amass/ZLMediaKit
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Merge branch 'master' of https://github.com/ZLMediaKit/ZLMediaKit

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This commit is contained in:
rayjay 2024-03-28 11:15:19 +08:00
commit a33c1d5a08
72 changed files with 2174 additions and 1172 deletions
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2
3rdpart/ZLToolKit

@ -1 +1 @@
Subproject commit fca6d2328871fc6af75e215f89c3f1092ba5bb21
Subproject commit 04d1c47d2568f5ce1ff84260cefaf2754e514a5e

2
3rdpart/media-server

@ -1 +1 @@
Subproject commit 043853ee7c004e3e5d5bf3d06f7a82d97155b0d1
Subproject commit 527c0f5117b489fda78fcd123d446370ddd9ec9a

6
AUTHORS
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@ -44,7 +44,6 @@ Xinghua Zhao <(holychaossword@hotmail.com>
[Dw9](https://github.com/Dw9)
明月惊鹊 <mingyuejingque@gmail.com>
cgm <2958580318@qq.com>
hejilin <1724010622@qq.com>
alexliyu7352 <liyu7352@gmail.com>
cgm <2958580318@qq.com>
[haorui wang](https://github.com/HaoruiWang)
@ -104,3 +103,8 @@ WuPeng <wp@zafu.edu.cn>
[sandro-qiang](https://github.com/sandro-qiang)
[Paul Philippov](https://github.com/themactep)
[张传峰](https://github.com/zhang-chuanfeng)
[lidaofu-hub](https://github.com/lidaofu-hub)
[huangcaichun](https://github.com/huangcaichun)
[jamesZHANG500](https://github.com/jamesZHANG500)
[weidelong](https://github.com/wdl1697454803)
[小强先生](https://github.com/linshangqiang)

10
CMakeLists.txt
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@ -141,8 +141,8 @@ if(GIT_FOUND)
endif()
configure_file(
${CMAKE_CURRENT_SOURCE_DIR}/version.h.ini
${CMAKE_CURRENT_BINARY_DIR}/version.h
${CMAKE_CURRENT_SOURCE_DIR}/ZLMVersion.h.ini
${CMAKE_CURRENT_BINARY_DIR}/ZLMVersion.h
@ONLY)
message(STATUS "Git version is ${BRANCH_NAME} ${COMMIT_HASH}/${COMMIT_TIME} ${BUILD_TIME}")
@ -532,3 +532,9 @@ endif ()
file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/www" DESTINATION ${EXECUTABLE_OUTPUT_PATH})
file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/conf/config.ini" DESTINATION ${EXECUTABLE_OUTPUT_PATH})
file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/default.pem" DESTINATION ${EXECUTABLE_OUTPUT_PATH})
# 拷贝VideoStack 无视频流时默认填充的背景图片
# Copy the default background image used by VideoStack when there is no video stream
if (ENABLE_FFMPEG AND ENABLE_X264)
file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/conf/novideo.yuv" DESTINATION ${EXECUTABLE_OUTPUT_PATH})
endif ()

5
README.md
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@ -358,6 +358,11 @@ bash build_docker_images.sh
[sandro-qiang](https://github.com/sandro-qiang)
[Paul Philippov](https://github.com/themactep)
[张传峰](https://github.com/zhang-chuanfeng)
[lidaofu-hub](https://github.com/lidaofu-hub)
[huangcaichun](https://github.com/huangcaichun)
[jamesZHANG500](https://github.com/jamesZHANG500)
[weidelong](https://github.com/wdl1697454803)
[小强先生](https://github.com/linshangqiang)
同时感谢JetBrains对开源项目的支持，本项目使用CLion开发与调试：

5
README_en.md
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@ -516,6 +516,11 @@ Thanks to all those who have supported this project in various ways, including b
[sandro-qiang](https://github.com/sandro-qiang)
[Paul Philippov](https://github.com/themactep)
[张传峰](https://github.com/zhang-chuanfeng)
[lidaofu-hub](https://github.com/lidaofu-hub)
[huangcaichun](https://github.com/huangcaichun)
[jamesZHANG500](https://github.com/jamesZHANG500)
[weidelong](https://github.com/wdl1697454803)
[小强先生](https://github.com/linshangqiang)
Also thank to JetBrains for their support for open source project, we developed and debugged zlmediakit with CLion:

0
version.h.ini → ZLMVersion.h.ini
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27
api/include/mk_events.h
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@ -177,6 +177,33 @@ typedef struct {
*/
void(API_CALL *on_mk_media_send_rtp_stop)(const char *vhost, const char *app, const char *stream, const char *ssrc, int err, const char *msg);
/**
* rtc sctp连接中/完成/失败/关闭回调
* @param rtc_transport 数据通道对象
*/
void(API_CALL *on_mk_rtc_sctp_connecting)(mk_rtc_transport rtc_transport);
void(API_CALL *on_mk_rtc_sctp_connected)(mk_rtc_transport rtc_transport);
void(API_CALL *on_mk_rtc_sctp_failed)(mk_rtc_transport rtc_transport);
void(API_CALL *on_mk_rtc_sctp_closed)(mk_rtc_transport rtc_transport);
/**
* rtc数据通道发送数据回调
* @param rtc_transport 数据通道对象
* @param msg 数据
* @param len 数据长度
*/
void(API_CALL *on_mk_rtc_sctp_send)(mk_rtc_transport rtc_transport, const uint8_t *msg, size_t len);
/**
* rtc数据通道接收数据回调
* @param rtc_transport 数据通道对象
* @param streamId 流id
* @param ppid 协议id
* @param msg 数据
* @param len 数据长度
*/
void(API_CALL *on_mk_rtc_sctp_received)(mk_rtc_transport rtc_transport, uint16_t streamId, uint32_t ppid, const uint8_t *msg, size_t len);
} mk_events;

14
api/include/mk_events_objects.h
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@ -352,6 +352,20 @@ API_EXPORT mk_auth_invoker API_CALL mk_auth_invoker_clone(const mk_auth_invoker
*/
API_EXPORT void API_CALL mk_auth_invoker_clone_release(const mk_auth_invoker ctx);
///////////////////////////////////////////WebRtcTransport/////////////////////////////////////////////
//WebRtcTransport对象的C映射
typedef struct mk_rtc_transport_t *mk_rtc_transport;
/**
* 发送rtc数据通道
* @param ctx 数据通道对象
* @param streamId 流id
* @param ppid 协议id
* @param msg 数据
* @param len 数据长度
*/
API_EXPORT void API_CALL mk_rtc_send_datachannel(const mk_rtc_transport ctx, uint16_t streamId, uint32_t ppid, const char* msg, size_t len);
#ifdef __cplusplus
}
#endif

9
api/source/mk_common.cpp
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@ -33,12 +33,11 @@ static TcpServer::Ptr shell_server;
#ifdef ENABLE_RTPPROXY
#include "Rtp/RtpServer.h"
static std::shared_ptr<RtpServer> rtpServer;
static RtpServer::Ptr rtpServer;
#endif
#ifdef ENABLE_WEBRTC
#include "../webrtc/WebRtcSession.h"
#include "../webrtc/WebRtcTransport.h"
static UdpServer::Ptr rtcServer_udp;
static TcpServer::Ptr rtcServer_tcp;
#endif
@ -305,10 +304,10 @@ API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(void *user_data, on_user_data
std::string offer_str = offer;
std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
auto args = std::make_shared<WebRtcArgsUrl>(url);
WebRtcPluginManager::Instance().getAnswerSdp(*session, type, *args,
[offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
WebRtcPluginManager::Instance().negotiateSdp(*session, type, *args, [offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
auto &handler = const_cast<WebRtcInterface &>(exchanger);
try {
auto sdp_answer = exchangeSdp(exchanger, offer_str);
auto sdp_answer = handler.getAnswerSdp(offer_str);
cb(ptr.get(), sdp_answer.data(), nullptr);
} catch (std::exception &ex) {
cb(ptr.get(), nullptr, ex.what());

41
api/source/mk_events.cpp
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@ -15,6 +15,10 @@
#include "Rtsp/RtspSession.h"
#include "Record/MP4Recorder.h"
#ifdef ENABLE_WEBRTC
#include "webrtc/WebRtcTransport.h"
#endif
using namespace toolkit;
using namespace mediakit;
@ -167,6 +171,43 @@ API_EXPORT void API_CALL mk_events_listen(const mk_events *events){
sender.getMediaTuple().stream.c_str(), ssrc.c_str(), ex.getErrCode(), ex.what());
}
});
#ifdef ENABLE_WEBRTC
NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpConnecting,[](BroadcastRtcSctpConnectArgs){
if (s_events.on_mk_rtc_sctp_connecting) {
s_events.on_mk_rtc_sctp_connecting((mk_rtc_transport)&sender);
}
});
NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpConnected,[](BroadcastRtcSctpConnectArgs){
if (s_events.on_mk_rtc_sctp_connected) {
s_events.on_mk_rtc_sctp_connected((mk_rtc_transport)&sender);
}
});
NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpFailed,[](BroadcastRtcSctpConnectArgs){
if (s_events.on_mk_rtc_sctp_failed) {
s_events.on_mk_rtc_sctp_failed((mk_rtc_transport)&sender);
}
});
NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpClosed,[](BroadcastRtcSctpConnectArgs){
if (s_events.on_mk_rtc_sctp_closed) {
s_events.on_mk_rtc_sctp_closed((mk_rtc_transport)&sender);
}
});
NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpSend,[](BroadcastRtcSctpSendArgs){
if (s_events.on_mk_rtc_sctp_send) {
s_events.on_mk_rtc_sctp_send((mk_rtc_transport)&sender, data, len);
}
});
NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpReceived,[](BroadcastRtcSctpReceivedArgs){
if (s_events.on_mk_rtc_sctp_received) {
s_events.on_mk_rtc_sctp_received((mk_rtc_transport)&sender, streamId, ppid, msg, len);
}
});
#endif
});
}

22
api/source/mk_events_objects.cpp
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@ -18,6 +18,10 @@
#include "Http/HttpClient.h"
#include "Rtsp/RtspSession.h"
#ifdef ENABLE_WEBRTC
#include "webrtc/WebRtcTransport.h"
#endif
using namespace toolkit;
using namespace mediakit;
@ -498,3 +502,21 @@ API_EXPORT void API_CALL mk_auth_invoker_clone_release(const mk_auth_invoker ctx
Broadcast::AuthInvoker *invoker = (Broadcast::AuthInvoker *)ctx;
delete invoker;
}
///////////////////////////////////////////WebRtcTransport/////////////////////////////////////////////
API_EXPORT void API_CALL mk_rtc_sendDatachannel(const mk_rtc_transport ctx, uint16_t streamId, uint32_t ppid, const char *msg, size_t len) {
#ifdef ENABLE_WEBRTC
assert(ctx && msg);
WebRtcTransport *transport = (WebRtcTransport *)ctx;
std::string msg_str(msg, len);
std::weak_ptr<WebRtcTransport> weak_trans = transport->shared_from_this();
transport->getPoller()->async([streamId, ppid, msg_str, weak_trans]() {
// 切换线程后再操作
if (auto trans = weak_trans.lock()) {
trans->sendDatachannel(streamId, ppid, msg_str.c_str(), msg_str.size());
}
});
#else
WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
#endif
}

11
conf/config.ini
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@ -277,6 +277,8 @@ sampleMS=500
fastStart=0
#MP4点播(rtsp/rtmp/http-flv/ws-flv)是否循环播放文件
fileRepeat=0
#MP4录制写文件格式是否采用fmp4，启用的话，断电未完成录制的文件也能正常打开
enableFmp4=0
[rtmp]
#rtmp必须在此时间内完成握手，否则服务器会断开链接，单位秒
@ -329,6 +331,13 @@ opus_pt=100
#如果不调用startSendRtp相关接口，可以置0节省内存
gop_cache=1
#国标发送g711 rtp 打包时，每个包的语音时长是多少，默认是100 ms，范围为20~180ms (gb28181-2016，c.2.4规定)，
#最好为20 的倍数，程序自动向20的倍数取整
rtp_g711_dur_ms = 100
#udp接收数据socket buffer大小配置
#4*1024*1024=4196304
udp_recv_socket_buffer=4194304
[rtc]
#rtc播放推流、播放超时时间
timeoutSec=15
@ -348,7 +357,7 @@ tcpPort = 8000
rembBitRate=0
#rtc支持的音频codec类型,在前面的优先级更高
#以下范例为所有支持的音频codec
preferredCodecA=PCMU,PCMA,opus,mpeg4-generic
preferredCodecA=PCMA,PCMU,opus,mpeg4-generic
#rtc支持的视频codec类型,在前面的优先级更高
#以下范例为所有支持的视频codec
preferredCodecV=H264,H265,AV1,VP9,VP8

1
conf/novideo.yuv Normal file
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File diff suppressed because one or more lines are too long

2
ext-codec/AACRtmp.cpp
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@ -17,7 +17,7 @@ using namespace toolkit;
namespace mediakit {
void AACRtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {
CHECK(pkt->size() > 2);
CHECK_RET(pkt->size() > 2);
if (pkt->isConfigFrame()) {
getTrack()->setExtraData((uint8_t *)pkt->data() + 2, pkt->size() - 2);
return;

36
ext-codec/G711Rtp.cpp
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@ -8,13 +8,27 @@ G711RtpEncoder::G711RtpEncoder(CodecId codec, uint32_t channels){
_channels = channels;
}
void G711RtpEncoder::setOpt(int opt, const toolkit::Any &param) {
if (opt == RTP_ENCODER_PKT_DUR_MS) {
if (param.is<uint32_t>()) {
auto dur = param.get<uint32_t>();
if (dur < 20 || dur > 180) {
WarnL << "set g711 rtp encoder duration ms failed for " << dur;
return;
}
// 向上 20ms 取整
_pkt_dur_ms = (dur + 19) / 20 * 20;
}
}
}
bool G711RtpEncoder::inputFrame(const Frame::Ptr &frame) {
auto dur = (_cache_frame->size() - _cache_frame->prefixSize()) / (8 * _channels);
auto next_pts = _cache_frame->pts() + dur;
if (next_pts == 0) {
_cache_frame->_pts = frame->pts();
} else {
if ((next_pts + 20) < frame->pts()) { // 有丢包超过20ms
if ((next_pts + _pkt_dur_ms) < frame->pts()) { // 有丢包超过20ms
_cache_frame->_pts = frame->pts() - dur;
}
}
@ -24,24 +38,20 @@ bool G711RtpEncoder::inputFrame(const Frame::Ptr &frame) {
auto ptr = _cache_frame->data() + _cache_frame->prefixSize();
auto len = _cache_frame->size() - _cache_frame->prefixSize();
auto remain_size = len;
auto max_size = 160 * _channels; // 20 ms per rtp
int n = 0;
bool mark = false;
size_t max_size = 160 * _channels * _pkt_dur_ms / 20; // 20 ms per 160 byte
size_t n = 0;
bool mark = true;
while (remain_size >= max_size) {
size_t rtp_size;
if (remain_size >= max_size) {
rtp_size = max_size;
} else {
break;
}
assert(remain_size >= max_size);
const size_t rtp_size = max_size;
n++;
stamp += 20;
RtpCodec::inputRtp(getRtpInfo().makeRtp(TrackAudio, ptr, rtp_size, mark, stamp), false);
stamp += _pkt_dur_ms;
RtpCodec::inputRtp(getRtpInfo().makeRtp(TrackAudio, ptr, rtp_size, mark, stamp), true);
ptr += rtp_size;
remain_size -= rtp_size;
}
_cache_frame->_buffer.erase(0, n * max_size);
_cache_frame->_pts += 20 * n;
_cache_frame->_pts += (uint64_t)_pkt_dur_ms * n;
return len > 0;
}

3
ext-codec/G711Rtp.h
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@ -36,8 +36,11 @@ public:
*/
bool inputFrame(const Frame::Ptr &frame) override;
void setOpt(int opt, const toolkit::Any &param) override;
private:
uint32_t _channels = 1;
uint32_t _pkt_dur_ms = 20;
FrameImp::Ptr _cache_frame;
};

8
ext-codec/H264Rtmp.cpp
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@ -14,14 +14,6 @@
using namespace std;
using namespace toolkit;
#define CHECK_RET(...) \
try { \
CHECK(__VA_ARGS__); \
} catch (AssertFailedException & ex) { \
WarnL << ex.what(); \
return; \
}
namespace mediakit {
void H264RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {

8
ext-codec/H265Rtmp.cpp
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@ -18,14 +18,6 @@
using namespace std;
using namespace toolkit;
#define CHECK_RET(...) \
try { \
CHECK(__VA_ARGS__); \
} catch (AssertFailedException & ex) { \
WarnL << ex.what(); \
return; \
}
namespace mediakit {
void H265RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {

4
player/CMakeLists.txt
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@ -31,7 +31,9 @@ if(PKG_CONFIG_FOUND)
list(APPEND LINK_LIBRARIES PkgConfig::SDL2)
message(STATUS "found library: ${SDL2_LIBRARIES}")
endif()
else()
endif()
if(NOT SDL2_FOUND)
find_package(SDL2 QUIET)
if(SDL2_FOUND)
include_directories(SYSTEM ${SDL2_INCLUDE_DIR})

79
postman/ZLMediaKit.postman_collection.json
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@ -1,11 +1,10 @@
{
"info": {
"_postman_id": "08e3bc35-5318-4949-81bb-90d854706194",
"_postman_id": "8b3cdc62-3e18-4700-9ddd-dc9f58ebce83",
"name": "ZLMediaKit",
"description": "媒体服务器",
"schema": "https://schema.getpostman.com/json/collection/v2.1.0/collection.json",
"_exporter_id": "29185956",
"_collection_link": "https://lively-station-598157.postman.co/workspace/%E6%B5%81%E5%AA%92%E4%BD%93%E6%9C%8D%E5%8A%A1~1e119172-45b0-4ed6-b1fc-8a15d0e2d5f8/collection/29185956-08e3bc35-5318-4949-81bb-90d854706194?action=share&source=collection_link&creator=29185956"
"_exporter_id": "26338564"
},
"item": [
{
@ -34,6 +33,72 @@
},
"response": []
},
{
"name": "关闭多屏拼接(stack/stop)",
"request": {
"method": "GET",
"header": [],
"url": {
"raw": "{{ZLMediaKit_URL}}/index/api/getApiList?secret={{ZLMediaKit_secret}}&id=stack_test",
"host": [
"{{ZLMediaKit_URL}}"
],
"path": [
"index",
"api",
"getApiList"
],
"query": [
{
"key": "secret",
"value": "{{ZLMediaKit_secret}}",
"description": "api操作密钥(配置文件配置)"
},
{
"key": "id",
"value": "stack_test"
}
]
}
},
"response": []
},
{
"name": "添加多屏拼接(stack/start)",
"request": {
"method": "POST",
"header": [],
"body": {
"mode": "raw",
"raw": "{\r\n \"gapv\": 0.002,\r\n \"gaph\": 0.001,\r\n \"width\": 1920,\r\n \"url\": [\r\n [\r\n \"rtsp://kkem.me/live/test3\",\r\n \"rtsp://kkem.me/live/cy1\",\r\n \"rtsp://kkem.me/live/cy1\",\r\n \"rtsp://kkem.me/live/cy2\"\r\n ],\r\n [\r\n \"rtsp://kkem.me/live/cy1\",\r\n \"rtsp://kkem.me/live/cy5\",\r\n \"rtsp://kkem.me/live/cy3\",\r\n \"rtsp://kkem.me/live/cy4\"\r\n ],\r\n [\r\n \"rtsp://kkem.me/live/cy5\",\r\n \"rtsp://kkem.me/live/cy6\",\r\n \"rtsp://kkem.me/live/cy7\",\r\n \"rtsp://kkem.me/live/cy8\"\r\n ],\r\n [\r\n \"rtsp://kkem.me/live/cy9\",\r\n \"rtsp://kkem.me/live/cy10\",\r\n \"rtsp://kkem.me/live/cy11\",\r\n \"rtsp://kkem.me/live/cy12\"\r\n ]\r\n ],\r\n \"id\": \"89\",\r\n \"row\": 4,\r\n \"col\": 4,\r\n \"height\": 1080,\r\n \"span\": [\r\n [\r\n [\r\n 0,\r\n 0\r\n ],\r\n [\r\n 1,\r\n 1\r\n ]\r\n ],\r\n [\r\n [\r\n 3,\r\n 0\r\n ],\r\n [\r\n 3,\r\n 1\r\n ]\r\n ],\r\n [\r\n [\r\n 2,\r\n 3\r\n ],\r\n [\r\n 3,\r\n 3\r\n ]\r\n ]\r\n ]\r\n}",
"options": {
"raw": {
"language": "json"
}
}
},
"url": {
"raw": "{{ZLMediaKit_URL}}/index/api/stack/start?secret={{ZLMediaKit_secret}}",
"host": [
"{{ZLMediaKit_URL}}"
],
"path": [
"index",
"api",
"stack",
"start"
],
"query": [
{
"key": "secret",
"value": "{{ZLMediaKit_secret}}",
"description": "api操作密钥(配置文件配置)"
}
]
}
},
"response": []
},
{
"name": "获取网络线程负载(getThreadsLoad)",
"request": {
@ -1470,9 +1535,9 @@
"disabled": true
},
{
"key": "only_audio",
"key": "only_track",
"value": "1",
"description": "是否为单音频track，用于语音对讲",
"description": "是否为单音频/单视频track，0：不设置，1：单音频，2：单视频",
"disabled": true
},
{
@ -1523,9 +1588,9 @@
"description": "该端口绑定的流id\n"
},
{
"key": "only_audio",
"key": "only_track",
"value": "0",
"description": "是否为单音频track，用于语音对讲",
"description": "是否为单音频/单视频track，0：不设置，1：单音频，2：单视频",
"disabled": true
},
{

