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>
Refactors the `srs_net_url_parse_tcurl` function to use the robust
`SrsHttpUri` class for URL parsing and implements a dedicated legacy
RTMP URL conversion function to handle various URL formats consistently.
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
This PR refactors the HTTP routing system by renaming "hijack"
terminology to "dynamic match" for improved code clarity and better
semantic meaning.
Interface and Class Renaming
* ISrsHttpMatchHijacker → ISrsHttpDynamicMatcher
* hijack() method → dynamic_match() method
* hijackers member variables → dynamic_matchers_
Method Renaming
* SrsHttpServeMux::hijack() → SrsHttpServeMux::add_dynamic_matcher()
* SrsHttpServeMux::unhijack() →
SrsHttpServeMux::remove_dynamic_matcher()
The new "dynamic match" terminology better reflects that this is a
legitimate routing mechanism, not a security bypass or interception.
This PR consolidates the SRT and RTC server functionality into the main
SrsServer class, eliminating the separate `SrsSrtServer` and
`SrsRtcServer` classes and their corresponding adapter classes. This
architectural change simplifies the codebase by removing the hybrid
server pattern and integrating all protocol handling directly into
`SrsServer`.
As unified connection manager (`_srs_conn_manager`) for all protocol
connections, all incoming connections are checked against the same
connection limit in `on_before_connection()`. This enables consistent
connection limits: `max_connections` now protects against resource
exhaustion from any protocol, not just RTMP.
Remove modules because it's not used now, so only keep the server
application module and main entry point. Remove the wait group to run
server, instead, directly run server and invoke the cycle method.
After this PR, the startup workflow and servers architecture should be
much easier to maintain.
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Try to fix#4450
The SRS transcode rtp packets, whose sequence number in range [start,
end], to one rtmp packet, but when the first rtp packet is empty, then
this crash happens.
check #4450 for details.
5.0release and 6.0release branch.
develop branch already has its own solution.
So this PR is targeting to **6.0release**.
find the first not empty rtp packet in seq range [start, end].
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Co-authored-by: winlin <winlinvip@gmail.com>
This PR introduces a comprehensive stream publish token system that
prevents race conditions when multiple publishers attempt to publish to
the same stream URL simultaneously across different protocols (RTMP,
WebRTC, SRT).
* Race Condition Issue: Multiple publishers could create duplicate
sources for the same stream when context switches occurred during source
initialization in SRS's coroutine-based architecture
* Cross-Protocol Conflicts: Different protocols (RTMP, RTC, SRT) could
simultaneously publish to the same stream URL without coordination
* Resource Management: No centralized mechanism to ensure exclusive
stream publishing access
---------
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>
This PR makes WebRTC a core feature of SRS and enforces C++98
compatibility by:
1. Always Enable WebRTC Support
- Remove `--rtc=on|off` configuration option - WebRTC is now always
enabled
- Eliminate all `#ifdef SRS_RTC` conditional compilation blocks
- Include WebRTC-related modules (RTC, SRTP, DTLS) in all builds
- Update build scripts to always link WebRTC dependencies
2. Enforce C++98 Compatibility
- Remove `--cxx11=on|off` and `--cxx14=on|off` configuration options
- Force `SRS_CXX11=NO` and `SRS_CXX14=NO` in build system
- Move these options to deprecated section with warnings
- Ensure codebase maintains C++98 standard compatibility
3. Remove Windows/Cygwin Support
- Remove all Windows and Cygwin64 conditional compilation blocks (#ifdef
_WIN32, #ifdef CYGWIN64)
- Delete Cygwin64 build configurations from build scripts (
auto/options.sh, auto/depends.sh, configure)
- Remove Cygwin64 assembly files and State Threads platform support (
md_cygwin64.S)
- Eliminate Windows-specific GitHub Actions workflows and CI/CD jobs
- Remove NSIS packaging files and Windows installer generation
- Delete Windows documentation and update feature lists to mark support
as removed in v7.0
- Simplify OS detection to only support Unix-like systems (Linux, macOS)
4. Code Cleanup
- Remove conditional WebRTC code blocks throughout the codebase
- Simplify build configuration by removing WebRTC-related conditionals
- Update constructor delegation patterns to be C++98 compatible
- Fix vector initialization to use C++98 syntax
- Eliminate Windows-specific implementations for file operations, time
handling, and networking
- Unified platform handling with consistent POSIX API usage
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
This PR removes the multi-threading infrastructure from SRS and
consolidates the codebase to use single-thread architecture exclusively.
