HTTP-TS: Support guess_has_av for audio only stream. v6.0.141 (#4063)

## 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>

trunk/doc/CHANGELOG.md

@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v6-changes"></a>
 
 ## SRS 6.0 Changelog
+* v6.0, 2024-07-24, Merge [#4063](https://github.com/ossrs/srs/pull/4063): let http-remux ts stream support guess_has_av feature;. v6.0.141 (#4063)
 * v6.0, 2024-07-24, Merge [#4116](https://github.com/ossrs/srs/pull/4116): Dockerfile: Consistently use proper ENV syntax using "=". v6.0.140 (#4116)
 * v6.0, 2024-07-24, Merge [#4126](https://github.com/ossrs/srs/pull/4126): Edge: Improve stability for state and fd closing. v6.0.139 (#4126)
 * v6.0, 2024-07-13, Merge [#4111](https://github.com/ossrs/srs/pull/4111): DASH: Fix time unit error for disposing. v6.0.138 (#4111)

trunk/src/app/srs_app_http_stream.cpp

@@ -257,11 +257,17 @@ void SrsTsStreamEncoder::set_has_audio(bool v)
 {
     enc->set_has_audio(v);
 }
+
 void SrsTsStreamEncoder::set_has_video(bool v)
 {
     enc->set_has_video(v);
 }
 
+void SrsTsStreamEncoder::set_guess_has_av(bool v)
+{
+    enc->set_guess_has_av(v);
+}
+
 SrsFlvStreamEncoder::SrsFlvStreamEncoder()
 {
     header_written = false;
@@ -677,6 +683,7 @@ srs_error_t SrsLiveStream::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMess
         enc_raw = new SrsTsStreamEncoder();
         ((SrsTsStreamEncoder*)enc_raw)->set_has_audio(has_audio);
         ((SrsTsStreamEncoder*)enc_raw)->set_has_video(has_video);
+        ((SrsTsStreamEncoder*)enc_raw)->set_guess_has_av(guess_has_av);
     } else {
         return srs_error_new(ERROR_HTTP_LIVE_STREAM_EXT, "invalid pattern=%s", entry->pattern.c_str());
     }

trunk/src/app/srs_app_http_stream.hpp

@@ -114,6 +114,7 @@ public:
 public:
     void set_has_audio(bool v);
     void set_has_video(bool v);
+    void set_guess_has_av(bool v);
 };
 
 // Transmux RTMP with AAC stream to HTTP AAC Streaming.

trunk/src/core/srs_core_version6.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       6
 #define VERSION_MINOR       0
-#define VERSION_REVISION    140
+#define VERSION_REVISION    141
 
 #endif

trunk/src/kernel/srs_kernel_ts.cpp

@@ -3176,6 +3176,7 @@ SrsTsTransmuxer::SrsTsTransmuxer()
     context = new SrsTsContext();
     tscw = NULL;
     has_audio_ = has_video_ = true;
+    guess_has_av_ = true;
 }
 
 SrsTsTransmuxer::~SrsTsTransmuxer()
@@ -3189,11 +3190,28 @@ SrsTsTransmuxer::~SrsTsTransmuxer()
 void SrsTsTransmuxer::set_has_audio(bool v)
 {
     has_audio_ = v;
+
+    if (tscw != NULL && !v) {
+        tscw->set_acodec(SrsAudioCodecIdForbidden);
+    }
 }
 
 void SrsTsTransmuxer::set_has_video(bool v)
 {
     has_video_ = v;
+
+    if (tscw != NULL && !v) {
+        tscw->set_vcodec(SrsVideoCodecIdForbidden);
+    }
+}
+
+void SrsTsTransmuxer::set_guess_has_av(bool v)
+{
+    guess_has_av_ = v;
+    if (tscw != NULL && v) {
+        tscw->set_acodec(SrsAudioCodecIdForbidden);
+        tscw->set_vcodec(SrsVideoCodecIdForbidden);
+    }
 }
 
 srs_error_t SrsTsTransmuxer::initialize(ISrsStreamWriter* fw)
@@ -3211,6 +3229,11 @@ srs_error_t SrsTsTransmuxer::initialize(ISrsStreamWriter* fw)
     SrsAudioCodecId acodec = has_audio_ ? SrsAudioCodecIdAAC : SrsAudioCodecIdForbidden;
     SrsVideoCodecId vcodec = has_video_ ? SrsVideoCodecIdAVC : SrsVideoCodecIdForbidden;
 
+    if (guess_has_av_) {
+        acodec = SrsAudioCodecIdForbidden;
+        vcodec = SrsVideoCodecIdForbidden;
+    }
+
     srs_freep(tscw);
     tscw = new SrsTsContextWriter(fw, context, acodec, vcodec);
 

trunk/src/kernel/srs_kernel_ts.hpp

@@ -1333,6 +1333,7 @@ private:
     ISrsStreamWriter* writer;
     bool has_audio_;
     bool has_video_;
+    bool guess_has_av_;
 private:
     SrsFormat* format;
     SrsTsMessageCache* tsmc;
@@ -1344,6 +1345,7 @@ public:
 public:
     void set_has_audio(bool v);
     void set_has_video(bool v);
+    void set_guess_has_av(bool v);
 public:
     // Initialize the underlayer file stream.
     // @param fw the writer to use for ts encoder, user must free it.