592
server/VideoStack.cpp Normal file
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@ -0,0 +1,592 @@
#if defined(ENABLE_X264) && defined(ENABLE_FFMPEG)
#include "VideoStack.h"
#include "Codec/Transcode.h"
#include "Common/Device.h"
#include "Util/logger.h"
#include "Util/util.h"
#include "json/value.h"
#include <Thread/WorkThreadPool.h>
#include <fstream>
#include <libavutil/pixfmt.h>
#include <memory>
#include <mutex>
// ITU-R BT.601
// #define RGB_TO_Y(R, G, B) ((( 66 * (R) + 129 * (G) + 25 * (B)+128) >> 8)+16)
// #define RGB_TO_U(R, G, B) (((-38 * (R) - 74 * (G) + 112 * (B)+128) >> 8)+128)
// #define RGB_TO_V(R, G, B) (((112 * (R) - 94 * (G) - 18 * (B)+128) >> 8)+128)
// ITU-R BT.709
#define RGB_TO_Y(R, G, B) (((47 * (R) + 157 * (G) + 16 * (B) + 128) >> 8) + 16)
#define RGB_TO_U(R, G, B) (((-26 * (R)-87 * (G) + 112 * (B) + 128) >> 8) + 128)
#define RGB_TO_V(R, G, B) (((112 * (R)-102 * (G)-10 * (B) + 128) >> 8) + 128)
INSTANCE_IMP(VideoStackManager)
Param::~Param()
{
VideoStackManager::Instance().unrefChannel(
id, width, height, pixfmt);
}
Channel::Channel(const std::string& id, int width, int height, AVPixelFormat pixfmt)
: _id(id)
, _width(width)
, _height(height)
, _pixfmt(pixfmt)
{
_tmp = std::make_shared<mediakit::FFmpegFrame>();
_tmp->get()->width = _width;
_tmp->get()->height = _height;
_tmp->get()->format = _pixfmt;
av_frame_get_buffer(_tmp->get(), 32);
memset(_tmp->get()->data[0], 0, _tmp->get()->linesize[0] * _height);
memset(_tmp->get()->data[1], 0, _tmp->get()->linesize[1] * _height / 2);
memset(_tmp->get()->data[2], 0, _tmp->get()->linesize[2] * _height / 2);
auto frame = VideoStackManager::Instance().getBgImg();
_sws = std::make_shared<mediakit::FFmpegSws>(_pixfmt, _width, _height);
_tmp = _sws->inputFrame(frame);
}
void Channel::addParam(const std::weak_ptr<Param>& p)
{
std::lock_guard<std::recursive_mutex> lock(_mx);
_params.push_back(p);
}
void Channel::onFrame(const mediakit::FFmpegFrame::Ptr& frame)
{
std::weak_ptr<Channel> weakSelf = shared_from_this();
_poller = _poller ? _poller : toolkit::WorkThreadPool::Instance().getPoller();
_poller->async([weakSelf, frame]() {
auto self = weakSelf.lock();
if (!self) {
return;
}
self->_tmp = self->_sws->inputFrame(frame);
self->forEachParam([self](const Param::Ptr& p) { self->fillBuffer(p); });
});
}
void Channel::forEachParam(const std::function<void(const Param::Ptr&)>& func)
{
for (auto& wp : _params) {
if (auto sp = wp.lock()) {
func(sp);
}
}
}
void Channel::fillBuffer(const Param::Ptr& p)
{
if (auto buf = p->weak_buf.lock()) {
copyData(buf, p);
}
}
void Channel::copyData(const mediakit::FFmpegFrame::Ptr& buf, const Param::Ptr& p)
{
switch (p->pixfmt) {
case AV_PIX_FMT_YUV420P: {
for (int i = 0; i < p->height; i++) {
memcpy(buf->get()->data[0] + buf->get()->linesize[0] * (i + p->posY) + p->posX,
_tmp->get()->data[0] + _tmp->get()->linesize[0] * i,
_tmp->get()->width);
}
//确保height为奇数时，也能正确的复制到最后一行uv数据
for (int i = 0; i < (p->height + 1) / 2; i++) {
// U平面
memcpy(buf->get()->data[1] + buf->get()->linesize[1] * (i + p->posY / 2) + p->posX / 2,
_tmp->get()->data[1] + _tmp->get()->linesize[1] * i,
_tmp->get()->width / 2);
// V平面
memcpy(buf->get()->data[2] + buf->get()->linesize[2] * (i + p->posY / 2) + p->posX / 2,
_tmp->get()->data[2] + _tmp->get()->linesize[2] * i,
_tmp->get()->width / 2);
}
break;
}
case AV_PIX_FMT_NV12: {
//TODO: 待实现
break;
}
default:
WarnL << "No support pixformat: " << av_get_pix_fmt_name(p->pixfmt);
break;
}
}
void StackPlayer::addChannel(const std::weak_ptr<Channel>& chn)
{
std::lock_guard<std::recursive_mutex> lock(_mx);
_channels.push_back(chn);
}
void StackPlayer::play()
{
auto url = _url;
//创建拉流 解码对象
_player = std::make_shared<mediakit::MediaPlayer>();
std::weak_ptr<mediakit::MediaPlayer> weakPlayer = _player;
std::weak_ptr<StackPlayer> weakSelf = shared_from_this();
(*_player)[mediakit::Client::kWaitTrackReady] = false;
(*_player)[mediakit::Client::kRtpType] = mediakit::Rtsp::RTP_TCP;
_player->setOnPlayResult([weakPlayer, weakSelf, url](const toolkit::SockException& ex) mutable {
TraceL << "StackPlayer: " << url << " OnPlayResult: " << ex.what();
auto strongPlayer = weakPlayer.lock();
if (!strongPlayer) {
return;
}
auto self = weakSelf.lock();
if (!self) {
return;
}
if (!ex) {
// 取消定时器
self->_timer.reset();
self->_failedCount = 0;
} else {
self->onDisconnect();
self->rePlay(url);
}
auto videoTrack = std::dynamic_pointer_cast<mediakit::VideoTrack>(strongPlayer->getTrack(mediakit::TrackVideo, false));
//auto audioTrack = std::dynamic_pointer_cast<mediakit::AudioTrack>(strongPlayer->getTrack(mediakit::TrackAudio, false));
if (videoTrack) {
//TODO:添加使用显卡还是cpu解码的判断逻辑
//auto decoder = std::make_shared<FFmpegDecoder>(videoTrack, 1, std::vector<std::string>{ "hevc_cuvid", "h264_cuvid"});
auto decoder = std::make_shared<mediakit::FFmpegDecoder>(videoTrack, 0, std::vector<std::string> { "h264", "hevc" });
decoder->setOnDecode([weakSelf](const mediakit::FFmpegFrame::Ptr& frame) mutable {
auto self = weakSelf.lock();
if (!self) {
return;
}
self->onFrame(frame);
});
videoTrack->addDelegate([decoder](const mediakit::Frame::Ptr& frame) {
return decoder->inputFrame(frame, false, true);
});
}
});
_player->setOnShutdown([weakPlayer, url, weakSelf](const toolkit::SockException& ex) {
TraceL << "StackPlayer: " << url << " OnShutdown: " << ex.what();
auto strongPlayer = weakPlayer.lock();
if (!strongPlayer) {
return;
}
auto self = weakSelf.lock();
if (!self) {
return;
}
self->onDisconnect();
self->rePlay(url);
});
_player->play(url);
}
void StackPlayer::onFrame(const mediakit::FFmpegFrame::Ptr& frame)
{
std::lock_guard<std::recursive_mutex> lock(_mx);
for (auto& weak_chn : _channels) {
if (auto chn = weak_chn.lock()) {
chn->onFrame(frame);
}
}
}
void StackPlayer::onDisconnect()
{
std::lock_guard<std::recursive_mutex> lock(_mx);
for (auto& weak_chn : _channels) {
if (auto chn = weak_chn.lock()) {
auto frame = VideoStackManager::Instance().getBgImg();
chn->onFrame(frame);
}
}
}
void StackPlayer::rePlay(const std::string& url)
{
_failedCount++;
auto delay = MAX(2 * 1000, MIN(_failedCount * 3 * 1000, 60 * 1000)); //步进延迟 重试间隔
std::weak_ptr<StackPlayer> weakSelf = shared_from_this();
_timer = std::make_shared<toolkit::Timer>(
delay / 1000.0f, [weakSelf, url]() {
auto self = weakSelf.lock();
if (!self) {
}
WarnL << "replay [" << self->_failedCount << "]:" << url;
self->_player->play(url);
return false;
},
nullptr);
}
VideoStack::VideoStack(const std::string& id, int width, int height, AVPixelFormat pixfmt, float fps, int bitRate)
: _id(id)
, _width(width)
, _height(height)
, _pixfmt(pixfmt)
, _fps(fps)
, _bitRate(bitRate)
{
_buffer = std::make_shared<mediakit::FFmpegFrame>();
_buffer->get()->width = _width;
_buffer->get()->height = _height;
_buffer->get()->format = _pixfmt;
av_frame_get_buffer(_buffer->get(), 32);
_dev = std::make_shared<mediakit::DevChannel>(mediakit::MediaTuple { DEFAULT_VHOST, "live", _id });
mediakit::VideoInfo info;
info.codecId = mediakit::CodecH264;
info.iWidth = _width;
info.iHeight = _height;
info.iFrameRate = _fps;
info.iBitRate = _bitRate;
_dev->initVideo(info);
//dev->initAudio(); //TODO:音频
_dev->addTrackCompleted();
_isExit = false;
}
VideoStack::~VideoStack()
{
_isExit = true;
if (_thread.joinable()) {
_thread.join();
}
}
void VideoStack::setParam(const Params& params)
{
if (_params) {
for (auto& p : (*_params)) {
if (!p)
continue;
p->weak_buf.reset();
}
}
initBgColor();
for (auto& p : (*params)) {
if (!p)
continue;
p->weak_buf = _buffer;
if (auto chn = p->weak_chn.lock()) {
chn->addParam(p);
chn->fillBuffer(p);
}
}
_params = params;
}
void VideoStack::start()
{
_thread = std::thread([&]() {
uint64_t pts = 0;
int frameInterval = 1000 / _fps;
auto lastEncTP = std::chrono::steady_clock::now();
while (!_isExit) {
if (std::chrono::steady_clock::now() - lastEncTP > std::chrono::milliseconds(frameInterval)) {
lastEncTP = std::chrono::steady_clock::now();
_dev->inputYUV((char**)_buffer->get()->data, _buffer->get()->linesize, pts);
pts += frameInterval;
}
}
});
}
void VideoStack::initBgColor()
{
//填充底色
auto R = 20;
auto G = 20;
auto B = 20;
double Y = RGB_TO_Y(R, G, B);
double U = RGB_TO_U(R, G, B);
double V = RGB_TO_V(R, G, B);
memset(_buffer->get()->data[0], Y, _buffer->get()->linesize[0] * _height);
memset(_buffer->get()->data[1], U, _buffer->get()->linesize[1] * _height / 2);
memset(_buffer->get()->data[2], V, _buffer->get()->linesize[2] * _height / 2);
}
Channel::Ptr VideoStackManager::getChannel(const std::string& id,
int width,
int height,
AVPixelFormat pixfmt)
{
std::lock_guard<std::recursive_mutex> lock(_mx);
auto key = id + std::to_string(width) + std::to_string(height) + std::to_string(pixfmt);
auto it = _channelMap.find(key);
if (it != _channelMap.end()) {
return it->second->acquire();
}
return createChannel(id, width, height, pixfmt);
}
void VideoStackManager::unrefChannel(const std::string& id,
int width,
int height,
AVPixelFormat pixfmt)
{
std::lock_guard<std::recursive_mutex> lock(_mx);
auto key = id + std::to_string(width) + std::to_string(height) + std::to_string(pixfmt);
auto chn_it = _channelMap.find(key);
if (chn_it != _channelMap.end() && chn_it->second->dispose()) {
_channelMap.erase(chn_it);
auto player_it = _playerMap.find(id);
if (player_it != _playerMap.end() && player_it->second->dispose()) {
_playerMap.erase(player_it);
}
}
}
int VideoStackManager::startVideoStack(const Json::Value& json)
{
std::string id;
int width, height;
auto params = parseParams(json, id, width, height);
if (!params) {
ErrorL << "Videostack parse params failed!";
return -1;
}
auto stack = std::make_shared<VideoStack>(id, width, height);
for (auto& p : (*params)) {
if (!p)
continue;
p->weak_chn = getChannel(p->id, p->width, p->height, p->pixfmt);
}
stack->setParam(params);
stack->start();
std::lock_guard<std::recursive_mutex> lock(_mx);
_stackMap[id] = stack;
return 0;
}
int VideoStackManager::resetVideoStack(const Json::Value& json)
{
std::string id;
int width, height;
auto params = parseParams(json, id, width, height);
if (!params) {
return -1;
}
VideoStack::Ptr stack;
{
std::lock_guard<std::recursive_mutex> lock(_mx);
auto it = _stackMap.find(id);
if (it == _stackMap.end()) {
return -2;
}
stack = it->second;
}
for (auto& p : (*params)) {
if (!p)
continue;
p->weak_chn = getChannel(p->id, p->width, p->height, p->pixfmt);
}
stack->setParam(params);
return 0;
}
int VideoStackManager::stopVideoStack(const std::string& id)
{
std::lock_guard<std::recursive_mutex> lock(_mx);
auto it = _stackMap.find(id);
if (it != _stackMap.end()) {
_stackMap.erase(it);
InfoL << "VideoStack stop: " << id;
return 0;
}
return -1;
}
mediakit::FFmpegFrame::Ptr VideoStackManager::getBgImg()
{
return _bgImg;
}
Params VideoStackManager::parseParams(const Json::Value& json,
std::string& id,
int& width,
int& height)
{
try {
id = json["id"].asString();
width = json["width"].asInt();
height = json["height"].asInt();
int rows = json["row"].asInt(); //堆叠行数
int cols = json["col"].asInt(); //堆叠列数
float gapv = json["gapv"].asFloat(); //垂直间距
float gaph = json["gaph"].asFloat(); //水平间距
//单个间距
int gaphPix = static_cast<int>(round(width * gaph));
int gapvPix = static_cast<int>(round(height * gapv));
// 根据间距计算格子宽高
int gridWidth = cols > 1 ? (width - gaphPix * (cols - 1)) / cols : width;
int gridHeight = rows > 1 ? (height - gapvPix * (rows - 1)) / rows : height;
auto params = std::make_shared<std::vector<Param::Ptr>>(rows * cols);
for (int row = 0; row < rows; row++) {
for (int col = 0; col < cols; col++) {
std::string url = json["url"][row][col].asString();
auto param = std::make_shared<Param>();
param->posX = gridWidth * col + col * gaphPix;
param->posY = gridHeight * row + row * gapvPix;
param->width = gridWidth;
param->height = gridHeight;
param->id = url;
(*params)[row * cols + col] = param;
}
}
//判断是否需要合并格子 （焦点屏）
if (!json["span"].empty() && json.isMember("span")) {
for (const auto& subArray : json["span"]) {
if (!subArray.isArray() || subArray.size() != 2) {
throw Json::LogicError("Incorrect 'span' sub-array format in JSON");
}
std::array<int, 4> mergePos;
int index = 0;
for (const auto& innerArray : subArray) {
if (!innerArray.isArray() || innerArray.size() != 2) {
throw Json::LogicError("Incorrect 'span' inner-array format in JSON");
}
for (const auto& number : innerArray) {
if (index < mergePos.size()) {
mergePos[index++] = number.asInt();
}
}
}
for (int i = mergePos[0]; i <= mergePos[2]; i++) {
for (int j = mergePos[1]; j <= mergePos[3]; j++) {
if (i == mergePos[0] && j == mergePos[1]) {
(*params)[i * cols + j]->width = (mergePos[3] - mergePos[1] + 1) * gridWidth + (mergePos[3] - mergePos[1]) * gapvPix;
(*params)[i * cols + j]->height = (mergePos[2] - mergePos[0] + 1) * gridHeight + (mergePos[2] - mergePos[0]) * gaphPix;
} else {
(*params)[i * cols + j] = nullptr;
}
}
}
}
}
return params;
} catch (const std::exception& e) {
ErrorL << "Videostack parse params failed! " << e.what();
return nullptr;
}
}
bool VideoStackManager::loadBgImg(const std::string& path)
{
_bgImg = std::make_shared<mediakit::FFmpegFrame>();
_bgImg->get()->width = 1280;
_bgImg->get()->height = 720;
_bgImg->get()->format = AV_PIX_FMT_YUV420P;
av_frame_get_buffer(_bgImg->get(), 32);
std::ifstream file(path, std::ios::binary);
if (!file.is_open()) {
return false;
}
file.read((char*)_bgImg->get()->data[0], _bgImg->get()->linesize[0] * _bgImg->get()->height); // Y
file.read((char*)_bgImg->get()->data[1], _bgImg->get()->linesize[1] * _bgImg->get()->height / 2); // U
file.read((char*)_bgImg->get()->data[2], _bgImg->get()->linesize[2] * _bgImg->get()->height / 2); // V
return true;
}
Channel::Ptr VideoStackManager::createChannel(const std::string& id,
int width,
int height,
AVPixelFormat pixfmt)
{
std::lock_guard<std::recursive_mutex> lock(_mx);
StackPlayer::Ptr player;
auto it = _playerMap.find(id);
if (it != _playerMap.end()) {
player = it->second->acquire();
} else {
player = createPlayer(id);
}
auto refChn = std::make_shared<RefWrapper<Channel::Ptr>>(std::make_shared<Channel>(id, width, height, pixfmt));
auto chn = refChn->acquire();
player->addChannel(chn);
_channelMap[id + std::to_string(width) + std::to_string(height) + std::to_string(pixfmt)] = refChn;
return chn;
}
StackPlayer::Ptr VideoStackManager::createPlayer(const std::string& id)
{
std::lock_guard<std::recursive_mutex> lock(_mx);
auto refPlayer = std::make_shared<RefWrapper<StackPlayer::Ptr>>(std::make_shared<StackPlayer>(id));
_playerMap[id] = refPlayer;
auto player = refPlayer->acquire();
if (!id.empty()) {
player->play();
}
return player;
}
#endif

208
server/VideoStack.h Normal file
View File

@ -0,0 +1,208 @@
#pragma once
#if defined(ENABLE_X264) && defined(ENABLE_FFMPEG)
#include "Codec/Transcode.h"
#include "Common/Device.h"
#include "Player/MediaPlayer.h"
#include "json/json.h"
#include <mutex>
template <typename T>
class RefWrapper {
public:
using Ptr = std::shared_ptr<RefWrapper<T>>;
template <typename... Args>
explicit RefWrapper(Args&&... args)
: _rc(0)
, _entity(std::forward<Args>(args)...)
{
}
T acquire()
{
++_rc;
return _entity;
}
bool dispose() { return --_rc <= 0; }
private:
T _entity;
std::atomic<int> _rc;
};
class Channel;
struct Param {
using Ptr = std::shared_ptr<Param>;
int posX = 0;
int posY = 0;
int width = 0;
int height = 0;
AVPixelFormat pixfmt = AV_PIX_FMT_YUV420P;
std::string id {};
// runtime
std::weak_ptr<Channel> weak_chn;
std::weak_ptr<mediakit::FFmpegFrame> weak_buf;
~Param();
};
using Params = std::shared_ptr<std::vector<Param::Ptr>>;
class Channel : public std::enable_shared_from_this<Channel> {
public:
using Ptr = std::shared_ptr<Channel>;
Channel(const std::string& id, int width, int height, AVPixelFormat pixfmt);
void addParam(const std::weak_ptr<Param>& p);
void onFrame(const mediakit::FFmpegFrame::Ptr& frame);
void fillBuffer(const Param::Ptr& p);
protected:
void forEachParam(const std::function<void(const Param::Ptr&)>& func);
void copyData(const mediakit::FFmpegFrame::Ptr& buf, const Param::Ptr& p);
private:
std::string _id;
int _width;
int _height;
AVPixelFormat _pixfmt;
mediakit::FFmpegFrame::Ptr _tmp;
std::recursive_mutex _mx;
std::vector<std::weak_ptr<Param>> _params;
mediakit::FFmpegSws::Ptr _sws;
toolkit::EventPoller::Ptr _poller;
};
class StackPlayer : public std::enable_shared_from_this<StackPlayer> {
public:
using Ptr = std::shared_ptr<StackPlayer>;
StackPlayer(const std::string& url)
: _url(url)
{
}
void addChannel(const std::weak_ptr<Channel>& chn);
void play();
void onFrame(const mediakit::FFmpegFrame::Ptr& frame);
void onDisconnect();
protected:
void rePlay(const std::string& url);
private:
std::string _url;
mediakit::MediaPlayer::Ptr _player;
//用于断线重连
toolkit::Timer::Ptr _timer;
int _failedCount = 0;
std::recursive_mutex _mx;
std::vector<std::weak_ptr<Channel>> _channels;
};
class VideoStack {
public:
using Ptr = std::shared_ptr<VideoStack>;
VideoStack(const std::string& url,
int width = 1920,
int height = 1080,
AVPixelFormat pixfmt = AV_PIX_FMT_YUV420P,
float fps = 25.0,
int bitRate = 2 * 1024 * 1024);
~VideoStack();
void setParam(const Params& params);
void start();
protected:
void initBgColor();
public:
Params _params;
mediakit::FFmpegFrame::Ptr _buffer;
private:
std::string _id;
int _width;
int _height;
AVPixelFormat _pixfmt;
float _fps;
int _bitRate;
mediakit::DevChannel::Ptr _dev;
bool _isExit;
std::thread _thread;
};
class VideoStackManager {
public:
static VideoStackManager& Instance();
Channel::Ptr getChannel(const std::string& id,
int width,
int height,
AVPixelFormat pixfmt);
void unrefChannel(const std::string& id,
int width,
int height,
AVPixelFormat pixfmt);
int startVideoStack(const Json::Value& json);
int resetVideoStack(const Json::Value& json);
int stopVideoStack(const std::string& id);
bool loadBgImg(const std::string& path);
mediakit::FFmpegFrame::Ptr getBgImg();
protected:
Params parseParams(const Json::Value& json,
std::string& id,
int& width,
int& height);
protected:
Channel::Ptr createChannel(const std::string& id,
int width,
int height,
AVPixelFormat pixfmt);
StackPlayer::Ptr createPlayer(const std::string& id);
private:
mediakit::FFmpegFrame::Ptr _bgImg;
private:
std::recursive_mutex _mx;
std::unordered_map<std::string, VideoStack::Ptr> _stackMap;
std::unordered_map<std::string, RefWrapper<Channel::Ptr>::Ptr> _channelMap;
std::unordered_map<std::string, RefWrapper<StackPlayer::Ptr>::Ptr> _playerMap;
};
#endif

