SRS supports including another configuration in the include package.
When generating configurations, we can only generate the changed
configurations, while the unchanged configurations are in the fixed
files, for example:
```nginx
listen 1935;
include server.conf;
```
In `server.conf`, we can manage the changing configurations with the
program:
```nginx
http_api { enabled on; }
```
However, during system initialization, we often create an empty
`server.conf`, and the content is generated only after the program
starts, so `server.conf` might be an empty file. This also makes it
convenient to use a script to confirm the existence of this file:
```bash
touch server.conf
```
Currently, SRS does not support empty configurations and will report an
error. This PR is to solve this problem, making it more convenient to
use include.
`TRANS_BY_GPT4`
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
For the DJI M30, there is a bug where empty NALU packets with a size of
zero are causing issues with HLS streaming. This bug leads to random
unpublish events due to the SRS disconnecting the connection for the HLS
module when it fails to handle empty NALU packets.
To address this bug, we have patched the system to ignore any empty NALU
packets with a size of zero. Additionally, we have created a tool in the
srs-bench to replay pcapng files captured by tcpdump or Wireshark. We
have also added utest using mprotect and asan to detect any memory
corruption.
It is important to note that this bug has been fixed in versions 4.0.271
6477f31004 and 5.0.170
939f6b484b. This patch specifically
addresses the issue in SRS 6.0.
Please be aware that there is another commit related to this bug that
partially fixes the issue but still leaves a small problem for asan to
detect memory corruption. This commit,
577cd299e1, only ignores empty NALU
packets but still reads beyond the memory.
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
* Replaced all occurrences of sprintf with snprintf to address deprecation warnings
* Ensured proper buffer size is passed to snprintf to prevent potential buffer overflows
* Ran tests to confirm that the changes do not introduce any new issues or regressions
---------
Co-authored-by: ChenGH <chengh_math@126.com>
* RTMP: Support enhanced RTMP specification for HEVC, v6.0.42.
* Player: Upgrade mpegts.js to support it.
Enhanced RTMP specification: https://github.com/veovera/enhanced-rtmp
First, start SRS `v6.0.42+` with HTTP-TS support:
```bash
./objs/srs -c conf/http.ts.live.conf
```
Then, you can use [OBS 29.1+](https://github.com/obsproject/obs-studio/releases) to push HEVC via RTMP.
Start OBS with the following settings in the `Settings > Stream` tab:
* Server: `rtmp://localhost/live`
* Stream Key: `livestream`
* Encoder: Please select the HEVC hardware encoder.
Finally, open the player http://localhost:8080/players/srs_player.html?stream=livestream.ts
Or use VLS or ffplay to play `http://localhost:8080/live/livestream.ts`
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
In dockerfile, we can set the default RTC candidate to env:
```
ENV SRS_RTC_SERVER_CANDIDATE=\$CANDIDATE
CMD ["./objs/srs", "-e"]
```
When starts a docker container, user can setup the candidate by env:
```
docker run --rm -it --env CANDIDATE=1.2.3.4 ossrs/srs:5
```
We should parse the content of SRS_RTC_SERVER_CANDIDATE as env variable name and parse it again.
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: pengfei.ma <pengfei.ma@ctechm.com>
For some use scenario, the publisher is invited when player want to view the stream:
1. Publisher connect to system, but does not publish any stream to SRS yet.
2. Player connect to system and start to request the stream.
3. System notifies publisher to publish stream to SRS.
4. Player play the stream from SRS.
Please notice that `system` means your business system, not SRS.
This is what we called `on-demand-live-streaming`, so when the last player stop to view the stream, what happends?
1. System needs to notify publisher to stop publish.
2. Or, SRS disconnect the publisher when idle(the last player stops playing).
This PR is for the solution 2, so that the cleanup is very simple, your system does not need to notify publisher to stop publish, because SRS has already disconnected the publihser.