This is a architectural simplification that aligns with SRS's
coroutine-based design philosophy.
* Simplified Architecture: Eliminates complexity of multi-threading
coordination
* Better Alignment: Matches SRS's coroutine-based single-thread design
philosophy
* Reduced Complexity: Removes potential race conditions and threading
bugs
* Cleaner Code: More focused modules with clear responsibilities
* Easier Maintenance: Fewer moving parts and clearer execution flow
---------
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
## Summary
Removes the deprecated `hls_acodec` and `hls_vcodec` configuration
options and implements automatic codec detection for HLS streams, fixing
issues with video-only streams incorrectly showing audio information.
## Problem
- When streaming video-only content via RTMP, HLS output incorrectly
contained audio track information due to hardcoded default codec
settings
- The static `hls_acodec` and `hls_vcodec` configurations were
inflexible and caused compatibility issues with some players
- Users had to manually configure `hls_acodec an` to fix video-only
streams
## Solution
- **Remove deprecated configs**: Eliminates `hls_acodec` and
`hls_vcodec` configuration options entirely
- **Dynamic codec detection**: HLS muxer now automatically detects and
uses actual stream codecs in real-time
- **Improved defaults**: Changes from hardcoded AAC/H.264 defaults to
disabled state, letting actual stream content determine codec
information
- **Real-time codec switching**: Supports codec changes during streaming
with proper logging
## Changes
- Remove `get_hls_acodec()` and `get_hls_vcodec()` from SrsConfig
- Update HLS muxer to use `latest_acodec_`/`latest_vcodec_` for codec
detection
- Add codec detection logic in `write_audio()` and `write_video()`
methods
- Remove deprecated config options from all configuration files
- Add comprehensive unit tests for codec detection functionality
Fixes#4223
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Fixes#3993 - WebRTC streams recorded to MP4 via DVR exhibit audio/video
synchronization issues, with audio typically ahead of video. **Note:
This issue is specific to MP4 format; FLV recordings are not affected.**
When WebRTC streams are converted to RTMP and then muxed to MP4, the
audio and video tracks may start at different timestamps. The MP4 muxer
was not accounting for this timing offset between the first audio and
video samples in the STTS (Sample Time-to-Sample) table, causing the
tracks to be misaligned in the final MP4 file.
Introduces `SrsMp4DvrJitter` class specifically for MP4 audio/video
synchronization:
- **Timestamp Tracking**: Records the DTS of the first audio and video
samples
- **Offset Calculation**: Computes the timing difference between track
start times
- **MP4 STTS Correction**: Sets appropriate `sample_delta` values in the
MP4 STTS table to maintain proper A/V sync
- Added `SrsMp4DvrJitter` class in `srs_kernel_mp4.hpp/cpp`
- Integrated jitter correction into `SrsMp4SampleManager::write_track()`
for MP4 format only
- Added comprehensive unit tests covering various timing scenarios
- **Scope**: Changes are isolated to MP4 kernel code and do not affect
FLV processing
This fix ensures that MP4 DVR recordings from WebRTC streams maintain
proper audio/video synchronization regardless of the relative timing of
the first audio and video frames, while leaving FLV format processing
unchanged.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: john <hondaxiao@tencent.com>
Co-authored-by: winlin <winlinvip@gmail.com>
Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Currently, SRS only supports HLS with MPEG-TS format segment files, but
for LL-HLS and HEVC, it requires the fMP4 format. See #4327 for details.
Furthermore, fMP4 has a smaller overhead compared to TS, and fMP4 can be
used for DVR. In short, fMP4 is definitely the future segment format for
HLS.