482
server/WebApi.cpp
View File

@ -59,7 +59,11 @@
#endif
#if defined(ENABLE_VERSION)
#include "version.h"
#include "ZLMVersion.h"
#endif
#if defined(ENABLE_X264) && defined (ENABLE_FFMPEG)
#include "VideoStack.h"
#endif
using namespace std;
@ -115,7 +119,7 @@ static HttpApi toApi(const function<void(API_ARGS_MAP_ASYNC)> &cb) {
//参数解析成map
auto args = getAllArgs(parser);
cb(sender, headerOut, HttpAllArgs<decltype(args)>(parser, args), val, invoker);
cb(sender, headerOut, ArgsMap(parser, args), val, invoker);
};
}
@ -143,7 +147,7 @@ static HttpApi toApi(const function<void(API_ARGS_JSON_ASYNC)> &cb) {
Json::Reader reader;
reader.parse(parser.content(), args);
cb(sender, headerOut, HttpAllArgs<decltype(args)>(parser, args), val, invoker);
cb(sender, headerOut, ArgsJson(parser, args), val, invoker);
};
}
@ -163,7 +167,7 @@ static HttpApi toApi(const function<void(API_ARGS_STRING_ASYNC)> &cb) {
Json::Value val;
val["code"] = API::Success;
cb(sender, headerOut, HttpAllArgs<string>(parser, (string &)parser.content()), val, invoker);
cb(sender, headerOut, ArgsString(parser, (string &)parser.content()), val, invoker);
};
}
@ -297,22 +301,71 @@ static inline void addHttpListener(){
});
}
template <typename Type>
class ServiceController {
public:
using Pointer = std::shared_ptr<Type>;
std::unordered_map<std::string, Pointer> _map;
mutable std::recursive_mutex _mtx;
void clear() {
decltype(_map) copy;
{
std::lock_guard<std::recursive_mutex> lck(_mtx);
copy.swap(_map);
}
}
size_t erase(const std::string &key) {
std::lock_guard<std::recursive_mutex> lck(_mtx);
return _map.erase(key);
}
Pointer find(const std::string &key) const {
std::lock_guard<std::recursive_mutex> lck(_mtx);
auto it = _map.find(key);
if (it == _map.end()) {
return nullptr;
}
return it->second;
}
template<class ..._Args>
Pointer make(const std::string &key, _Args&& ...__args) {
// assert(!find(key));
auto server = std::make_shared<Type>(std::forward<_Args>(__args)...);
std::lock_guard<std::recursive_mutex> lck(_mtx);
auto it = _map.emplace(key, server);
assert(it.second);
return server;
}
template<class ..._Args>
Pointer makeWithAction(const std::string &key, function<void(Pointer)> action, _Args&& ...__args) {
// assert(!find(key));
auto server = std::make_shared<Type>(std::forward<_Args>(__args)...);
action(server);
std::lock_guard<std::recursive_mutex> lck(_mtx);
auto it = _map.emplace(key, server);
assert(it.second);
return server;
}
};
//拉流代理器列表
static unordered_map<string, PlayerProxy::Ptr> s_proxyMap;
static recursive_mutex s_proxyMapMtx;
static ServiceController<PlayerProxy> s_player_proxy;
//推流代理器列表
static unordered_map<string, PusherProxy::Ptr> s_proxyPusherMap;
static recursive_mutex s_proxyPusherMapMtx;
static ServiceController<PusherProxy> s_pusher_proxy;
//FFmpeg拉流代理器列表
static unordered_map<string, FFmpegSource::Ptr> s_ffmpegMap;
static recursive_mutex s_ffmpegMapMtx;
static ServiceController<FFmpegSource> s_ffmpeg_src;
#if defined(ENABLE_RTPPROXY)
//rtp服务器列表
static unordered_map<string, RtpServer::Ptr> s_rtpServerMap;
static recursive_mutex s_rtpServerMapMtx;
static ServiceController<RtpServer> s_rtp_server;
#endif
static inline string getProxyKey(const string &vhost, const string &app, const string &stream) {
@ -415,47 +468,24 @@ Value makeMediaSourceJson(MediaSource &media){
}
#if defined(ENABLE_RTPPROXY)
uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio, bool multiplex) {
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
if (s_rtpServerMap.find(stream_id) != s_rtpServerMap.end()) {
uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc, int only_track, bool multiplex) {
if (s_rtp_server.find(stream_id)) {
//为了防止RtpProcess所有权限混乱的问题，不允许重复添加相同的stream_id
return 0;
}
RtpServer::Ptr server = std::make_shared<RtpServer>();
server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_audio, multiplex);
auto server = s_rtp_server.makeWithAction(stream_id, [&](RtpServer::Ptr server) {
server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_track, multiplex);
});
server->setOnDetach([stream_id]() {
//设置rtp超时移除事件
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
s_rtpServerMap.erase(stream_id);
s_rtp_server.erase(stream_id);
});
//保存对象
s_rtpServerMap.emplace(stream_id, server);
//回复json
return server->getPort();
}
void connectRtpServer(const string &stream_id, const string &dst_url, uint16_t dst_port, const function<void(const SockException &ex)> &cb) {
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
auto it = s_rtpServerMap.find(stream_id);
if (it == s_rtpServerMap.end()) {
cb(SockException(Err_other, "未找到rtp服务"));
return;
}
it->second->connectToServer(dst_url, dst_port, cb);
}
bool closeRtpServer(const string &stream_id) {
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
auto it = s_rtpServerMap.find(stream_id);
if (it == s_rtpServerMap.end()) {
return false;
}
auto server = it->second;
s_rtpServerMap.erase(it);
return true;
}
#endif
void getStatisticJson(const function<void(Value &val)> &cb) {
@ -546,23 +576,23 @@ void addStreamProxy(const string &vhost, const string &app, const string &stream
const ProtocolOption &option, int rtp_type, float timeout_sec, const mINI &args,
const function<void(const SockException &ex, const string &key)> &cb) {
auto key = getProxyKey(vhost, app, stream);
lock_guard<recursive_mutex> lck(s_proxyMapMtx);
if (s_proxyMap.find(key) != s_proxyMap.end()) {
if (s_player_proxy.find(key)) {
//已经在拉流了
cb(SockException(Err_other, "This stream already exists"), key);
return;
}
//添加拉流代理
auto player = std::make_shared<PlayerProxy>(vhost, app, stream, option, retry_count);
s_proxyMap[key] = player;
auto player = s_player_proxy.make(key, vhost, app, stream, option, retry_count);
// 先透传参数
player->mINI::operator=(args);
// 先透传拷贝参数
for (auto &pr : args) {
(*player)[pr.first] = pr.second;
}
//指定RTP over TCP(播放rtsp时有效)
(*player)[Client::kRtpType] = rtp_type;
if (timeout_sec > 0.1) {
if (timeout_sec > 0.1f) {
//播放握手超时时间
(*player)[Client::kTimeoutMS] = timeout_sec * 1000;
}
@ -570,28 +600,69 @@ void addStreamProxy(const string &vhost, const string &app, const string &stream
//开始播放，如果播放失败或者播放中止，将会自动重试若干次，默认一直重试
player->setPlayCallbackOnce([cb, key](const SockException &ex) {
if (ex) {
lock_guard<recursive_mutex> lck(s_proxyMapMtx);
s_proxyMap.erase(key);
s_player_proxy.erase(key);
}
cb(ex, key);
});
//被主动关闭拉流
player->setOnClose([key](const SockException &ex) {
lock_guard<recursive_mutex> lck(s_proxyMapMtx);
s_proxyMap.erase(key);
s_player_proxy.erase(key);
});
player->play(url);
};
template <typename Type>
static void getArgsValue(const HttpAllArgs<ApiArgsType> &allArgs, const string &key, Type &value) {
auto val = allArgs[key];
if (!val.empty()) {
value = (Type)val;
void addStreamPusherProxy(const string &schema,
const string &vhost,
const string &app,
const string &stream,
const string &url,
int retry_count,
int rtp_type,
float timeout_sec,
const function<void(const SockException &ex, const string &key)> &cb) {
auto key = getPusherKey(schema, vhost, app, stream, url);
auto src = MediaSource::find(schema, vhost, app, stream);
if (!src) {
cb(SockException(Err_other, "can not find the source stream"), key);
return;
}
if (s_pusher_proxy.find(key)) {
//已经在推流了
cb(SockException(Err_success), key);
return;
}
//添加推流代理
auto pusher = s_pusher_proxy.make(key, src, retry_count);
//指定RTP over TCP(播放rtsp时有效)
pusher->emplace(Client::kRtpType, rtp_type);
if (timeout_sec > 0.1f) {
//推流握手超时时间
pusher->emplace(Client::kTimeoutMS, timeout_sec * 1000);
}
//开始推流，如果推流失败或者推流中止，将会自动重试若干次，默认一直重试
pusher->setPushCallbackOnce([cb, key, url](const SockException &ex) {
if (ex) {
WarnL << "Push " << url << " failed, key: " << key << ", err: " << ex;
s_pusher_proxy.erase(key);
}
cb(ex, key);
});
//被主动关闭推流
pusher->setOnClose([key, url](const SockException &ex) {
WarnL << "Push " << url << " failed, key: " << key << ", err: " << ex;
s_pusher_proxy.erase(key);
});
pusher->publish(url);
}
/**
* 安装api接口
* 所有api都支持GET和POST两种方式
@ -657,7 +728,7 @@ void installWebApi() {
CHECK_SECRET();
auto &ini = mINI::Instance();
int changed = API::Success;
for (auto &pr : allArgs.getArgs()) {
for (auto &pr : allArgs.args) {
if (ini.find(pr.first) == ini.end()) {
#if 1
//没有这个key
@ -973,59 +1044,6 @@ void installWebApi() {
val["count_hit"] = (Json::UInt64)count_hit;
});
static auto addStreamPusherProxy = [](const string &schema,
const string &vhost,
const string &app,
const string &stream,
const string &url,
int retry_count,
int rtp_type,
float timeout_sec,
const function<void(const SockException &ex, const string &key)> &cb) {
auto key = getPusherKey(schema, vhost, app, stream, url);
auto src = MediaSource::find(schema, vhost, app, stream);
if (!src) {
cb(SockException(Err_other, "can not find the source stream"), key);
return;
}
lock_guard<recursive_mutex> lck(s_proxyPusherMapMtx);
if (s_proxyPusherMap.find(key) != s_proxyPusherMap.end()) {
//已经在推流了
cb(SockException(Err_success), key);
return;
}
//添加推流代理
auto pusher = std::make_shared<PusherProxy>(src, retry_count);
s_proxyPusherMap[key] = pusher;
//指定RTP over TCP(播放rtsp时有效)
(*pusher)[Client::kRtpType] = rtp_type;
if (timeout_sec > 0.1) {
//推流握手超时时间
(*pusher)[Client::kTimeoutMS] = timeout_sec * 1000;
}
//开始推流，如果推流失败或者推流中止，将会自动重试若干次，默认一直重试
pusher->setPushCallbackOnce([cb, key, url](const SockException &ex) {
if (ex) {
WarnL << "Push " << url << " failed, key: " << key << ", err: " << ex;
lock_guard<recursive_mutex> lck(s_proxyPusherMapMtx);
s_proxyPusherMap.erase(key);
}
cb(ex, key);
});
//被主动关闭推流
pusher->setOnClose([key, url](const SockException &ex) {
WarnL << "Push " << url << " failed, key: " << key << ", err: " << ex;
lock_guard<recursive_mutex> lck(s_proxyPusherMapMtx);
s_proxyPusherMap.erase(key);
});
pusher->publish(url);
};
//动态添加rtsp/rtmp推流代理
//测试url http://127.0.0.1/index/api/addStreamPusherProxy?schema=rtmp&vhost=__defaultVhost__&app=proxy&stream=0&dst_url=rtmp://127.0.0.1/live/obs
api_regist("/index/api/addStreamPusherProxy", [](API_ARGS_MAP_ASYNC) {
@ -1058,8 +1076,7 @@ void installWebApi() {
api_regist("/index/api/delStreamPusherProxy", [](API_ARGS_MAP) {
CHECK_SECRET();
CHECK_ARGS("key");
lock_guard<recursive_mutex> lck(s_proxyPusherMapMtx);
val["data"]["flag"] = s_proxyPusherMap.erase(allArgs["key"]) == 1;
val["data"]["flag"] = s_pusher_proxy.erase(allArgs["key"]) == 1;
});
//动态添加rtsp/rtmp拉流代理
@ -1069,7 +1086,7 @@ void installWebApi() {
CHECK_ARGS("vhost","app","stream","url");
mINI args;
for (auto &pr : allArgs.getArgs()) {
for (auto &pr : allArgs.args) {
args.emplace(pr.first, pr.second);
}
@ -1100,8 +1117,7 @@ void installWebApi() {
api_regist("/index/api/delStreamProxy",[](API_ARGS_MAP){
CHECK_SECRET();
CHECK_ARGS("key");
lock_guard<recursive_mutex> lck(s_proxyMapMtx);
val["data"]["flag"] = s_proxyMap.erase(allArgs["key"]) == 1;
val["data"]["flag"] = s_player_proxy.erase(allArgs["key"]) == 1;
});
static auto addFFmpegSource = [](const string &ffmpeg_cmd_key,
@ -1112,25 +1128,21 @@ void installWebApi() {
bool enable_mp4,
const function<void(const SockException &ex, const string &key)> &cb) {
auto key = MD5(dst_url).hexdigest();
lock_guard<decltype(s_ffmpegMapMtx)> lck(s_ffmpegMapMtx);
if (s_ffmpegMap.find(key) != s_ffmpegMap.end()) {
if (s_ffmpeg_src.find(key)) {
//已经在拉流了
cb(SockException(Err_success), key);
return;
}
FFmpegSource::Ptr ffmpeg = std::make_shared<FFmpegSource>();
s_ffmpegMap[key] = ffmpeg;
auto ffmpeg = s_ffmpeg_src.make(key);
ffmpeg->setOnClose([key]() {
lock_guard<decltype(s_ffmpegMapMtx)> lck(s_ffmpegMapMtx);
s_ffmpegMap.erase(key);
s_ffmpeg_src.erase(key);
});
ffmpeg->setupRecordFlag(enable_hls, enable_mp4);
ffmpeg->play(ffmpeg_cmd_key, src_url, dst_url, timeout_ms, [cb, key](const SockException &ex) {
if (ex) {
lock_guard<decltype(s_ffmpegMapMtx)> lck(s_ffmpegMapMtx);
s_ffmpegMap.erase(key);
s_ffmpeg_src.erase(key);
}
cb(ex, key);
});
@ -1164,15 +1176,14 @@ void installWebApi() {
api_regist("/index/api/delFFmpegSource",[](API_ARGS_MAP){
CHECK_SECRET();
CHECK_ARGS("key");
lock_guard<decltype(s_ffmpegMapMtx)> lck(s_ffmpegMapMtx);
val["data"]["flag"] = s_ffmpegMap.erase(allArgs["key"]) == 1;
val["data"]["flag"] = s_ffmpeg_src.erase(allArgs["key"]) == 1;
});
//新增http api下载可执行程序文件接口
//测试url http://127.0.0.1/index/api/downloadBin
api_regist("/index/api/downloadBin",[](API_ARGS_MAP_ASYNC){
CHECK_SECRET();
invoker.responseFile(allArgs.getParser().getHeader(),StrCaseMap(),exePath());
invoker.responseFile(allArgs.parser.getHeader(), StrCaseMap(), exePath());
});
#if defined(ENABLE_RTPPROXY)
@ -1198,12 +1209,17 @@ void installWebApi() {
//兼容老版本请求，新版本去除enable_tcp参数并新增tcp_mode参数
tcp_mode = 1;
}
auto only_track = allArgs["only_track"].as<int>();
if (allArgs["only_audio"].as<bool>()) {
// 兼容老版本请求，新版本去除only_audio参数并新增only_track参数
only_track = 1;
}
std::string local_ip = "::";
if (!allArgs["local_ip"].empty()) {
local_ip = allArgs["local_ip"];
}
auto port = openRtpServer(allArgs["port"], stream_id, tcp_mode, local_ip, allArgs["re_use_port"].as<bool>(),
allArgs["ssrc"].as<uint32_t>(), allArgs["only_audio"].as<bool>());
allArgs["ssrc"].as<uint32_t>(), only_track);
if (port == 0) {
throw InvalidArgsException("该stream_id已存在");
}
@ -1220,11 +1236,16 @@ void installWebApi() {
// 兼容老版本请求，新版本去除enable_tcp参数并新增tcp_mode参数
tcp_mode = 1;
}
auto only_track = allArgs["only_track"].as<int>();
if (allArgs["only_audio"].as<bool>()) {
// 兼容老版本请求，新版本去除only_audio参数并新增only_track参数
only_track = 1;
}
std::string local_ip = "::";
if (!allArgs["local_ip"].empty()) {
local_ip = allArgs["local_ip"];
}
auto port = openRtpServer(allArgs["port"], stream_id, tcp_mode, local_ip, true, 0, allArgs["only_audio"].as<bool>(),true);
auto port = openRtpServer(allArgs["port"], stream_id, tcp_mode, local_ip, true, 0, only_track,true);
if (port == 0) {
throw InvalidArgsException("该stream_id已存在");
}
@ -1235,22 +1256,27 @@ void installWebApi() {
api_regist("/index/api/connectRtpServer", [](API_ARGS_MAP_ASYNC) {
CHECK_SECRET();
CHECK_ARGS("stream_id", "dst_url", "dst_port");
connectRtpServer(
allArgs["stream_id"], allArgs["dst_url"], allArgs["dst_port"],
[val, headerOut, invoker](const SockException &ex) mutable {
if (ex) {
val["code"] = API::OtherFailed;
val["msg"] = ex.what();
}
invoker(200, headerOut, val.toStyledString());
});
auto cb = [val, headerOut, invoker](const SockException &ex) mutable {
if (ex) {
val["code"] = API::OtherFailed;
val["msg"] = ex.what();
}
invoker(200, headerOut, val.toStyledString());
};
auto server = s_rtp_server.find(allArgs["stream_id"]);
if (!server) {
cb(SockException(Err_other, "未找到rtp服务"));
return;
}
server->connectToServer(allArgs["dst_url"], allArgs["dst_port"], cb);
});
api_regist("/index/api/closeRtpServer",[](API_ARGS_MAP){
CHECK_SECRET();
CHECK_ARGS("stream_id");
if(!closeRtpServer(allArgs["stream_id"])){
if(s_rtp_server.erase(allArgs["stream_id"]) == 0){
val["hit"] = 0;
return;
}
@ -1261,19 +1287,18 @@ void installWebApi() {
CHECK_SECRET();
CHECK_ARGS("stream_id", "ssrc");
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
auto it = s_rtpServerMap.find(allArgs["stream_id"]);
if (it == s_rtpServerMap.end()) {
auto server = s_rtp_server.find(allArgs["stream_id"]);
if (!server) {
throw ApiRetException("RtpServer not found by stream_id", API::NotFound);
}
it->second->updateSSRC(allArgs["ssrc"]);
server->updateSSRC(allArgs["ssrc"]);
});
api_regist("/index/api/listRtpServer",[](API_ARGS_MAP){
CHECK_SECRET();
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
for (auto &pr : s_rtpServerMap) {
std::lock_guard<std::recursive_mutex> lck(s_rtp_server._mtx);
for (auto &pr : s_rtp_server._map) {
Value obj;
obj["stream_id"] = pr.first;
obj["port"] = pr.second->getPort();
@ -1289,7 +1314,11 @@ void installWebApi() {
if (!src) {
throw ApiRetException("can not find the source stream", API::NotFound);
}
auto type = allArgs["type"].as<int>();
if (!allArgs["use_ps"].empty()) {
// 兼容之前的use_ps参数
type = allArgs["use_ps"].as<int>();
}
MediaSourceEvent::SendRtpArgs args;
args.passive = false;
args.dst_url = allArgs["dst_url"];
@ -1299,11 +1328,11 @@ void installWebApi() {
args.is_udp = allArgs["is_udp"];
args.src_port = allArgs["src_port"];
args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
args.use_ps = allArgs["use_ps"].empty() ? true : allArgs["use_ps"].as<bool>();
args.type = (MediaSourceEvent::SendRtpArgs::Type)type;
args.only_audio = allArgs["only_audio"].as<bool>();
args.udp_rtcp_timeout = allArgs["udp_rtcp_timeout"];
args.recv_stream_id = allArgs["recv_stream_id"];
TraceL << "startSendRtp, pt " << int(args.pt) << " ps " << args.use_ps << " audio " << args.only_audio;
TraceL << "startSendRtp, pt " << int(args.pt) << " rtp type " << type << " audio " << args.only_audio;
src->getOwnerPoller()->async([=]() mutable {
src->startSendRtp(args, [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable {
@ -1325,6 +1354,11 @@ void installWebApi() {
if (!src) {
throw ApiRetException("can not find the source stream", API::NotFound);
}
auto type = allArgs["type"].as<int>();
if (!allArgs["use_ps"].empty()) {
// 兼容之前的use_ps参数
type = allArgs["use_ps"].as<int>();
}
MediaSourceEvent::SendRtpArgs args;
args.passive = true;
@ -1332,12 +1366,12 @@ void installWebApi() {
args.is_udp = false;
args.src_port = allArgs["src_port"];
args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
args.use_ps = allArgs["use_ps"].empty() ? true : allArgs["use_ps"].as<bool>();
args.type = (MediaSourceEvent::SendRtpArgs::Type)type;
args.only_audio = allArgs["only_audio"].as<bool>();
args.recv_stream_id = allArgs["recv_stream_id"];
//tcp被动服务器等待链接超时时间
args.tcp_passive_close_delay_ms = allArgs["close_delay_ms"];
TraceL << "startSendRtpPassive, pt " << int(args.pt) << " ps " << args.use_ps << " audio " << args.only_audio;
TraceL << "startSendRtpPassive, pt " << int(args.pt) << " rtp type " << type << " audio " << args.only_audio;
src->getOwnerPoller()->async([=]() mutable {
src->startSendRtp(args, [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable {
@ -1499,18 +1533,11 @@ void installWebApi() {
api_regist("/index/api/getProxyPusherInfo", [](API_ARGS_MAP_ASYNC) {
CHECK_SECRET();
CHECK_ARGS("key");
decltype(s_proxyPusherMap.end()) it;
{
lock_guard<recursive_mutex> lck(s_proxyPusherMapMtx);
it = s_proxyPusherMap.find(allArgs["key"]);
}
if (it == s_proxyPusherMap.end()) {
auto pusher = s_pusher_proxy.find(allArgs["key"]);
if (!pusher) {
throw ApiRetException("can not find pusher", API::NotFound);
}
auto pusher = it->second;
val["data"]["status"] = pusher->getStatus();
val["data"]["liveSecs"] = pusher->getLiveSecs();
val["data"]["rePublishCount"] = pusher->getRePublishCount();
@ -1520,18 +1547,11 @@ void installWebApi() {
api_regist("/index/api/getProxyInfo", [](API_ARGS_MAP_ASYNC) {
CHECK_SECRET();
CHECK_ARGS("key");
decltype(s_proxyMap.end()) it;
{
lock_guard<recursive_mutex> lck(s_proxyMapMtx);
it = s_proxyMap.find(allArgs["key"]);
}
if (it == s_proxyMap.end()) {
auto proxy = s_player_proxy.find(allArgs["key"]);
if (!proxy) {
throw ApiRetException("can not find the proxy", API::NotFound);
}
auto proxy = it->second;
val["data"]["status"] = proxy->getStatus();
val["data"]["liveSecs"] = proxy->getLiveSecs();
val["data"]["rePullCount"] = proxy->getRePullCount();
@ -1670,7 +1690,7 @@ void installWebApi() {
//截图存在，且未过期，那么返回之
res_old_snap = true;
responseSnap(path, allArgs.getParser().getHeader(), invoker);
responseSnap(path, allArgs.parser.getHeader(), invoker);
//中断遍历
return false;
});
@ -1701,7 +1721,7 @@ void installWebApi() {
File::delete_file(new_snap);
rename(new_snap_tmp.data(), new_snap.data());
}
responseSnap(new_snap, allArgs.getParser().getHeader(), invoker, err_msg);
responseSnap(new_snap, allArgs.parser.getHeader(), invoker, err_msg);
});
});
@ -1716,7 +1736,7 @@ void installWebApi() {
#ifdef ENABLE_WEBRTC
class WebRtcArgsImp : public WebRtcArgs {
public:
WebRtcArgsImp(const HttpAllArgs<string> &args, std::string session_id)
WebRtcArgsImp(const ArgsString &args, std::string session_id)
: _args(args)
, _session_id(std::move(session_id)) {}
~WebRtcArgsImp() override = default;
@ -1734,40 +1754,26 @@ void installWebApi() {
CHECK_ARGS("app", "stream");
return StrPrinter << "rtc://" << _args["Host"] << "/" << _args["app"] << "/"
<< _args["stream"] << "?" << _args.getParser().params() + "&session=" + _session_id;
<< _args["stream"] << "?" << _args.parser.params() + "&session=" + _session_id;
}
private:
HttpAllArgs<string> _args;
ArgsString _args;
std::string _session_id;
};
api_regist("/index/api/webrtc",[](API_ARGS_STRING_ASYNC){
CHECK_ARGS("type");
auto type = allArgs["type"];
auto offer = allArgs.getArgs();
auto offer = allArgs.args;
CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
std::string host = allArgs.getParser()["Host"];
std::string localIp = host.substr(0, host.find(':'));
auto isVaildIP = [](std::string ip)-> bool {
int a,b,c,d;
return sscanf(ip.c_str(),"%d.%d.%d.%d", &a, &b, &c, &d) == 4;
};
if (!isVaildIP(localIp) || localIp=="127.0.0.1") {
localIp = "";
}
auto &session = static_cast<Session&>(sender);
auto args = std::make_shared<WebRtcArgsImp>(allArgs, sender.getIdentifier());
WebRtcPluginManager::Instance().getAnswerSdp(static_cast<Session&>(sender), type, *args, [invoker, val, offer, headerOut, localIp](const WebRtcInterface &exchanger) mutable {
//设置返回类型
headerOut["Content-Type"] = HttpFileManager::getContentType(".json");
//设置跨域
headerOut["Access-Control-Allow-Origin"] = "*";
WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, val, offer, headerOut](const WebRtcInterface &exchanger) mutable {
auto &handler = const_cast<WebRtcInterface &>(exchanger);
try {
setLocalIp(exchanger,localIp);
val["sdp"] = exchangeSdp(exchanger, offer);
val["sdp"] = handler.getAnswerSdp(offer);
val["id"] = exchanger.getIdentifier();
val["type"] = "answer";
invoker(200, headerOut, val.toStyledString());
@ -1781,26 +1787,24 @@ void installWebApi() {
static constexpr char delete_webrtc_url [] = "/index/api/delete_webrtc";
static auto whip_whep_func = [](const char *type, API_ARGS_STRING_ASYNC) {
auto offer = allArgs.getArgs();
auto offer = allArgs.args;
CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
auto &session = static_cast<Session&>(sender);
auto location = std::string("http") + (session.overSsl() ? "s" : "") + "://" + allArgs["host"] + delete_webrtc_url;
auto location = std::string(session.overSsl() ? "https://" : "http://") + allArgs["host"] + delete_webrtc_url;
auto args = std::make_shared<WebRtcArgsImp>(allArgs, sender.getIdentifier());
WebRtcPluginManager::Instance().getAnswerSdp(session, type, *args,
[invoker, offer, headerOut, location](const WebRtcInterface &exchanger) mutable {
// 设置跨域
headerOut["Access-Control-Allow-Origin"] = "*";
try {
// 设置返回类型
headerOut["Content-Type"] = "application/sdp";
headerOut["Location"] = location + "?id=" + exchanger.getIdentifier() + "&token=" + exchanger.deleteRandStr();
invoker(201, headerOut, exchangeSdp(exchanger, offer));
} catch (std::exception &ex) {
headerOut["Content-Type"] = "text/plain";
invoker(406, headerOut, ex.what());
}
});
WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, offer, headerOut, location](const WebRtcInterface &exchanger) mutable {
auto &handler = const_cast<WebRtcInterface &>(exchanger);
try {
// 设置返回类型
headerOut["Content-Type"] = "application/sdp";
headerOut["Location"] = location + "?id=" + exchanger.getIdentifier() + "&token=" + exchanger.deleteRandStr();
invoker(201, headerOut, handler.getAnswerSdp(offer));
} catch (std::exception &ex) {
headerOut["Content-Type"] = "text/plain";
invoker(406, headerOut, ex.what());
}
});
};
api_regist("/index/api/whip", [](API_ARGS_STRING_ASYNC) { whip_whep_func("push", API_ARGS_VALUE, invoker); });
@ -1808,7 +1812,7 @@ void installWebApi() {
api_regist(delete_webrtc_url, [](API_ARGS_MAP_ASYNC) {
CHECK_ARGS("id", "token");
CHECK(allArgs.getParser().method() == "DELETE", "http method is not DELETE: " + allArgs.getParser().method());
CHECK(allArgs.parser.method() == "DELETE", "http method is not DELETE: " + allArgs.parser.method());
auto obj = WebRtcTransportManager::Instance().getItem(allArgs["id"]);
if (!obj) {
invoker(404, headerOut, "id not found");
@ -1858,7 +1862,7 @@ void installWebApi() {
std::set<std::string> ret;
auto vec = toolkit::split(str, ";");
for (auto &item : vec) {
auto root = File::absolutePath(item, "", true);
auto root = File::absolutePath("", item, true);
ret.emplace(std::move(root));
}
return ret;
@ -1894,44 +1898,50 @@ void installWebApi() {
if (!save_name.empty()) {
res_header.emplace("Content-Disposition", "attachment;filename=\"" + save_name + "\"");
}
invoker.responseFile(allArgs.getParser().getHeader(), res_header, allArgs["file_path"]);
invoker.responseFile(allArgs.parser.getHeader(), res_header, allArgs["file_path"]);
}
};
bool flag = NOTICE_EMIT(BroadcastHttpAccessArgs, Broadcast::kBroadcastHttpAccess, allArgs.getParser(), file_path, false, file_invoker, sender);
bool flag = NOTICE_EMIT(BroadcastHttpAccessArgs, Broadcast::kBroadcastHttpAccess, allArgs.parser, file_path, false, file_invoker, sender);
if (!flag) {
// 文件下载鉴权事件无人监听，不允许下载
invoker(401, StrCaseMap {}, "None http access event listener");
}
});
#if defined(ENABLE_X264) && defined(ENABLE_FFMPEG)
VideoStackManager::Instance().loadBgImg("novideo.yuv");
NoticeCenter::Instance().addListener(nullptr, Broadcast::kBroadcastStreamNoneReader, [](BroadcastStreamNoneReaderArgs) {
auto id = sender.getMediaTuple().stream;
VideoStackManager::Instance().stopVideoStack(id);
});
api_regist("/index/api/stack/start", [](API_ARGS_JSON_ASYNC) {
CHECK_SECRET();
auto ret = VideoStackManager::Instance().startVideoStack(allArgs.args());
val["code"] = ret;
val["msg"] = ret ? "failed" : "success";
invoker(200, headerOut, val.toStyledString());
});
api_regist("/index/api/stack/stop", [](API_ARGS_MAP_ASYNC) {
CHECK_SECRET();
CHECK_ARGS("id");
auto ret = VideoStackManager::Instance().stopVideoStack(allArgs["id"]);
val["code"] = ret;
val["msg"] = ret ? "failed" : "success";
invoker(200, headerOut, val.toStyledString());
});
#endif
}
void unInstallWebApi(){
{
lock_guard<recursive_mutex> lck(s_proxyMapMtx);
auto proxyMap(std::move(s_proxyMap));
proxyMap.clear();
}
{
lock_guard<recursive_mutex> lck(s_ffmpegMapMtx);
auto ffmpegMap(std::move(s_ffmpegMap));
ffmpegMap.clear();
}
{
lock_guard<recursive_mutex> lck(s_proxyPusherMapMtx);
auto proxyPusherMap(std::move(s_proxyPusherMap));
proxyPusherMap.clear();
}
{
s_player_proxy.clear();
s_ffmpeg_src.clear();
s_pusher_proxy.clear();
#if defined(ENABLE_RTPPROXY)
RtpSelector::Instance().clear();
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
auto rtpServerMap(std::move(s_rtpServerMap));
rtpServerMap.clear();
s_rtp_server.clear();
#endif
}
NoticeCenter::Instance().delListener(&web_api_tag);
}

