReport, _media_info, _total_bytes, duration, !_is_pusher,
static_cast<SockInfo &>(*this));
}
}
......@@ -149,6 +149,11 @@ std::shared_ptr<SockInfo> SrtTransportImp::getOriginSock(mediakit::MediaSource &
return static_pointer_cast<SockInfo>(getSession());
}
toolkit::EventPoller::Ptr SrtTransportImp::getOwnerPoller(MediaSource &sender){
auto session = getSession();
return session ? session->getPoller() : nullptr;
}
void SrtTransportImp::emitOnPublish() {
std::weak_ptr<SrtTransportImp> weak_self = static_pointer_cast<SrtTransportImp>(shared_from_this());
Broadcast::PublishAuthInvoker invoker = [weak_self](const std::string &err, const ProtocolOption &option) {
......@@ -282,7 +287,10 @@ std::string SrtTransportImp::getIdentifier() const {
bool SrtTransportImp::inputFrame(const Frame::Ptr &frame) {
if (_muxer) {
return _muxer->inputFrame(frame);
//TraceL<<"before type "<<frame->getCodecName()<<" dts "<<frame->dts()<<" pts "<<frame->pts();
auto frame_tmp = std::make_shared<FrameStamp>(frame, _type_to_stamp[frame->getTrackType()],false);
//TraceL<<"after type "<<frame_tmp->getCodecName()<<" dts "<<frame_tmp->dts()<<" pts "<<frame_tmp->pts();
return _muxer->inputFrame(frame_tmp);
}
if (_cached_func.size() > 200) {
WarnL << "cached frame of track(" << frame->getCodecName() << ") is too much, now dropped";
......@@ -290,11 +298,17 @@ bool SrtTransportImp::inputFrame(const Frame::Ptr &frame) {
}
auto frame_cached = Frame::getCacheAbleFrame(frame);
lock_guard<recursive_mutex> lck(_func_mtx);
_cached_func.emplace_back([this, frame_cached]() { _muxer->inputFrame(frame_cached); });
_cached_func.emplace_back([this, frame_cached]() {
//TraceL<<"before type "<<frame_cached->getCodecName()<<" dts "<<frame_cached->dts()<<" pts "<<frame_cached->pts();
auto frame_tmp = std::make_shared<FrameStamp>(frame_cached, _type_to_stamp[frame_cached->getTrackType()],false);
//TraceL<<"after type "<<frame_tmp->getCodecName()<<" dts "<<frame_tmp->dts()<<" pts "<<frame_tmp->pts();
_muxer->inputFrame(frame_tmp);
});
return true;
}
bool SrtTransportImp::addTrack(const Track::Ptr &track) {
_type_to_stamp.emplace(track->getTrackType(),Stamp());
if (_muxer) {
return _muxer->addTrack(track);
}
......@@ -311,6 +325,9 @@ void SrtTransportImp::addTrackCompleted() {
lock_guard<recursive_mutex> lck(_func_mtx);
_cached_func.emplace_back([this]() { _muxer->addTrackCompleted(); });
}
if(_type_to_stamp.size() >1){
_type_to_stamp[TrackType::TrackAudio].syncTo(_type_to_stamp[TrackType::TrackVideo]);
}
}
void SrtTransportImp::doCachedFunc() {
......
srt/SrtTransportImp.hpp
......@@ -59,6 +59,8 @@ protected:
std::string getOriginUrl(mediakit::MediaSource &sender) const override;
// 获取媒体源客户端相关信息
std::shared_ptr<SockInfo> getOriginSock(mediakit::MediaSource &sender) const override;
// get poller
toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) override;
///////MediaSinkInterface override///////
void resetTracks() override {};
......@@ -87,6 +89,8 @@ private:
DecoderImp::Ptr _decoder;
std::recursive_mutex _func_mtx;
std::deque<std::function<void()>> _cached_func;
std::unordered_map<int, Stamp> _type_to_stamp;
};
} // namespace SRT
......
webrtc/Sdp.cpp
......@@ -771,7 +771,6 @@ void RtcSession::loadFrom(const string &str) {
session_name = sdp.getSessionName();
session_info = sdp.getSessionInfo();
connection = sdp.getConnection();
bandwidth = sdp.getBandwidth();
time = sdp.getSessionTime();
msid_semantic = sdp.getItemClass<SdpAttrMsidSemantic>('a', "msid-semantic");
for (auto &media : sdp.medias) {
......@@ -783,6 +782,7 @@ void RtcSession::loadFrom(const string &str) {
rtc_media.type = mline.type;
rtc_media.port = mline.port;
rtc_media.addr = media.getItemClass<SdpConnection>('c');
rtc_media.bandwidth = media.getItemClass<SdpBandwidth>('b');
rtc_media.ice_ufrag = media.getStringItem('a', "ice-ufrag");
rtc_media.ice_pwd = media.getStringItem('a', "ice-pwd");
rtc_media.role = media.getItemClass<SdpAttrSetup>('a', "setup").role;
......@@ -1060,9 +1060,6 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{
if(connection.empty()){
sdp.addItem(std::make_shared<SdpConnection>(connection));
}
if (!bandwidth.empty()) {
sdp.addItem(std::make_shared<SdpBandwidth>(bandwidth));
}
sdp.addAttr(std::make_shared<SdpAttrGroup>(group));
sdp.addAttr(std::make_shared<SdpAttrMsidSemantic>(msid_semantic));
for (auto &m : media) {
......@@ -1080,6 +1077,9 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{
}
sdp_media.addItem(std::move(mline));
sdp_media.addItem(std::make_shared<SdpConnection>(m.addr));
if (!m.bandwidth.empty() && m.type != TrackAudio) {
sdp_media.addItem(std::make_shared<SdpBandwidth>(m.bandwidth));
}
if (!m.rtcp_addr.empty()) {
sdp_media.addAttr(std::make_shared<SdpAttrRtcp>(m.rtcp_addr));
}
......@@ -1631,6 +1631,7 @@ RETRY:
answer_media.proto = offer_media.proto;
answer_media.port = offer_media.port;
answer_media.addr = offer_media.addr;
answer_media.bandwidth = offer_media.bandwidth;
answer_media.rtcp_addr = offer_media.rtcp_addr;
answer_media.rtcp_mux = offer_media.rtcp_mux && configure.rtcp_mux;
answer_media.rtcp_rsize = offer_media.rtcp_rsize && configure.rtcp_rsize;
......
webrtc/Sdp.h
......@@ -612,6 +612,7 @@ public:
std::string mid;
uint16_t port{0};
SdpConnection addr;
SdpBandwidth bandwidth;
std::string proto;
RtpDirection direction{RtpDirection::invalid};
std::vector<RtcCodecPlan> plan;
......@@ -666,7 +667,6 @@ public:
std::string session_info;
SdpTime time;
SdpConnection connection;
SdpBandwidth bandwidth;
SdpAttrMsidSemantic msid_semantic;
std::vector<RtcMedia> media;
SdpAttrGroup group;
......
www/ZLMRTCClient.js.map
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www/webrtc/ZLMRTCClient.js.map
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