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>
In the scenario of converting WebRTC to RTMP, this conversion will not
proceed until an SenderReport is received; for reference, see:
https://github.com/ossrs/srs/pull/2470.
Thus, if HTTP-FLV streaming is attempted before the SR is received, the
FLV Header will contain only audio, devoid of video content.
This error can be resolved by disabling `guess_has_av` in the
configuration file, since we can guarantee that both audio and video are
present in the test cases.
However, in the original regression tests, the
`TestRtcPublish_HttpFlvPlay` test case contains a bug:
5a404c089b/trunk/3rdparty/srs-bench/srs/rtc_test.go (L2421-L2424)
The test would pass when `hasAudio` is true and `hasVideo` is false,
which is actually incorrect. Therefore, it has been revised so that the
test now only passes if both values are true.
---------
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
Co-authored-by: winlin <winlinvip@gmail.com>
To enable H.265 support for the WebRTC protocol, upgrade the pion/webrtc
library to version 4.
---------
Co-authored-by: john <hondaxiao@tencent.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>
>
https://github.com/marketplace/actions/delete-artifact?version=v5.0.0#-compatibility
The current version of `actions/upload-artifact` is `v4`, and the
corresponding version for `delete-artifact` should be `v5`.
---------
`TRANS_BY_GPT4`
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: winlin <winlinvip@gmail.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>
## 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>
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>
Please note that the proxy server is a new architecture or the next
version of the Origin Cluster, which allows the publication of multiple
streams. The SRS origin cluster consists of a group of origin servers
designed to handle a large number of streams.
```text
+-----------------------+
+---+ SRS Proxy(Deployment) +------+---------------------+
+-----------------+ | +-----------+-----------+ + +
| LB(K8s Service) +--+ +(Redis/MESH) + SRS Origin Servers +
+-----------------+ | +-----------+-----------+ + (Deployment) +
+---+ SRS Proxy(Deployment) +------+---------------------+
+-----------------------+
```
The new origin cluster is designed as a collection of proxy servers. For
more information, see [Discussion
#3634](https://github.com/ossrs/srs/discussions/3634). If you prefer to
use the old origin cluster, please switch to a version before SRS 6.0.
A proxy server can be used for a set of origin servers, which are
isolated and dedicated origin servers. The main improvement in the new
architecture is to store the state for origin servers in the proxy
server, rather than using MESH to communicate between origin servers.
With a proxy server, you can deploy origin servers as stateless servers,
such as in a Kubernetes (K8s) deployment.
Now that the proxy server is a stateful server, it uses Redis to store
the states. For faster development, we use Go to develop the proxy
server, instead of C/C++. Therefore, the proxy server itself is also
stateless, with all states stored in the Redis server or cluster. This
makes the new origin cluster architecture very powerful and robust.
The proxy server is also an architecture designed to solve multiple
process bottlenecks. You can run hundreds of SRS origin servers with one
proxy server on the same machine. This solution can utilize multi-core
machines, such as servers with 128 CPUs. Thus, we can keep SRS
single-threaded and very simple. See
https://github.com/ossrs/srs/discussions/3665#discussioncomment-6474441
for details.
```text
+--------------------+
+-------+ SRS Origin Server +
+ +--------------------+
+
+-----------------------+ + +--------------------+
+ SRS Proxy(Deployment) +------+-------+ SRS Origin Server +
+-----------------------+ + +--------------------+
+
+ +--------------------+
+-------+ SRS Origin Server +
+--------------------+
```
Keep in mind that the proxy server for the Origin Cluster is designed to
handle many streams. To address the issue of many viewers, we will
enhance the Edge Cluster to support more protocols.
```text
+------------------+ +--------------------+
+ SRS Edge Server +--+ +-------+ SRS Origin Server +
+------------------+ + + +--------------------+
+ +
+------------------+ + +-----------------------+ + +--------------------+
+ SRS Edge Server +--+-----+ SRS Proxy(Deployment) +------+-------+ SRS Origin Server +
+------------------+ + +-----------------------+ + +--------------------+
+ +
+------------------+ + + +--------------------+
+ SRS Edge Server +--+ +-------+ SRS Origin Server +
+------------------+ +--------------------+
```
With the new Origin Cluster and Edge Cluster, you have a media system
capable of supporting a large number of streams and viewers. For
example, you can publish 10,000 streams, each with 100,000 viewers.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
The heartbeat of SRS is a timer that requests an HTTP URL. We can use
this heartbeat to report the necessary information for registering the
backend server with the proxy server.
```text
SRS(backend) --heartbeat---> Proxy server
```
A proxy server is a specialized load balancer for media servers. It
operates at the application level rather than the TCP level. For more
information about the proxy server, see issue #4158.
Note that we will merge this PR into SRS 5.0+, allowing the use of SRS
5.0+ as the backend server, not limited to SRS 7.0. However, the proxy
server is introduced in SRS 7.0.
It's also possible to implement a registration service, allowing you to
use other media servers as backend servers. For example, if you gather
information about an nginx-rtmp server and register it with the proxy
server, the proxy will forward RTMP streams to nginx-rtmp. The backend
server is not limited to SRS.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
1. Do not create a source when mounting FLV because it may not unmount
FLV when freeing the source. If you access the FLV stream without any
publisher, then wait for source cleanup and review the FLV stream again,
there is an annoying warning message.