81
server/WebApi.h
View File

@ -115,72 +115,41 @@ std::string getValue(const mediakit::Parser &parser, Args &args, const First &fi
template<typename Args>
class HttpAllArgs {
mediakit::Parser* _parser = nullptr;
Args* _args = nullptr;
public:
HttpAllArgs(const mediakit::Parser &parser, Args &args) {
_get_args = [&args]() {
return (void *) &args;
};
_get_parser = [&parser]() -> const mediakit::Parser & {
return parser;
};
_get_value = [](HttpAllArgs &that, const std::string &key) {
return getValue(that.getParser(), that.getArgs(), key);
};
_clone = [&](HttpAllArgs &that) {
that._get_args = [args]() {
return (void *) &args;
};
that._get_parser = [parser]() -> const mediakit::Parser & {
return parser;
};
that._get_value = [](HttpAllArgs &that, const std::string &key) {
return getValue(that.getParser(), that.getArgs(), key);
};
that._cache_able = true;
};
}
const mediakit::Parser& parser;
Args& args;
HttpAllArgs(const HttpAllArgs &that) {
if (that._cache_able) {
_get_args = that._get_args;
_get_parser = that._get_parser;
_get_value = that._get_value;
_cache_able = true;
} else {
that._clone(*this);
HttpAllArgs(const mediakit::Parser &p, Args &a): parser(p), args(a) {}
HttpAllArgs(const HttpAllArgs &that): _parser(new mediakit::Parser(that.parser)),
_args(new Args(that.args)),
parser(*_parser), args(*_args) {}
~HttpAllArgs() {
if (_parser) {
delete _parser;
}
if (_args) {
delete _args;
}
}
template<typename Key>
toolkit::variant operator[](const Key &key) const {
return (toolkit::variant)_get_value(*(HttpAllArgs*)this, key);
return (toolkit::variant)getValue(parser, args, key);
}
const mediakit::Parser &getParser() const {
return _get_parser();
}
Args &getArgs() {
return *((Args *) _get_args());
}
const Args &getArgs() const {
return *((Args *) _get_args());
}
private:
bool _cache_able = false;
std::function<void *() > _get_args;
std::function<const mediakit::Parser &() > _get_parser;
std::function<std::string(HttpAllArgs &that, const std::string &key)> _get_value;
std::function<void(HttpAllArgs &that) > _clone;
};
#define API_ARGS_MAP toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<ApiArgsType> &allArgs, Json::Value &val
using ArgsMap = HttpAllArgs<ApiArgsType>;
using ArgsJson = HttpAllArgs<Json::Value>;
using ArgsString = HttpAllArgs<std::string>;
#define API_ARGS_MAP toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsMap &allArgs, Json::Value &val
#define API_ARGS_MAP_ASYNC API_ARGS_MAP, const mediakit::HttpSession::HttpResponseInvoker &invoker
#define API_ARGS_JSON toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<Json::Value> &allArgs, Json::Value &val
#define API_ARGS_JSON toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsJson &allArgs, Json::Value &val
#define API_ARGS_JSON_ASYNC API_ARGS_JSON, const mediakit::HttpSession::HttpResponseInvoker &invoker
#define API_ARGS_STRING toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<std::string> &allArgs, Json::Value &val
#define API_ARGS_STRING toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsString &allArgs, Json::Value &val
#define API_ARGS_STRING_ASYNC API_ARGS_STRING, const mediakit::HttpSession::HttpResponseInvoker &invoker
#define API_ARGS_VALUE sender, headerOut, allArgs, val
@ -233,9 +202,7 @@ void installWebApi();
void unInstallWebApi();
#if defined(ENABLE_RTPPROXY)
uint16_t openRtpServer(uint16_t local_port, const std::string &stream_id, int tcp_mode, const std::string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio, bool multiplex=false);
void connectRtpServer(const std::string &stream_id, const std::string &dst_url, uint16_t dst_port, const std::function<void(const toolkit::SockException &ex)> &cb);
bool closeRtpServer(const std::string &stream_id);
uint16_t openRtpServer(uint16_t local_port, const std::string &stream_id, int tcp_mode, const std::string &local_ip, bool re_use_port, uint32_t ssrc, int only_track, bool multiplex=false);
#endif
Json::Value makeMediaSourceJson(mediakit::MediaSource &media);

6
server/main.cpp
View File

@ -38,7 +38,7 @@
#endif
#if defined(ENABLE_VERSION)
#include "version.h"
#include "ZLMVersion.h"
#endif
#if !defined(_WIN32)
@ -392,8 +392,8 @@ int start_main(int argc,char *argv[]) {
#endif//defined(ENABLE_WEBRTC)
#if defined(ENABLE_SRT)
// srt udp服务器
if(srtPort) { srtSrv->start<SRT::SrtSession>(srtPort); }
// srt udp服务器
if (srtPort) { srtSrv->start<SRT::SrtSession>(srtPort); }
#endif//defined(ENABLE_SRT)
} catch (std::exception &ex) {

4
src/Common/MediaSink.cpp
View File

@ -133,7 +133,7 @@ void MediaSink::checkTrackIfReady() {
}
GET_CONFIG(uint32_t, kMaxAddTrackMS, General::kWaitAddTrackMS);
if (_track_map.size() == 1 && _ticker.elapsedTime() > kMaxAddTrackMS) {
if (_track_map.size() == 1 && (_ticker.elapsedTime() > kMaxAddTrackMS || !_enable_audio)) {
// 如果只有一个Track，那么在该Track添加后，我们最多还等待若干时间(可能后面还会添加Track)
emitAllTrackReady();
return;
@ -187,6 +187,8 @@ void MediaSink::emitAllTrackReady() {
pr.second.for_each([&](const Frame::Ptr &frame) { MediaSink::inputFrame(frame); });
}
_frame_unread.clear();
} else {
throw toolkit::SockException(toolkit::Err_shutdown, "no vaild track data");
}
}

23
src/Common/MediaSource.h
View File

@ -92,10 +92,11 @@ public:
class SendRtpArgs {
public:
enum Type { kRtpRAW = 0, kRtpPS = 1, kRtpTS = 2 };
// 是否采用udp方式发送rtp
bool is_udp = true;
// rtp采用ps还是es方式
bool use_ps = true;
// rtp类型
Type type = kRtpPS;
//发送es流时指定是否只发送纯音频流
bool only_audio = false;
//tcp被动方式
@ -135,6 +136,15 @@ private:
toolkit::Timer::Ptr _async_close_timer;
};
template <typename MAP, typename KEY, typename TYPE>
static void getArgsValue(const MAP &allArgs, const KEY &key, TYPE &value) {
auto val = ((MAP &)allArgs)[key];
if (!val.empty()) {
value = (TYPE)val;
}
}
class ProtocolOption {
public:
ProtocolOption();
@ -242,15 +252,6 @@ public:
GET_OPT_VALUE(stream_replace);
GET_OPT_VALUE(max_track);
}
private:
template <typename MAP, typename KEY, typename TYPE>
static void getArgsValue(const MAP &allArgs, const KEY &key, TYPE &value) {
auto val = ((MAP &)allArgs)[key];
if (!val.empty()) {
value = (TYPE)val;
}
}
};
//该对象用于拦截感兴趣的MediaSourceEvent事件

14
src/Common/MultiMediaSourceMuxer.cpp
View File

@ -44,6 +44,7 @@ public:
}
void resetTimer(const EventPoller::Ptr &poller) {
std::lock_guard<std::recursive_mutex> lck(_mtx);
std::weak_ptr<FramePacedSender> weak_self = shared_from_this();
_timer = std::make_shared<Timer>(_paced_sender_ms / 1000.0f, [weak_self]() {
if (auto strong_self = weak_self.lock()) {
@ -55,6 +56,7 @@ public:
}
bool inputFrame(const Frame::Ptr &frame) override {
std::lock_guard<std::recursive_mutex> lck(_mtx);
if (!_timer) {
setCurrentStamp(frame->dts());
resetTimer(EventPoller::getCurrentPoller());
@ -66,6 +68,7 @@ public:
private:
void onTick() {
std::lock_guard<std::recursive_mutex> lck(_mtx);
auto dst = _cache.empty() ? 0 : _cache.back().first;
while (!_cache.empty()) {
auto &front = _cache.front();
@ -110,6 +113,7 @@ private:
OnFrame _cb;
Ticker _ticker;
Timer::Ptr _timer;
std::recursive_mutex _mtx;
std::list<std::pair<uint64_t, Frame::Ptr>> _cache;
};
@ -593,15 +597,17 @@ void MultiMediaSourceMuxer::resetTracks() {
}
}
bool MultiMediaSourceMuxer::onTrackFrame(const Frame::Ptr &frame) {
bool MultiMediaSourceMuxer::onTrackFrame(const Frame::Ptr &frame_in) {
auto frame = frame_in;
if (_option.modify_stamp != ProtocolOption::kModifyStampOff) {
// 时间戳不采用原始的绝对时间戳
const_cast<Frame::Ptr&>(frame) = std::make_shared<FrameStamp>(frame, _stamps[frame->getIndex()], _option.modify_stamp);
frame = std::make_shared<FrameStamp>(frame, _stamps[frame->getIndex()], _option.modify_stamp);
}
return _paced_sender ? _paced_sender->inputFrame(frame) : onTrackFrame_l(frame);
}
bool MultiMediaSourceMuxer::onTrackFrame_l(const Frame::Ptr &frame) {
bool MultiMediaSourceMuxer::onTrackFrame_l(const Frame::Ptr &frame_in) {
auto frame = frame_in;
bool ret = false;
if (_rtmp) {
ret = _rtmp->inputFrame(frame) ? true : ret;
@ -629,7 +635,7 @@ bool MultiMediaSourceMuxer::onTrackFrame_l(const Frame::Ptr &frame) {
}
if (_ring) {
// 此场景由于直接转发，可能存在切换线程引起的数据被缓存在管道，所以需要CacheAbleFrame
const_cast<Frame::Ptr &>(frame) = Frame::getCacheAbleFrame(frame);
frame = Frame::getCacheAbleFrame(frame);
if (frame->getTrackType() == TrackVideo) {
// 视频时，遇到第一帧配置帧或关键帧则标记为gop开始处
auto video_key_pos = frame->keyFrame() || frame->configFrame();

4
src/Common/Parser.cpp
View File

@ -294,8 +294,8 @@ void RtspUrl::setup(bool is_ssl, const string &url, const string &user, const st
splitUrl(ip, ip, port);
_url = std::move(url);
_user = strCoding::UrlDecodeComponent(std::move(user));
_passwd = strCoding::UrlDecodeComponent(std::move(passwd));
_user = strCoding::UrlDecodeComponent(user);
_passwd = strCoding::UrlDecodeComponent(passwd);
_host = std::move(ip);
_port = port;
_is_ssl = is_ssl;

2
src/Common/Parser.h
View File

@ -30,7 +30,7 @@ struct StrCaseCompare {
class StrCaseMap : public std::multimap<std::string, std::string, StrCaseCompare> {
public:
using Super = multimap<std::string, std::string, StrCaseCompare>;
using Super = std::multimap<std::string, std::string, StrCaseCompare>;
std::string &operator[](const std::string &k) {
auto it = find(k);

