## How to reproduce?
1. Refer this commit, which contains the web demo to capture screen as
video stream through RTC.
2. Copy the `trunk/research/players/whip.html` and
`trunk/research/players/js/srs.sdk.js` to replace the `develop` branch
source code.
3. `./configure && make`
4. `./objs/srs -c conf/rtc2rtmp.conf`
5. open `http://localhost:8080/players/whip.html?schema=http`
6. check `Screen` radio option.
7. click `publish`, then check the screen to share.
8. play the rtmp live stream: `rtmp://localhost/live/livestream`
9. check the video stuttering.
## Cause
When capture screen by the chrome web browser, which send RTP packet
with empty payload frequently, then all the cached RTP packets are
dropped before next key frame arrive in this case.
The OBS screen stream and camera stream do not have such problem.
## Add screen stream to WHIP demo
><img width="581" alt="Screenshot 2024-08-28 at 2 49 46 PM"
src="https://github.com/user-attachments/assets/9557dbd2-c799-4dfd-b336-5bbf2e4f8fb8">
---------
Co-authored-by: winlin <winlinvip@gmail.com>
For coroutine, we should use `__sanitizer_start_switch_fiber` which
similar to`VALGRIND_STACK_REGISTER`, see
https://github.com/google/sanitizers/issues/189#issuecomment-1346243598
for details. If not fix this, asan will output warning:
```
==72269==WARNING: ASan is ignoring requested __asan_handle_no_return: stack type: default top: 0x00016f638000; bottom 0x000106bec000; size: 0x000068a4c000 (1755627520)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
```
It will cause asan failed to get the stack, see
`research/st/asan-switch.cpp` for example:
```
==71611==ERROR: AddressSanitizer: heap-buffer-overflow on address 0x000103600733 at pc 0x0001009d3d7c bp 0x000100b4bd40 sp 0x000100b4bd38
WRITE of size 1 at 0x000103600733 thread T0
#0 0x1009d3d78 in foo(void*) asan-switch.cpp:13
```
After fix this issue, it should provide the full stack when crashing:
```
==73437==ERROR: AddressSanitizer: heap-buffer-overflow on address 0x000103300733 at pc 0x000100693d7c bp 0x00016f76f550 sp 0x00016f76f548
WRITE of size 1 at 0x000103300733 thread T0
#0 0x100693d78 in foo(void*) asan-switch.cpp:13
#1 0x100693df4 in main asan-switch.cpp:23
#2 0x195aa20dc (<unknown module>)
```
For primordial coroutine, if not set the stack by
`st_set_primordial_stack`, then the stack is NULL and asan can't get the
stack tracing. Note that it's optional and only make it fail to display
the stack information, no other errors.
---
Co-authored-by: john <hondaxiao@tencent.com>
Improvements for ST(State Threads):
1. ST: Use g++ for CXX compiler.
2. ST: Remove macros for clist.
3. ST: Remove macros for global thread and vp.
4. ST: Remove macros for vp queue operations.
5. ST: Remove macros for context switch.
6. ST: Remove macros for setjmp/longjmp.
7. ST: Remove macro for stack pad.
8. ST: Refine macro for valgrind.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
1. `trunk/research/st/exceptions.cpp` About exceptions with ST, works
well on linux and mac, not work on cygwin.
2. `trunk/research/st/pthreads.cpp` About pthreads with ST, works well
on all platforms.
3. `trunk/research/st/hello.cpp` Hello world, without ST, works well on
all platforms.
4. `trunk/research/st/hello-world.cpp` Hello world, with ST, works well
on all platforms.
5. `trunk/research/st/hello-st.cpp` A very simple version for hello
world with ST, works well on all platforms.
HLS typically has a delay of around 30 seconds, roughly comprising three
segments, each lasting 10 seconds. We can reduce the delay to about 5
seconds by lowering the segment duration to 2 seconds and starting
playback from the last segment, achieving a stable delay.
Of course, this requires setting the OBS's GOP to 1 second, and the
profile to baseline, preset to fast, and tune to zerolatency.
Additionally, updating a few configurations in the hls.js player is
necessary, such as setting it to start playback from the last segment,
setting the maximum buffer, and initiating accelerated playback to
reduce latency.
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: john <hondaxiao@tencent.com>
### Description
When converting between AAC and Opus formats (aac2opus or opus2aac), the
`av_frame_get_buffer` API is frequently called.
### Objective
The goal is to optimize the code logic and reduce the frequent
allocation and deallocation of memory.
In the case of aac2opus, av_frame_get_buffer is still frequently called.
In the case of opus2aac, the goal is to avoid calling
av_frame_get_buffer and reduce memory allocations.
### Additional Note
Before calling the `av_audio_fifo_read` API, use
`av_frame_make_writable` to check if the frame is writable. If it is not
writable, create a new frame.
---------
Co-authored-by: john <hondaxiao@tencent.com>
By default, caching is enabled during compilation, which means that data
is cached in Docker. This helps to avoid compiling third-party
dependency libraries. However, sometimes when updating third-party
libraries, it's necessary to disable caching to temporarily verify if
the pipeline can succeed. Therefore, a configure option should be added.
When this option is enabled, the compilation cache will not be used, and
all third-party libraries will be compiled from scratch.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
When accessing the SRS Stack, you should log in and use a token for each
request, or utilize the HTTP API with a secret Bearer token included in
every request. The SRS Stack HTTP API proxies both /api/v1 and /rtc/v1
to the SRS HTTP API while ensuring secure authentication. Additionally,
there is a console in the SRS Stack that requires the same token to
request the SRS Stack HTTP API, which is then proxied to the SRS HTTP
API.
The SRS Stack runs SRS with the HTTP API listening at 127.0.0.1:1985 on
the local loopback interface, allowing only the SRS Stack to access it
without authentication. All other users must login and access the SRS
Stack through its interface, rather than directly accessing the SRS HTTP
API within the SRS Stack.
---------
Co-authored-by: panda <542638787@qq.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>
* 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>
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.
* Forward: add backend config and demo server for dynamic create forwarder to other server.(#1342)
* Forward: if call forward backend failed, then return directly.
* Forward: add API description and change return value format.
* Forward: add backend conf file and wrapper function for backend service.
* Forward: add backend comment in full.conf and update forward.backend.conf.
* Forward: rename backend param and add comment tips.