```bash
# View HTTP FLV stream by curl, wait for stream to be ready.
# curl http://localhost:8080/live/livestream.flv -v >/dev/null
HTTP #0 127.0.0.1:58026 GET http://localhost:8080/live/livestream.flv, content-length=-1
new live source, stream_url=/live/livestream
http: mount flv stream for sid=/live/livestream, mount=/live/livestream.flv
# Cancel the curl and trigger source cleanup without http unmount.
client disconnect peer. ret=1007
Live: cleanup die source, id=[], total=1
# View the stream again, it fails.
# curl http://localhost:8080/live/livestream.flv -v >/dev/null
HTTP #0 127.0.0.1:58040 GET http://localhost:8080/live/livestream.flv, content-length=-1
serve error code=1097(NoSource)(No source found) : process request=0 : cors serve : serve http : no source for /live/livestream
serve_http() [srs_app_http_stream.cpp:641]
```
> Note: There is an inconsistency. The first time, you can access the
FLV stream and wait for the publisher, but the next time, you cannot.
2. Create a source when starting to serve the FLV client. We do not need
to create the source when creating the HTTP handler. Instead, we should
try to create the source in the cache or stream. Because the source
cleanup does not unmount the HTTP handler, the handler remains after the
source is destroyed. The next time you access the FLV stream, the source
is not found.
```cpp
srs_error_t SrsHttpStreamServer::hijack(ISrsHttpMessage* request, ISrsHttpHandler** ph) {
SrsSharedPtr<SrsLiveSource> live_source;
if ((err = _srs_sources->fetch_or_create(r.get(), server, live_source)) != srs_success) { }
if ((err = http_mount(r.get())) != srs_success) { }
srs_error_t SrsBufferCache::cycle() {
SrsSharedPtr<SrsLiveSource> live_source = _srs_sources->fetch(req);
if (!live_source.get()) {
return srs_error_new(ERROR_NO_SOURCE, "no source for %s", req->get_stream_url().c_str());
}
srs_error_t SrsLiveStream::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r) {
SrsSharedPtr<SrsLiveSource> live_source = _srs_sources->fetch(req);
if (!live_source.get()) {
return srs_error_new(ERROR_NO_SOURCE, "no source for %s", req->get_stream_url().c_str());
}
```
> Note: We should not create the source in hijack, instead, we create it
in cache or stream:
```cpp
srs_error_t SrsHttpStreamServer::hijack(ISrsHttpMessage* request, ISrsHttpHandler** ph) {
if ((err = http_mount(r.get())) != srs_success) { }
srs_error_t SrsBufferCache::cycle() {
SrsSharedPtr<SrsLiveSource> live_source;
if ((err = _srs_sources->fetch_or_create(req, server_, live_source)) != srs_success) { }
srs_error_t SrsLiveStream::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r) {
SrsSharedPtr<SrsLiveSource> live_source;
if ((err = _srs_sources->fetch_or_create(req, server_, live_source)) != srs_success) { }
```
> Note: This fixes the failure and annoying warning message, and
maintains consistency by always waiting for the stream to be ready if
there is no publisher.
3. Fail the http request if the HTTP handler is disposing, and also keep
the handler entry when disposing the stream, because we should dispose
the handler entry and stream at the same time.
```cpp
srs_error_t SrsHttpStreamServer::http_mount(SrsRequest* r) {
entry = streamHandlers[sid];
if (entry->disposing) {
return srs_error_new(ERROR_STREAM_DISPOSING, "stream is disposing");
}
void SrsHttpStreamServer::http_unmount(SrsRequest* r) {
std::map<std::string, SrsLiveEntry*>::iterator it = streamHandlers.find(sid);
SrsUniquePtr<SrsLiveEntry> entry(it->second);
entry->disposing = true;
```
> Note: If the disposal process takes a long time, this will prevent
unexpected behavior or access to the resource that is being disposed of.
4. In edge mode, the edge ingester will unpublish the source when the
last consumer quits, which is actually triggered by the HTTP stream.
While it also waits for the stream to quit when the HTTP unmounts, there
is a self-destruction risk: the HTTP live stream object destroys itself.
```cpp
srs_error_t SrsLiveStream::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r) {
SrsUniquePtr<SrsLiveConsumer> consumer(consumer_raw); // Trigger destroy.
void SrsHttpStreamServer::http_unmount(SrsRequest* r) {
for (;;) { if (!cache->alive() && !stream->alive()) { break; } // A circle reference.
mux.unhandle(entry->mount, stream.get()); // Free the SrsLiveStream itself.
```
> Note: It also introduces a circular reference in the object
relationships, the stream reference to itself when unmount:
```text
SrsLiveStream::serve_http
-> SrsLiveConsumer::~SrsLiveConsumer -> SrsEdgeIngester::stop
-> SrsLiveSource::on_unpublish -> SrsHttpStreamServer::http_unmount
-> SrsLiveStream::alive
```
> Note: We should use an asynchronous worker to perform the cleanup to
avoid the stream destroying itself and to prevent self-referencing.
```cpp
void SrsHttpStreamServer::http_unmount(SrsRequest* r) {
entry->disposing = true;
if ((err = async_->execute(new SrsHttpStreamDestroy(&mux, &streamHandlers, sid))) != srs_success) { }
```
> Note: This also ensures there are no circular references and no
self-destruction.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Edge FLV is not working because it is stuck in an infinite loop waiting.
Previously, there was no need to wait for exit since resources were not
being cleaned up. Now, since resources need to be cleaned up, it must
wait for all active connections to exit, which causes this issue.