Start SRS with the config file that enables HLS with fMP4:
```
./objs/srs -c conf/hls.mp4.conf
```
Publish stream by FFmpeg:
```
ffmpeg -re -i doc/source.flv -c copy -f flv rtmp://localhost/live/livestream
```
Play the stream by SRS player:
[http://localhost:8080/live/livestream.m3u8](http://localhost:8080/players/srs_player.html?stream=livestream.m3u8)
Finished by AI:
* [AI: Change init.mp4 to the same directory of
m3u8.](17621c8442)
* [AI: Fix the error handling
bug.](af3758a592)
* [AI: Fix Chrome stuttering
problem.](aaab60c314)
---------
Co-authored-by: winlin <winlinvip@gmail.com>
Rtp packets may be retransmitted, disordered, jittery, delayed,
etc.There may be abnormalities when converting to rtmp.
To reproduce this problem, you need to set the network reordering by
[tc-ui](https://github.com/ossrs/tc-ui). Note that you need a linux
server, and start it by docker:
```bash
docker run --network=host --privileged -it --restart always -d \
--name tc -v /lib/modules:/lib/modules:ro ossrs/tc-ui:1
```
Set up 5% packet reordering and a 1ms delay; then you will notice that
the audio is stuttering, somewhat noisy, and lacks fluency.
```bash
curl http://localhost:2023/tc/api/v1/config/raw -X POST \
-d 'tcset ens5 --direction incoming --delay 40ms --reordering 5% --port 8000'
```
> Note: Even without network conditions, the natural state can also
cause packet reordering, especially in public cloud platforms such as
AWS EC2.
> Note: You can use command `curl
http://localhost:2023/tc/api/v1/config/raw -X POST -d 'tcdel --all
ens5'` to reset the network condition settings.
Check the web console, you will see the reordering setup:
<img width="500" alt="TC Settings"
src="https://github.com/user-attachments/assets/b278fdf4-9fcc-4aac-b534-dfa34e28c371"
/>
Then, publish stream via WHIP: http://localhost:8080/players/whip.html
And, play via HTTP-FLV: http://localhost:8080/players/srs_player.html
Finished by AI:
* [AI: Extract audio jitter buffer to class
AudioPacketCache](a4097d9374)
* [AI: Add utest and fix
bug.](c919227af5)
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: winlin <winlinvip@gmail.com>
## Introduce
This PR adds support for viewing streams via the RTSP protocol. Note
that it only supports viewing streams, not publishing streams via RTSP.
Currently, only publishing via RTMP is supported, which is then
converted to RTSP. Further work is needed to support publishing RTC/SRT
streams and converting them to RTSP.
## Usage
Build and run SRS with RTSP support:
```
cd srs/trunk && ./configure --rtsp=on && make -j16
./objs/srs -c conf/rtsp.conf
```
Push stream via RTMP by FFmpeg:
```
ffmpeg -re -i doc/source.flv -c copy -f flv rtmp://localhost/live/livestream
```
View the stream via RTSP protocol, try UDP first, then use TCP:
```
ffplay -i rtsp://localhost:8554/live/livestream
```
Or specify the transport protocol with TCP:
```
ffplay -rtsp_transport tcp -i rtsp://localhost:8554/live/livestream
```
## Unit Test
Run utest for RTSP:
```
./configure --utest=on & make utest -j16
./objs/srs_utest
```
## Regression Test
You need to start SRS for regression testing.
```
./objs/srs -c conf/regression-test-for-clion.conf
```
Then run regression tests for RTSP.
```
cd srs/trunk/3rdparty/srs-bench
go test ./srs -mod=vendor -v -count=1 -run=TestRtmpPublish_RtspPlay
```
## Blackbox Test
For blackbox testing, SRS will be started by utest, so there is no need
to start SRS manually.
```
cd srs/trunk/3rdparty/srs-bench
go test ./blackbox -mod=vendor -v -count=1 -run=TestFast_RtmpPublish_RtspPlay_Basic
```
## UDP Transport
As UDP requires port allocation, this PR doesn't support delivering
media stream via UDP transport, so it will fail if you try to use UDP as
transport:
```
ffplay -rtsp_transport udp -i rtsp://localhost:8554/live/livestream
[rtsp @ 0x7fbc99a14880] method SETUP failed: 461 Unsupported Transport
rtsp://localhost:8554/live/livestream: Protocol not supported
[2025-07-05 21:30:52.738][WARN][14916][7d7gf623][35] RTSP: setup failed: code=2057
(RtspTransportNotSupported) : UDP transport not supported, only TCP/interleaved mode is supported
```
There are no plans to support UDP transport for RTSP. In the real world,
UDP is rarely used; the vast majority of RTSP traffic uses TCP.
