fix #409: support pure video hls. 2.0.172.

@@ -342,6 +342,7 @@ Remark:
 ## History
 
 ### SRS 2.0 history
+* v2.0, 2015-05-29, fix [#409](https://github.com/simple-rtmp-server/srs/issues/409) support pure video hls. 2.0.172.
 * v2.0, 2015-05-28, support [srs-dolphin][srs-dolphin], the multiple-process SRS.
 * v2.0, 2015-05-24, fix [#404](https://github.com/simple-rtmp-server/srs/issues/404) register handler then start http thread. 2.0.167.
 * v2.0, 2015-05-23, refine the thread, protocol, kbps code. 2.0.166

@@ -608,7 +608,7 @@ vhost with-hls.srs.com {
         # when codec changed, write the PAT/PMT table, but maybe ok util next ts.
         # so user can set the default codec for mp3.
         # the available audio codec: 
-        #       aac, mp3
+        #       aac, mp3, an
         # default: aac
         hls_acodec      aac;
         # the default video codec of hls.

@@ -420,6 +420,9 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
         } else if (default_acodec_str == "aac") {
             default_acodec = SrsCodecAudioAAC;
             srs_info("hls: use default aac acodec");
+        } else if (default_acodec_str == "an") {
+            default_acodec = SrsCodecAudioDisabled;
+            srs_info("hls: use default an acodec for pure video");
         } else {
             srs_warn("hls: use aac for other codec=%s", default_acodec_str.c_str());
         }

@@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // current release version
 #define VERSION_MAJOR       2
 #define VERSION_MINOR       0
-#define VERSION_REVISION    171
+#define VERSION_REVISION    172
 
 // server info.
 #define RTMP_SIG_SRS_KEY "SRS"

@@ -136,6 +136,9 @@ enum SrsCodecAudio
     // set to the max value to reserved, for array map.
     SrsCodecAudioReserved1                = 16,
     
+    // for user to disable audio, for example, use pure video hls.
+    SrsCodecAudioDisabled                   = 17,
+    
     SrsCodecAudioLinearPCMPlatformEndian             = 0,
     SrsCodecAudioADPCM                                 = 1,
     SrsCodecAudioMP3                                 = 2,

@@ -215,13 +215,13 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #define ERROR_HTTP_DVR_CREATE_REQUEST       3053
 #define ERROR_HTTP_DVR_NO_TAEGET            3054
 #define ERROR_ADTS_ID_NOT_AAC               3055
-// HDS error code
 #define ERROR_HDS_OPEN_F4M_FAILED           3056
 #define ERROR_HDS_WRITE_F4M_FAILED          3057
 #define ERROR_HDS_OPEN_BOOTSTRAP_FAILED     3058
 #define ERROR_HDS_WRITE_BOOTSTRAP_FAILED    3059
 #define ERROR_HDS_OPEN_FRAGMENT_FAILED      3060
 #define ERROR_HDS_WRITE_FRAGMENT_FAILED     3061
+#define ERROR_HLS_NO_STREAM                 3062
 
 ///////////////////////////////////////////////////////
 // HTTP/StreamCaster protocol error.

@@ -302,10 +302,12 @@ int SrsTsContext::encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo
             vs = SrsTsStreamVideoH264; 
             video_pid = TS_VIDEO_AVC_PID;
             break;
+        case SrsCodecVideoDisabled:
+            vs = SrsTsStreamReserved;
+            break;
         case SrsCodecVideoReserved:
         case SrsCodecVideoReserved1:
         case SrsCodecVideoReserved2:
-        case SrsCodecVideoDisabled:
         case SrsCodecVideoSorensonH263:
         case SrsCodecVideoScreenVideo:
         case SrsCodecVideoOn2VP6:
@@ -323,6 +325,9 @@ int SrsTsContext::encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo
             as = SrsTsStreamAudioMp3; 
             audio_pid = TS_AUDIO_MP3_PID;
             break;
+        case SrsCodecAudioDisabled:
+            as = SrsTsStreamReserved;
+            break;
         case SrsCodecAudioReserved1:
         case SrsCodecAudioLinearPCMPlatformEndian:
         case SrsCodecAudioADPCM:
@@ -340,6 +345,12 @@ int SrsTsContext::encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo
             break;
     }
     
+    if (as == SrsTsStreamReserved && vs == SrsTsStreamReserved) {
+        ret = ERROR_HLS_NO_STREAM;
+        srs_error("hls: no video or audio stream, vcodec=%d, acodec=%d. ret=%d", vc, ac, ret);
+        return ret;
+    }
+    
     // when any codec changed, write PAT/PMT table.
     if (vcodec != vc || acodec != ac) {
         vcodec = vc;
@@ -360,6 +371,12 @@ int SrsTsContext::encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo
 int SrsTsContext::encode_pat_pmt(SrsFileWriter* writer, int16_t vpid, SrsTsStream vs, int16_t apid, SrsTsStream as)
 {
     int ret = ERROR_SUCCESS;
+    
+    if (vs != SrsTsStreamVideoH264 && as != SrsTsStreamAudioAAC && as != SrsTsStreamAudioMp3) {
+        ret = ERROR_HLS_NO_STREAM;
+        srs_error("hls: no pmt pcr pid, vs=%d, as=%d. ret=%d", vs, as, ret);
+        return ret;
+    }
 
     int16_t pmt_number = TS_PMT_NUMBER;
     int16_t pmt_pid = TS_PMT_PID;
@@ -754,15 +771,17 @@ SrsTsPacket* SrsTsPacket::create_pmt(SrsTsContext* context, int16_t pmt_number,
     pmt->last_section_number = 0;
     pmt->program_info_length = 0;
     
-    // use audio to carray pcr by default.
-    // for hls, there must be atleast one audio channel.
-    pmt->PCR_PID = apid;
-    pmt->infos.push_back(new SrsTsPayloadPMTESInfo(as, apid));
-    
     // if h.264 specified, use video to carry pcr.
     if (vs == SrsTsStreamVideoH264) {
         pmt->PCR_PID = vpid;
         pmt->infos.push_back(new SrsTsPayloadPMTESInfo(vs, vpid));
+    } else if (as == SrsTsStreamAudioAAC || as == SrsTsStreamAudioMp3) {
+        // use audio to carray pcr by default.
+        // for hls, there must be atleast one audio channel.
+        pmt->PCR_PID = apid;
+        pmt->infos.push_back(new SrsTsPayloadPMTESInfo(as, apid));
+    } else {
+        srs_assert(false);
     }
     
     pmt->CRC_32 = 0; // calc in encode.