To reproduce the issue, start SRS edge, run the bellow command and press
`CTRL+C` to stop the request:
```bash
curl http://localhost:8080/live/livestream.flv -v >/dev/null
```
It will cause edge to fetch stream from origin, and free the consumer
when client quit. When `SrsLiveStream::do_serve_http` return, it will
free the consumer:
```cpp
srs_error_t SrsLiveStream::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r) {
SrsUniquePtr<SrsLiveConsumer> consumer(consumer_raw);
```
Keep in mind that in this moment, the stream is alive, because only set
to not alive after this function return:
```cpp
alive_viewers_++;
err = do_serve_http(w, r); // Free 'this' alive stream.
alive_viewers_--; // Crash here, because 'this' is freed.
```
When freeing the consumer, it will cause the source to unpublish and
attempt to free the HTTP handler, which ultimately waits for the stream
not to be alive:
```cpp
SrsLiveConsumer::~SrsLiveConsumer() {
source_->on_consumer_destroy(this);
void SrsLiveSource::on_consumer_destroy(SrsLiveConsumer* consumer) {
if (consumers.empty()) {
play_edge->on_all_client_stop();
void SrsLiveSource::on_unpublish() {
handler->on_unpublish(req);
void SrsHttpStreamServer::http_unmount(SrsRequest* r) {
if (stream->entry) stream->entry->enabled = false;
for (; i < 1024; i++) {
if (!cache->alive() && !stream->alive()) {
break;
}
srs_usleep(100 * SRS_UTIME_MILLISECONDS);
}
```
After 120 seconds, it will free the stream and cause SRS to crash
because the stream is still active. In order to track this potential
issue, also add an important warning log:
```cpp
srs_warn("http: try to free a alive stream, cache=%d, stream=%d", cache->alive(), stream->alive());
```
SRS may crash if got this log.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
If SRS responds with this empty data packet, FFmpeg will receive an
empty stream, like `Stream #0:0: Data: none` in following logs:
```bash
ffmpeg -i rtmp://localhost:11935/live/livestream
# Stream #0:0: Data: none
# Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 30 kb/s
# Stream #0:2: Video: h264 (High), yuv420p(progressive), 768x320 [SAR 1:1 DAR 12:5], 212 kb/s, 25 fps, 25 tbr, 1k tbn
```
This won't cause the player to fail, but it will inconvenience the user
significantly. It may also cause FFmpeg slower to analysis the stream,
see #3767
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
When stopping the stream, it will wait for the HTTP Streaming to exit.
If the HTTP Streaming goroutine hangs, it will not exit automatically.
```cpp
void SrsHttpStreamServer::http_unmount(SrsRequest* r)
{
SrsUniquePtr<SrsLiveStream> stream(entry->stream);
if (stream->entry) stream->entry->enabled = false;
srs_usleep(...); // Wait for about 120s.
mux.unhandle(entry->mount, stream.get()); // Free stream.
}
srs_error_t SrsLiveStream::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
{
err = do_serve_http(w, r); // If stuck in here for 120s+
alive_viewers_--; // Crash at here, because stream has been deleted.
```
We should notify http stream connection to interrupt(expire):
```cpp
void SrsHttpStreamServer::http_unmount(SrsRequest* r)
{
SrsUniquePtr<SrsLiveStream> stream(entry->stream);
if (stream->entry) stream->entry->enabled = false;
stream->expire(); // Notify http stream to interrupt.
```
Note that we should notify all viewers pulling stream from this http
stream.
Note that we have tried to fix this issue, but only try to wait for all
viewers to quit, without interrupting the viewers, see
https://github.com/ossrs/srs/pull/4144
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
1. Remove the srs_global_dispose, which causes the crash when still
publishing when quit.
2. Always call _srs_thread_pool->initialize for single thread.
3. Support `--signal-api` to send signal by HTTP API, because CLion
eliminate the signals.
---
Co-authored-by: Jacob Su <suzp1984@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.
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>
New features for valgrind:
1. ST: Support /api/v1/valgrind for leaking check.
2. ST: Support /api/v1/valgrind?check=full|added|changed|new|quick
To use Valgrind to detect memory leaks in SRS, even though Valgrind
hooks are supported in ST, there are still many false positives. A more
reasonable approach is to have Valgrind report incremental memory leaks.
This way, global and static variables can be avoided, and detection can
be achieved without exiting the program. Follow these steps:
1. Compile SRS with Valgrind support: `./configure --valgrind=on &&
make`
2. Start SRS with memory leak detection enabled: `valgrind
--leak-check=full ./objs/srs -c conf/console.conf`
3. Trigger memory detection by using curl to access the API and generate
calibration data. There will still be many false positives, but these
can be ignored: `curl http://127.0.0.1:1985/api/v1/valgrind?check=added`
4. Perform load testing or test the suspected leaking functionality,
such as RTMP streaming: `ffmpeg -re -i doc/source.flv -c copy -f flv
rtmp://127.0.0.1/live/livestream`
5. Stop streaming and wait for SRS to clean up the Source memory,
approximately 30 seconds.
6. Perform incremental memory leak detection. The reported leaks will be
very accurate at this point: `curl
http://127.0.0.1:1985/api/v1/valgrind?check=added`
> Note: To avoid interference from the HTTP request itself on Valgrind,
SRS uses a separate coroutine to perform periodic checks. Therefore,
after accessing the API, you may need to wait a few seconds for the
detection to be triggered.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
I did some preliminary code inspection. The two playback endpoints share
the same `SrsLiveStream` instance. After the first one disconnects,
`alive_` is set to false.
