Merge branch 'develop' into pr-develop

.vscode/launch.json

@@ -2,7 +2,7 @@
     "version": "0.2.0",
     "configurations": [
         {
-          "name": "Launch SRS",
+          "name": "Launch SRS with conf/console.conf",
           "type": "cppdbg",
           "request": "launch",
           "program": "${workspaceFolder}/trunk/cmake/build/srs",
@@ -29,6 +29,34 @@
             "engineLogging": true
           }
         },
+        {
+          "name": "Launch SRS with conf/rtc.conf",
+          "type": "cppdbg",
+          "request": "launch",
+          "program": "${workspaceFolder}/trunk/cmake/build/srs",
+          "args": ["-c", "conf/rtc.conf"],
+          "stopAtEntry": false,
+          "cwd": "${workspaceFolder}/trunk",
+          "environment": [],
+          "externalConsole": false,
+          "linux": {
+            "MIMode": "gdb"
+          },
+          "osx": {
+            "MIMode": "lldb"
+          },
+          "setupCommands": [
+            {
+              "description": "Enable pretty-printing for gdb",
+              "text": "-enable-pretty-printing",
+              "ignoreFailures": true
+            }
+          ],
+          "preLaunchTask": "build",
+          "logging": {
+            "engineLogging": true
+          }
+        },
         {
           "name": "Launch srs-proxy",
           "type": "go",

trunk/doc/CHANGELOG.md

@@ -7,16 +7,18 @@ The changelog for SRS.
 <a name="v7-changes"></a>
 
 ## SRS 7.0 Changelog
+* v7.0, 2025-10-30, WebRTC: Use realtime for TWCC timestamp accuracy. v7.0.114
+* v7.0, 2025-10-30, AI: HLS: Fix crash when segment is not open by adding NULL checks. v7.0.113 (#3431)
 * v7.0, 2025-10-29, AI: AAC: Fix mono audio reported as stereo in HTTP API. v7.0.112 (#3556)
-* v7.0, 2025-10-27, HLS/DASH: Fix dispose to skip unpublish when not enabled, and add forbidden directory protection to SrsPath::unlink. v7.0.111
+* v7.0, 2025-10-27, HLS/DASH: Skip unpublish if disabled; add protection in SrsPath::unlink. v7.0.111
-* v7.0, 2025-10-27, AI: HTTP-FLV: Enforce minimum 10ms sleep to prevent CPU busy-wait when mw_latency=0. v7.0.110 (#3963)
+* v7.0, 2025-10-27, AI: HTTP-FLV: Add 10 ms sleep to prevent busy-wait when mw_latency=0. v7.0.110 (#3963)
-* v7.0, 2025-10-26, AI: Edge: Fix stream names with dots being incorrectly truncated in source URL generation. v7.0.109 (#4011)
+* v7.0, 2025-10-26, AI: Edge: Fix truncation of stream names containing dots. v7.0.109 (#4011)
-* v7.0, 2025-10-26, AI: HTTPS: Handle SSL_ERROR_ZERO_RETURN as graceful connection closure. v7.0.108 (#4036)
+* v7.0, 2025-10-26, AI: HTTPS: Treat SSL_ERROR_ZERO_RETURN as graceful close. v7.0.108 (#4036)
-* v7.0, 2025-10-26, AI: API: Add clients field to on_play/on_stop webhooks and total field to HTTP API. v7.0.107 (#4147)
+* v7.0, 2025-10-26, AI: API: Add clients to webhooks and total to stats API. v7.0.107 (#4147)
-* v7.0, 2025-10-26, AI: WebRTC: Fix camera/microphone not released after closing publisher. v7.0.106 (#4261)
+* v7.0, 2025-10-26, AI: WebRTC: Fix camera/mic not released after closing. v7.0.106 (#4261)
-* v7.0, 2025-10-26, AI: Build: Improve dependency checking to report all missing dependencies at once. v7.0.105 (#4293)
+* v7.0, 2025-10-26, AI: Build: Report all missing dependencies at once. v7.0.105 (#4293)
-* v7.0, 2025-10-26, AI: HLS: Support hls_master_m3u8_path_relative for reverse proxy compatibility. v7.0.104 (#4338)
+* v7.0, 2025-10-26, AI: HLS: Support hls_master_m3u8_path_relative for reverse proxy. v7.0.104 (#4338)
-* v7.0, 2025-10-25, AI: API: Remove minimum limit of 10 for count parameter in /api/v1/streams and /api/v1/clients. v7.0.103 (#4358)
+* v7.0, 2025-10-25, AI: API: Remove minimum count=10 limit in stream/client queries. v7.0.103 (#4358)
 * v7.0, 2025-10-22, AI: Only support AAC/MP3/Opus audio codec. v7.0.102 (#4516)
 * v7.0, 2025-10-22, AI: Fix AAC audio sample rate reporting in API. v7.0.101 (#4518)
 * v7.0, 2025-10-20, Merge [#4537](https://github.com/ossrs/srs/pull/4537): Forward: Reject RTMPS destinations with clear error message. v7.0.100 (#4537)

trunk/src/app/srs_app_hls.cpp

@@ -847,6 +847,11 @@ std::string SrsHlsFmp4Muxer::generate_m4s_filename()
     return m4s_file;
 }
 