12
src/Common/config.cpp
View File

@ -58,6 +58,12 @@ const string kBroadcastStreamNoneReader = "kBroadcastStreamNoneReader";
const string kBroadcastHttpBeforeAccess = "kBroadcastHttpBeforeAccess";
const string kBroadcastSendRtpStopped = "kBroadcastSendRtpStopped";
const string kBroadcastRtpServerTimeout = "kBroadcastRtpServerTimeout";
const string kBroadcastRtcSctpConnecting = "kBroadcastRtcSctpConnecting";
const string kBroadcastRtcSctpConnected = "kBroadcastRtcSctpConnected";
const string kBroadcastRtcSctpFailed = "kBroadcastRtcSctpFailed";
const string kBroadcastRtcSctpClosed = "kBroadcastRtcSctpClosed";
const string kBroadcastRtcSctpSend = "kBroadcastRtcSctpSend";
const string kBroadcastRtcSctpReceived = "kBroadcastRtcSctpReceived";
} // namespace Broadcast
@ -291,6 +297,7 @@ const string kSampleMS = RECORD_FIELD "sampleMS";
const string kFileBufSize = RECORD_FIELD "fileBufSize";
const string kFastStart = RECORD_FIELD "fastStart";
const string kFileRepeat = RECORD_FIELD "fileRepeat";
const string kEnableFmp4 = RECORD_FIELD "enableFmp4";
static onceToken token([]() {
mINI::Instance()[kAppName] = "record";
@ -298,6 +305,7 @@ static onceToken token([]() {
mINI::Instance()[kFileBufSize] = 64 * 1024;
mINI::Instance()[kFastStart] = false;
mINI::Instance()[kFileRepeat] = false;
mINI::Instance()[kEnableFmp4] = false;
});
} // namespace Record
@ -338,6 +346,8 @@ const string kH265PT = RTP_PROXY_FIELD "h265_pt";
const string kPSPT = RTP_PROXY_FIELD "ps_pt";
const string kOpusPT = RTP_PROXY_FIELD "opus_pt";
const string kGopCache = RTP_PROXY_FIELD "gop_cache";
const string kRtpG711DurMs = RTP_PROXY_FIELD "rtp_g711_dur_ms";
const string kUdpRecvSocketBuffer = RTP_PROXY_FIELD "udp_recv_socket_buffer";
static onceToken token([]() {
mINI::Instance()[kDumpDir] = "";
@ -348,6 +358,8 @@ static onceToken token([]() {
mINI::Instance()[kPSPT] = 96;
mINI::Instance()[kOpusPT] = 100;
mINI::Instance()[kGopCache] = 1;
mINI::Instance()[kRtpG711DurMs] = 100;
mINI::Instance()[kUdpRecvSocketBuffer] = 4 * 1024 * 1024;
});
} // namespace RtpProxy

22
src/Common/config.h
View File

@ -109,6 +109,21 @@ extern const std::string kBroadcastReloadConfig;
extern const std::string kBroadcastRtpServerTimeout;
#define BroadcastRtpServerTimeoutArgs uint16_t &local_port, const string &stream_id,int &tcp_mode, bool &re_use_port, uint32_t &ssrc
// rtc transport sctp 连接状态
extern const std::string kBroadcastRtcSctpConnecting;
extern const std::string kBroadcastRtcSctpConnected;
extern const std::string kBroadcastRtcSctpFailed;
extern const std::string kBroadcastRtcSctpClosed;
#define BroadcastRtcSctpConnectArgs WebRtcTransport& sender
// rtc transport sctp 发送数据
extern const std::string kBroadcastRtcSctpSend;
#define BroadcastRtcSctpSendArgs WebRtcTransport& sender, const uint8_t *&data, size_t& len
// rtc transport sctp 接收数据
extern const std::string kBroadcastRtcSctpReceived;
#define BroadcastRtcSctpReceivedArgs WebRtcTransport& sender, uint16_t &streamId, uint32_t &ppid, const uint8_t *&msg, size_t &len
#define ReloadConfigTag ((void *)(0xFF))
#define RELOAD_KEY(arg, key) \
do { \
@ -339,6 +354,8 @@ extern const std::string kFileBufSize;
extern const std::string kFastStart;
// mp4文件是否重头循环读取
extern const std::string kFileRepeat;
// mp4录制文件是否采用fmp4格式
extern const std::string kEnableFmp4;
} // namespace Record
////////////HLS相关配置///////////
@ -382,6 +399,11 @@ extern const std::string kPSPT;
extern const std::string kOpusPT;
// RtpSender相关功能是否提前开启gop缓存优化级联秒开体验，默认开启
extern const std::string kGopCache;
//国标发送g711 rtp 打包时，每个包的语音时长是多少，默认是100 ms，范围为20~180ms (gb28181-2016，c.2.4规定)，
//最好为20 的倍数，程序自动向20的倍数取整
extern const std::string kRtpG711DurMs;
// udp recv socket buffer size
extern const std::string kUdpRecvSocketBuffer;
} // namespace RtpProxy
/**

2
src/Common/macros.cpp
View File

@ -14,7 +14,7 @@
using namespace toolkit;
#if defined(ENABLE_VERSION)
#include "version.h"
#include "ZLMVersion.h"
#endif
extern "C" {

10
src/Common/macros.h
View File

@ -36,6 +36,16 @@
#define CHECK(exp, ...) ::mediakit::Assert_ThrowCpp(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__, ##__VA_ARGS__)
#endif // CHECK
#ifndef CHECK_RET
#define CHECK_RET(...) \
try { \
CHECK(__VA_ARGS__); \
} catch (AssertFailedException & ex) { \
WarnL << ex.what(); \
return; \
}
#endif
#ifndef MAX
#define MAX(a, b) ((a) > (b) ? (a) : (b))
#endif // MAX

67
src/Common/strCoding.cpp
View File

@ -53,22 +53,6 @@ char HexStrToBin(const char *str) {
return (high << 4) | low;
}
string strCoding::UrlEncode(const string &str) {
string out;
size_t len = str.size();
for (size_t i = 0; i < len; ++i) {
char ch = str[i];
if (isalnum((uint8_t) ch)) {
out.push_back(ch);
} else {
char buf[4];
sprintf(buf, "%%%X%X", (uint8_t) ch >> 4, (uint8_t) ch & 0x0F);
out.append(buf);
}
}
return out;
}
string strCoding::UrlEncodePath(const string &str) {
const char *dont_escape = "!#&'*+:=?@/._-$,;~()";
string out;
@ -79,7 +63,7 @@ string strCoding::UrlEncodePath(const string &str) {
out.push_back(ch);
} else {
char buf[4];
sprintf(buf, "%%%X%X", (uint8_t) ch >> 4, (uint8_t) ch & 0x0F);
snprintf(buf, 4, "%%%X%X", (uint8_t) ch >> 4, (uint8_t) ch & 0x0F);
out.append(buf);
}
}
@ -96,39 +80,13 @@ string strCoding::UrlEncodeComponent(const string &str) {
out.push_back(ch);
} else {
char buf[4];
sprintf(buf, "%%%X%X", (uint8_t) ch >> 4, (uint8_t) ch & 0x0F);
snprintf(buf, 4, "%%%X%X", (uint8_t) ch >> 4, (uint8_t) ch & 0x0F);
out.append(buf);
}
}
return out;
}
string strCoding::UrlDecode(const string &str) {
string output;
size_t i = 0, len = str.length();
while (i < len) {
if (str[i] == '%') {
if (i + 3 > len) {
// %后面必须还有两个字节才会反转义
output.append(str, i, len - i);
break;
}
char ch = HexStrToBin(&(str[i + 1]));
if (ch == -1) {
// %后面两个字节不是16进制字符串，转义失败；直接拼接3个原始字符
output.append(str, i, 3);
} else {
output += ch;
}
i += 3;
} else {
output += str[i];
++i;
}
}
return output;
}
string strCoding::UrlDecodePath(const string &str) {
const char *dont_unescape = "#$&+,/:;=?@";
string output;
@ -185,27 +143,6 @@ std::string strCoding::UrlDecodeComponent(const std::string &str) {
return output;
}
#if 0
#include "Util/onceToken.h"
static toolkit::onceToken token([]() {
auto str0 = strCoding::UrlDecode(
"rtsp%3A%2F%2Fadmin%3AJm13317934%25jm%40111.47.84.69%3A554%2FStreaming%2FChannels%2F101%3Ftransportmode%3Dunicast%26amp%3Bprofile%3DProfile_1");
auto str1 = strCoding::UrlDecode("%j1"); // 测试%后面两个字节不是16进制字符串
auto str2 = strCoding::UrlDecode("%a"); // 测试%后面字节数不够
auto str3 = strCoding::UrlDecode("%"); // 测试只有%
auto str4 = strCoding::UrlDecode("%%%"); // 测试多个%
auto str5 = strCoding::UrlDecode("%%%%40"); // 测试多个非法%后恢复正常解析
auto str6 = strCoding::UrlDecode("Jm13317934%jm"); // 测试多个非法%后恢复正常解析
cout << str0 << endl;
cout << str1 << endl;
cout << str2 << endl;
cout << str3 << endl;
cout << str4 << endl;
cout << str5 << endl;
cout << str6 << endl;
});
#endif
///////////////////////////////windows专用///////////////////////////////////
#if defined(_WIN32)
void UnicodeToGB2312(char* pOut, wchar_t uData)

2
src/Common/strCoding.h
View File

@ -18,10 +18,8 @@ namespace mediakit {
class strCoding {
public:
[[deprecated]] static std::string UrlEncode(const std::string &str); //url utf8编码, deprecated
static std::string UrlEncodePath(const std::string &str); //url路径 utf8编码
static std::string UrlEncodeComponent(const std::string &str); // url参数 utf8编码
[[deprecated]] static std::string UrlDecode(const std::string &str); //url utf8解码, deprecated
static std::string UrlDecodePath(const std::string &str); //url路径 utf8解码
static std::string UrlDecodeComponent(const std::string &str); // url参数 utf8解码
#if defined(_WIN32)

12
src/Http/HttpRequestSplitter.cpp
View File

@ -65,18 +65,18 @@ void HttpRequestSplitter::input(const char *data,size_t len) {
_content_len = onRecvHeader(header_ptr, header_size);
}
if(_remain_data_size <= 0){
//没有剩余数据，清空缓存
_remain_data.clear();
return;
}
/*
* 恢复末尾字节
* 移动到这来，目的是防止HttpRequestSplitter::reset()导致内存失效
*/
tail_ref = tail_tmp;
if(_remain_data_size <= 0){
//没有剩余数据，清空缓存
_remain_data.clear();
return;
}
if(_content_len == 0){
//尚未找到http头，缓存定位到剩余数据部分
_remain_data.assign(ptr,_remain_data_size);

12
src/Http/HttpSession.cpp
View File

@ -683,18 +683,6 @@ void HttpSession::sendResponse(int code,
AsyncSender::onSocketFlushed(data);
}
string HttpSession::urlDecode(const string &str) {
auto ret = strCoding::UrlDecode(str);
#ifdef _WIN32
GET_CONFIG(string, charSet, Http::kCharSet);
bool isGb2312 = !strcasecmp(charSet.data(), "gb2312");
if (isGb2312) {
ret = strCoding::UTF8ToGB2312(ret);
}
#endif // _WIN32
return ret;
}
string HttpSession::urlDecodePath(const string &str) {
auto ret = strCoding::UrlDecodePath(str);
#ifdef _WIN32

1
src/Http/HttpSession.h
View File

@ -44,7 +44,6 @@ public:
void onRecv(const toolkit::Buffer::Ptr &) override;
void onError(const toolkit::SockException &err) override;
void onManager() override;
[[deprecated]] static std::string urlDecode(const std::string &str);
static std::string urlDecodePath(const std::string &str);
static std::string urlDecodeComponent(const std::string &str);
void setTimeoutSec(size_t second);

2
src/Record/HlsMakerImp.cpp
View File

@ -72,7 +72,7 @@ void HlsMakerImp::clearCache(bool immediately, bool eof) {
std::list<std::string> lst;
lst.emplace_back(_path_hls);
lst.emplace_back(_path_hls_delay);
if (!_path_init.empty()) {
if (!_path_init.empty() && eof) {
lst.emplace_back(_path_init);
}
for (auto &pr : _segment_file_paths) {

3
src/Record/MP4Muxer.cpp
View File

@ -31,7 +31,8 @@ void MP4Muxer::openMP4(const string &file) {
MP4FileIO::Writer MP4Muxer::createWriter() {
GET_CONFIG(bool, mp4FastStart, Record::kFastStart);
return _mp4_file->createWriter(mp4FastStart ? MOV_FLAG_FASTSTART : 0, false);
GET_CONFIG(bool, recordEnableFmp4, Record::kEnableFmp4);
return _mp4_file->createWriter(mp4FastStart ? MOV_FLAG_FASTSTART : 0, recordEnableFmp4);
}
void MP4Muxer::closeMP4() {

10
src/Record/MP4Recorder.cpp
View File

@ -117,11 +117,13 @@ bool MP4Recorder::inputFrame(const Frame::Ptr &frame) {
if (!(_have_video && frame->getTrackType() == TrackAudio)) {
//如果有视频且输入的是音频，那么应该忽略切片逻辑
if (_last_dts == 0 || _last_dts > frame->dts()) {
//极少情况下dts时间戳可能回退
_last_dts = frame->dts();
//b帧情况下dts时间戳可能回退
_last_dts = MAX(frame->dts(), _last_dts);
}
auto duration = 5; // 默认至少一帧5ms
if (frame->dts() > 0 && frame->dts() > _last_dts) {
duration = MAX(duration, frame->dts() - _last_dts);
}
auto duration = frame->dts() - _last_dts;
if (!_muxer || ((duration > _max_second * 1000) && (!_have_video || (_have_video && frame->keyFrame())))) {
//成立条件
// 1、_muxer为空

8
src/Rtmp/RtmpMediaSourceMuxer.h
View File

@ -29,7 +29,13 @@ public:
getRtmpRing()->setDelegate(_media_src);
}
~RtmpMediaSourceMuxer() override { RtmpMuxer::flush(); }
~RtmpMediaSourceMuxer() override {
try {
RtmpMuxer::flush();
} catch (std::exception &ex) {
WarnL << ex.what();
}
}
void setListener(const std::weak_ptr<MediaSourceEvent> &listener){
setDelegate(listener);

12
src/Rtp/PSEncoder.cpp
View File

@ -19,9 +19,17 @@ using namespace toolkit;
namespace mediakit{
PSEncoderImp::PSEncoderImp(uint32_t ssrc, uint8_t payload_type) : MpegMuxer(true) {
GET_CONFIG(uint32_t,video_mtu,Rtp::kVideoMtuSize);
PSEncoderImp::PSEncoderImp(uint32_t ssrc, uint8_t payload_type, bool ps_or_ts) : MpegMuxer(ps_or_ts) {
GET_CONFIG(uint32_t, s_video_mtu, Rtp::kVideoMtuSize);
_rtp_encoder = std::make_shared<CommonRtpEncoder>();
auto video_mtu = s_video_mtu;
if (!ps_or_ts) {
// 确保ts rtp负载部分长度是188的倍数
video_mtu = RtpPacket::kRtpHeaderSize + (s_video_mtu - (s_video_mtu % 188));
if (video_mtu > s_video_mtu) {
video_mtu -= 188;
}
}
_rtp_encoder->setRtpInfo(ssrc, video_mtu, 90000, payload_type);
auto ring = std::make_shared<RtpRing::RingType>();
ring->setDelegate(std::make_shared<RingDelegateHelper>([this](RtpPacket::Ptr rtp, bool is_key) { onRTP(std::move(rtp), is_key); }));

14
src/Rtp/PSEncoder.h
View File

@ -16,11 +16,19 @@
#include "Record/MPEG.h"
#include "Common/MediaSink.h"
namespace mediakit{
namespace mediakit {
class CommonRtpEncoder;
class PSEncoderImp : public MpegMuxer{
class PSEncoderImp : public MpegMuxer {
public:
PSEncoderImp(uint32_t ssrc, uint8_t payload_type = 96);
/**
* 创建psh或ts rtp编码器
* @param ssrc rtp的ssrc
* @param payload_type rtp的pt
* @param ps_or_ts true: ps, false: ts
*/
PSEncoderImp(uint32_t ssrc, uint8_t payload_type = 96, bool ps_or_ts = true);
~PSEncoderImp() override;
protected:

6
src/Rtp/RawEncoder.cpp
View File

@ -34,6 +34,12 @@ bool RawEncoderImp::addTrack(const Track::Ptr &track) {
auto ring = std::make_shared<RtpRing::RingType>();
ring->setDelegate(std::make_shared<RingDelegateHelper>([this](RtpPacket::Ptr rtp, bool is_key) { onRTP(std::move(rtp), true); }));
_rtp_encoder->setRtpRing(std::move(ring));
if (track->getCodecId() == CodecG711A || track->getCodecId() == CodecG711U) {
GET_CONFIG(uint32_t, dur_ms, RtpProxy::kRtpG711DurMs);
Any param;
param.set<uint32_t>(dur_ms);
_rtp_encoder->setOpt(RtpCodec::RTP_ENCODER_PKT_DUR_MS, param);
}
return true;
}

7
src/Rtp/RtpCache.h
View File

@ -40,7 +40,9 @@ private:
class RtpCachePS : public RtpCache, public PSEncoderImp {
public:
RtpCachePS(onFlushed cb, uint32_t ssrc, uint8_t payload_type = 96) : RtpCache(std::move(cb)), PSEncoderImp(ssrc, payload_type) {};
RtpCachePS(onFlushed cb, uint32_t ssrc, uint8_t payload_type = 96, bool ps_or_ts = true) :
RtpCache(std::move(cb)), PSEncoderImp(ssrc, ps_or_ts ? payload_type : Rtsp::PT_MP2T, ps_or_ts) {};
void flush() override;
protected:
@ -56,6 +58,7 @@ protected:
void onRTP(toolkit::Buffer::Ptr rtp, bool is_key = false) override;
};
}//namespace mediakit
} //namespace mediakit
#endif//ENABLE_RTPPROXY
#endif //ZLMEDIAKIT_RTPCACHE_H

10
src/Rtp/RtpProcess.cpp
View File

@ -199,8 +199,8 @@ void RtpProcess::setStopCheckRtp(bool is_check){
}
}
void RtpProcess::setOnlyAudio(bool only_audio){
_only_audio = only_audio;
void RtpProcess::setOnlyTrack(OnlyTrack only_track) {
_only_track = only_track;
}
void RtpProcess::onDetach() {
@ -259,8 +259,10 @@ void RtpProcess::emitOnPublish() {
if (!option.stream_replace.empty()) {
RtpSelector::Instance().addStreamReplace(strong_self->_media_info.stream, option.stream_replace);
}
if (strong_self->_only_audio) {
strong_self->_muxer->setOnlyAudio();
switch (strong_self->_only_track) {
case kOnlyAudio: strong_self->_muxer->setOnlyAudio(); break;
case kOnlyVideo: strong_self->_muxer->enableAudio(false); break;
default: break;
}
strong_self->_muxer->setMediaListener(strong_self);
strong_self->doCachedFunc();

7
src/Rtp/RtpProcess.h
View File

@ -24,6 +24,7 @@ public:
friend class RtpProcessHelper;
RtpProcess(const std::string &stream_id);
~RtpProcess();
enum OnlyTrack { kAll = 0, kOnlyAudio = 1, kOnlyVideo = 2 };
/**
* 输入rtp
@ -58,10 +59,10 @@ public:
void setStopCheckRtp(bool is_check=false);
/**
* 设置为单track，单音频时可以加快媒体注册速度
* 设置为单track，单音频/单视频时可以加快媒体注册速度
* 请在inputRtp前调用此方法，否则可能会是空操作
*/
void setOnlyAudio(bool only_audio);
void setOnlyTrack(OnlyTrack only_track);
/**
* flush输出缓存
@ -93,7 +94,7 @@ private:
void doCachedFunc();
private:
bool _only_audio = false;
OnlyTrack _only_track = kAll;
std::string _auth_err;
uint64_t _dts = 0;
uint64_t _total_bytes = 0;

9
src/Rtp/RtpSender.cpp
View File

@ -40,10 +40,11 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
if (!_interface) {
//重连时不重新创建对象
auto lam = [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); };
if (args.use_ps) {
_interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt);
} else {
_interface = std::make_shared<RtpCacheRaw>(lam, atoi(args.ssrc.data()), args.pt, args.only_audio);
switch (args.type) {
case MediaSourceEvent::SendRtpArgs::kRtpPS: _interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt, true); break;
case MediaSourceEvent::SendRtpArgs::kRtpTS: _interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt, false); break;
case MediaSourceEvent::SendRtpArgs::kRtpRAW: _interface = std::make_shared<RtpCacheRaw>(lam, atoi(args.ssrc.data()), args.pt, args.only_audio); break;
default: CHECK(0, "invalid rtp type:" + to_string(args.type)); break;
}
}

20
src/Rtp/RtpServer.cpp
View File

@ -42,12 +42,12 @@ public:
}
}
void setRtpServerInfo(uint16_t local_port,RtpServer::TcpMode mode,bool re_use_port,uint32_t ssrc, bool only_audio) {
void setRtpServerInfo(uint16_t local_port, RtpServer::TcpMode mode, bool re_use_port, uint32_t ssrc, int only_track) {
_local_port = local_port;
_tcp_mode = mode;
_re_use_port = re_use_port;
_ssrc = ssrc;
_only_audio = only_audio;
_only_track = only_track;
}
void setOnDetach(function<void()> cb) {
@ -61,7 +61,7 @@ public:
void onRecvRtp(const Socket::Ptr &sock, const Buffer::Ptr &buf, struct sockaddr *addr) {
if (!_process) {
_process = RtpSelector::Instance().getProcess(_stream_id, true);
_process->setOnlyAudio(_only_audio);
_process->setOnlyTrack((RtpProcess::OnlyTrack)_only_track);
_process->setOnDetach(std::move(_on_detach));
cancelDelayTask();
}
@ -142,7 +142,7 @@ private:
private:
bool _re_use_port = false;
bool _only_audio = false;
int _only_track = 0;
uint16_t _local_port = 0;
uint32_t _ssrc = 0;
RtpServer::TcpMode _tcp_mode = RtpServer::NONE;
@ -156,7 +156,7 @@ private:
EventPoller::DelayTask::Ptr _delay_task;
};
void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc, bool only_audio, bool multiplex) {
void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc, int only_track, bool multiplex) {
//创建udp服务器
Socket::Ptr rtp_socket = Socket::createSocket(nullptr, true);
Socket::Ptr rtcp_socket = Socket::createSocket(nullptr, true);
@ -174,7 +174,8 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
}
//设置udp socket读缓存
SockUtil::setRecvBuf(rtp_socket->rawFD(), 4 * 1024 * 1024);
GET_CONFIG(int, udpRecvSocketBuffer, RtpProxy::kUdpRecvSocketBuffer);
SockUtil::setRecvBuf(rtp_socket->rawFD(), udpRecvSocketBuffer);
TcpServer::Ptr tcp_server;
_tcp_mode = tcp_mode;
@ -183,7 +184,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
tcp_server = std::make_shared<TcpServer>(rtp_socket->getPoller());
(*tcp_server)[RtpSession::kStreamID] = stream_id;
(*tcp_server)[RtpSession::kSSRC] = ssrc;
(*tcp_server)[RtpSession::kOnlyAudio] = only_audio;
(*tcp_server)[RtpSession::kOnlyTrack] = only_track;
if (tcp_mode == PASSIVE) {
tcp_server->start<RtpSession>(local_port, local_ip);
} else if (stream_id.empty()) {
@ -200,7 +201,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
//指定了流id，那么一个端口一个流(不管是否包含多个ssrc的多个流，绑定rtp源后，会筛选掉ip端口不匹配的流)
helper = std::make_shared<RtcpHelper>(std::move(rtcp_socket), stream_id);
helper->startRtcp();
helper->setRtpServerInfo(local_port, tcp_mode, re_use_port, ssrc, only_audio);
helper->setRtpServerInfo(local_port, tcp_mode, re_use_port, ssrc, only_track);
bool bind_peer_addr = false;
auto ssrc_ptr = std::make_shared<uint32_t>(ssrc);
_ssrc = ssrc_ptr;
@ -222,7 +223,8 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
} else {
//单端口多线程接收多个流，根据ssrc区分流
udp_server = std::make_shared<UdpServer>(rtp_socket->getPoller());
(*udp_server)[RtpSession::kOnlyAudio] = only_audio;
(*udp_server)[RtpSession::kOnlyTrack] = only_track;
(*udp_server)[RtpSession::kUdpRecvBuffer] = udpRecvSocketBuffer;
udp_server->start<RtpSession>(local_port, local_ip);
rtp_socket = nullptr;
}

