Skip to content
项目
群组
代码片段
帮助
当前项目
正在载入...
登录 / 注册
切换导航面板
Z
ZLMediaKit
概览
Overview
Details
Activity
Cycle Analytics
版本库
Repository
Files
Commits
Branches
Tags
Contributors
Graph
Compare
Charts
问题
0
Issues
0
列表
Board
标记
里程碑
合并请求
0
Merge Requests
0
CI / CD
CI / CD
流水线
作业
日程表
图表
维基
Wiki
代码片段
Snippets
成员
Collapse sidebar
Close sidebar
活动
图像
聊天
创建新问题
作业
提交
Issue Boards
Open sidebar
张翔宇
ZLMediaKit
Commits
d5eb486f
Commit
d5eb486f
authored
May 08, 2021
by
xia-chu
Browse files
Options
Browse Files
Download
Email Patches
Plain Diff
修改rtp的ext id
优化代码结构
parent
9f95f743
隐藏空白字符变更
内嵌
并排
正在显示
2 个修改的文件
包含
58 行增加
和
29 行删除
+58
-29
webrtc/WebRtcTransport.cpp
+53
-26
webrtc/WebRtcTransport.h
+5
-3
没有找到文件。
webrtc/WebRtcTransport.cpp
查看文件 @
d5eb486f
...
...
@@ -390,10 +390,8 @@ bool WebRtcTransportImp::canRecvRtp() const{
}
void
WebRtcTransportImp
::
onStartWebRTC
()
{
//获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息
for
(
auto
&
m
:
getSdp
(
SdpType
::
offer
).
media
)
{
if
(
m
.
type
==
TrackVideo
)
{
_recv_video_ssrc
=
m
.
rtp_ssrc
.
ssrc
;
}
for
(
auto
&
plan
:
m
.
plan
)
{
auto
hit_pan
=
getSdp
(
SdpType
::
answer
).
getMedia
(
m
.
type
)
->
getPlan
(
plan
.
pt
);
if
(
!
hit_pan
)
{
...
...
@@ -413,6 +411,14 @@ void WebRtcTransportImp::onStartWebRTC() {
onBeforeSortedRtp
(
ref
,
rtp
);
});
}
if
(
m
.
type
!=
TrackApplication
)
{
//记录rtp ext类型与id的关系,方便接收或发送rtp时修改rtp ext id
for
(
auto
&
pr
:
m
.
extmap
)
{
auto
ext_type
=
RtpExt
::
getExtType
(
pr
.
second
.
ext
);
_rtp_ext_id_to_type
.
emplace
(
pr
.
second
.
id
,
ext_type
);
_rtp_ext_type_to_id
.
emplace
(
ext_type
,
pr
.
second
.
id
);
}
}
}
if
(
canRecvRtp
())
{
...
...
@@ -431,6 +437,19 @@ void WebRtcTransportImp::onStartWebRTC() {
strongSelf
->
onSendRtp
(
rtp
,
++
i
==
pkt
->
size
());
});
});
RtcSession
rtsp_send_sdp
;
rtsp_send_sdp
.
loadFrom
(
_play_src
->
getSdp
(),
false
);
for
(
auto
&
m
:
getSdp
(
SdpType
::
answer
).
media
)
{
if
(
m
.
type
==
TrackApplication
)
{
continue
;
}
auto
rtsp_media
=
rtsp_send_sdp
.
getMedia
(
m
.
type
);
if
(
rtsp_media
&&
getCodecId
(
rtsp_media
->
plan
[
0
].
codec
)
==
getCodecId
(
m
.
plan
[
0
].
codec
))
{
//记录发送rtp时约定的pt,届时发送rtp时需要修改pt
_send_rtp_pt
[
m
.
type
]
=
m
.
plan
[
0
].
pt
;
}
}
}
}
...
...
@@ -440,6 +459,7 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){
return
;
}
//修改sdp的ip、端口信息
GET_CONFIG
(
string
,
extern_ip
,
RTC
::
kExternIP
);
for
(
auto
&
m
:
sdp
.
media
)
{
m
.
addr
.
reset
();
...
...