## Play Before Publish
RTSP supports audio with AAC and OPUS codecs, which is significantly
different from RTMP or WebRTC.
RTSP uses commands to exchange SDP and specify the audio track to play,
unlike WHEP or HTTP-FLV, which use the query string of the URL. RTSP
depends on the player’s behavior, making it very difficult to use and
describe.
Considering the feature that allows playing the stream before publishing
it, it requires generating some default parameters in the SDP. For OPUS,
the sample rate is 48 kHz with 2 channels, while AAC is more complex,
especially regarding the sample rate, which may be 44.1 kHz, 32 kHz, or
48 kHz.
Therefore, for RTSP, we cannot support play-then-publish. Instead, there
must already be a stream when playing it, so that the audio codec is
determined.
## Opus Codec
No Opus codec support for RTSP, because for RTC2RTSP, it always converts
RTC to RTMP frames, then converts them to RTSP packets. Therefore, the
audio codec is always AAC after converting RTC to RTMP.
This means the bridge architecture needs some changes. We need a new
bridge that binds to the target protocol. For example, RTC2RTMP converts
the audio codec, but RTC2RTSP keeps the original audio codec.
Furthermore, the RTC2RTMP bridge should also support bypassing the Opus
codec if we use enhanced-RTMP, which supports the Opus audio codec. I
think it should be configurable to either transcode or bypass the audio
codec. However, this is not relevant to RTSP.
## AI Contributor
Below commits are contributed by AI:
* [AI: Remove support for media transport via
UDP.](755686229f)
* [AI: Add crutial logs for each RTSP
stage.](9c8cbe7bde)
* [AI: Support AAC doec for
RTSP.](7d7cc12bae)
* [AI: Add option --rtsp for
RTSP.](f67414d9ee)
* [AI: Extract SrsRtpVideoBuilder for RTC and
RTSP.](562e76b904)
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: winlin <winlinvip@gmail.com>
For H.264, only when the NAL Type is 1, 2, 3, or 4 is it possible for
B-frames to be present; that is, non-IDR pictures and slice data.
The current `SrsVideoFrame::parse_avc_bframe()` function uses incorrect
logic to determine if a NALU can contain B-frames. The original
implementation only checked for specific NALU types (IDR, SPS, PPS) to
mark as non-B-frames, but this approach misses many other NALU types
that cannot contain B-frames according to the H.264 specification.
According to H.264 specification (ISO_IEC_14496-10-AVC-2012.pdf, Table
7-1), B-frames can **only** exist in these specific NALU types:
- Type 1: Non-IDR coded slice (`SrsAvcNaluTypeNonIDR`)
- Type 2: Coded slice data partition A (`SrsAvcNaluTypeDataPartitionA`)
- Type 3: Coded slice data partition B (`SrsAvcNaluTypeDataPartitionB`)
- Type 4: Coded slice data partition C (`SrsAvcNaluTypeDataPartitionC`)
All other NALU types (IDR=5, SEI=6, SPS=7, PPS=8, AUD=9, etc.) cannot
contain B-frames by definition.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: winlin <winlinvip@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>
1. When the chunk message header employs type 1 and type 2, the extended
timestamp denotes the time delta.
2. When the DTS (Decoding Time Stamp) experiences a jump and exceeds
16777215, there can be errors in DTS calculation, and if the audio and
video delta differs, it may result in audio-video synchronization
issues.
---------
`TRANS_BY_GPT4`
---------
Co-authored-by: 彭治湘 <zuolengchan@douyu.tv>
Co-authored-by: Haibo Chen(陈海博) <495810242@qq.com>
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
Co-authored-by: winlin <winlinvip@gmail.com>
SrsUniquePtr does not support array or object created by malloc, because
we only use delete to dispose the resource. You can use a custom
function to free the memory allocated by malloc or other allocators.