+bool SrsHlsFmp4Muxer::is_segment_open()
+{
+    return current_ != NULL;
+}
+
 srs_error_t SrsHlsFmp4Muxer::on_sequence_header()
 {
     return srs_success;
@@ -854,7 +859,10 @@ srs_error_t SrsHlsFmp4Muxer::on_sequence_header()
 
 bool SrsHlsFmp4Muxer::is_segment_overflow()
 {
-    srs_assert(current_);
+    // If segment is not open, it cannot overflow.
+    if (!is_segment_open()) {
+        return false;
+    }
 
     // to prevent very small segment.
     if (current_->duration() < 2 * SRS_HLS_SEGMENT_MIN_DURATION) {
@@ -876,7 +884,10 @@ bool SrsHlsFmp4Muxer::wait_keyframe()
 
 bool SrsHlsFmp4Muxer::is_segment_absolutely_overflow()
 {
-    srs_assert(current_);
+    // If segment is not open, it cannot overflow.
+    if (!is_segment_open()) {
+        return false;
+    }
 
     // to prevent very small segment.
     if (current_->duration() < 2 * SRS_HLS_SEGMENT_MIN_DURATION) {
@@ -890,6 +901,10 @@ bool SrsHlsFmp4Muxer::is_segment_absolutely_overflow()
 
 void SrsHlsFmp4Muxer::update_duration(uint64_t dts)
 {
+    // If segment is not open, ignore the update event.
+    if (!is_segment_open()) {
+        return;
+    }
     current_->append(dts / 90);
 }
 
@@ -1652,7 +1667,11 @@ srs_error_t SrsHlsMuxer::on_sequence_header()
 {
     srs_error_t err = srs_success;
 
-    srs_assert(current_);
+    // If segment is not open, ignore the sequence header event.
+    if (!is_segment_open()) {
+        srs_warn("sequence header ignored, for segment is not open.");
+        return err;
+    }
 
     // set the current segment to sequence header,
     // when close the segement, it will write a discontinuity to m3u8 file.
@@ -1661,9 +1680,17 @@ srs_error_t SrsHlsMuxer::on_sequence_header()
     return err;
 }
 
+bool SrsHlsMuxer::is_segment_open()
+{
+    return current_ != NULL;
+}
+
 bool SrsHlsMuxer::is_segment_overflow()
 {
-    srs_assert(current_);
+    // If segment is not open, it cannot overflow.
+    if (!is_segment_open()) {
+        return false;
+    }
 
     // to prevent very small segment.
     if (current_->duration() < 2 * SRS_HLS_SEGMENT_MIN_DURATION) {
@@ -1685,7 +1712,10 @@ bool SrsHlsMuxer::wait_keyframe()
 
 bool SrsHlsMuxer::is_segment_absolutely_overflow()
 {
-    srs_assert(current_);
+    // If segment is not open, it cannot overflow.
+    if (!is_segment_open()) {
+        return false;
+    }
 
     // to prevent very small segment.
     if (current_->duration() < 2 * SRS_HLS_SEGMENT_MIN_DURATION) {
@@ -1760,6 +1790,10 @@ srs_error_t SrsHlsMuxer::flush_video(SrsTsMessageCache *cache)
 
 void SrsHlsMuxer::update_duration(uint64_t dts)
 {
+    // If segment is not open, ignore the update event.
+    if (!is_segment_open()) {
+        return;
+    }
     current_->append(dts / 90);
 }
 

trunk/src/app/srs_app_hls.hpp

@@ -358,6 +358,7 @@ public:
 // clang-format off
 SRS_DECLARE_PRIVATE: // clang-format on
     virtual std::string generate_ts_filename();
+    virtual bool is_segment_open();
 
 public:
     virtual srs_error_t on_sequence_header();
@@ -541,6 +542,7 @@ public:
 // clang-format off
 SRS_DECLARE_PRIVATE: // clang-format on
     virtual std::string generate_m4s_filename();
+    virtual bool is_segment_open();
 
 public:
     virtual srs_error_t on_sequence_header();