@@ -455,9 +475,6 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){
return
;
}
RtcSession
rtsp_send_sdp
;
rtsp_send_sdp
.
loadFrom
(
_play_src
->
getSdp
(),
false
);
for
(
auto
&
m
:
sdp
.
media
)
{
if
(
m
.
type
==
TrackApplication
)
{
continue
;
...
...
@@ -470,11 +487,6 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){
m
.
rtx_ssrc
.
ssrc
=
RTX_SSRC_OFFSET
+
m
.
rtp_ssrc
.
ssrc
;
m
.
rtx_ssrc
.
cname
=
RTX_CNAME
;
}
auto
rtsp_media
=
rtsp_send_sdp
.
getMedia
(
m
.
type
);
if
(
rtsp_media
&&
getCodecId
(
rtsp_media
->
plan
[
0
].
codec
)
==
getCodecId
(
m
.
plan
[
0
].
codec
))
{
//记录发送rtp的pt
_send_rtp_pt
[
m
.
type
]
=
m
.
plan
[
0
].
pt
;
}
}
}
...
...
@@ -557,6 +569,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
it
->
second
->
rtcp_context_recv
->
onRtcp
(
sr
);
auto
rr
=
it
->
second
->
rtcp_context_recv
->
createRtcpRR
(
sr
->
items
.
ssrc
,
sr
->
ssrc
);
sendRtcpPacket
(
rr
->
data
(),
rr
->
size
(),
true
);
}
else
{
WarnL
<<
"未识别的sr rtcp包:"
<<
sr
->
ssrc
;
}
break
;
}
...
...
@@ -568,6 +582,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
if
(
it
!=
_rtp_info_ssrc
.
end
())
{
auto
sr
=
it
->
second
->
rtcp_context_send
->
createRtcpSR
(
rr
->
items
.
ssrc
);
sendRtcpPacket
(
sr
->
data
(),
sr
->
size
(),
true
);
}
else
{
WarnL
<<
"未识别的rr rtcp包:"
<<
rr
->
ssrc
;
}
break
;
}
...
...
@@ -577,6 +593,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
for
(
auto
ssrc
:
bye
->
getSSRC
())
{
auto
it
=
_rtp_info_ssrc
.
find
(
*
ssrc
);
if
(
it
==
_rtp_info_ssrc
.
end
())
{
WarnL
<<
"未识别的bye rtcp包:"
<<
*
ssrc
;
continue
;
}
_rtp_info_pt
.
erase
(
it
->
second
->
plan
->
pt
);
...
...
@@ -587,7 +604,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
}
case
RtcpType
:
:
RTCP_PSFB
:
case
RtcpType
:
:
RTCP_RTPFB
:
{
InfoL
<<
"
\n
"
<<
rtcp
->
dumpString
();
break
;
}
default
:
break
;
...
...
@@ -617,36 +633,46 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr
//todo rtx/red/ulpfec类型的rtp先未处理
return
;
}
if
(
_pli_ticker
.
elapsedTime
()
>
2000
)
{
if
(
info
.
media
->
type
==
TrackVideo
&&
_pli_ticker
.
elapsedTime
()
>
2000
)
{
//定期发送pli请求关键帧,方便非rtc等协议
_pli_ticker
.
resetTime
();
sendRtcpPli
(
_recv_video_ssrc
);
sendRtcpPli
(
rtp
->
getSSRC
()
);
//开启remb,则发送remb包调节比特率
GET_CONFIG
(
size_t
,
remb_bit_rate
,
RTC
::
kRembBitRate
);
if
(
remb_bit_rate
&&
getSdp
(
SdpType
::
answer
).
supportRtcpFb
(
SdpConst
::
kRembRtcpFb
))
{
sendRtcpRemb
(
_recv_video_ssrc
,
remb_bit_rate
);
sendRtcpRemb
(
rtp
->
getSSRC
()
,
remb_bit_rate
);
}
}
if
(
_push_src
)
{
_push_src
->
onWrite
(
std
::
move
(
rtp
),
false
);
}
}
void
setExtType
(
RtpExt
&
ext
,
uint8_t
tp
)
{}
void
setExtType
(
RtpExt
&
ext
,
RtpExtType
tp
)
{
ext
.
