mirror of
https://github.com/ossrs/srs.git
synced 2025-02-15 04:42:04 +00:00
1. Add live benchmark support in srs-bench, which only connects and disconnects without any media transport, to test source creation and disposal and verify source memory leaks. 2. SmartPtr: Support cleanup of HTTP-FLV stream. Unregister the HTTP-FLV handler for the pattern and clean up the objects and resources. 3. Support benchmarking RTMP/SRT with srs-bench by integrating the gosrt and oryx RTMP libraries. 4. Refine SRT and RTC sources by using a timer to clean up the sources, following the same strategy as the Live source. --------- Co-authored-by: Haibo Chen <495810242@qq.com> Co-authored-by: Jacob Su <suzp1984@gmail.com>
210 lines
6.3 KiB
Go
210 lines
6.3 KiB
Go
// The MIT License (MIT)
|
|
//
|
|
// # Copyright (c) 2021 Winlin
|
|
//
|
|
// Permission is hereby granted, free of charge, to any person obtaining a copy of
|
|
// this software and associated documentation files (the "Software"), to deal in
|
|
// the Software without restriction, including without limitation the rights to
|
|
// use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
|
|
// the Software, and to permit persons to whom the Software is furnished to do so,
|
|
// subject to the following conditions:
|
|
//
|
|
// The above copyright notice and this permission notice shall be included in all
|
|
// copies or substantial portions of the Software.
|
|
//
|
|
// THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
// IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
|
|
// FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
|
|
// COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
|
|
// IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
|
|
// CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
|
|
package live
|
|
|
|
import (
|
|
"context"
|
|
"fmt"
|
|
"math/rand"
|
|
"net"
|
|
"net/url"
|
|
"strconv"
|
|
"strings"
|
|
"time"
|
|
|
|
"github.com/haivision/srtgo"
|
|
"github.com/ossrs/go-oryx-lib/amf0"
|
|
"github.com/ossrs/go-oryx-lib/errors"
|
|
"github.com/ossrs/go-oryx-lib/logger"
|
|
"github.com/ossrs/go-oryx-lib/rtmp"
|
|
)
|
|
|
|
func startPublish(ctx context.Context, r string, closeAfterPublished bool) error {
|
|
ctx = logger.WithContext(ctx)
|
|
logger.Tf(ctx, "Run publish url=%v, cap=%v", r, closeAfterPublished)
|
|
|
|
u, err := url.Parse(r)
|
|
if err != nil {
|
|
return errors.Wrapf(err, "parse %v", r)
|
|
}
|
|
|
|
if u.Scheme == "rtmp" {
|
|
return startPublishRTMP(ctx, u, closeAfterPublished)
|
|
} else if u.Scheme == "srt" {
|
|
return startPublishSRT(ctx, u, closeAfterPublished)
|
|
}
|
|
|
|
return fmt.Errorf("invalid schema %v of %v", u.Scheme, r)
|
|
}
|
|
|
|
func startPublishSRT(ctx context.Context, u *url.URL, closeAfterPublished bool) (err error) {
|
|
// Parse host and port.
|
|
port := 1935
|
|
if u.Port() != "" {
|
|
if port, err = strconv.Atoi(u.Port()); err != nil {
|
|
return errors.Wrapf(err, "parse port %v", u.Port())
|
|
}
|
|
}
|
|
|
|
ips, err := net.LookupIP(u.Hostname())
|
|
if err != nil {
|
|
return errors.Wrapf(err, "lookup %v", u.Hostname())
|
|
}
|
|
if len(ips) == 0 {
|
|
return errors.Errorf("no ips for %v", u.Hostname())
|
|
}
|
|
logger.Tf(ctx, "Parse url %v to host=%v, ip=%v, port=%v",
|
|
u.String(), u.Hostname(), ips[0], port)
|
|
|
|
// Setup libsrt.
|
|
client := srtgo.NewSrtSocket(ips[0].To4().String(), uint16(port),
|
|
map[string]string{
|
|
"transtype": "live",
|
|
"tsbpdmode": "false",
|
|
"tlpktdrop": "false",
|
|
"latency": "0",
|
|
"streamid": fmt.Sprintf("#%v", u.Fragment),
|
|
},
|
|
)
|
|
defer client.Close()
|
|
|
|
if err := client.Connect(); err != nil {
|
|
return errors.Wrapf(err, "SRT connect to %v:%v", u.Hostname(), port)
|
|
}
|
|
logger.Tf(ctx, "Connect to SRT server %v:%v success", u.Hostname(), port)
|
|
|
|
// We should wait for a while after connected to SRT server before quit. Because SRT server use timeout
|
|
// to detect UDP connection status, so we should never reconnect very fast.
