external/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-02-13 03:41:55 +00:00
Code Issues Releases Wiki Activity

for #133, show more info about rtp.

Browse source
This commit is contained in:
winlin 2015-02-18 07:27:11 +08:00
parent 45c0b12958
commit 6a64164985
4 changed files with 41 additions and 25 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

23
trunk/src/app/srs_app_rtsp.cpp
View file

@ -128,15 +128,19 @@ int SrsRtspConn::do_cycle()
}
} else if (req->is_announce()) {
srs_assert(req->sdp);
video_id = req->sdp->video_stream_id;
audio_id = req->sdp->audio_stream_id;
video_id = ::atoi(req->sdp->video_stream_id.c_str());
audio_id = ::atoi(req->sdp->audio_stream_id.c_str());
video_codec = req->sdp->video_codec;
audio_codec = req->sdp->audio_codec;
audio_sample_rate = ::atoi(req->sdp->audio_sample_rate.c_str());
audio_channel = ::atoi(req->sdp->audio_channel.c_str());
sps = req->sdp->video_sps;
pps = req->sdp->video_pps;
asc = req->sdp->audio_sh;
srs_trace("rtsp: video(#%s, %s), audio(#%s, %s, %sHZ %schannels)",
req->sdp->video_stream_id.c_str(), req->sdp->video_codec.c_str(),
req->sdp->audio_stream_id.c_str(), req->sdp->audio_codec.c_str(),
req->sdp->audio_sample_rate.c_str(), req->sdp->audio_channel.c_str()
srs_trace("rtsp: video(#%d, %s, %s/%s), audio(#%d, %s, %s/%s, %dHZ %dchannels)",
video_id, video_codec.c_str(), req->sdp->video_protocol.c_str(), req->sdp->video_transport_format.c_str(),
audio_id, audio_codec.c_str(), req->sdp->audio_protocol.c_str(), req->sdp->audio_transport_format.c_str(),
audio_sample_rate, audio_channel
);
SrsRtspResponse* res = new SrsRtspResponse(req->seq);
@ -167,7 +171,12 @@ int SrsRtspConn::do_cycle()
srs_error("rtsp: rtp listen at port=%d failed. ret=%d", lpm, ret);
return ret;
}
srs_trace("rtsp: rtp listen at port=%d ok.", lpm);
srs_trace("rtsp: #%d %s over %s/%s/%s %s client-port=%d-%d, server-port=%d-%d",
req->stream_id, (req->stream_id == video_id)? "Video":"Audio",
req->transport->transport.c_str(), req->transport->profile.c_str(), req->transport->lower_transport.c_str(),
req->transport->cast_type.c_str(), req->transport->client_port_min, req->transport->client_port_max,
lpm, lpm + 1
);
// create session.
if (session.empty()) {

8
trunk/src/app/srs_app_rtsp.hpp
View file

@ -76,10 +76,14 @@ private:
private:
std::string session;
// video stream.
std::string video_id;
int video_id;
std::string video_codec;
SrsRtpConn* video_rtp;
// audio stream.
std::string audio_id;
int audio_id;
std::string audio_codec;
int audio_sample_rate;
int audio_channel;
SrsRtpConn* audio_rtp;
// video sequence header.
std::string sps;

11
trunk/src/protocol/srs_rtsp_stack.cpp
View file

@ -461,6 +461,7 @@ SrsRtspRequest::SrsRtspRequest()
{
seq = 0;
content_length = 0;
stream_id = 0;
sdp = NULL;
transport = NULL;
}
@ -766,13 +767,15 @@ int SrsRtspStack::do_recv_message(SrsRtspRequest* req)
// for setup, parse the stream id from uri.
if (req->is_setup()) {
size_t pos = string::npos;
std::string stream_id;
if ((pos = req->uri.rfind("/")) != string::npos) {
req->stream_id = req->uri.substr(pos + 1);
stream_id = req->uri.substr(pos + 1);
}
if ((pos = req->stream_id.find("=")) != string::npos) {
req->stream_id = req->stream_id.substr(pos + 1);
if ((pos = stream_id.find("=")) != string::npos) {
stream_id = stream_id.substr(pos + 1);
}
srs_info("rtsp: setup stream id=%s", req->stream_id.c_str());
req->stream_id = ::atoi(stream_id.c_str());
srs_info("rtsp: setup stream id=%d", req->stream_id);
}
// parse rdp body.

