Fix log flooding issue when processing SRT streams containing SCTE-35
PIDs or other unrecognized stream types.
The `SrsSrtFormat::on_srt_packet()` method continuously parses TS
packets throughout the entire stream lifetime. The TS parser logs
warnings for every unrecognized stream type (like SCTE-35) in the PMT,
causing log flooding.
However, `SrsFormat` is only used to detect audio/video codec
information. Once both codecs are detected, there's no need to continue
parsing TS packets.
Note: This fix mitigates the problem - there will still be some warning
logs during the initial codec detection phase (typically 5-10 seconds),
but the continuous log flooding after codec detection is completely
eliminated.
This PR adds separate audio and video frame counting to the HTTP API
(`/api/v1/streams/`) for better stream observability. The API now
reports three frame fields:
- `frames` - Total frames (video + audio)
- `video_frames` - Video frames/packets only
- `audio_frames` - Audio frames/packets only
This enhancement provides better visibility into stream composition and
helps detect issues with CBR/VBR streams, audio/video sync problems, and
codec-specific behavior.
**Before:**
```json
{
"streams": [
{
"frames": 0, // video frames.
}
]
}
```
**After:**
```json
{
"streams": [
{
"frames": 6912, // video frames.
"audio_frames": 5678, // audio frames.
"video_frames": 1234, // video frames.
}
]
}
```
Frame Counting Strategy
- All protocols report frames every N frames to balance accuracy and
performance
- Frames are counted at the protocol-specific message/packet level:
- RTMP: Counts RTMP messages (video/audio)
- WebRTC: Counts RTP packets (video/audio)
- SRT: Counts MPEG-TS messages (H.264/HEVC/AAC)
This PR refactors the stream bridge architecture in SRS to improve code
organization, type safety, and maintainability by replacing the generic
ISrsStreamBridge interface with protocol-specific bridge classes and
target interfaces.
1. New Target Interface Architecture:
- Introduces ISrsFrameTarget for AV frame consumers (RTMP sources)
- Introduces ISrsRtpTarget for RTP packet consumers (RTC sources)
- Introduces ISrsSrtTarget for SRT packet consumers (SRT sources)
2. Protocol-Specific Bridge Classes:
- SrsRtmpBridge: Converts RTMP frames to RTC/RTSP protocols
- SrsSrtBridge: Converts SRT packets to RTMP/RTC protocols
- SrsRtcBridge: Converts RTC packets to RTMP protocol
3. Simplified Bridge Management:
- Removes the generic SrsCompositeBridge chain pattern
- Each source type now uses its appropriate bridge type directly
With this improvement, you are able to implement very complex bridge and
protocol converting, for example, you can bridge RTMP to RTC with opus
audio when you support enhanced RTMP with opus.
Another plan is to support bridging RTC to RTSP, directly without
converting RTP to media frame packet, but directly deliver RTP packet
from RTC source to RTSP source.
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
try to fix#4428.
## Cause
rtmp do not support hevc, rtmp enhanced do.
## How to reproduce
1. start srs.
`./objs/srs -c conf/srt.conf`
2. publish hevc (h.265) stream to srs by srt.
`ffmpeg -re -i ./doc/source.flv -c:v libx265 -crf 28 -preset medium -c:a
copy -pes_payload_size 0 -f mpegts
'srt://127.0.0.1:10080?streamid=#!::r=live/livestream,m=publish'`
3. probe the rtmp stream
`ffprobe rtmp://localhost/live/livestream`
## About the Failed BlackBox test
The failed blackbox test: `TestSlow_SrtPublish_RtmpPlay_HEVC_Basic`
`TestSlow_SrtPublish_HttpFlvPlay_HEVC_Basic`
### Cause:
The ffmpeg 5 is used to record a piece of video (DRV), the ffmpeg will
transcode the enhanced flv format to TS format, but ffmpeg 5 don't
support enhanced rtmp (or flv) in this case.
The solution is to replace the ffmpeg to version 7 in those 2 test
cases.
### why not upgrade ffmpeg to version 7?
The black tests dependency on ffmpeg 5 will fail, and there are a few of
them are not easy to resolve in ffmpeg 7.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
This PR introduces anonymous coroutine macros for easier coroutine
creation and improves the State Threads (ST) mutex and condition
variable handling in SRS.
- **Added coroutine macros**: `SRS_COROUTINE_GO`, `SRS_COROUTINE_GO2`,
`SRS_COROUTINE_GO_CTX`, `SRS_COROUTINE_GO_CTX2`
- **Added `SrsCoroutineChan`**: Channel for sharing data between
coroutines with coroutine-safe operations
- **Simplified coroutine creation**: Go-like syntax for creating
anonymous coroutines with code blocks
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
This PR introduces a major refactoring to replace `SrsSharedPtrMessage`
with `SrsMediaPacket` throughout the SRS codebase, providing a more
unified and cleaner approach to media packet handling.
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
This PR modernizes the memory management architecture in SRS by
refactoring RTMP message handling to use shared pointers
(SrsSharedPtr<SrsMemoryBlock>) instead of manual memory management. This
change improves memory safety, reduces the risk of memory leaks, and
provides a cleaner abstraction for message payload handling.
