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张翔宇
ZLMediaKit
Commits
33d2e713
Commit
33d2e713
authored
Apr 02, 2021
by
ziyue
Browse files
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合并代码,去除冗余代码
parents
1978748f
a328520c
隐藏空白字符变更
内嵌
并排
正在显示
8 个修改的文件
包含
45 行增加
和
51 行删除
+45
-51
server/Process.cpp
+7
-0
src/Extension/G711.h
+1
-1
src/Rtmp/RtmpSession.cpp
+5
-10
src/Rtmp/RtmpSession.h
+1
-1
src/Rtp/RtpProcess.cpp
+3
-0
webrtc/Sdp.cpp
+17
-15
webrtc/Sdp.h
+0
-5
webrtc/WebRtcTransport.cpp
+11
-19
没有找到文件。
server/Process.cpp
查看文件 @
33d2e713
...
...
@@ -108,10 +108,17 @@ void Process::run(const string &cmd, const string &log_file_tmp) {
}
fprintf
(
stderr
,
"
\r\n\r\n
#### pid=%d,cmd=%s #####
\r\n\r\n
"
,
getpid
(),
cmd
.
data
());
#ifndef ANDROID
//关闭父进程继承的fd
for
(
int
i
=
3
;
i
<
getdtablesize
();
i
++
)
{
::
close
(
i
);
}
#else
//关闭父进程继承的fd
for
(
int
i
=
3
;
i
<
1024
;
i
++
)
{
::
close
(
i
);
}
#endif
auto
params
=
split
(
cmd
,
" "
);
// memory leak in child process, it's ok.
...
...
src/Extension/G711.h
查看文件 @
33d2e713
...
...
@@ -22,7 +22,7 @@ namespace mediakit{
class
G711Track
:
public
AudioTrackImp
{
public
:
using
Ptr
=
std
::
shared_ptr
<
G711Track
>
;
G711Track
(
CodecId
codecId
,
int
sample_rate
,
int
channels
,
int
sample_bit
)
:
AudioTrackImp
(
codecId
,
sample_rate
,
channels
,
sample_bit
)
{}
G711Track
(
CodecId
codecId
,
int
sample_rate
,
int
channels
,
int
sample_bit
)
:
AudioTrackImp
(
codecId
,
8000
,
1
,
16
)
{}
private
:
Sdp
::
Ptr
getSdp
()
override
;
...
...
src/Rtmp/RtmpSession.cpp
查看文件 @
33d2e713
...
...
@@ -415,17 +415,11 @@ void RtmpSession::onCmd_pause(AMFDecoder &dec) {
}
void
RtmpSession
::
setMetaData
(
AMFDecoder
&
dec
)
{
if
(
!
_publisher_src
)
{
throw
std
::
runtime_error
(
"not a publisher"
);
}
std
::
string
type
=
dec
.
load
<
std
::
string
>
();
if
(
type
!=
"onMetaData"
)
{
throw
std
::
runtime_error
(
"can only set metadata"
);
}
auto
metadata
=
dec
.
load
<
AMFValue
>
();
// dumpMetadata(metadata);
_publisher_src
->
setMetaData
(
metadata
);
_set_meta_data
=
true
;
_publisher_metadata
=
dec
.
load
<
AMFValue
>
();
}
void
RtmpSession
::
onProcessCmd
(
AMFDecoder
&
dec
)
{
...
...
@@ -478,7 +472,8 @@ void RtmpSession::onRtmpChunk(RtmpPacket::Ptr packet) {
case
MSG_AUDIO
:
case
MSG_VIDEO
:
{
if
(
!
_publisher_src
)
{
throw
std
::
runtime_error
(
"Not a rtmp publisher!"
);
WarnL
<<
"Not a rtmp publisher!"
;
return
;
}
GET_CONFIG
(
bool
,
rtmp_modify_stamp
,
Rtmp
::
kModifyStamp
);
if
(
rtmp_modify_stamp
)
{
...
...
@@ -487,9 +482,9 @@ void RtmpSession::onRtmpChunk(RtmpPacket::Ptr packet) {
chunk_data
.
time_stamp
=
(
uint32_t
)
dts_out
;
}
if
(
!
_set_meta_data
&&
!
chunk_data
.
isCfgFrame
()
)
{
if
(
!
_set_meta_data
)
{
_set_meta_data
=
true
;
_publisher_src
->
setMetaData
(
TitleMeta
().
getMetadata
());
_publisher_src
->
setMetaData
(
_publisher_metadata
?
_publisher_metadata
:
TitleMeta
().
getMetadata
());
}
_publisher_src
->
onWrite
(
std
::
move
(
packet
));
break
;
...
...
src/Rtmp/RtmpSession.h
查看文件 @
33d2e713
...
...