```cpp
char* p = (char*)malloc(1024);
SrsUniquePtr<char> ptr(p, your_free_chars);
```
This is used to replace the SrsAutoFreeH. For example:
```cpp
addrinfo* r = NULL;
SrsAutoFreeH(addrinfo, r, freeaddrinfo);
getaddrinfo("127.0.0.1", NULL, &hints, &r);
```
Now, this can be replaced by:
```cpp
addrinfo* r = NULL;
getaddrinfo("127.0.0.1", NULL, &hints, &r);
SrsUniquePtr<addrinfo> r2(r, freeaddrinfo);
```
Please aware that there is a slight difference between SrsAutoFreeH and
SrsUniquePtr. SrsAutoFreeH will track the address of pointer, while
SrsUniquePtr will not.
```cpp
addrinfo* r = NULL;
SrsAutoFreeH(addrinfo, r, freeaddrinfo); // r will be freed even r is changed later.
SrsUniquePtr<addrinfo> ptr(r, freeaddrinfo); // crash because r is an invalid pointer.
```
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: john <hondaxiao@tencent.com>
```
../../../src/utest/srs_utest_st.cpp:27: Failure
Expected: (st_time_2 - st_time_1) <= (100), actual: 119 vs 100
[ FAILED ] StTest.StUtimeInMicroseconds (0 ms)
```
Maybe github's vm, running the action jobs, is slower. I notice this
error happens frequently, so let the UT pass by increase the number.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: winlin <winlinvip@gmail.com>
try to fix#3978
**Background**
check #3978
**Research**
I referred the Android platform's solution, because I have android
background, and there is a loop to handle message inside android.
ff007a03c0/core/java/android/os/Handler.java (L701-L706C6)
```
public final boolean sendMessageDelayed(@NonNull Message msg, long delayMillis) {
if (delayMillis < 0) {
delayMillis = 0;
}
return sendMessageAtTime(msg, SystemClock.uptimeMillis() + delayMillis);
}
```
59d9dc1f50/libutils/SystemClock.cpp (L37-L51)
```
/*
* native public static long uptimeMillis();
*/
int64_t uptimeMillis()
{
return nanoseconds_to_milliseconds(uptimeNanos());
}
/*
* public static native long uptimeNanos();
*/
int64_t uptimeNanos()
{
return systemTime(SYSTEM_TIME_MONOTONIC);
}
```
59d9dc1f50/libutils/Timers.cpp (L32-L55)
```
#if defined(__linux__)
nsecs_t systemTime(int clock) {
checkClockId(clock);
static constexpr clockid_t clocks[] = {CLOCK_REALTIME, CLOCK_MONOTONIC,
CLOCK_PROCESS_CPUTIME_ID, CLOCK_THREAD_CPUTIME_ID,
CLOCK_BOOTTIME};
static_assert(clock_id_max == arraysize(clocks));
timespec t = {};
clock_gettime(clocks[clock], &t);
return nsecs_t(t.tv_sec)*1000000000LL + t.tv_nsec;
}
#else
nsecs_t systemTime(int clock) {
// TODO: is this ever called with anything but REALTIME on mac/windows?
checkClockId(clock);
// Clock support varies widely across hosts. Mac OS doesn't support
// CLOCK_BOOTTIME (and doesn't even have clock_gettime until 10.12).
// Windows is windows.
timeval t = {};
gettimeofday(&t, nullptr);
return nsecs_t(t.tv_sec)*1000000000LL + nsecs_t(t.tv_usec)*1000LL;
}
#endif
```
For Linux system, we can use `clock_gettime` api, but it's first
appeared in Mac OSX 10.12.
`man clock_gettime`
The requirement is to find an alternative way to get the timestamp in
microsecond unit, but the `clock_gettime` get nanoseconds, the math
formula is the nanoseconds / 1000 = microsecond. Then I check the
performance of this api + math division.
I used those code to check the `clock_gettime` performance.