4
src/Rtp/RtpServer.h
View File

@ -44,7 +44,7 @@ public:
* @param multiplex 多路复用
*/
void start(uint16_t local_port, const std::string &stream_id = "", TcpMode tcp_mode = PASSIVE,
const char *local_ip = "::", bool re_use_port = true, uint32_t ssrc = 0, bool only_audio = false, bool multiplex = false);
const char *local_ip = "::", bool re_use_port = true, uint32_t ssrc = 0, int only_track = 0, bool multiplex = false);
/**
* 连接到tcp服务(tcp主动模式)
@ -81,7 +81,7 @@ protected:
std::shared_ptr<RtcpHelper> _rtcp_helper;
std::function<void()> _on_cleanup;
bool _only_audio = false;
int _only_track = 0;
//用于tcp主动模式
TcpMode _tcp_mode = NONE;
};

17
src/Rtp/RtpSession.cpp
View File

@ -23,7 +23,8 @@ namespace mediakit{
const string RtpSession::kStreamID = "stream_id";
const string RtpSession::kSSRC = "ssrc";
const string RtpSession::kOnlyAudio = "only_audio";
const string RtpSession::kOnlyTrack = "only_track";
const string RtpSession::kUdpRecvBuffer = "udp_recv_socket_buffer";
void RtpSession::attachServer(const Server &server) {
setParams(const_cast<Server &>(server));
@ -32,7 +33,13 @@ void RtpSession::attachServer(const Server &server) {
void RtpSession::setParams(mINI &ini) {
_stream_id = ini[kStreamID];
_ssrc = ini[kSSRC];
_only_audio = ini[kOnlyAudio];
_only_track = ini[kOnlyTrack];
int udp_socket_buffer = ini[kUdpRecvBuffer];
if (_is_udp) {
// 设置udp socket读缓存
SockUtil::setRecvBuf(getSock()->rawFD(),
(udp_socket_buffer > 0) ? udp_socket_buffer : (4 * 1024 * 1024));
}
}
RtpSession::RtpSession(const Socket::Ptr &sock)
@ -40,10 +47,6 @@ RtpSession::RtpSession(const Socket::Ptr &sock)
socklen_t addr_len = sizeof(_addr);
getpeername(sock->rawFD(), (struct sockaddr *)&_addr, &addr_len);
_is_udp = sock->sockType() == SockNum::Sock_UDP;
if (_is_udp) {
// 设置udp socket读缓存
SockUtil::setRecvBuf(getSock()->rawFD(), 4 * 1024 * 1024);
}
}
RtpSession::~RtpSession() = default;
@ -122,7 +125,7 @@ void RtpSession::onRtpPacket(const char *data, size_t len) {
_delay_close = true;
return;
}
_process->setOnlyAudio(_only_audio);
_process->setOnlyTrack((RtpProcess::OnlyTrack)_only_track);
_process->setDelegate(static_pointer_cast<RtpSession>(shared_from_this()));
}
try {

5
src/Rtp/RtpSession.h
View File

@ -24,7 +24,8 @@ class RtpSession : public toolkit::Session, public RtpSplitter, public MediaSour
public:
static const std::string kStreamID;
static const std::string kSSRC;
static const std::string kOnlyAudio;
static const std::string kOnlyTrack;
static const std::string kUdpRecvBuffer;
RtpSession(const toolkit::Socket::Ptr &sock);
~RtpSession() override;
@ -51,7 +52,7 @@ private:
bool _is_udp = false;
bool _search_rtp = false;
bool _search_rtp_finished = false;
bool _only_audio = false;
int _only_track = 0;
uint32_t _ssrc = 0;
toolkit::Ticker _ticker;
std::string _stream_id;

11
src/Rtsp/RtpCodec.h
View File

@ -93,6 +93,17 @@ public:
RtpInfo &getRtpInfo() { return *_rtp_info; }
enum {
RTP_ENCODER_PKT_DUR_MS = 1 // 主要应用于g711 rtp 打包器每个包的时间长度，option_value 为int*, option_len 为4
};
/**
* @brief 设置rtp打包器与解包器的相关参数，主要应用与g711 rtp 打包器，使用方法类似setsockopt
*
* @param opt 设置的选项
* @param param 设置的参数
*/
virtual void setOpt(int opt, const toolkit::Any &param) {};
private:
std::unique_ptr<RtpInfo> _rtp_info;
};

12
src/Rtsp/Rtsp.cpp
View File

@ -352,12 +352,20 @@ public:
}
void makeSockPair(std::pair<Socket::Ptr, Socket::Ptr> &pair, const string &local_ip, bool re_use_port, bool is_udp) {
auto &sock0 = pair.first;
auto &sock1 = pair.second;
auto sock_pair = getPortPair();
if (!sock_pair) {
throw runtime_error("none reserved port in pool");
}
makeSockPair_l(sock_pair, pair, local_ip, re_use_port, is_udp);
// 确保udp和tcp模式都能打开
auto new_pair = std::make_pair(Socket::createSocket(), Socket::createSocket());
makeSockPair_l(sock_pair, new_pair, local_ip, re_use_port, !is_udp);
}
void makeSockPair_l(const std::shared_ptr<uint16_t> &sock_pair, std::pair<Socket::Ptr, Socket::Ptr> &pair, const string &local_ip, bool re_use_port, bool is_udp) {
auto &sock0 = pair.first;
auto &sock1 = pair.second;
if (is_udp) {
if (!sock0->bindUdpSock(2 * *sock_pair, local_ip.data(), re_use_port)) {
// 分配端口失败

78
srt/SrtSession.cpp
View File

@ -16,41 +16,17 @@ SrtSession::SrtSession(const Socket::Ptr &sock)
// TraceL<<"after addr len "<<addr_len<<" family "<<_peer_addr.ss_family;
}
EventPoller::Ptr SrtSession::queryPoller(const Buffer::Ptr &buffer) {
uint8_t *data = (uint8_t *)buffer->data();
size_t size = buffer->size();
if (DataPacket::isDataPacket(data, size)) {
uint32_t socket_id = DataPacket::getSocketID(data, size);
auto trans = SrtTransportManager::Instance().getItem(std::to_string(socket_id));
return trans ? trans->getPoller() : nullptr;
}
if (HandshakePacket::isHandshakePacket(data, size)) {
auto type = HandshakePacket::getHandshakeType(data, size);
if (type == HandshakePacket::HS_TYPE_INDUCTION) {
// 握手第一阶段
return nullptr;
} else if (type == HandshakePacket::HS_TYPE_CONCLUSION) {
// 握手第二阶段
uint32_t sync_cookie = HandshakePacket::getSynCookie(data, size);
auto trans = SrtTransportManager::Instance().getHandshakeItem(std::to_string(sync_cookie));
return trans ? trans->getPoller() : nullptr;
} else {
WarnL << " not reach there";
}
} else {
uint32_t socket_id = ControlPacket::getSocketID(data, size);
auto trans = SrtTransportManager::Instance().getItem(std::to_string(socket_id));
return trans ? trans->getPoller() : nullptr;
}
return nullptr;
}
void SrtSession::attachServer(const toolkit::Server &server) {
SockUtil::setRecvBuf(getSock()->rawFD(), 1024 * 1024);
}
extern SrtTransport::Ptr querySrtTransport(uint8_t *data, size_t size, const EventPoller::Ptr& poller);
EventPoller::Ptr SrtSession::queryPoller(const Buffer::Ptr &buffer) {
auto transport = querySrtTransport((uint8_t *)buffer->data(), buffer->size(), nullptr);
return transport ? transport->getPoller() : nullptr;
}
void SrtSession::onRecv(const Buffer::Ptr &buffer) {
uint8_t *data = (uint8_t *)buffer->data();
size_t size = buffer->size();
@ -58,45 +34,7 @@ void SrtSession::onRecv(const Buffer::Ptr &buffer) {
if (_find_transport) {
//只允许寻找一次transport
_find_transport = false;
if (DataPacket::isDataPacket(data, size)) {
uint32_t socket_id = DataPacket::getSocketID(data, size);
auto trans = SrtTransportManager::Instance().getItem(std::to_string(socket_id));
if (trans) {
_transport = std::move(trans);
} else {
WarnL << " data packet not find transport ";
}
}
if (HandshakePacket::isHandshakePacket(data, size)) {
auto type = HandshakePacket::getHandshakeType(data, size);
if (type == HandshakePacket::HS_TYPE_INDUCTION) {
// 握手第一阶段
_transport = std::make_shared<SrtTransportImp>(getPoller());
} else if (type == HandshakePacket::HS_TYPE_CONCLUSION) {
// 握手第二阶段
uint32_t sync_cookie = HandshakePacket::getSynCookie(data, size);
auto trans = SrtTransportManager::Instance().getHandshakeItem(std::to_string(sync_cookie));
if (trans) {
_transport = std::move(trans);
} else {
WarnL << " hanshake packet not find transport ";
}
} else {
WarnL << " not reach there";
}
} else {
uint32_t socket_id = ControlPacket::getSocketID(data, size);
auto trans = SrtTransportManager::Instance().getItem(std::to_string(socket_id));
if (trans) {
_transport = std::move(trans);
} else {
WarnL << " not find transport";
}
}
_transport = querySrtTransport(data, size, getPoller());
if (_transport) {
_transport->setSession(static_pointer_cast<Session>(shared_from_this()));
}

34
srt/SrtTransport.cpp
View File

@ -61,7 +61,7 @@ void SrtTransport::switchToOtherTransport(uint8_t *buf, int len, uint32_t socket
BufferRaw::Ptr tmp = BufferRaw::create();
struct sockaddr_storage tmp_addr = *addr;
tmp->assign((char *)buf, len);
auto trans = SrtTransportManager::Instance().getItem(std::to_string(socketid));
auto trans = SrtTransportManager::Instance().getItem(socketid);
if (trans) {
trans->getPoller()->async([tmp, tmp_addr, trans] {
trans->inputSockData((uint8_t *)tmp->data(), tmp->size(), (struct sockaddr_storage *)&tmp_addr);
@ -700,30 +700,30 @@ void SrtTransport::sendPacket(Buffer::Ptr pkt, bool flush) {
}
}
std::string SrtTransport::getIdentifier() {
std::string SrtTransport::getIdentifier() const {
return _selected_session ? _selected_session->getIdentifier() : "";
}
void SrtTransport::registerSelfHandshake() {
SrtTransportManager::Instance().addHandshakeItem(std::to_string(_sync_cookie), shared_from_this());
SrtTransportManager::Instance().addHandshakeItem(_sync_cookie, shared_from_this());
}
void SrtTransport::unregisterSelfHandshake() {
if (_sync_cookie == 0) {
return;
}
SrtTransportManager::Instance().removeHandshakeItem(std::to_string(_sync_cookie));
SrtTransportManager::Instance().removeHandshakeItem(_sync_cookie);
}
void SrtTransport::registerSelf() {
if (_socket_id == 0) {
return;
}
SrtTransportManager::Instance().addItem(std::to_string(_socket_id), shared_from_this());
SrtTransportManager::Instance().addItem(_socket_id, shared_from_this());
}
void SrtTransport::unregisterSelf() {
SrtTransportManager::Instance().removeItem(std::to_string(_socket_id));
SrtTransportManager::Instance().removeItem(_socket_id);
}
void SrtTransport::onShutdown(const SockException &ex) {
@ -739,7 +739,7 @@ void SrtTransport::onShutdown(const SockException &ex) {
}
}
size_t SrtTransport::getPayloadSize() {
size_t SrtTransport::getPayloadSize() const {
size_t ret = (_mtu - 28 - 16) / 188 * 188;
return ret;
}
@ -792,15 +792,13 @@ SrtTransportManager &SrtTransportManager::Instance() {
return s_instance;
}
void SrtTransportManager::addItem(const std::string &key, const SrtTransport::Ptr &ptr) {
void SrtTransportManager::addItem(const uint32_t key, const SrtTransport::Ptr &ptr) {
std::lock_guard<std::mutex> lck(_mtx);
_map[key] = ptr;
}
SrtTransport::Ptr SrtTransportManager::getItem(const std::string &key) {
if (key.empty()) {
return nullptr;
}
SrtTransport::Ptr SrtTransportManager::getItem(const uint32_t key) {
assert(key > 0);
std::lock_guard<std::mutex> lck(_mtx);
auto it = _map.find(key);
if (it == _map.end()) {
@ -809,25 +807,23 @@ SrtTransport::Ptr SrtTransportManager::getItem(const std::string &key) {
return it->second.lock();
}
void SrtTransportManager::removeItem(const std::string &key) {
void SrtTransportManager::removeItem(const uint32_t key) {
std::lock_guard<std::mutex> lck(_mtx);
_map.erase(key);
}
void SrtTransportManager::addHandshakeItem(const std::string &key, const SrtTransport::Ptr &ptr) {
void SrtTransportManager::addHandshakeItem(const uint32_t key, const SrtTransport::Ptr &ptr) {
std::lock_guard<std::mutex> lck(_handshake_mtx);
_handshake_map[key] = ptr;
}
void SrtTransportManager::removeHandshakeItem(const std::string &key) {
void SrtTransportManager::removeHandshakeItem(const uint32_t key) {
std::lock_guard<std::mutex> lck(_handshake_mtx);
_handshake_map.erase(key);
}
SrtTransport::Ptr SrtTransportManager::getHandshakeItem(const std::string &key) {
if (key.empty()) {
return nullptr;
}
SrtTransport::Ptr SrtTransportManager::getHandshakeItem(const uint32_t key) {
assert(key > 0);
std::lock_guard<std::mutex> lck(_handshake_mtx);
auto it = _handshake_map.find(key);
if (it == _handshake_map.end()) {

20
srt/SrtTransport.hpp
View File

@ -45,7 +45,7 @@ public:
virtual void inputSockData(uint8_t *buf, int len, struct sockaddr_storage *addr);
virtual void onSendTSData(const Buffer::Ptr &buffer, bool flush);
std::string getIdentifier();
std::string getIdentifier() const;
void unregisterSelf();
void unregisterSelfHandshake();
@ -89,7 +89,7 @@ private:
void sendShutDown();
void sendMsgDropReq(uint32_t first, uint32_t last);
size_t getPayloadSize();
size_t getPayloadSize() const;
void createTimerForCheckAlive();
@ -164,23 +164,23 @@ private:
class SrtTransportManager {
public:
static SrtTransportManager &Instance();
SrtTransport::Ptr getItem(const std::string &key);
void addItem(const std::string &key, const SrtTransport::Ptr &ptr);
void removeItem(const std::string &key);
SrtTransport::Ptr getItem(const uint32_t key);
void addItem(const uint32_t key, const SrtTransport::Ptr &ptr);
void removeItem(const uint32_t key);
void addHandshakeItem(const std::string &key, const SrtTransport::Ptr &ptr);
void removeHandshakeItem(const std::string &key);
SrtTransport::Ptr getHandshakeItem(const std::string &key);
void addHandshakeItem(const uint32_t key, const SrtTransport::Ptr &ptr);
void removeHandshakeItem(const uint32_t key);
SrtTransport::Ptr getHandshakeItem(const uint32_t key);
private:
SrtTransportManager() = default;
private:
std::mutex _mtx;
std::unordered_map<std::string, std::weak_ptr<SrtTransport>> _map;
std::unordered_map<uint32_t , std::weak_ptr<SrtTransport>> _map;
std::mutex _handshake_mtx;
std::unordered_map<std::string, std::weak_ptr<SrtTransport>> _handshake_map;
std::unordered_map<uint32_t, std::weak_ptr<SrtTransport>> _handshake_map;
};
} // namespace SRT

26
srt/SrtTransportImp.cpp
View File

@ -24,6 +24,32 @@ SrtTransportImp::~SrtTransportImp() {
}
}
SrtTransport::Ptr querySrtTransport(uint8_t *data, size_t size, const EventPoller::Ptr& poller) {
if (DataPacket::isDataPacket(data, size)) {
uint32_t socket_id = DataPacket::getSocketID(data, size);
return SrtTransportManager::Instance().getItem(socket_id);
}
if (HandshakePacket::isHandshakePacket(data, size)) {
auto type = HandshakePacket::getHandshakeType(data, size);
if (type == HandshakePacket::HS_TYPE_INDUCTION) {
// 握手第一阶段
return poller ? std::make_shared<SrtTransportImp>(poller) : nullptr;
}
if (type == HandshakePacket::HS_TYPE_CONCLUSION) {
// 握手第二阶段
uint32_t sync_cookie = HandshakePacket::getSynCookie(data, size);
return SrtTransportManager::Instance().getHandshakeItem(sync_cookie);
}
}
uint32_t socket_id = ControlPacket::getSocketID(data, size);
return SrtTransportManager::Instance().getItem(socket_id);
}
void SrtTransportImp::onHandShakeFinished(std::string &streamid, struct sockaddr_storage *addr) {
SrtTransport::onHandShakeFinished(streamid,addr);
// TODO parse stream id like this zlmediakit.com/live/test?token=1213444&type=push

3
webrtc/DtlsTransport.cpp
View File

@ -741,8 +741,7 @@ namespace RTC
if (!IsRunning())
{
MS_ERROR("cannot process data while not running");
MS_WARN_TAG(nullptr,"cannot process data while not running");
return;
}