```
alive_ = true;
err = do_serve_http(w, r);
alive_ = false;
```
In the `SrsHttpStreamServer::http_unmount(SrsRequest* r)` function,
`stream->alive()` is already false, so `mux.unhandle` will free the
`SrsLiveStream`. This causes the other connection coroutine to return to
its execution environment after the `SrsLiveStream` instance has already
been freed.
```
// Wait for cache and stream to stop.
int i = 0;
for (; i < 1024; i++) {
if (!cache->alive() && !stream->alive()) {
break;
}
srs_usleep(100 * SRS_UTIME_MILLISECONDS);
}
// Unmount the HTTP handler, which will free the entry. Note that we must free it after cache and
// stream stopped for it uses it.
mux.unhandle(entry->mount, stream.get());
```
`alive_` was changed from a `bool` to an `int` to ensure that
`mux.unhandle` is only executed after each connection's `serve_http` has
exited.
---------
Co-authored-by: liumengte <liumengte@visionular.com>
Co-authored-by: winlin <winlinvip@gmail.com>
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>
When unpublishing, the handler callback that will stop the coroutine:
```cpp
_can_publish = true;
handler->on_unpublish(req);
```
In this handler, the `http_unmount` will be called:
```cpp
void SrsHttpStreamServer::http_unmount(SrsRequest* r)
cache->stop();
```
In this `http_unmount` function, there could be context switching. In
such a situation, a new connection might publish the stream while the
unpublish process is freeing the stream, leading to a crash.
To prevent a new publisher, we should change the state only after all
handlers and hooks are completed.
---------
Co-authored-by: liumengte <liumengte@visionular.com>
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>
Sorry this is another pull request with same intention. But there is
more variants of h264 und h265 codecs and I think it is good to support
them all.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
Currently only libx264 ffmpeg encoder is supported. This pull request
add also h264_qsv. But maybe a more generic solution with oder encoders
would be useful to.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
For #3369 to support an external powerful SIP server, do not use the
embedded SIP server of SRS.
For more information, detailed steps, system architecture, and
background explanation, please see
https://ossrs.net/lts/zh-cn/docs/v6/doc/gb28181#external-sip
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
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>
## Describe ##
http_remux feature support config `has_audio`, `has_video` &
`guess_has_av` prop.
282d94d7bb/trunk/src/app/srs_app_http_stream.cpp (L630-L632)
Take `http_flv` as example, `srs` can accept both RTMP streams with only
audio, only video or both audio and video streams. It is controlled by
above three properties.
But `guess_has_av` is not implemented by `http_ts`. The problem is that
if I want publish a RTMP stream with audio or video track, the
`has_audio` and `has_video`, which are default true/on, must to be
config to match the RTMP stream, otherwise the `mpegts.js` player can't
play the `http-ts` stream.
## How to reproduce ##
1. `export SRS_VHOST_HTTP_REMUX_HAS_AUDIO=on; export
SRS_VHOST_HTTP_REMUX_HAS_VIDEO=on; export
SRS_VHOST_HTTP_REMUX_GUESS_HAS_AV=on; ./objs/srs -c
conf/http.ts.live.conf`
2. publish rtmp stream without video: `ffmpeg -re -stream_loop -1 -i
srs/trunk/doc/source.200kbps.768x320.flv -vn -acodec copy -f flv
rtmp://localhost/live/livestream`
3. open chrome browser, open
`http://localhost:8080/players/srs_player.html?schema=http`, go to
`LivePlayer`, input URL: `http://localhost:8080/live/livestream.ts`,
click play.
4. the `http://localhost:8080/live/livestream.ts` can not play.
## Solution ##
Let `http-ts` support `guess_has_av`, `http-flv` already supported. The
`guess_has_av` default value is ture/on, so the `http-ts|flv` can play
any streams with audio, video or both.
---------
Co-authored-by: Winlin <winlinvip@gmail.com>
This not only silences a deprecation warning by docker build, but also
makes it consistent as the other ENV statement already uses the new
syntax.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
1. Should always stop coroutine before close fd, see #511, #1784
2. When edge forwarder coroutine quit, always set the error code.
3. Do not unpublish if invalid state.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
## Cause
dash auto dispose is configured by seconds, but the code compare by
usecond, 1 second = 1,000,000 useconds.
releated to #4097
Bug introduced after #4097 supported Dash auto dispose after a timeout
without media data.
## How to reproduce
1. `./objs/srs -c conf/dash.conf`
2. publish a rtmp stream.
3. play dash stream. -> no dash stream, always 404 error.
---------
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. fix redundant null check, there is no potential risks by the way,
just redundant null check.
2. Potential use pointer after free, that's not true. So we can ignore
this one, or find a way to make stupid security tool happy.
---------
Co-authored-by: winlin <winlinvip@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>
Try to fix two blackbox test:
1. TestSlow_SrtPublish_HttpTsPlay_HEVC_Basic: fixed by enlarge the wait
time from 3 seconds to 4 before run ffprobe task, which will record the
stream by ffmpeg first.
2 TestSlow_SrtPublish_HlsPlay_HEVC_Basic: fixed by wait 16 seconds
before run ffprobe task.
About the 2 case: it seems ridiculous to add 16 seconds delay before run
ffprobe task.