24
trunk/src/protocol/srs_rtsp_stack.hpp
View file

@ -48,7 +48,7 @@ class ISrsProtocolReaderWriter;
// SP = <US-ASCII SP, space (32)>
#define __SRS_RTSP_SP ' ' // 0x20
// 4 RTSP Message
// 4 RTSP Message, @see rtsp-rfc2326-1998.pdf, page 37
// Lines are terminated by CRLF, but
// receivers should be prepared to also interpret CR and LF by
// themselves as line terminators.
@ -100,7 +100,7 @@ enum SrsRtspSdpState
};
/**
* 10 Method Definitions
* 10 Method Definitions, @see rtsp-rfc2326-1998.pdf, page 57
* The method token indicates the method to be performed on the resource
* identified by the Request-URI. The method is case-sensitive. New
* methods may be defined in the future. Method names may not start with
@ -147,7 +147,7 @@ enum SrsRtspTokenState
};
/**
* the sdp in announce.
* the sdp in announce, @see rtsp-rfc2326-1998.pdf, page 159
* Appendix C: Use of SDP for RTSP Session Descriptions
* The Session Description Protocol (SDP, RFC 2327 [6]) may be used to
* describe streams or presentations in RTSP.
@ -241,7 +241,7 @@ private:
/**
* the rtsp transport.
* 12.39 Transport
* 12.39 Transport, @see rtsp-rfc2326-1998.pdf, page 115
* This request header indicates which transport protocol is to be used
* and configures its parameters such as destination address,
* compression, multicast time-to-live and destination port for a single
@ -288,7 +288,7 @@ public:
/**
* the rtsp request message.
* 6 Request
* 6 Request, @see rtsp-rfc2326-1998.pdf, page 39
* A request message from a client to a server or vice versa includes,
* within the first line of that message, the method to be applied to
* the resource, the identifier of the resource, and the protocol
@ -322,14 +322,14 @@ public:
*/
long seq;
/**
* 12.16 Content-Type
* 12.16 Content-Type, @see rtsp-rfc2326-1998.pdf, page 99
* See [H14.18]. Note that the content types suitable for RTSP are
* likely to be restricted in practice to presentation descriptions and
* parameter-value types.
*/
std::string content_type;
/**
* 12.14 Content-Length
* 12.14 Content-Length, @see rtsp-rfc2326-1998.pdf, page 99
* This field contains the length of the content of the method (i.e.
* after the double CRLF following the last header). Unlike HTTP, it
* MUST be included in all messages that carry content beyond the header
@ -353,7 +353,7 @@ public:
/**
* for setup message, parse the stream id from uri.
*/
std::string stream_id;
int stream_id;
public:
SrsRtspRequest();
virtual ~SrsRtspRequest();
@ -366,7 +366,7 @@ public:
/**
* the rtsp response message.
* 7 Response
* 7 Response, @see rtsp-rfc2326-1998.pdf, page 43
* [H6] applies except that HTTP-Version is replaced by RTSP-Version.
* Also, RTSP defines additional status codes and does not define some
* HTTP codes. The valid response codes and the methods they can be used
@ -396,7 +396,7 @@ public:
// @see about the status of rtsp, see SRS_CONSTS_RTSP_OK
int status;
/**
* 12.17 CSeq
* 12.17 CSeq, @see rtsp-rfc2326-1998.pdf, page 99
* The CSeq field specifies the sequence number for an RTSP requestresponse
* pair. This field MUST be present in all requests and
* responses. For every RTSP request containing the given sequence
@ -426,7 +426,7 @@ protected:
};
/**
* 10.1 OPTIONS
* 10.1 OPTIONS, @see rtsp-rfc2326-1998.pdf, page 59
* The behavior is equivalent to that described in [H9.2]. An OPTIONS
* request may be issued at any time, e.g., if the client is about to
* try a nonstandard request. It does not influence server state.
@ -446,7 +446,7 @@ protected:
};
/**
* 10.4 SETUP
* 10.4 SETUP, @see rtsp-rfc2326-1998.pdf, page 65
* The SETUP request for a URI specifies the transport mechanism to be
* used for the streamed media. A client can issue a SETUP request for a
* stream that is already playing to change transport parameters, which