```
#include <sys/time.h>
#include <time.h>
#include <stdio.h>
#include <unistd.h>
int main() {
struct timeval tv;
struct timespec ts;
clock_t start;
clock_t end;
long t;
while (1) {
start = clock();
gettimeofday(&tv, NULL);
end = clock();
printf("gettimeofday clock is %lu\n", end - start);
printf("gettimeofday is %lld\n", (tv.tv_sec * 1000000LL + tv.tv_usec));
start = clock();
clock_gettime(CLOCK_MONOTONIC, &ts);
t = ts.tv_sec * 1000000L + ts.tv_nsec / 1000L;
end = clock();
printf("clock_monotonic clock is %lu\n", end - start);
printf("clock_monotonic: seconds is %ld, nanoseconds is %ld, sum is %ld\n", ts.tv_sec, ts.tv_nsec, t);
start = clock();
clock_gettime(CLOCK_MONOTONIC_RAW, &ts);
t = ts.tv_sec * 1000000L + ts.tv_nsec / 1000L;
end = clock();
printf("clock_monotonic_raw clock is %lu\n", end - start);
printf("clock_monotonic_raw: nanoseconds is %ld, sum is %ld\n", ts.tv_nsec, t);
sleep(3);
}
return 0;
}
```
Here is output:
env: Mac OS M2 chip.
```
gettimeofday clock is 11
gettimeofday is 1709775727153949
clock_monotonic clock is 2
clock_monotonic: seconds is 1525204, nanoseconds is 409453000, sum is 1525204409453
clock_monotonic_raw clock is 2
clock_monotonic_raw: nanoseconds is 770493000, sum is 1525222770493
```
We can see the `clock_gettime` is faster than `gettimeofday`, so there
are no performance risks.
**MacOS solution**
`clock_gettime` api only available until mac os 10.12, for the mac os
older than 10.12, just keep the `gettimeofday`.
check osx version in `auto/options.sh`, then add MACRO in
`auto/depends.sh`, the MACRO is `MD_OSX_HAS_NO_CLOCK_GETTIME`.
**CYGWIN**
According to google search, it seems the
`clock_gettime(CLOCK_MONOTONIC)` is not support well at least 10 years
ago, but I didn't own an windows machine, so can't verify it. so keep
win's solution.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
By setting the env `ASAN_OPTIONS=halt_on_error=0`, we can ignore memory
leaks, see
https://github.com/google/sanitizers/wiki/AddressSanitizerFlags
By setting env `ASAN_OPTIONS=detect_leaks=0`, we can disable memory
leaking detection in parent process when forking for daemon.
1. don't use static variable to store the result;
2. add more UT to handle the multi value and values with whitespaces;
related to #409216e569d823/trunk/src/app/srs_app_config.cpp (L71-L82)
`static SrsConfDirective* dir` removed, this static var here is to avoid
the memory leak, I add the `SrsConfDirective` instance to the `env_dirs`
directive container, which will destroy itself inside `SrsConfig`
destructor.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
1. add on_connect & on_close directives to conf/full.conf;
2. let http_hooks env overwrite support multi values; e.g.
SRS_VHOST_HTTP_HOOKS_ON_CONNECT="http://127.0.0.1/api/connecthttp://localhost/api/connect"
related to
https://github.com/ossrs/srs/issues/1222#issuecomment-2170424703
Above comments said `http_hook` env may not works as expected, as I
found there are still has some issue in `http_hooks` env configuration,
but this PR may not target above problem.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
The session HLS manifest file lacks a terminating newline in the final
line.
This may cause strict players to reject it.
---------
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>
The object relations:
Session manages SIP and Media object using shared resource or shared
ptr. Note that I actually use SrsExecutorCoroutine to delete the object
when each coroutine is done, because there is always a dedicate
coroutine for each object.
For SIP and Media object, they directly use the session by raw pointer,
it's safe because session always live longer than session and media
object.
---
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Fix#4037 SRS should not send the publish start message
`onStatus(NetStream.Publish.Start)` if hooks fail, which causes OBS to
repeatedly reconnect.
Note that this fix does not send an RTMP error message when publishing
fails, because neither OBS nor FFmpeg process this specific error
message; they only display a general error.
Apart from the order of messages, nothing else has been changed.
Previously, we sent the publish start message
`onStatus(NetStream.Publish.Start)` before the HTTP hook `on_publish`;
now, we have modified it to send this message after the HTTP hook.