trunk/src/app/srs_app_rtc_conn.cpp

@@ -1311,8 +1311,11 @@ srs_error_t SrsRtcPublishStream::initialize(ISrsRequest *r, SrsRtcSourceDescript
 
     int twcc_id = -1;
     uint32_t media_ssrc = 0;
-    // because audio_track_desc have not twcc id, for example, h5demo
+    // TWCC is transport-wide, so audio and video share the same extension ID.
-    // fetch twcc_id from video track description,
+    // We fetch the TWCC ID from video track, which is sufficient because:
+    //   1. Standard WebRTC clients use the same TWCC ID for both audio and video.
+    //   2. The ID is used to parse TWCC extension from all RTP packets (audio+video).
+    //   3. TWCC feedback will include both audio and video packets.
     for (int i = 0; i < (int)stream_desc->video_track_descs_.size(); ++i) {
         SrsRtcTrackDescription *desc = stream_desc->video_track_descs_.at(i);
         twcc_id = desc->get_rtp_extension_id(kTWCCExt);
@@ -1529,7 +1532,10 @@ srs_error_t SrsRtcPublishStream::on_twcc(uint16_t sn)
 {
     srs_error_t err = srs_success;
 
-    srs_utime_t now = srs_time_now_cached();
+    // To get more accurate timestamp, and avoid deviation caused by coroutine scheduler, 
+    // we use realtime for TWCC.
+    srs_utime_t now = srs_time_now_realtime();
+    
     err = rtcp_twcc_->recv_packet(sn, now);
 
     return err;
@@ -1581,10 +1587,6 @@ srs_error_t SrsRtcPublishStream::on_rtp_plaintext(char *plaintext, int nb_plaint
 {
     srs_error_t err = srs_success;
 
-    if (_srs_blackhole->blackhole_) {
-        _srs_blackhole->sendto(plaintext, nb_plaintext);
-    }
-
     // Allocate packet form cache.
     SrsRtpPacket *pkt = new SrsRtpPacket();
 
@@ -2663,6 +2665,10 @@ srs_error_t SrsRtcConnection::send_rtcp(char *data, int nb_data)
 
     ++_srs_pps_srtcps->sugar_;
 
+    if (_srs_blackhole->blackhole_) {
+        _srs_blackhole->sendto(data, nb_data);
+    }
+
     int nb_buf = nb_data;
     if ((err = networks_->available()->protect_rtcp(data, &nb_buf)) != srs_success) {
         return srs_error_wrap(err, "protect rtcp");
@@ -2818,10 +2824,6 @@ srs_error_t SrsRtcConnection::send_rtcp_fb_pli(uint32_t ssrc, const SrsContextId
                   nn, pli_epp_->nn_count_, stream.pos());
     }
 
-    if (_srs_blackhole->blackhole_) {
-        _srs_blackhole->sendto(stream.data(), stream.pos());
-    }
-
     return send_rtcp(stream.data(), stream.pos());
 }
 

trunk/src/app/srs_app_rtmp_source.cpp

@@ -1008,11 +1008,12 @@ srs_error_t SrsOriginHub::on_audio(SrsMediaPacket *shared_audio)
             sample_rate = srs_audio_sample_rate_from_number(srs_aac_srates[c->aac_sample_rate_]);
         }
 
-        // For AAC, use aac_channels_ from AudioSpecificConfig instead of sound_type_ from FLV tag.
+        // The FLV specification force to be 1(stereo) for AAC codec, see E.4.2 Audio Tags,
-        // The FLV sound_type field is often incorrect for AAC streams (e.g., FFmpeg sets it to stereo
+        // video_file_format_spec_v10_1.pdf, page 77.
-        // even for mono streams). The AAC AudioSpecificConfig channelConfiguration is the authoritative
+        //      If the SoundFormat indicates AAC, the SoundType should be 1 (stereo) and the
-        // source: 1=mono, 2=stereo. Map to SrsAudioChannels: 0=mono, 1=stereo.
+        //      SoundRate should be 3 (44 kHz).
-        // For MP3 and other codecs, use sound_type_ from FLV tag.
+        // FFmpeg also follows this specification, that is why it always send fixed channels
+        // and sample rate to SRS.
         SrsAudioChannels channels = c->sound_type_;
         if (format->acodec_->id_ == SrsAudioCodecIdAAC) {
             channels = (c->aac_channels_ == 1) ? SrsAudioChannelsMono : SrsAudioChannelsStereo;

trunk/src/core/srs_core_version7.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR 7
 #define VERSION_MINOR 0
-#define VERSION_REVISION 112
+#define VERSION_REVISION 114
 
 #endif