437
webrtc/Sdp.cpp
View File

File diff suppressed because it is too large Load Diff

459
webrtc/Sdp.h
View File

@ -22,97 +22,87 @@
namespace mediakit {
//https://datatracker.ietf.org/doc/rfc4566/?include_text=1
//https://blog.csdn.net/aggresss/article/details/109850434
//https://aggresss.blog.csdn.net/article/details/106436703
//Session description
// v= (protocol version)
// o= (originator and session identifier)
// s= (session name)
// i=* (session information)
// u=* (URI of description)
// e=* (email address)
// p=* (phone number)
// c=* (connection information -- not required if included in
// all media)
// b=* (zero or more bandwidth information lines)
// One or more time descriptions ("t=" and "r=" lines; see below)
// z=* (time zone adjustments)
// k=* (encryption key)
// a=* (zero or more session attribute lines)
// Zero or more media descriptions
// https://datatracker.ietf.org/doc/rfc4566/?include_text=1
// https://blog.csdn.net/aggresss/article/details/109850434
// https://aggresss.blog.csdn.net/article/details/106436703
// Session description
// v= (protocol version)
// o= (originator and session identifier)
// s= (session name)
// i=* (session information)
// u=* (URI of description)
// e=* (email address)
// p=* (phone number)
// c=* (connection information -- not required if included in
// all media)
// b=* (zero or more bandwidth information lines)
// One or more time descriptions ("t=" and "r=" lines; see below)
// z=* (time zone adjustments)
// k=* (encryption key)
// a=* (zero or more session attribute lines)
// Zero or more media descriptions
//
// Time description
// t= (time the session is active)
// r=* (zero or more repeat times)
// Time description
// t= (time the session is active)
// r=* (zero or more repeat times)
//
// Media description, if present
// m= (media name and transport address)
// i=* (media title)
// c=* (connection information -- optional if included at
// session level)
// b=* (zero or more bandwidth information lines)
// k=* (encryption key)
// a=* (zero or more media attribute lines)
// Media description, if present
// m= (media name and transport address)
// i=* (media title)
// c=* (connection information -- optional if included at
// session level)
// b=* (zero or more bandwidth information lines)
// k=* (encryption key)
// a=* (zero or more media attribute lines)
enum class RtpDirection {
invalid = -1,
//只发送
// 只发送
sendonly,
//只接收
// 只接收
recvonly,
//同时发送接收
// 同时发送接收
sendrecv,
//禁止发送数据
// 禁止发送数据
inactive
};
enum class DtlsRole {
invalid = -1,
//客户端
// 客户端
active,
//服务端
// 服务端
passive,
//既可作做客户端也可以做服务端
// 既可作做客户端也可以做服务端
actpass,
};
enum class SdpType {
invalid = -1,
offer,
answer
};
enum class SdpType { invalid = -1, offer, answer };
DtlsRole getDtlsRole(const std::string &str);
const char* getDtlsRoleString(DtlsRole role);
const char *getDtlsRoleString(DtlsRole role);
RtpDirection getRtpDirection(const std::string &str);
const char* getRtpDirectionString(RtpDirection val);
const char *getRtpDirectionString(RtpDirection val);
class SdpItem {
public:
using Ptr = std::shared_ptr<SdpItem>;
virtual ~SdpItem() = default;
virtual void parse(const std::string &str) {
value = str;
}
virtual std::string toString() const {
return value;
}
virtual const char* getKey() const = 0;
virtual void parse(const std::string &str) { value = str; }
virtual std::string toString() const { return value; }
virtual const char *getKey() const = 0;
void reset() {
value.clear();
}
void reset() { value.clear(); }
protected:
mutable std::string value;
};
template <char KEY>
class SdpString : public SdpItem{
class SdpString : public SdpItem {
public:
SdpString() = default;
SdpString(std::string val) {value = std::move(val);}
SdpString(std::string val) { value = std::move(val); }
// *=*
const char* getKey() const override { static std::string key(1, KEY); return key.data();}
};
@ -126,34 +116,34 @@ public:
this->value = std::move(val);
}
const char* getKey() const override { return key.data();}
const char *getKey() const override { return key.data(); }
};
class SdpTime : public SdpItem{
class SdpTime : public SdpItem {
public:
//5.9. Timing ("t=")
// t=<start-time> <stop-time>
uint64_t start {0};
uint64_t stop {0};
// 5.9. Timing ("t=")
// t=<start-time> <stop-time>
uint64_t start { 0 };
uint64_t stop { 0 };
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "t";}
const char *getKey() const override { return "t"; }
};
class SdpOrigin : public SdpItem{
class SdpOrigin : public SdpItem {
public:
// 5.2. Origin ("o=")
// o=jdoe 2890844526 2890842807 IN IP4 10.47.16.5
// o=<username> <sess-id> <sess-version> <nettype> <addrtype> <unicast-address>
std::string username {"-"};
std::string username { "-" };
std::string session_id;
std::string session_version;
std::string nettype {"IN"};
std::string addrtype {"IP4"};
std::string address {"0.0.0.0"};
std::string nettype { "IN" };
std::string addrtype { "IP4" };
std::string address { "0.0.0.0" };
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "o";}
const char *getKey() const override { return "o"; }
bool empty() const {
return username.empty() || session_id.empty() || session_version.empty()
|| nettype.empty() || addrtype.empty() || address.empty();
@ -165,28 +155,28 @@ public:
// 5.7. Connection Data ("c=")
// c=IN IP4 224.2.17.12/127
// c=<nettype> <addrtype> <connection-address>
std::string nettype {"IN"};
std::string addrtype {"IP4"};
std::string address {"0.0.0.0"};
std::string nettype { "IN" };
std::string addrtype { "IP4" };
std::string address { "0.0.0.0" };
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "c";}
bool empty() const {return address.empty();}
const char *getKey() const override { return "c"; }
bool empty() const { return address.empty(); }
};
class SdpBandwidth : public SdpItem {
public:
//5.8. Bandwidth ("b=")
//b=<bwtype>:<bandwidth>
// 5.8. Bandwidth ("b=")
// b=<bwtype>:<bandwidth>
//AS、CT
std::string bwtype {"AS"};
uint32_t bandwidth {0};
// AS、CT
std::string bwtype { "AS" };
uint32_t bandwidth { 0 };
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "b";}
bool empty() const {return bandwidth == 0;}
const char *getKey() const override { return "b"; }
bool empty() const { return bandwidth == 0; }
};
class SdpMedia : public SdpItem {
@ -195,287 +185,284 @@ public:
// m=<media> <port> <proto> <fmt> ...
TrackType type;
uint16_t port;
//RTP/AVP：应用场景为视频/音频的 RTP 协议。参考 RFC 3551
//RTP/SAVP：应用场景为视频/音频的 SRTP 协议。参考 RFC 3711
//RTP/AVPF: 应用场景为视频/音频的 RTP 协议，支持 RTCP-based Feedback。参考 RFC 4585
//RTP/SAVPF: 应用场景为视频/音频的 SRTP 协议，支持 RTCP-based Feedback。参考 RFC 5124
// RTP/AVP：应用场景为视频/音频的 RTP 协议。参考 RFC 3551
// RTP/SAVP：应用场景为视频/音频的 SRTP 协议。参考 RFC 3711
// RTP/AVPF: 应用场景为视频/音频的 RTP 协议，支持 RTCP-based Feedback。参考 RFC 4585
// RTP/SAVPF: 应用场景为视频/音频的 SRTP 协议，支持 RTCP-based Feedback。参考 RFC 5124
std::string proto;
std::vector<std::string> fmts;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "m";}
const char *getKey() const override { return "m"; }
};
class SdpAttr : public SdpItem{
class SdpAttr : public SdpItem {
public:
using Ptr = std::shared_ptr<SdpAttr>;
//5.13. Attributes ("a=")
//a=<attribute>
//a=<attribute>:<value>
// 5.13. Attributes ("a=")
// a=<attribute>
// a=<attribute>:<value>
SdpItem::Ptr detail;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "a";}
const char *getKey() const override { return "a"; }
};
class SdpAttrGroup : public SdpItem{
class SdpAttrGroup : public SdpItem {
public:
//a=group:BUNDLE line with all the 'mid' identifiers part of the
// BUNDLE group is included at the session-level.
//a=group:LS session level attribute MUST be included wth the 'mid'
// identifiers that are part of the same lip sync group.
std::string type {"BUNDLE"};
// a=group:BUNDLE line with all the 'mid' identifiers part of the
// BUNDLE group is included at the session-level.
// a=group:LS session level attribute MUST be included wth the 'mid'
// identifiers that are part of the same lip sync group.
std::string type { "BUNDLE" };
std::vector<std::string> mids;
void parse(const std::string &str) override ;
std::string toString() const override ;
const char* getKey() const override { return "group";}
void parse(const std::string &str) override;
std::string toString() const override;
const char *getKey() const override { return "group"; }
};
class SdpAttrMsidSemantic : public SdpItem {
public:
//https://tools.ietf.org/html/draft-alvestrand-rtcweb-msid-02#section-3
//3. The Msid-Semantic Attribute
// https://tools.ietf.org/html/draft-alvestrand-rtcweb-msid-02#section-3
// 3. The Msid-Semantic Attribute
//
// In order to fully reproduce the semantics of the SDP and SSRC
// grouping frameworks, a session-level attribute is defined for
// signalling the semantics associated with an msid grouping.
// In order to fully reproduce the semantics of the SDP and SSRC
// grouping frameworks, a session-level attribute is defined for
// signalling the semantics associated with an msid grouping.
//
// This OPTIONAL attribute gives the message ID and its group semantic.
// a=msid-semantic: examplefoo LS
// This OPTIONAL attribute gives the message ID and its group semantic.
// a=msid-semantic: examplefoo LS
//
//
// The ABNF of msid-semantic is:
// The ABNF of msid-semantic is:
//
// msid-semantic-attr = "msid-semantic:" " " msid token
// token = <as defined in RFC 4566>
// msid-semantic-attr = "msid-semantic:" " " msid token
// token = <as defined in RFC 4566>
//
// The semantic field may hold values from the IANA registries
// "Semantics for the "ssrc-group" SDP Attribute" and "Semantics for the
// "group" SDP Attribute".
//a=msid-semantic: WMS 616cfbb1-33a3-4d8c-8275-a199d6005549
std::string msid{"WMS"};
// The semantic field may hold values from the IANA registries
// "Semantics for the "ssrc-group" SDP Attribute" and "Semantics for the
// "group" SDP Attribute".
// a=msid-semantic: WMS 616cfbb1-33a3-4d8c-8275-a199d6005549
std::string msid { "WMS" };
std::string token;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "msid-semantic";}
bool empty() const {
return msid.empty();
}
const char *getKey() const override { return "msid-semantic"; }
bool empty() const { return msid.empty(); }
};
class SdpAttrRtcp : public SdpItem {
public:
// a=rtcp:9 IN IP4 0.0.0.0
uint16_t port{0};
std::string nettype {"IN"};
std::string addrtype {"IP4"};
std::string address {"0.0.0.0"};
void parse(const std::string &str) override;;
uint16_t port { 0 };
std::string nettype { "IN" };
std::string addrtype { "IP4" };
std::string address { "0.0.0.0" };
void parse(const std::string &str) override;
;
std::string toString() const override;
const char* getKey() const override { return "rtcp";}
bool empty() const {
return address.empty() || !port;
}
const char *getKey() const override { return "rtcp"; }
bool empty() const { return address.empty() || !port; }
};
class SdpAttrIceUfrag : public SdpItem {
public:
SdpAttrIceUfrag() = default;
SdpAttrIceUfrag(std::string str) {value = std::move(str);}
//a=ice-ufrag:sXJ3
const char* getKey() const override { return "ice-ufrag";}
SdpAttrIceUfrag(std::string str) { value = std::move(str); }
// a=ice-ufrag:sXJ3
const char *getKey() const override { return "ice-ufrag"; }
};
class SdpAttrIcePwd : public SdpItem {
public:
SdpAttrIcePwd() = default;
SdpAttrIcePwd(std::string str) {value = std::move(str);}
//a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV
const char* getKey() const override { return "ice-pwd";}
SdpAttrIcePwd(std::string str) { value = std::move(str); }
// a=ice-pwd:yEclOTrLg1gEubBFefOqtmyV
const char *getKey() const override { return "ice-pwd"; }
};
class SdpAttrIceOption : public SdpItem {
public:
//a=ice-options:trickle
bool trickle{false};
bool renomination{false};
// a=ice-options:trickle
bool trickle { false };
bool renomination { false };
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "ice-options";}
const char *getKey() const override { return "ice-options"; }
};
class SdpAttrFingerprint : public SdpItem {
public:
//a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79
// a=fingerprint:sha-256 22:14:B5:AF:66:12:C7:C7:8D:EF:4B:DE:40:25:ED:5D:8F:17:54:DD:88:33:C0:13:2E:FD:1A:FA:7E:7A:1B:79
std::string algorithm;
std::string hash;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "fingerprint";}
const char *getKey() const override { return "fingerprint"; }
bool empty() const { return algorithm.empty() || hash.empty(); }
};
class SdpAttrSetup : public SdpItem {
public:
//a=setup:actpass
// a=setup:actpass
SdpAttrSetup() = default;
SdpAttrSetup(DtlsRole r) { role = r; }
DtlsRole role{DtlsRole::actpass};
DtlsRole role { DtlsRole::actpass };
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "setup";}
const char *getKey() const override { return "setup"; }
};
class SdpAttrMid : public SdpItem {
public:
SdpAttrMid() = default;
SdpAttrMid(std::string val) { value = std::move(val); }
//a=mid:audio
const char* getKey() const override { return "mid";}
// a=mid:audio
const char *getKey() const override { return "mid"; }
};
class SdpAttrExtmap : public SdpItem {
public:
//https://aggresss.blog.csdn.net/article/details/106436703
//a=extmap:1[/sendonly] urn:ietf:params:rtp-hdrext:ssrc-audio-level
// https://aggresss.blog.csdn.net/article/details/106436703
// a=extmap:1[/sendonly] urn:ietf:params:rtp-hdrext:ssrc-audio-level
uint8_t id;
RtpDirection direction{RtpDirection::invalid};
RtpDirection direction { RtpDirection::invalid };
std::string ext;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "extmap";}
const char *getKey() const override { return "extmap"; }
};
class SdpAttrRtpMap : public SdpItem {
public:
//a=rtpmap:111 opus/48000/2
// a=rtpmap:111 opus/48000/2
uint8_t pt;
std::string codec;
uint32_t sample_rate;
uint32_t channel {0};
uint32_t channel { 0 };
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "rtpmap";}
const char *getKey() const override { return "rtpmap"; }
};
class SdpAttrRtcpFb : public SdpItem {
public:
//a=rtcp-fb:98 nack pli
//a=rtcp-fb:120 nack 支持 nack 重传，nack (Negative-Acknowledgment) 。
//a=rtcp-fb:120 nack pli 支持 nack 关键帧重传，PLI (Picture Loss Indication) 。
//a=rtcp-fb:120 ccm fir 支持编码层关键帧请求，CCM (Codec Control Message)，FIR (Full Intra Request )，通常与 nack pli 有同样的效果，但是 nack pli 是用于重传时的关键帧请求。
//a=rtcp-fb:120 goog-remb 支持 REMB (Receiver Estimated Maximum Bitrate) 。
//a=rtcp-fb:120 transport-cc 支持 TCC (Transport Congest Control) 。
// a=rtcp-fb:98 nack pli
// a=rtcp-fb:120 nack 支持 nack 重传，nack (Negative-Acknowledgment) 。
// a=rtcp-fb:120 nack pli 支持 nack 关键帧重传，PLI (Picture Loss Indication) 。
// a=rtcp-fb:120 ccm fir 支持编码层关键帧请求，CCM (Codec Control Message)，FIR (Full Intra Request )，通常与 nack pli 有同样的效果，但是 nack pli
// 是用于重传时的关键帧请求。 a=rtcp-fb:120 goog-remb 支持 REMB (Receiver Estimated Maximum Bitrate) 。 a=rtcp-fb:120 transport-cc 支持 TCC (Transport
// Congest Control) 。
uint8_t pt;
std::string rtcp_type;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "rtcp-fb";}
const char *getKey() const override { return "rtcp-fb"; }
};
class SdpAttrFmtp : public SdpItem {
public:
//fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
// fmtp:96 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
uint8_t pt;
std::map<std::string/*key*/, std::string/*value*/, StrCaseCompare> fmtp;
std::map<std::string /*key*/, std::string /*value*/, StrCaseCompare> fmtp;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "fmtp";}
const char *getKey() const override { return "fmtp"; }
};
class SdpAttrSSRC : public SdpItem {
public:
//a=ssrc:3245185839 cname:Cx4i/VTR51etgjT7
//a=ssrc:3245185839 msid:cb373bff-0fea-4edb-bc39-e49bb8e8e3b9 0cf7e597-36a2-4480-9796-69bf0955eef5
//a=ssrc:3245185839 mslabel:cb373bff-0fea-4edb-bc39-e49bb8e8e3b9
//a=ssrc:3245185839 label:0cf7e597-36a2-4480-9796-69bf0955eef5
//a=ssrc:<ssrc-id> <attribute>
//a=ssrc:<ssrc-id> <attribute>:<value>
//cname 是必须的，msid/mslabel/label 这三个属性都是 WebRTC 自创的，或者说 Google 自创的，可以参考 https://tools.ietf.org/html/draft-ietf-mmusic-msid-17，
// 理解它们三者的关系需要先了解三个概念：RTP stream / MediaStreamTrack / MediaStream ：
//一个 a=ssrc 代表一个 RTP stream ；
//一个 MediaStreamTrack 通常包含一个或多个 RTP stream，例如一个视频 MediaStreamTrack 中通常包含两个 RTP stream，一个用于常规传输，一个用于 nack 重传；
//一个 MediaStream 通常包含一个或多个 MediaStreamTrack ，例如 simulcast 场景下，一个 MediaStream 通常会包含三个不同编码质量的 MediaStreamTrack ；
//这种标记方式并不被 Firefox 认可，在 Firefox 生成的 SDP 中一个 a=ssrc 通常只有一行，例如：
//a=ssrc:3245185839 cname:Cx4i/VTR51etgjT7
// a=ssrc:3245185839 cname:Cx4i/VTR51etgjT7
// a=ssrc:3245185839 msid:cb373bff-0fea-4edb-bc39-e49bb8e8e3b9 0cf7e597-36a2-4480-9796-69bf0955eef5
// a=ssrc:3245185839 mslabel:cb373bff-0fea-4edb-bc39-e49bb8e8e3b9
// a=ssrc:3245185839 label:0cf7e597-36a2-4480-9796-69bf0955eef5
// a=ssrc:<ssrc-id> <attribute>
// a=ssrc:<ssrc-id> <attribute>:<value>
// cname 是必须的，msid/mslabel/label 这三个属性都是 WebRTC 自创的，或者说 Google 自创的，可以参考 https://tools.ietf.org/html/draft-ietf-mmusic-msid-17，
// 理解它们三者的关系需要先了解三个概念：RTP stream / MediaStreamTrack / MediaStream ：
// 一个 a=ssrc 代表一个 RTP stream ；
// 一个 MediaStreamTrack 通常包含一个或多个 RTP stream，例如一个视频 MediaStreamTrack 中通常包含两个 RTP stream，一个用于常规传输，一个用于 nack 重传；
// 一个 MediaStream 通常包含一个或多个 MediaStreamTrack ，例如 simulcast 场景下，一个 MediaStream 通常会包含三个不同编码质量的 MediaStreamTrack ；
// 这种标记方式并不被 Firefox 认可，在 Firefox 生成的 SDP 中一个 a=ssrc 通常只有一行，例如：
// a=ssrc:3245185839 cname:Cx4i/VTR51etgjT7
uint32_t ssrc;
std::string attribute;
std::string attribute_value;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "ssrc";}
const char *getKey() const override { return "ssrc"; }
};
class SdpAttrSSRCGroup : public SdpItem {
public:
//a=ssrc-group 定义参考 RFC 5576(https://tools.ietf.org/html/rfc5576) ，用于描述多个 ssrc 之间的关联，常见的有两种：
//a=ssrc-group:FID 2430709021 3715850271
// FID (Flow Identification) 最初用在 FEC 的关联中，WebRTC 中通常用于关联一组常规 RTP stream 和 重传 RTP stream 。
//a=ssrc-group:SIM 360918977 360918978 360918980
// 在 Chrome 独有的 SDP munging 风格的 simulcast 中使用，将三组编码质量由低到高的 MediaStreamTrack 关联在一起。
std::string type{"FID"};
// a=ssrc-group 定义参考 RFC 5576(https://tools.ietf.org/html/rfc5576) ，用于描述多个 ssrc 之间的关联，常见的有两种：
// a=ssrc-group:FID 2430709021 3715850271
// FID (Flow Identification) 最初用在 FEC 的关联中，WebRTC 中通常用于关联一组常规 RTP stream 和 重传 RTP stream 。
// a=ssrc-group:SIM 360918977 360918978 360918980
// 在 Chrome 独有的 SDP munging 风格的 simulcast 中使用，将三组编码质量由低到高的 MediaStreamTrack 关联在一起。
std::string type { "FID" };
std::vector<uint32_t> ssrcs;
bool isFID() const { return type == "FID"; }
bool isSIM() const { return type == "SIM"; }
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "ssrc-group";}
const char *getKey() const override { return "ssrc-group"; }
};
class SdpAttrSctpMap : public SdpItem {
public:
//https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-05
//a=sctpmap:5000 webrtc-datachannel 1024
//a=sctpmap: sctpmap-number media-subtypes [streams]
// https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-05
// a=sctpmap:5000 webrtc-datachannel 1024
// a=sctpmap: sctpmap-number media-subtypes [streams]
uint16_t port = 0;
std::string subtypes;
uint32_t streams = 0;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "sctpmap";}
const char *getKey() const override { return "sctpmap"; }
bool empty() const { return port == 0 && subtypes.empty() && streams == 0; }
};
class SdpAttrCandidate : public SdpItem {
public:
using Ptr = std::shared_ptr<SdpAttrCandidate>;
//https://tools.ietf.org/html/rfc5245
//15.1. "candidate" Attribute
//a=candidate:4 1 udp 2 192.168.1.7 58107 typ host
//a=candidate:<foundation> <component-id> <transport> <priority> <address> <port> typ <cand-type>
// https://tools.ietf.org/html/rfc5245
// 15.1. "candidate" Attribute
// a=candidate:4 1 udp 2 192.168.1.7 58107 typ host
// a=candidate:<foundation> <component-id> <transport> <priority> <address> <port> typ <cand-type>
std::string foundation;
//传输媒体的类型,1代表RTP;2代表 RTCP。
// 传输媒体的类型,1代表RTP;2代表 RTCP。
uint32_t component;
std::string transport {"udp"};
std::string transport { "udp" };
uint32_t priority;
std::string address;
uint16_t port;
std::string type;
std::vector<std::pair<std::string, std::string> > arr;
std::vector<std::pair<std::string, std::string>> arr;
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "candidate";}
const char *getKey() const override { return "candidate"; }
};
class SdpAttrMsid : public SdpItem{
class SdpAttrMsid : public SdpItem {
public:
const char* getKey() const override { return "msid";}
const char *getKey() const override { return "msid"; }
};
class SdpAttrExtmapAllowMixed : public SdpItem{
class SdpAttrExtmapAllowMixed : public SdpItem {
public:
const char* getKey() const override { return "extmap-allow-mixed";}
const char *getKey() const override { return "extmap-allow-mixed"; }
};
class SdpAttrSimulcast : public SdpItem{
class SdpAttrSimulcast : public SdpItem {
public:
//https://www.meetecho.com/blog/simulcast-janus-ssrc/
//https://tools.ietf.org/html/draft-ietf-mmusic-sdp-simulcast-14
const char* getKey() const override { return "simulcast";}
// https://www.meetecho.com/blog/simulcast-janus-ssrc/
// https://tools.ietf.org/html/draft-ietf-mmusic-sdp-simulcast-14
const char *getKey() const override { return "simulcast"; }
void parse(const std::string &str) override;
std::string toString() const override;
bool empty() const { return rids.empty(); }
@ -483,11 +470,11 @@ public:
std::vector<std::string> rids;
};
class SdpAttrRid : public SdpItem{
class SdpAttrRid : public SdpItem {
public:
void parse(const std::string &str) override;
std::string toString() const override;
const char* getKey() const override { return "rid";}
const char *getKey() const override { return "rid"; }
std::string direction;
std::string rid;
};
@ -507,8 +494,8 @@ public:
RtpDirection getDirection() const;
template<typename cls>
cls getItemClass(char key, const char *attr_key = nullptr) const{
template <typename cls>
cls getItemClass(char key, const char *attr_key = nullptr) const {
auto item = std::dynamic_pointer_cast<cls>(getItem(key, attr_key));
if (!item) {
return cls();
@ -516,7 +503,7 @@ public:
return *item;
}
std::string getStringItem(char key, const char *attr_key = nullptr) const{
std::string getStringItem(char key, const char *attr_key = nullptr) const {
auto item = getItem(key, attr_key);
if (!item) {
return "";
@ -526,7 +513,7 @@ public:
SdpItem::Ptr getItem(char key, const char *attr_key = nullptr) const;
template<typename cls>
template <typename cls>
std::vector<cls> getAllItem(char key_c, const char *attr_key = nullptr) const {
std::vector<cls> ret;
std::string key(1, key_c);
@ -555,7 +542,7 @@ private:
std::vector<SdpItem::Ptr> items;
};
class RtcSessionSdp : public RtcSdpBase{
class RtcSessionSdp : public RtcSdpBase {
public:
using Ptr = std::shared_ptr<RtcSessionSdp>;
int getVersion() const;
@ -580,45 +567,45 @@ public:
//////////////////////////////////////////////////////////////////
//ssrc相关信息
class RtcSSRC{
// ssrc相关信息
class RtcSSRC {
public:
uint32_t ssrc {0};
uint32_t rtx_ssrc {0};
uint32_t ssrc { 0 };
uint32_t rtx_ssrc { 0 };
std::string cname;
std::string msid;
std::string mslabel;
std::string label;
bool empty() const {return ssrc == 0 && cname.empty();}
bool empty() const { return ssrc == 0 && cname.empty(); }
};
//rtc传输编码方案
class RtcCodecPlan{
// rtc传输编码方案
class RtcCodecPlan {
public:
using Ptr = std::shared_ptr<RtcCodecPlan>;
uint8_t pt;
std::string codec;
uint32_t sample_rate;
//音频时有效
// 音频时有效
uint32_t channel = 0;
//rtcp反馈
// rtcp反馈
std::set<std::string> rtcp_fb;
std::map<std::string/*key*/, std::string/*value*/, StrCaseCompare> fmtp;
std::map<std::string /*key*/, std::string /*value*/, StrCaseCompare> fmtp;
std::string getFmtp(const char *key) const;
};
//rtc 媒体描述
class RtcMedia{
// rtc 媒体描述
class RtcMedia {
public:
TrackType type{TrackType::TrackInvalid};
TrackType type { TrackType::TrackInvalid };
std::string mid;
uint16_t port{0};
uint16_t port { 0 };
SdpConnection addr;
SdpBandwidth bandwidth;
std::string proto;
RtpDirection direction{RtpDirection::invalid};
RtpDirection direction { RtpDirection::invalid };
std::vector<RtcCodecPlan> plan;
//////// rtp ////////
@ -629,20 +616,20 @@ public:
std::vector<std::string> rtp_rids;
//////// rtcp ////////
bool rtcp_mux{false};
bool rtcp_rsize{false};
bool rtcp_mux { false };
bool rtcp_rsize { false };
SdpAttrRtcp rtcp_addr;
//////// ice ////////
bool ice_trickle{false};
bool ice_lite{false};
bool ice_renomination{false};
bool ice_trickle { false };
bool ice_lite { false };
bool ice_renomination { false };
std::string ice_ufrag;
std::string ice_pwd;
std::vector<SdpAttrCandidate> candidate;
//////// dtls ////////
DtlsRole role{DtlsRole::invalid};
DtlsRole role { DtlsRole::invalid };
SdpAttrFingerprint fingerprint;
//////// extmap ////////
@ -650,7 +637,7 @@ public:
//////// sctp ////////////
SdpAttrSctpMap sctpmap;
uint32_t sctp_port{0};
uint32_t sctp_port { 0 };
void checkValid() const;
const RtcCodecPlan *getPlan(uint8_t pt) const;
@ -679,7 +666,7 @@ public:
void checkValid() const;
std::string toString() const;
std::string toRtspSdp() const;
const RtcMedia *getMedia(TrackType type) const;
const RtcMedia *getMedia(TrackType type) const;
bool supportRtcpFb(const std::string &name, TrackType type = TrackType::TrackVideo) const;
bool supportSimulcast() const;
bool isOnlyDatachannel() const;
@ -705,7 +692,7 @@ public:
std::string ice_ufrag;
std::string ice_pwd;
RtpDirection direction{RtpDirection::invalid};
RtpDirection direction { RtpDirection::invalid };
SdpAttrFingerprint fingerprint;
std::set<std::string> rtcp_fb;
@ -752,6 +739,6 @@ private:
~SdpConst() = delete;
};
}// namespace mediakit
} // namespace mediakit
#endif //ZLMEDIAKIT_SDP_H
#endif // ZLMEDIAKIT_SDP_H