setType
(
tp
);
}
template
<
typename
Type
>
static
void
changeRtpExtId
(
const
RtpPacket
::
Ptr
&
rtp
,
const
Type
&
map
)
{
auto
header
=
rtp
->
getHeader
();
auto
ext_map
=
RtpExt
::
getExtValue
(
header
,
*
(
info
.
media
)
);
auto
ext_map
=
RtpExt
::
getExtValue
(
header
);
for
(
auto
&
pr
:
ext_map
)
{
if
(
rtp
->
type
==
TrackVideo
)
{
InfoL
<<
pr
.
second
.
dumpString
();
}
else
{
DebugL
<<
pr
.
second
.
dumpString
()
;
auto
it
=
map
.
find
((
Type
::
key_type
)
pr
.
first
);
if
(
it
==
map
.
end
())
{
WarnL
<<
"未处理的rtp ext, 类型不识别:"
<<
(
int
)
pr
.
first
;
continue
;
}
//推流时修改ext id为统一的id,播放时再修改为对方设置的ext id
pr
.
second
.
setExtId
((
uint8_t
)
pr
.
first
);
}
if
(
_push_src
)
{
_push_src
->
onWrite
(
std
::
move
(
rtp
),
false
);
setExtType
(
pr
.
second
,
it
->
first
);
setExtType
(
pr
.
second
,
it
->
second
);
pr
.
second
.
setExtId
((
uint8_t
)
it
->
second
);
}
}
void
WebRtcTransportImp
::
onBeforeSortedRtp
(
const
RtpPayloadInfo
&
info
,
const
RtpPacket
::
Ptr
&
rtp
)
{
changeRtpExtId
(
rtp
,
_rtp_ext_id_to_type
);
//统计rtp收到的情况,好做rr汇报
info
.
rtcp_context_recv
->
onRtp
(
rtp
->
getSeq
(),
rtp
->
getStampMS
(),
rtp
->
size
()
-
RtpPacket
::
kRtpTcpHeaderSize
);
}
...
...
@@ -657,6 +683,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
//忽略,对方不支持该编码类型
return
;
}
changeRtpExtId
(
rtp
,
_rtp_ext_type_to_id
);
_bytes_usage
+=
rtp
->
size
()
-
RtpPacket
::
kRtpTcpHeaderSize
;
sendRtpPacket
(
rtp
->
data
()
+
RtpPacket
::
kRtpTcpHeaderSize
,
rtp
->
size
()
-
RtpPacket
::
kRtpTcpHeaderSize
,
flush
,
pt
);
//统计rtp发送情况,好做sr汇报
...
...
webrtc/WebRtcTransport.h
查看文件 @
d5eb486f
...
...
@@ -211,8 +211,6 @@ private:
Ticker
_alive_ticker
;
//pli rtcp计时器
Ticker
_pli_ticker
;
//rtc rtp推流的视频ssrc
uint32_t
_recv_video_ssrc
;
//记录协商的rtp的pt类型
uint8_t
_send_rtp_pt
[
2
]
=
{
0xFF
,
0xFF
};
//复合udp端口,接收一切rtp与rtcp
...
...
@@ -225,8 +223,12 @@ private:
RtspMediaSource
::
RingType
::
RingReader
::
Ptr
_reader
;
//根据rtp的pt获取相关信息
unordered_map
<
uint8_t
,
RtpPayloadInfo
>
_rtp_info_pt
;
//根据推流端rtp的ssrc获取相关信息
//根据推流端rt
c
p的ssrc获取相关信息
unordered_map
<
uint32_t
,
RtpPayloadInfo
*>
_rtp_info_ssrc
;
//发送rtp时需要修改rtp ext id
unordered_map
<
RtpExtType
,
uint8_t
>
_rtp_ext_type_to_id
;
//接收rtp时需要修改rtp ext id
unordered_map
<
uint8_t
,
RtpExtType
>
_rtp_ext_id_to_type
;
};
...
...
编写
预览
Markdown
格式
0%
重试
或
添加新文件
添加附件
取消
您添加了
0
人
到此讨论。请谨慎行事。
请先完成此评论的编辑!
取消
请
注册
或者
登录
后发表评论