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srs/trunk/3rdparty/srs-bench/live/publisher.go
Winlin 1f9309ae25
SmartPtr: Support load test for source by srs-bench. v6.0.130 (#4097)
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>
2024-06-21 07:13:12 +08:00

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
}