> So, I start #4088 to check the github action workflow process, I start
this branch from a very earlier version `6.0.113
(srs/core/srs_core_version6.hpp)`, what I found it that, inside `SRS
blackbox-test`, the srs version `6.0.128`, the latest version, was
printed. That's really wired.
I confirmed this is not the SRS source code's problem, check
https://github.com/suzp1984/srs/actions/runs/9511600525/job/26218025559
the srs code 6.0.113 was checkout and running actions based on them,
still met same problem.
---------
Co-authored-by: winlin <winlinvip@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.
Fix#3967 There is an API `SSL_use_certificate_chain_file`, which can load the
certification chain and also single certificate.
---------
Co-authored-by: winlin <winlinvip@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.
When debugging the RTMP protocol, we can capture packets using tcpdump
and then replay the pcap file. For example:
```bash
cd ~/git/srs/trunk/3rdparty/srs-bench/pcap
tcpdump -i any -w t.pcap tcp port 1935
go run . -f ./t.pcap -s 127.0.0.1:1935
```
However, sometimes due to poor network conditions between the server and
the client, there may be many retransmitted packets. In such cases,
setting up a transparent TCP proxy that listens on port 1935 and
forwards to port 19350 can be a solution:
```bash
./objs/srs -c conf/origin.conf
cd 3rdparty/srs-bench/tcpproxy/ && go run main.go
tcpdump -i any -w t.pcap tcp port 19350
```
This approach allows for the implementation of packet dumping,
multipoint replication, or the provision of detailed timestamps and byte
information at the proxy. It enables the collection of debugging
information without the need to modify the server.
---------
`TRANS_BY_GPT4`
---------
Co-authored-by: john <hondaxiao@tencent.com>
1. When converting RTC to RTMP, it is necessary to synchronize the audio
and video timestamps. When the synchronization status changes, whether
it is unsynchronized or synchronized, print logs to facilitate
troubleshooting of such issues.
2. Chrome uses the STAP-A packet, which means a single RTP packet
contains SPS/PPS information. OBS WHIP, on the other hand, sends SPS and
PPS in separate RTP packets. Therefore, SPS and PPS are in two
independent RTP packets, and SRS needs to cache these two packets.
---------
Co-authored-by: john <hondaxiao@tencent.com>
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>
For WebRTC:
when player before publisher, it will happen track pt didn't change.
- At source change step, change track pt
---------
Co-authored-by: mingche.tsai <w41203208.work@gmail.com>
Co-authored-by: john <hondaxiao@tencent.com>
Accommodate certain complex parameters that include the "=" character,
for example.
`configure --extra-flags="-O2 -D_FORTIFY_SOURCE=2"`
---------
Co-authored-by: john <hondaxiao@tencent.com>
Description
A crash occurs when a forward relay connection has not been established
and an unpublish event is triggered simultaneously. For instance, if DVR
and forward are configured with a specified DVR path that already
exists, initiating a stream will trigger a crash.
Objective
Fix the crash caused by the forward mechanism.
Additional Information
For detailed reproduction steps, please refer to issue #3901.
---------
Co-authored-by: john <hondaxiao@tencent.com>
In an SDK that supports RTC Opus stereo, the parameter "stereo=1" may
appear. SRS (Spatial Reference System) needs to handle this correctly
and return an answer to enable WebRTC stereo support.
---------
`TRANS_BY_GPT4`
Security is the built-in IP whitelist feature of SRS, which allows and
denies certain IP and IP range users. Previously, it only supported
RTMP, but this PR now supports HTTP-FLV, HLS, WebRTC, SRT, and other
protocols.
See https://ossrs.io/lts/en-us/docs/v6/doc/security as example.
---------
Co-authored-by: john <hondaxiao@tencent.com>
The `-` character, when placed in the middle of a regular expression, is
interpreted as a range. It must be placed at the beginning or end to be
interpreted as a literal character.
---------
`TRANS_BY_GPT4`
---------
Co-authored-by: john <hondaxiao@tencent.com>
The `ffmpeg-opus` tool allows you to control the delay using the
`opus_delay` option. The minimum delay can be set to 2.5ms. However, in
practice, you cannot set it this low. You need to set at least 10 frames
to allow the audio encoder to lookahead. Otherwise, the sound will be
distorted.
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
In pure audio mode, there are no keyframes. Therefore, we can only rely
on the length of the slice to determine whether it should be output.
`hls_aof_ratio` is the coefficient that, once reached, will generate a
new slice.
In scenarios with video, if the `hls_aof_ratio` is too small, for
example 1.2, and the GOP (Group of Pictures) is 10 seconds, then a slice
will definitely be generated at 12 seconds. At this point, if there are
no keyframes, it will cause the next slice to start with a non-keyframe.
A safer coefficient is twice the GOP (Group of Pictures). This way, it
won't trigger incorrectly and prevent the individual transcoding of a ts
segment file.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
1. After enabling FFmpeg opus, the transcoding time for each opus packet
is around 4ms.
2. To speed up case execution, our test publisher sends 400 opus packets
at intervals of 1ms.
3. After the publisher starts, wait for 30ms, then the player starts.
4. Due to the lengthy processing time for each opus packet, SRS
continuously receives packets from the publisher, so it doesn't switch
coroutines and can't accept the player's connection.
5. Only after all opus packets are processed will it accept the player
connection. Therefore, the player doesn't receive any data, leading to
the failure of the case.