---------
Co-authored-by: winlin <winlin@vip.126.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
For compatibility, transform
rtmp://ip/app...vhost...VHOST/stream
to typical format:
rtmp://ip/app/stream?vhost=VHOST
This is used for some legacy devices, which does not
support standard HTTP url query string.
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: john <hondaxiao@tencent.com>
1. SRT: Fix srt to rtmp crash when sps or pps empty. v5.0.112 (#3323)
2. GB28181: Fix memory overlap for small packets. v5.0.111 (#3315)
3. FLV: Support set default has_av and disable guessing. v5.0.110 (#3311)
4. FLV: Drop packet if header flag is not matched. v5.0.109 (#3306)
5. FLV: Reset has_audio or has_video if only sequence header. (#3310)
* FLV: Support set default has_av and disable guessing. v5.0.110
1. Support config default has_audio and has_video.
2. Support disable guessing has_audio or has_video.
* FLV: Reset to false if start to guess has_av.
* FLV: Add regression test for FLV header av metadata.
1. Ignore audo or video packets if FLV header disable it.
2. Run: Add regression test config and run for IDEA.
3. Test: Refine regression test to allow no audio/video for FLV
4. Config: Whether drop packet if not match header.
* Add utc time utility
* Fix calculate duration in fmp4
* Refine dash code, use segment template timeline
* Shrink m4s file and cleanup
* Support play by dash.js
* Use SegmentTemplate timeline mode with $Number$
Co-authored-by: winlin <winlin@vip.126.com>
* Add utc time utility
* Fix calculate duration in fmp4
* Refine dash code, use segment template timeline
* Shrink m4s file and cleanup
* Support play by dash.js
* Use SegmentTemplate timeline mode with $Number$
Co-authored-by: winlin <winlin@vip.126.com>
1. Update TS video codec to HEVC during streaming.
2. Return error when HEVC is disabled.
3. Parse HEVC NALU type by SrsHevcNaluTypeParse.
4. Show message when codec change for TS.
Co-authored-by: runner365 <shi.weibd@hotmail.com>
1. Support configure with --h265=on.
2. Parse HEVC(H.265) from FLV or RTMP packet.
3. Support HEVC over RTMP or HTTP-FLV.
Co-authored-by: runner365 <shi.weibd@hotmail.com>
1. Support cygwin by '--cygwin64=on'
2. Detect cygwin automatically.
3. Disalbe sanitizer, srt and srtp with openssl.
4. Disable multiple threads, use single threads.
5. Support utest for cygwin64.
6. Query features for windows by API.
7. Disable stat APIs for cygwin.
8. Use ST select event driver.
Co-authored-by: wenjie.zhao <740936897@qq.com>
* MP4: Fix memory leak when error.
* Kernel: Support free global objects for utest.
* HTTP: Fix memory leak when error.
* MP4: Support more sample rate for audio.
* RTMP: Support free field for utest.
* UTest: Support address sanitizer.
* Exporter: metrics support cpu gauge.
* Exporter: metrics support memory and uname..
* Exporter: Ignore error when uname fail.
Co-authored-by: winlin <winlin@vip.126.com>
1. Support Java/log4j log level text.
2. Support configuring by `--log-new-level=on` which is enabled by default.
3. Support `--log-new-level=off` to use SRS 4.0 log level for compatibility.
01. Support GB config as StreamCaster.
02. Support disable GB by --gb28181=off.
03. Add utests for SIP examples.
04. Wireshark plugin to decode TCP/9000 as rtp.rfc4571
05. Support MPEGPS program stream codec.
06. Add utest for PS stream codec.
07. Decode MPEGPS packet stream.
08. Carry RTP and PS packet as helper in PS message.
09. Support recover from error mode.
10. Support process by a pack of PS/TS messages.
11. Add statistic for recovered and msgs dropped.
12. Recover from err position fastly.
13. Define state machine for GB session.
14. Bind context to GB session.
15. Re-invite when media disconnected.
16. Update GitHub actions with GB28181.
17. Support parse CANDIDATE by env or pip.
18. Support mux GB28181 to RTMP.
19. Support regression test by srs-bench.