1
webrtc/WebRtcEchoTest.h
View File

@ -27,7 +27,6 @@ protected:
void onRtp(const char *buf, size_t len, uint64_t stamp_ms) override;
void onRtcp(const char *buf, size_t len) override;
void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) override {};
void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) override {};

8
webrtc/WebRtcPlayer.cpp
View File

@ -17,9 +17,8 @@ namespace mediakit {
WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const MediaInfo &info,
bool preferred_tcp) {
WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info, preferred_tcp), [](WebRtcPlayer *ptr) {
const MediaInfo &info) {
WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info), [](WebRtcPlayer *ptr) {
ptr->onDestory();
delete ptr;
});
@ -29,8 +28,7 @@ WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const MediaInfo &info,
bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
const MediaInfo &info) : WebRtcTransportImp(poller) {
_media_info = info;
_play_src = src;
CHECK(src);

5
webrtc/WebRtcPlayer.h
View File

@ -19,7 +19,7 @@ namespace mediakit {
class WebRtcPlayer : public WebRtcTransportImp {
public:
using Ptr = std::shared_ptr<WebRtcPlayer>;
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool preferred_tcp = false);
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
MediaInfo getMediaInfo() { return _media_info; }
protected:
@ -27,10 +27,9 @@ protected:
void onStartWebRTC() override;
void onDestory() override;
void onRtcConfigure(RtcConfigure &configure) const override;
void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
private:
WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool preferred_tcp);
WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
private:
//媒体相关元数据

8
webrtc/WebRtcPusher.cpp
View File

@ -20,9 +20,8 @@ WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const std::shared_ptr<void> &ownership,
const MediaInfo &info,
const ProtocolOption &option,
bool preferred_tcp) {
WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option,preferred_tcp), [](WebRtcPusher *ptr) {
const ProtocolOption &option) {
WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
ptr->onDestory();
delete ptr;
});
@ -34,8 +33,7 @@ WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const std::shared_ptr<void> &ownership,
const MediaInfo &info,
const ProtocolOption &option,
bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
const ProtocolOption &option) : WebRtcTransportImp(poller) {
_media_info = info;
_push_src = src;
_push_src_ownership = ownership;

5
webrtc/WebRtcPusher.h
View File

@ -20,8 +20,7 @@ class WebRtcPusher : public WebRtcTransportImp, public MediaSourceEvent {
public:
using Ptr = std::shared_ptr<WebRtcPusher>;
static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src,
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool preferred_tcp = false);
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
protected:
///////WebRtcTransportImp override///////
@ -53,7 +52,7 @@ protected:
private:
WebRtcPusher(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src,
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool preferred_tcp);
const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
private:
bool _simulcast = false;

186
webrtc/WebRtcTransport.cpp
View File

@ -31,7 +31,6 @@
#define RTP_CNAME "zlmediakit-rtp"
#define RTP_LABEL "zlmediakit-label"
#define RTP_MSLABEL "zlmediakit-mslabel"
#define RTP_MSID RTP_MSLABEL " " RTP_LABEL
using namespace std;
@ -55,6 +54,9 @@ const string kStartBitrate = RTC_FIELD "start_bitrate";
const string kMaxBitrate = RTC_FIELD "max_bitrate";
const string kMinBitrate = RTC_FIELD "min_bitrate";
// 数据通道设置
const string kDataChannelEcho = RTC_FIELD "datachannel_echo";
static onceToken token([]() {
mINI::Instance()[kTimeOutSec] = 15;
mINI::Instance()[kExternIP] = "";
@ -65,6 +67,8 @@ static onceToken token([]() {
mINI::Instance()[kStartBitrate] = 0;
mINI::Instance()[kMaxBitrate] = 0;
mINI::Instance()[kMinBitrate] = 0;
mINI::Instance()[kDataChannelEcho] = true;
});
} // namespace RTC
@ -250,22 +254,47 @@ void WebRtcTransport::OnDtlsTransportApplicationDataReceived(
#ifdef ENABLE_SCTP
void WebRtcTransport::OnSctpAssociationConnecting(RTC::SctpAssociation *sctpAssociation) {
TraceL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpConnecting, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationConnected(RTC::SctpAssociation *sctpAssociation) {
InfoL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpConnected, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationFailed(RTC::SctpAssociation *sctpAssociation) {
WarnL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpFailed, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationClosed(RTC::SctpAssociation *sctpAssociation) {
InfoL << getIdentifier();
try {
NOTICE_EMIT(BroadcastRtcSctpConnectArgs, Broadcast::kBroadcastRtcSctpClosed, *this);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
void WebRtcTransport::OnSctpAssociationSendData(
RTC::SctpAssociation *sctpAssociation, const uint8_t *data, size_t len) {
try {
NOTICE_EMIT(BroadcastRtcSctpSendArgs, Broadcast::kBroadcastRtcSctpSend, *this, data, len);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
_dtls_transport->SendApplicationData(data, len);
}
@ -274,8 +303,18 @@ void WebRtcTransport::OnSctpAssociationMessageReceived(
InfoL << getIdentifier() << " " << streamId << " " << ppid << " " << len << " " << string((char *)msg, len);
RTC::SctpStreamParameters params;
params.streamId = streamId;
// 回显数据
_sctp->SendSctpMessage(params, ppid, msg, len);
GET_CONFIG(bool, datachannel_echo, Rtc::kDataChannelEcho);
if (datachannel_echo) {
// 回显数据
_sctp->SendSctpMessage(params, ppid, msg, len);
}
try {
NOTICE_EMIT(BroadcastRtcSctpReceivedArgs, Broadcast::kBroadcastRtcSctpReceived, *this, streamId, ppid, msg, len);
} catch (std::exception &ex) {
WarnL << "Exception occurred: " << ex.what();
}
}
#endif
@ -339,6 +378,12 @@ void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote) {
}
void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
SdpAttrFingerprint fingerprint;
fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
configure.setDefaultSetting(
_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
// 开启remb后关闭twcc，因为开启twcc后remb无效
GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
configure.enableTWCC(!remb_bit_rate);
@ -368,12 +413,7 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer) {
setRemoteDtlsFingerprint(*_offer_sdp);
//// sdp 配置 ////
SdpAttrFingerprint fingerprint;
fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
RtcConfigure configure;
configure.setDefaultSetting(
_ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
onRtcConfigure(configure);
//// 生成answer sdp ////
@ -392,10 +432,6 @@ static bool isDtls(char *buf) {
return ((*buf > 19) && (*buf < 64));
}
static string getPeerAddress(RTC::TransportTuple *tuple) {
return tuple->get_peer_ip();
}
void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tuple) {
if (RTC::StunPacket::IsStun((const uint8_t *)buf, len)) {
std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *)buf, len));
@ -412,7 +448,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
}
if (isRtp(buf, len)) {
if (!_srtp_session_recv) {
WarnL << "received rtp packet when dtls not completed from:" << getPeerAddress(tuple);
WarnL << "received rtp packet when dtls not completed from:" << tuple->get_peer_ip();
return;
}
if (_srtp_session_recv->DecryptSrtp((uint8_t *)buf, &len)) {
@ -422,7 +458,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
}
if (isRtcp(buf, len)) {
if (!_srtp_session_recv) {
WarnL << "received rtcp packet when dtls not completed from:" << getPeerAddress(tuple);
WarnL << "received rtcp packet when dtls not completed from:" << tuple->get_peer_ip();
return;
}
if (_srtp_session_recv->DecryptSrtcp((uint8_t *)buf, &len)) {
@ -494,8 +530,7 @@ void WebRtcTransportImp::OnDtlsTransportApplicationDataReceived(const RTC::DtlsT
#endif
}
WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp)
: WebRtcTransport(poller), _preferred_tcp(preferred_tcp) {
WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
InfoL << getIdentifier();
}
@ -635,7 +670,7 @@ void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
});
for (auto &m : sdp.media) {
m.addr.reset();
m.addr.address = extern_ips.empty() ? _localIp.empty() ? SockUtil::get_local_ip() : _localIp : extern_ips[0];
m.addr.address = extern_ips.empty() ? _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip : extern_ips[0];
m.rtcp_addr.reset();
m.rtcp_addr.address = m.addr.address;
@ -667,9 +702,9 @@ void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
// 发送的ssrc我们随便定义，因为在发送rtp时会修改为此值
ssrc.ssrc = m.type + RTP_SSRC_OFFSET;
ssrc.cname = RTP_CNAME;
ssrc.label = RTP_LABEL;
ssrc.label = std::string(RTP_LABEL) + '-' + m.mid;
ssrc.mslabel = RTP_MSLABEL;
ssrc.msid = RTP_MSID;
ssrc.msid = ssrc.mslabel + ' ' + ssrc.label;
if (m.getRelatedRtxPlan(m.plan[0].pt)) {
// rtx ssrc
@ -730,7 +765,7 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
return ret;
});
if (extern_ips.empty()) {
std::string local_ip = _localIp.empty() ? SockUtil::get_local_ip() : _localIp;
std::string local_ip = _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip;
if (local_udp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_udp_port, 120, "udp")); }
if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_tcp_port, _preferred_tcp ? 125 : 115, "tcp")); }
} else {
@ -744,12 +779,16 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
}
}
void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
_cands = std::move(cands);
void WebRtcTransportImp::setPreferredTcp(bool flag) {
_preferred_tcp = flag;
}
void WebRtcTransportImp::setLocalIp(const std::string &localIp) {
_localIp = localIp;
void WebRtcTransportImp::setLocalIp(std::string local_ip) {
_local_ip = std::move(local_ip);
}
void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
_cands = std::move(cands);
}
///////////////////////////////////////////////////////////////////
@ -1239,21 +1278,14 @@ void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
_map_creator[type] = std::move(cb);
}
std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& offer) {
return const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer);
}
void setLocalIp(const WebRtcInterface& exchanger, const std::string& localIp) {
return const_cast<WebRtcInterface &>(exchanger).setLocalIp(localIp);
}
void WebRtcPluginManager::setListener(Listener cb) {
lock_guard<mutex> lck(_mtx_creator);
_listener = std::move(cb);
}
void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateRtc &cb_in) {
onCreateRtc cb;
void WebRtcPluginManager::negotiateSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateWebRtc &cb_in) {
onCreateWebRtc cb;
lock_guard<mutex> lck(_mtx_creator);
if (_listener) {
auto listener = _listener;
@ -1269,21 +1301,19 @@ void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, cons
auto it = _map_creator.find(type);
if (it == _map_creator.end()) {
cb(WebRtcException(SockException(Err_other, "the type can not supported")));
cb_in(WebRtcException(SockException(Err_other, "the type can not supported")));
return;
}
it->second(sender, args, cb);
}
void echo_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void echo_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
}
void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void push_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
MediaInfo info(args["url"]);
bool preferred_tcp = args["preferred_tcp"];
Broadcast::PublishAuthInvoker invoker = [cb, info, preferred_tcp](const string &err, const ProtocolOption &option) mutable {
Broadcast::PublishAuthInvoker invoker = [cb, info](const string &err, const ProtocolOption &option) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
@ -1322,7 +1352,7 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
push_src_ownership = push_src->getOwnership();
push_src->setProtocolOption(option);
}
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option, preferred_tcp);
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option);
push_src->setListener(rtc);
cb(*rtc);
};
@ -1335,12 +1365,10 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
}
}
void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void play_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
MediaInfo info(args["url"]);
bool preferred_tcp = args["preferred_tcp"];
auto session_ptr = static_pointer_cast<Session>(sender.shared_from_this());
Broadcast::AuthInvoker invoker = [cb, info, session_ptr, preferred_tcp](const string &err) mutable {
Broadcast::AuthInvoker invoker = [cb, info, session_ptr](const string &err) mutable {
if (!err.empty()) {
cb(WebRtcException(SockException(Err_other, err)));
return;
@ -1356,7 +1384,7 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
}
// 还原成rtc，目的是为了hook时识别哪种播放协议
info.schema = "rtc";
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info, preferred_tcp);
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info);
cb(*rtc);
});
};
@ -1369,39 +1397,63 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
}
}
static void set_webrtc_cands(const WebRtcArgs &args, const WebRtcInterface &rtc) {
vector<SdpAttrCandidate> cands;
static void setWebRtcArgs(const WebRtcArgs &args, WebRtcInterface &rtc) {
{
auto cand_str = trim(args["cand_udp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
static auto is_vaild_ip = [](const std::string &ip) -> bool {
int a, b, c, d;
return sscanf(ip.c_str(), "%d.%d.%d.%d", &a, &b, &c, &d) == 4;
};
std::string host = args["Host"];
if (!host.empty()) {
auto local_ip = host.substr(0, host.find(':'));
if (!is_vaild_ip(local_ip) || local_ip == "127.0.0.1") {
local_ip = "";
}
rtc.setLocalIp(std::move(local_ip));
}
}
bool preferred_tcp = args["preferred_tcp"];
{
rtc.setPreferredTcp(preferred_tcp);
}
{
vector<SdpAttrCandidate> cands;
{
auto cand_str = trim(args["cand_udp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
// udp优先
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 100 : 120, "udp");
cands.emplace_back(std::move(*ice_cand));
}
}
{
auto cand_str = trim(args["cand_tcp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
// tcp模式
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 120 : 100, "tcp");
cands.emplace_back(std::move(*ice_cand));
}
}
if (!cands.empty()) {
// udp优先
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), 120, "udp");
cands.emplace_back(std::move(*ice_cand));
rtc.setIceCandidate(std::move(cands));
}
}
{
auto cand_str = trim(args["cand_tcp"]);
auto ip_port = toolkit::split(cand_str, ":");
if (ip_port.size() == 2) {
// tcp模式
auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), 100, "tcp");
cands.emplace_back(std::move(*ice_cand));
}
}
if (!cands.empty()) {
// udp优先
const_cast<WebRtcInterface &>(rtc).setIceCandidate(std::move(cands));
}
}
static onceToken s_rtc_auto_register([]() {
#if !defined (NDEBUG)
// debug模式才开启echo插件
WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
#endif
WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
WebRtcPluginManager::Instance().setListener([](Session &sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc) {
set_webrtc_cands(args, rtc);
setWebRtcArgs(args, const_cast<WebRtcInterface&>(rtc));
});
});

32
webrtc/WebRtcTransport.h
View File

@ -42,13 +42,10 @@ public:
virtual const std::string& getIdentifier() const = 0;
virtual const std::string& deleteRandStr() const { static std::string s_null; return s_null; }
virtual void setIceCandidate(std::vector<SdpAttrCandidate> cands) {}
virtual void setLocalIp(const std::string &localIp) {}
virtual void setLocalIp(std::string localIp) {}
virtual void setPreferredTcp(bool flag) {}
};
std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& offer);
void setLocalIp(const WebRtcInterface &exchanger, const std::string &localIp);
class WebRtcException : public WebRtcInterface {
public:
WebRtcException(const SockException &ex) : _ex(ex) {};
@ -88,7 +85,7 @@ public:
* @param offer offer sdp
* @return answer sdp
*/
std::string getAnswerSdp(const std::string &offer) override;
std::string getAnswerSdp(const std::string &offer) override final;
/**
* 获取对象唯一id
@ -252,14 +249,16 @@ public:
void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
void setIceCandidate(std::vector<SdpAttrCandidate> cands) override;
void removeTuple(RTC::TransportTuple* tuple);
void safeShutdown(const SockException &ex);
void setLocalIp(const std::string &localIp) override;
void setPreferredTcp(bool flag) override;
void setLocalIp(std::string local_ip) override;
void setIceCandidate(std::vector<SdpAttrCandidate> cands) override;
protected:
void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override;
WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp = false);
WebRtcTransportImp(const EventPoller::Ptr &poller);
void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
void onStartWebRTC() override;
void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) override;
@ -273,7 +272,7 @@ protected:
void onCreate() override;
void onDestory() override;
void onShutdown(const SockException &ex) override;
virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) = 0;
virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) {}
void updateTicker();
float getLossRate(TrackType type);
void onRtcpBye() override;
@ -289,7 +288,7 @@ private:
void onCheckAnswer(RtcSession &sdp);
private:
bool _preferred_tcp;
bool _preferred_tcp = false;
uint16_t _rtx_seq[2] = {0, 0};
//用掉的总流量
uint64_t _bytes_usage = 0;
@ -310,8 +309,8 @@ private:
//根据接收rtp的pt获取相关信息
std::unordered_map<uint8_t/*pt*/, std::unique_ptr<WrappedMediaTrack>> _pt_to_track;
std::vector<SdpAttrCandidate> _cands;
//源访问的hostip
std::string _localIp;
//http访问时的host ip
std::string _local_ip;
};
class WebRtcTransportManager {
@ -333,21 +332,20 @@ private:
class WebRtcArgs : public std::enable_shared_from_this<WebRtcArgs> {
public:
virtual ~WebRtcArgs() = default;
virtual variant operator[](const std::string &key) const = 0;
};
using onCreateWebRtc = std::function<void(const WebRtcInterface &rtc)>;
class WebRtcPluginManager {
public:
using onCreateRtc = std::function<void(const WebRtcInterface &rtc)>;
using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateRtc &cb)>;
using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb)>;
using Listener = std::function<void(Session &sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc)>;
static WebRtcPluginManager &Instance();
void registerPlugin(const std::string &type, Plugin cb);
void getAnswerSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateRtc &cb);
void setListener(Listener cb);
void negotiateSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateWebRtc &cb);
private:
WebRtcPluginManager() = default;

30
www/webrtc/index.html
View File

@ -115,17 +115,10 @@
document.getElementsByName("method").forEach((el,idx) => {
el.checked = el.value === type;
el.onclick = function(e) {
let url = new URL(document.getElementById('streamUrl').value);
const url = new URL(document.getElementById('streamUrl').value);
url.searchParams.set("type",el.value);
document.getElementById('streamUrl').value = url.toString();
if(el.value == "play"){
recvOnly = true;
}else if(el.value == "echo"){
recvOnly = false;
}else{
recvOnly = false;
}
recvOnly = 'play' === el.value;
};
});
@ -145,6 +138,25 @@
let h = parseInt(res.pop());
let w = parseInt(res.pop());
const url = new URL(document.getElementById('streamUrl').value);
const newUrl = new URL(window.location.href);
let count = 0;
if (url.searchParams.has('app')) {
newUrl.searchParams.set('app', url.searchParams.get('app'));
count++;
}
if (url.searchParams.has('stream')) {
newUrl.searchParams.set('stream', url.searchParams.get('stream'));
count++;
}
if (url.searchParams.has('type')) {
newUrl.searchParams.set('type', url.searchParams.get('type'));
count++;
}
if (count > 0) {
window.history.pushState(null, null, newUrl);
}
player = new ZLMRTCClient.Endpoint(
{
element: document.getElementById('video'),// video 标签