---------
Co-authored-by: winlin <winlinvip@gmail.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>
Follow the example in FFmpeg's doc, before calling the API
`avcodec_send_frame`, always use `av_frame_alloc` to create a new frame.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
SRS supports TCP WebRTC by reading 2 bytes of length, like `read(buf,
2)`. However, in some cases, it might receive 1 byte, causing subsequent
data to be incorrect and making it unable to push or play streams.
---------
Co-authored-by: john <hondaxiao@tencent.com>
Checking the HTTPS API or UDP connectivity for WHIP tests can be
difficult. For example, if the UDP port isn't available but the API is
fine, OBS only says it can't connect to the server. It's hard to see the
HTTPS API response or check if the UDP port is available.
This feature lets you set the ice username and password in SRS. You can
then send a STUN request using nc and see the response, making it easier
to check UDP port connectivity.
1. Use curl to test the WHIP API, including ice-frag and ice-pwd
queries.
2. Use nc to send a STUN binding request to test UDP connectivity.
3. If both the API and UDP are working, you should get a STUN response.
---------
Co-authored-by: john <hondaxiao@tencent.com>
When using Docker, logs are usually printed to console (stdout and
stderr). However, since Docker detection occurs late, after log
initialization, the default log output may be incorrect. In Docker, logs
may still be written to a file instead of the console as expected.
Additionally, the Dockerfile has been improved with a new environment
variable `SRS_IN_DOCKER=on` to clearly indicate a Docker environment. If
automatic Docker detection fails, the configuration will be read, and
this variable will correctly inform SRS that it's in a Docker
environment.
Lastly, the default configuration values have been improved for Docker
environments. By default, `SRS_LOG_TANK=console` and daemon mode is
disabled.
---------
Co-authored-by: john <hondaxiao@tencent.com>
The fix is for the DH_set_length error. As shown in lines 2-5, OpenSSL
3.0 added a check for length, which allowed this issue to be exposed.
```
1 if (dh->params.q == NULL) {
2 /* secret exponent length, must satisfy 2^(l-1) <= p */
3 if (dh->length != 0
4 && dh->length >= BN_num_bits(dh->params.p))
5 goto err;
6 l = dh->length ? dh->length : BN_num_bits(dh->params.p) - 1;
7 if (!BN_priv_rand_ex(priv_key, l, BN_RAND_TOP_ONE,
8 BN_RAND_BOTTOM_ANY, 0, ctx))
9 goto err;
... ...
}
```
---------
Co-authored-by: john <hondaxiao@tencent.com>
1. The comment on the ratio configuration says it can affect the slice
duration, but there is no effect after configuring it.
2. The default hls_td_ratio is 1.5, and after setting it to 1, the
duration is still slightly more than 10 seconds.
3. Even if the GOP is an integer, like 1 second, the slice is still a
non-integer, like 0.998 seconds, which seems a bit unreliable.
4. In the duration of the TS in the m3u8 file, it is one frame less than
the duration of the slice.
5. Set hls_dispose to 120s to dispose HLS files when no stream.
6. Use docker.conf for docker.
Before this patch:
```
#EXTINF:10.983, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
After this patch:
```
#EXTINF:10.000, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
Note: If the fragment is set to 10 seconds, but the GOP size cannot be
divided by 10, such as not 1, 2, 5, or 10, then the duration of ts will
still be more than 10 seconds.
---------
Co-authored-by: john <hondaxiao@tencent.com>
When I compile on OpenHarmony, I encounter an error at the
pthread_setname_np function:
```
./src/app/srs_app_threads.cpp:53:10: error: functions that differ only in their return type cannot be overloaded
void pthread_setname_np(pthread_t trd, const char* name) {
/data/local/ohos-sdk/linux/native/llvm/bin/../../sysroot/usr/include/pthread.h:379:5: note: previous declaration is here
int pthread_setname_np(pthread_t, const char *);
```
Our libc is using musl-libc and has no defined __GLIBC__, so we wanted
to add a judgment that __GLIBC__ already defined.
Please see https://github.com/ossrs/srs/issues/3803 for detail:
1. When using FFmpeg with the `-map 0` option, there may be a 4-byte
timecode in the AMF0 Data.
2. SRS should be able to handle this packet without causing a parsing
error, as it's generally expected to be an AMF0 string, not a 4-byte
timecode.
3. Disregard the timecode since SRS doesn't utilize it.
See [Error submitting a packet to the muxer: Broken pipe, Error muxing a
packet](https://trac.ffmpeg.org/ticket/10565)
---------
Co-authored-by: john <hondaxiao@tencent.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>
ISO_IEC_14496-10-AVC-2012.pdf, page 65
7.4.1.1 Encapsulation of an SODB within an RBSP (informative)
... 00 00 03 xx, the 03 byte should be drop where xx represents any 2
bit pattern: 00, 01, 10, or 11.