|
|
select {
|
|
case <-ctx.Done():
|
|
case <-time.After(3 * time.Second):
|
|
logger.Tf(ctx, "SRT publish stream success, stream=%v", u.Fragment)
|
|
}
|
|
|
|
if closeAfterPublished {
|
|
logger.Tf(ctx, "Close connection after published")
|
|
return nil
|
|
}
|
|
|
|
return nil
|
|
}
|
|
|
|
func startPublishRTMP(ctx context.Context, u *url.URL, closeAfterPublished bool) (err error) {
|
|
parts := strings.Split(u.Path, "/")
|
|
if len(parts) == 0 {
|
|
return errors.Errorf("invalid path %v", u.Path)
|
|
}
|
|
app, stream := strings.Join(parts[:len(parts)-1], "/"), parts[len(parts)-1]
|
|
|
|
// Parse host and port.
|
|
port := 1935
|
|
if u.Port() != "" {
|
|
if port, err = strconv.Atoi(u.Port()); err != nil {
|
|
return errors.Wrapf(err, "parse port %v", u.Port())
|
|
}
|
|
}
|
|
|
|
ips, err := net.LookupIP(u.Hostname())
|
|
if err != nil {
|
|
return errors.Wrapf(err, "lookup %v", u.Hostname())
|
|
}
|
|
if len(ips) == 0 {
|
|
return errors.Errorf("no ips for %v", u.Hostname())
|
|
}
|
|
logger.Tf(ctx, "Parse url %v to host=%v, ip=%v, port=%v, app=%v, stream=%v",
|
|
u.String(), u.Hostname(), ips[0], port, app, stream)
|
|
|
|
// Connect via TCP client.
|
|
c, err := net.DialTCP("tcp", nil, &net.TCPAddr{IP: ips[0], Port: port})
|
|
if err != nil {
|
|
return errors.Wrapf(err, "dial %v %v", u.Hostname(), u.Port())
|
|
}
|
|
defer c.Close()
|
|
logger.Tf(ctx, "Connect to RTMP server %v:%v success", u.Hostname(), port)
|
|
|
|
// RTMP Handshake.
|
|
rd := rand.New(rand.NewSource(time.Now().UnixNano()))
|
|
hs := rtmp.NewHandshake(rd)
|
|
|
|
if err := hs.WriteC0S0(c); err != nil {
|
|
return errors.Wrap(err, "write c0")
|
|
}
|
|
if err := hs.WriteC1S1(c); err != nil {
|
|
return errors.Wrap(err, "write c1")
|
|
}
|
|
|
|
if _, err = hs.ReadC0S0(c); err != nil {
|
|
return errors.Wrap(err, "read s1")
|
|
}
|
|
s1, err := hs.ReadC1S1(c)
|
|
if err != nil {
|
|
return errors.Wrap(err, "read s1")
|
|
}
|
|
if _, err = hs.ReadC2S2(c); err != nil {
|
|
return errors.Wrap(err, "read s2")
|
|
}
|
|
|
|
if err := hs.WriteC2S2(c, s1); err != nil {
|
|
return errors.Wrap(err, "write c2")
|
|
}
|
|
logger.Tf(ctx, "RTMP handshake with %v:%v success", ips[0], port)
|
|
|
|
// Do connect and publish.
|
|
client := rtmp.NewProtocol(c)
|
|
|
|
connectApp := rtmp.NewConnectAppPacket()
|
|
tcURL := fmt.Sprintf("rtmp://%v%v", u.Hostname(), app)
|
|
connectApp.CommandObject.Set("tcUrl", amf0.NewString(tcURL))
|
|
if err = client.WritePacket(connectApp, 1); err != nil {
|
|
return errors.Wrap(err, "write connect app")
|
|
}
|
|
|
|
var connectAppRes *rtmp.ConnectAppResPacket
|
|
if _, err = client.ExpectPacket(&connectAppRes); err != nil {
|
|
return errors.Wrap(err, "expect connect app res")
|
|
}
|
|
logger.Tf(ctx, "RTMP connect app success, tcUrl=%v", tcURL)
|
|
|
|
createStream := rtmp.NewCreateStreamPacket()
|
|
if err = client.WritePacket(createStream, 1); err != nil {
|
|
return errors.Wrap(err, "write create stream")
|
|
}
|
|
|
|
var createStreamRes *rtmp.CreateStreamResPacket
|
|
if _, err = client.ExpectPacket(&createStreamRes); err != nil {
|
|
return errors.Wrap(err, "expect create stream res")
|
|
}
|
|
logger.Tf(ctx, "RTMP create stream success")
|
|
|
|
publish := rtmp.NewPublishPacket()
|
|
publish.StreamName = *amf0.NewString(stream)
|
|
if err = client.WritePacket(publish, 1); err != nil {
|
|
return errors.Wrap(err, "write publish")
|
|
}
|
|
logger.Tf(ctx, "RTMP publish stream success, stream=%v", stream)
|
|
|
|
if closeAfterPublished {
|
|
logger.Tf(ctx, "Close connection after published")
|
|
return nil
|
|
}
|
|
|
|
return nil
|
|
}
|