@@ -98,8 +98,8 @@ private:
//数据接收超时计时器
Ticker
_ticker
;
MediaInfo
_media_info
;
std
::
weak_ptr
<
RtmpMediaSource
>
_player_src
;
AMFValue
_publisher_metadata
;
std
::
shared_ptr
<
RtmpMediaSourceImp
>
_publisher_src
;
RtmpMediaSource
::
RingType
::
RingReader
::
Ptr
_ring_reader
;
};
...
...
src/Rtp/RtpProcess.cpp
查看文件 @
33d2e713
...
...
@@ -145,6 +145,9 @@ bool RtpProcess::alive() {
void
RtpProcess
::
setStopCheckRtp
(
bool
is_check
){
_stop_rtp_check
=
is_check
;
if
(
!
is_check
)
{
_last_frame_time
.
resetTime
();
}
}
void
RtpProcess
::
onDetach
()
{
...
...
webrtc/Sdp.cpp
查看文件 @
33d2e713
...
...
@@ -1150,11 +1150,9 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
rtcp_mux
=
true
;
rtcp_rsize
=
false
;
group_bundle
=
true
;
unified_plan
=
false
;
support_rtx
=
true
;
support_red
=
false
;
support_ulpfec
=
false
;
support_simulcast
=
false
;
ice_lite
=
true
;
ice_trickle
=
true
;
ice_renomination
=
false
;
...
...
@@ -1226,26 +1224,19 @@ void RtcConfigure::addCandidate(const SdpAttrCandidate &candidate, TrackType typ
}
}
shared_ptr
<
RtcSession
>
RtcConfigure
::
createOffer
(){
shared_ptr
<
RtcSession
>
ret
=
std
::
make_shared
<
RtcSession
>
();
ret
->
version
=
0
;
ret
->
origin
.
parse
(
"- 0 0 IN IP4 0.0.0.0"
);
ret
->
session_name
=
"zlmediakit_webrtc_session"
;
ret
->
session_info
=
"zlmediakit_webrtc_session"
;
ret
->
connection
.
parse
(
"IN IP4 0.0.0.0"
);
ret
->
msid_semantic
.
parse
(
"WMS *"
);
return
nullptr
;
}
shared_ptr
<
RtcSession
>
RtcConfigure
::
createAnswer
(
const
RtcSession
&
offer
){
shared_ptr
<
RtcSession
>
ret
=
std
::
make_shared
<
RtcSession
>
();
ret
->
version
=
offer
.
version
;
//todo 此处设置会话id与会话地址,貌似没什么作用
ret
->
origin
=
offer
.
origin
;
ret
->
session_name
=
offer
.
session_name
;
ret
->
msid_semantic
=
offer
.
msid_semantic
;
matchMedia
(
ret
,
TrackAudio
,
offer
.
media
,
audio
);
matchMedia
(
ret
,
TrackVideo
,
offer
.
media
,
video
);
matchMedia
(
ret
,
TrackApplication
,
offer
.
media
,
application
);
if
(
ret
->
media
.
empty
())
{
throw
std
::
invalid_argument
(
"生成的answer sdp中媒体个数为0"
);
}
return
ret
;
}
...
...
@@ -1259,7 +1250,7 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
continue
;
}
if
(
offer_media
.
ice_lite
&&
configure
.
ice_lite
)
{
WarnL
<<
"
offer sdp开启了ice_lite模式,但是answer sdp配置为不支持
"
;
WarnL
<<
"
answer sdp配置为ice_lite模式,与offer sdp中的ice_lite模式冲突
"
;
continue
;
}
const
RtcCodecPlan
*
offer_plan_ptr
=
nullptr
;
...
...
@@ -1283,16 +1274,21 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
answer_media
.
type
=
offer_media
.
type
;
answer_media
.
mid
=
offer_media
.
mid
;
answer_media
.
proto
=
offer_media
.
proto
;
//todo(此处设置rtp端口,貌似没什么作用)
answer_media
.
port
=
offer_media
.
port
;
//todo(此处设置rtp的ip地址,貌似没什么作用)
answer_media
.
addr
=
offer_media
.
addr
;
//todo(此处设置rtcp地址,貌似没什么作用)
answer_media
.
rtcp_addr
=
offer_media
.
rtcp_addr
;
answer_media
.
rtcp_mux
=
offer_media
.
rtcp_mux
&&
configure
.
rtcp_mux
;
answer_media
.
rtcp_rsize
=
offer_media
.
rtcp_rsize
&&
configure
.
rtcp_rsize
;
answer_media
.
rtcp_addr
=
offer_media
.
rtcp_addr
;
answer_media
.
ice_trickle
=
offer_media
.
ice_trickle
&&
configure
.
ice_trickle
;
answer_media
.
ice_renomination
=
offer_media
.
ice_renomination
&&
configure
.
ice_renomination
;
answer_media
.
ice_ufrag
=
configure
.
ice_ufrag
;
answer_media
.
ice_pwd
=
configure
.
ice_pwd
;
answer_media
.
fingerprint
=
configure
.
fingerprint
;
answer_media
.
ice_lite
=
configure
.
ice_lite
;
answer_media
.
candidate
=
configure
.
candidate
;
switch
(
offer_media
.
role
)
{
case
DtlsRole
:
:
actpass
:
case
DtlsRole
:
:
active
:
{
...