---------
Co-authored-by: john <hondaxiao@tencent.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: winlin <winlin@vip.126.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>
remove unreachable links by python scripts:
```
def is_delete_issue(link):
try:
response = requests.get(link)
except RequestException as e:
print(f"An error occurred while trying to get the link: {e}")
return False
return "This issue has been deleted." in response.text
def remove_unreachable_links(dir):
string_to_search = re.compile(r'// @see https://github\.com/ossrs/srs/issues/.*')
file_list = util.find_files_with_extension(dir, ".cpp", True)
for file in file_list:
lines = []
with open(file, "r", encoding="utf-8") as f:
lines = f.readlines()
with open(file, "w", encoding="utf-8", newline="\n") as f:
for line in lines:
if string_to_search.search(line):
result = re.search(r'https://github\.com/ossrs/srs/issues/\d+', line)
if result:
link = result.group()
if is_delete_issue(link):
print("is_delete_issue link: file: %s, line: %s" % (file, line))
continue
f.write(line)
if __name__ == "__main__":
remove_unreachable_links("srs/trunk/src/")
```
I got an error when cross-compiling the aarch64 platform, the log is as
follows:
`./libavutil/libm.h:54:32: error: static declaration of 'cbrt' follows
non-static declaration`
I see that there are such compilation errors in the
trunk/auto/depends.sh file that have been resolved for the ARM and MIPSE
platforms, and it is recommended to add the ARCH64 platform
## Reload Error Ignore
During a Reload, several stages will be passed through:
1. Parsing new configurations: Parse.
2. Transforming configurations: Transform.
3. Applying configurations: Apply.
Previously, any error at any stage would result in a direct exit, making
the system completely dependent on configuration checks:
```bash
./objs/srs -c conf/srs.conf -t
echo $?
#0
```
Optimized to: If an error occurs before applying the configuration, it
can be ignored. If an error occurs during the application of the
configuration, some of the configuration may have already taken effect,
leading to unpredictable behavior, so SRS will exit directly.
## Reload Fetch API
Added a new HTTP API to query the result of the reload.
```nginx
http_api {
enabled on;
raw_api {
enabled on;
allow_reload on;
}
}
```
```bash
curl http://localhost:1985/api/v1/raw?rpc=reload-fetch
```
```json
{
"code": 0,
"data": {
"err": 0,
"msg": "Success",
"state": 0,
"rid": "0s6y0n9"
}
}
{
"code": 0,
"data": {
"err": 1023,
"msg": "code=1023(ConfigInvalid) : parse file : parse buffer containers/conf/srs.release-local.conf : root parse : parse dir : parse include buffer containers/data/config/srs.vhost.conf : read token, line=0, state=0 : line 3: unexpected end of file, expecting ; or \"}\"",
"state": 1,
"rid": "0g4z471"
}
}
```
This way, you can know if the last reload of the system was successful.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
SrsContextId object creation can be improved on `srs_protocol_log.cpp`,
No need to create one, then assign it back. It seems a common mistake
for Cpp programmers.
---------
Co-authored-by: john <hondaxiao@tencent.com>
on_hls and hls_dispose are two coroutines, with potential race
conditions. That is, during on_hls, if the API Server being accessed is
slower, it will switch to the hls_dispose coroutine to start cleaning
up. However, when the API Server is processing the slice, a situation
may occur where the slice does not exist, resulting in the following
log:
```
[2023-08-22 12:03:20.309][WARN][40][x5l48q7b][11] ignore task failed code=4005(HttpStatus)(Invalid HTTP status code) : callback on_hls http://localhost:2024/terraform/v1/hooks/srs/hls : http: post http://localhost:2024/terraform/v1/hooks/srs/hls with {"server_id":"vid-5d7dxn8","service_id":"cu153o7g","action":"on_hls","client_id":"x5l48q7b","ip":"172.17.0.1","vhost":"__defaultVhost__","app":"live","tcUrl":"srt://172.17.0.2/live","stream":"stream-44572-2739617660809856576","param":"secret=1ed8e0ffbc53439c8fc8da30ab8c19f0","duration":4.57,"cwd":"/usr/local/srs-stack/platform","file":"./objs/nginx/html/live/stream-44572-2739617660809856576-1.ts","url":"live/stream-44572-2739617660809856576-1.ts","m3u8":"./objs/nginx/html/live/stream-44572-2739617660809856576.m3u8","m3u8_url":"live/stream-44572-2739617660809856576.m3u8","seq_no":1,"stream_url":"/live/stream-44572-2739617660809856576","stream_id":"vid-0n9zoz3"}, status=500, res=invalid ts file ./objs/nginx/html/live/stream-44572-2739617660809856576-1.ts: stat ./objs/nginx/html/live/stream-44572-2739617660809856576-1.ts: no such file or directory
thread [40][x5l48q7b]: call() [./src/app/srs_app_hls.cpp:122][errno=11]
thread [40][x5l48q7b]: on_hls() [./src/app/srs_app_http_hooks.cpp:401][errno=11]
thread [40][x5l48q7b]: do_post() [./src/app/srs_app_http_hooks.cpp:638][errno=11]
[error] 2023/08/22 12:03:20.076984 [52][1001] Serve /terraform/v1/hooks/srs/hls failed, err is stat ./objs/nginx/html/live/stream-44572-2739617660809856576-1.ts: no such file or directory
invalid ts file ./objs/nginx/html/live/stream-44572-2739617660809856576-1.ts
main.handleOnHls.func1.1
/g/platform/srs-hooks.go:684
main.handleOnHls.func1
/g/platform/srs-hooks.go:720
net/http.HandlerFunc.ServeHTTP
/usr/local/go/src/net/http/server.go:2084
net/http.(*ServeMux).ServeHTTP
/usr/local/go/src/net/http/server.go:2462
net/http.serverHandler.ServeHTTP
/usr/local/go/src/net/http/server.go:2916
net/http.(*conn).serve
/usr/local/go/src/net/http/server.go:1966
runtime.goexit
/usr/local/go/src/runtime/asm_amd64.s:1571
```
Similarly, when stopping the stream, on_hls will also be called to
handle the last slice. If the API Server is slower at this time, it will
enter hls_dispose and call unpublish repeatedly. Since the previous
unpublish is still blocked in on_hls, the following interference log
will appear:
```
[2023-08-22 12:03:18.748][INFO][40][6498088c] hls cycle to dispose hls /live/stream-44572-2739617660809856576, timeout=10000000ms
[2023-08-22 12:03:18.752][WARN][40][6498088c][115] flush audio ignored, for segment is not open.