* Introduced `SrsMemoryBlock`: A dedicated class for managing memory
buffers with size information
* Replaced manual memory management: `SrsCommonMessage` and
`SrsSharedPtrMessage` now use `SrsSharedPtr<SrsMemoryBlock>` instead of
raw pointers
* Updated `SrsRtpPacket`: Now uses `SrsSharedPtr<SrsMemoryBlock>` for
shared buffer management
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
This PR fixes a critical race condition in SRS source managers where
multiple coroutines could create duplicate sources for the same stream.
- **Atomic source creation**: Source lookup, creation, and pool
insertion now happen atomically within lock scope
- **Consistent interface**: Standardize on `ISrsRequest*` interface
throughout codebase
- **Handler simplification**: Remove `ISrsLiveSourceHandler*` parameter,
obtain from global server instance
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
**Introduce**
This pull request builds upon the foundation laid in
https://github.com/ossrs/srs/pull/4289 . While the previous work solely
implemented unidirectional HEVC support from RTMP to RTC, this
submission further enhances it by introducing support for the RTC to
RTMP direction.
**Usage**
Launch SRS with `rtc2rtmp.conf`
```bash
./objs/srs -c conf/rtc2rtmp.conf
```
**Push with WebRTC**
Upgrade browser to Chrome(136+) or Safari(18+), then open [WHIP
encoder](http://localhost:8080/players/whip.html?schema=http&&codec=hevc),
push stream with URL that enables HEVC by query string `codec=hevc`:
```bash
http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream&codec=hevc
```
This query string `codec=hevc` is used to select the video codec, and
generate lines in the answer SDP.
```
m=video 9 UDP/TLS/RTP/SAVPF 49 123
a=rtpmap:49 H265/90000
```
The encoder log also show the codec:
```
Audio: opus, 48000HZ, channels: 2, pt: 111
Video: H265, 90000HZ, pt: 49
```
**Play with RTMP**
Play HEVC stream via RTMP.
```bash
ffplay -i rtmp://localhost/live/livestream
```
You will see the codec in logs:
```
Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp
Stream #0:1: Video: hevc (Main), yuv420p(tv, bt709), 320x240, 30 fps, 30 tbr, 1k tbn
```
You can also use [WHEP
player](http://localhost:8080/players/whep.html?schema=http&&codec=hevc)
to play the stream.
Important refactor with AI:
* [AI: Refactor packet cache for RTC frame
builder.](b8ffa1630e)
* [AI: Refactor the packet copy and free for
SrsRtcFrameBuilder](f3487b45d7)
* [AI: Refactor the frame detector for
SrsRtcFrameBuilder](4ffc1526b9)
* [AI: Refactor the packet_video_rtmp for
SrsRtcFrameBuilder](81f6aef4ed)
* [AI: Add utests for
SrsCodecPayload.codec](61eb1c0bfc)
* [AI: Add utests for VideoPacketCache in
SrsRtcFrameBuilder.](fd25480dfa)
* [AI: Add utests for VideoFrameDetector in
SrsRtcFrameBuilder.](b4aa977bbd)
* [AI: Add regression test for RTC2RTMP with
HEVC.](5259a2aac3)
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: winlin <winlinvip@gmail.com>
To manage an object:
```cpp
// Before
MyClass* ptr = new MyClass();
SrsAutoFree(MyClass, ptr);
ptr->do_something();
// Now
SrsUniquePtr<MyClass> ptr(new MyClass());
ptr->do_something();
```
To manage an array of objects:
```cpp
// Before
char* ptr = new char[10];
SrsAutoFreeA(char, ptr);
ptr[0] = 0xf;
// Now
SrsUniquePtr<char[]> ptr(new char[10]);
ptr[0] = 0xf;
```
In fact, SrsUniquePtr is a limited subset of SrsAutoFree, mainly
managing pointers and arrays. SrsUniquePtr is better than SrsAutoFree
because it has the same API to standard unique ptr.
```cpp
SrsUniquePtr<MyClass> ptr(new MyClass());
ptr->do_something();
MyClass* p = ptr.get();
```
SrsAutoFree actually uses a pointer to a pointer, so it can be set to
NULL, allowing the pointer's value to be changed later (this usage is
different from SrsUniquePtr).
```cpp
// OK to free ptr correctly.
MyClass* ptr;
SrsAutoFree(MyClass, ptr);
ptr = new MyClass();
// Crash because ptr is an invalid pointer.
MyClass* ptr;
SrsUniquePtr<MyClass> ptr(ptr);
ptr = new MyClass();
```
Additionally, SrsAutoFreeH can use specific release functions, which
SrsUniquePtr does not support.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
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>
When the srs have multiple pps in hevc.the srs can't parse for this.
problem fixed this #3604
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: john <hondaxiao@tencent.com>
* SRT: print warning log when audio duration too large
* Fix#3164: SRT: Choppy when audio ts gap is too large. v5.0.65
Co-authored-by: winlin <winlin@vip.126.com>