...
@@ -1332,7 +1328,11 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
}
default
:
continue
;
}
//添加媒体plan
answer_media
.
plan
.
emplace_back
(
*
offer_plan_ptr
);
//添加rtx,red,ulpfec plan
if
(
configure
.
support_red
||
configure
.
support_rtx
||
configure
.
support_ulpfec
)
{
for
(
auto
&
plan
:
offer_media
.
plan
)
{
if
(
!
strcasecmp
(
plan
.
codec
.
data
(),
"rtx"
))
{
...
...
@@ -1382,6 +1382,8 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
offer_rtcp_fb
.
emplace_back
(
fp
);
}
}
//修改为我们支持的rtcp-fb类型
answer_media
.
plan
[
0
].
rtcp_fb
.
swap
(
offer_rtcp_fb
);
ret
->
media
.
emplace_back
(
answer_media
);
return
;
...
...
webrtc/Sdp.h
查看文件 @
33d2e713
...
...
@@ -599,7 +599,6 @@ public:
RtcSSRC
rtx_ssrc
;
//////// simulcast ////////
bool
simulcast
{
false
};
RtcSSRC
rtp_ssrc_low
;
RtcSSRC
rtp_ssrc_mid
;
RtcSSRC
rtp_ssrc_high
;
...
...
@@ -663,12 +662,9 @@ public:
bool
rtcp_mux
;
bool
rtcp_rsize
;
bool
group_bundle
;
bool
unified_plan
;
bool
support_rtx
;
bool
support_red
;
bool
support_ulpfec
;
bool
support_simulcast
;
bool
ice_lite
;
bool
ice_trickle
;
bool
ice_renomination
;
...
...
@@ -696,7 +692,6 @@ public:
const
SdpAttrFingerprint
&
fingerprint
);
void
addCandidate
(
const
SdpAttrCandidate
&
candidate
,
TrackType
type
=
TrackInvalid
);
shared_ptr
<
RtcSession
>
createOffer
();
shared_ptr
<
RtcSession
>
createAnswer
(
const
RtcSession
&
offer
);
private
:
...
...
webrtc/WebRtcTransport.cpp
查看文件 @
33d2e713
...
...
@@ -88,6 +88,17 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer){
RtcConfigure
configure
;
configure
.
setDefaultSetting
(
ice_server_
->
GetUsernameFragment
(),
ice_server_
->
GetPassword
(),
RtpDirection
::
recvonly
,
fingerprint
);
SdpAttrCandidate
candidate
;
candidate
.
foundation
=
"udpcandidate"
;
candidate
.
component
=
1
;
candidate
.
transport
=
"udp"
;
candidate
.
priority
=
getSSRC
();
candidate
.
address
=
getIP
();
candidate
.
port
=
getPort
();
candidate
.
type
=
"host"
;
configure
.
addCandidate
(
candidate
);
_answer_sdp
=
configure
.
createAnswer
(
*
_offer_sdp
);
//设置远端dtls签名
...
...
@@ -104,23 +115,6 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer){
throw
std
::
invalid_argument
(
"支持group BUNDLE模式"
);
}
SdpAttrCandidate
candidate
;
candidate
.
foundation
=
"udpcandidate"
;
candidate
.
component
=
1
;
candidate
.
transport
=
"udp"
;
candidate
.
priority
=
getSSRC
();
candidate
.
address
=
getIP
();
candidate
.
port
=
getPort
();
candidate
.
type
=
"host"
;
for
(
auto
&
m
:
_answer_sdp
->
media
)
{
m
.
candidate
.
emplace_back
(
candidate
);
m
.
port
=
getPort
();
m
.
rtcp_addr
.
address
=
getIP
();
m
.
rtcp_addr
.
port
=
getPort
();
m
.
addr
.
address
=
getIP
();
}
_answer_sdp
->
connection
.
address
=
getIP
();
auto
str
=
_answer_sdp
->
toString
();
InfoL
<<
"
\r\n
"
<<
str
;
return
str
;
...
...
@@ -156,12 +150,10 @@ void WebRtcTransport::OnInputDataPacket(char *buf, size_t len, RTC::TransportTup
}
if
(
is_rtp
(
buf
))
{
RtpHeader
*
header
=
(
RtpHeader
*
)
buf
;
InfoL
<<
"rtp:"
<<
header
->
dumpString
(
len
);
return
;
}
if
(
is_rtcp
(
buf
))
{
RtcpHeader
*
header
=
(
RtcpHeader
*
)
buf
;
// InfoL << "rtcp:" << header->dumpString();
return
;
}
}
...
...
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