[2023-08-22 12:03:18.752][WARN][40][6498088c][115] ignore the segment close, for segment is not open.
```
Although this log will not cause problems, it can interfere with
judgment.
The solution is to add an 'unpublishing' status. If it is in the
'unpublishing' status, then do not clean up the slices.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
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>
When reloading HLS, it directly operates unpublish and publish. At this
time, if HLS is pushed, an exception may occur.
The reason is that these two coroutines operated on the HLS object at
the same time, causing a null pointer.
Solution: Use asynchronous reload. During reload, only set variables and
let the message processing coroutine implement the reload.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.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>
1. Introduce a novel Docker tag in the x.y.z format, akin to the HELM
format, such as ossrs/srs:5.0.155.
2. Incorporate the SRS_PLATFORM flag for containers initiated through
HELM.
---------
`TRANS_BY_GPT3`
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Correct the word errors by changing "ShowCoroutines" to "ShowCoroutines".
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: john <hondaxiao@tencent.com>
It is advised to eliminate any instances of _srs_rtc_manager that occur
multiple times.
---------
Co-authored-by: john <hondaxiao@tencent.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
1. Ignore SDP GROUP LS.
2. Support ice in global session info.
3. Support audio codec "OPUS" or "opus".
---------
Co-authored-by: Johnny <hellojinqiang@gmail.com>
1. The MTU is effective, with the certificate being split into two DTLS records to comply with the limit.
2. The issue occurs when using BIO_get_mem_data, which retrieves all DTLS packets in a single call, even though each is smaller than the MTU.
3. An alternative callback is available for using BIO_new with BIO_s_mem.
4. Improvements to the MTU setting were made, including adding the DTLS_set_link_mtu function and removing the SSL_set_max_send_fragment function.
5. The handshake process was refined, calling SSL_do_handshake only after ICE completion, and using SSL_read to handle handshake messages.
6. The session close code was improved to enable immediate closure upon receiving an SSL CloseNotify or fatal message.
------
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>
* Improve README with AI and add new features
1. Update README file with AI to make it more informative and user-friendly
2. Add a detailed table of contents (TOC) with an introduction for easy navigation
3. Introduce an auto-detecting Automake feature that displays the correct installation command
4. Add support for SRT to HTTP-TS config file
5. Refine the WHIP delete location URL
6. Add support for disabling encryption for WHIP or WHEP
This pull request aims to enhance the quality of the project by introducing innovative features and making the necessary updates. These updates will help users navigate the project more efficiently while also improving the overall project's quality.
---------
Co-authored-by: ChenGH <chengh_math@126.com>
Co-authored-by: john <hondaxiao@tencent.com>
This PR improves the functionality of the HTTP DELETE method used by WHIP to notify the server when the client stops publishing. The URL is parsed from the location header returned by SRS, and the URL is refined with the addition of the action=delete parameter to ensure more accurate identification of the DELETE request.
Furthermore, SRS will disconnect and close the session, enabling the client to publish the stream again quickly and easily. This update eliminates the approximately 30-second waiting period previously required for republishing the stream after an unpublish event.
Overall, this update provides a more effective and efficient method for notifying the server about unpublish events and will enhance the workflow experience for users of the WHIP platform.
-------
Co-authored-by: Haibo Chen <495810242@qq.com>
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 WebRTC, SRS expect the h.264 codec is:
```
a=rtpmap:106 H264/90000
a=fmtp:106 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
```
But sometimes, the device does not support the profile, for example only bellow:
```
a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e033
a=fmtp:122 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=420033
a=fmtp:121 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640033
a=fmtp:120 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0033
```
So we should warning user about the profile missmatch, because it might not work.
----------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: LiPeng <lipeng19811218@gmail.com>
If your OS is not CentOS, Ubuntu, macOS, cygwin64, run of configure will fail with:
```
Your OS Linux is not supported.
```
For other linux systems, we should support an option:
```
./configure --generic-linux=on
```
Please note that you might still fail for other issues while configuring or building.
-------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: ChenGH <chengh_math@126.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. Enable blackbox test for each PR and push.
2. Refine Makefile and README for srs-bench.
3. Add blackbox using FFmpeg and ffprobe.
4. Add blackbox basic test for RTMP stream.
5. Add blackbox basic test for HTTP-FLV stream.
6. Fix utest rand seed issue.
PICK 2141d220b4
* Asan: Disable asan for CentOS and use statically link if possible. v5.0.127 (#3347)
1. Disable asan for all CentOS by default, however user could enable it.
2. Link asan statically if possible.
* Update version to v5.0.127
Co-authored-by: winlin <winlin@vip.126.com>
PICK dd0f398296
1. Build regression test tool for gb28181.
2. Run regression test for gb28181.
3. Format go code and eliminate logs.
4. Change base docker to ubuntu20.
PICK 7750bdae10
1. Docker use srs.conf and env variables.
2. Show help if run SRS without any options.
3. Do not guess config file, use whatever from user.
PICK 07a9a005d5
