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winlinvip edited this page Nov 10, 2014 · 30 revisions

Delivery HLS

SRS supports RTMP and HLS, the most popular delivery protocol for internet.

RTMP is Adobe RTMP(Realtime Message Protocol), for low latency live streaming, and the standard protocol for internet encoder, and the best protocol for flash on PC.

HLS is Apple HLS(Http Live Streaming), for both live and vod streaming over HTTP, and the standard protocol on Apple platform.

Server deliverying HLS and RTMP can support all screen. For RTMP, see: Delivery RTMP。

For information about compare RTMP and HLS, read RTMP PK HLS.

For information about how to deploy SRS to support HLS, read Usage: HLS.

Use Scenario

The main use scenario of HLS:

  • Cross Platform: Live streaming on PC is RTMP, and some as library can play HLS on flash right now. Android 3.0 can play HLS, and Apple always support HLS.
  • Industrial Strength on Apple: The most stable live streaming protocol for OSX/IOS is HLS, similar the RTMP for flash.
  • Friendly for CDN: The HLS over HTTP is friendly for CDN to delivery HLS.
  • Simple: The HLS is open protocol and there are lots of tools for ts.

In a word, HLS is the best delivery protocol when user donot care about the latency, for both PC and mobile(Android and IOS).

Delivering Streams

The table bellow describes the different protocols for PC and mobile platform.

Delivery Platform Protocol Inventor Description
RTMP Windows Flash RTMP Adobe The RTMP is used to delivery low latency stream over internet, especially for the web explorer application for flash on PC. The RTMP allows you to publish stream to server very stable and for a long time.
HLS Apple/
Android		 HTTP Apple/
Google		 HLS lantency >= 10s. Android3+ supports HLS. All Apple platforms support HLS.
HDS - HTTP Adobe HDS is a protocol similar to HLS, develped by Adobe. HDS is complex and no benifits, so SRS never support it.
DASH - HTTP - Dash(Dynamic Adaptive Streaming over HTTP), developed by some company similar to HLS. It's a new protocol, and SRS maybe support it when it already used by many users.

HLS

HLS is a http m3u8 url, which can be play in Apple Safari directly. For example:

http://demo.srs.com/live/livestream.m3u8

The m3u8 url must embed in HTML5 for Android. For example:

<!-- livestream.html -->
<video width="640" height="360"
        autoplay controls autobuffer 
        src="http://demo.srs.com/live/livestream.m3u8"
        type="application/vnd.apple.mpegurl">
</video>

The m3u8 of HLS is a play list actually. For example:

#EXTM3U
#EXT-X-VERSION:3
#EXT-X-MEDIA-SEQUENCE:64
#EXT-X-TARGETDURATION:12
#EXTINF:11.550
livestream-64.ts
#EXTINF:5.250
livestream-65.ts
#EXTINF:7.700
livestream-66.ts
#EXTINF:6.850
livestream-67.ts

The important data item can be configed in SRS config file:

  • EXT-X-TARGETDURATION: It's calc by SRS automatically. The EXT-X-TARGETDURATION tag specifies the maximum media segment duration. The EXTINF duration of each media segment in the Playlist file, when rounded to the nearest integer, MUST be less than or equal to the target duration. This tag MUST appear once in a Media Playlist file. It applies to the entire Playlist file.
  • EXTINF: It's calc by SRS automatically, and the max value is configed in SRS hls_fragment. The EXTINF tag specifies the duration of a media segment. It applies only to the media segment that follows it, and MUST be followed by a media segment URI. Each media segment MUST be preceded by an EXTINF tag.
  • Number of ts in m3u8: The hls_window in seconds specifies how many ts files in m3u8, and SRS will delete the old ts files.
  • Name of ts filename: For example, livestream-67.ts, SRS will increase the number, the next ts for instance is livestream-68.ts.

For example, the hls_fragment is 10s, and the hls_window is 60s, then there about 60/10=6 ts files i m3u8, and the old ts files are automatically deleted by SRS.

HLS Workflow

HLS的主要流程是：

  1. FFMPEG或FMLE或编码器，推送RTMP流到SRS，编码为H264/AAC（其他编码需要SRS转码）
  2. SRS将RTMP切片成TS，并生成M3U8。若流非H264和AAC，则停止输出HLS（可使用SRS转码到SRS其他vhost或流，然后再切HLS）。
  3. 访问m3u8，srs内置的http服务器（或者通用http服务器）提供HTTP服务。
    注意：SRS只需要在Vhost上配置HLS，会自动根据流的app创建目录，但是配置的hls_path必须自己创建

配置方法

conf/full.conf中的with-hls.vhost.com是HLS配置的实例，可以拷贝到默认的Vhost，例如：

vhost __defaultVhost__ {
    hls {
        # whether the hls is enabled.
        # if off, donot write hls(ts and m3u8) when publish.
        # default: off
        enabled         on;
        # the hls output path.
        # the app dir is auto created under the hls_path.
        # for example, for rtmp stream:
        #   rtmp://127.0.0.1/live/livestream
        #   http://127.0.0.1/live/livestream.m3u8
        # where hls_path is /hls, srs will create the following files:
        #   /hls/live       the app dir for all streams.
        #   /hls/live/livestream.m3u8   the HLS m3u8 file.
        #   /hls/live/livestream-1.ts   the HLS media/ts file.
        # in a word, the hls_path is for vhost.
        # default: ./objs/nginx/html
        hls_path        /data/nginx/html;
        # the hls fragment in seconds, the duration of a piece of ts.
        # default: 10
        hls_fragment    10;
        # the hls window in seconds, the number of ts in m3u8.
        # default: 60
        hls_window      60;
    }
}

其中hls配置就是HLS的配置，主要配置项如下：

  • enabled：是否开启HLS，on/off，默认off。
  • hls_path：HLS的m3u8和ts文件保存的路径。SRS会自动加上app和stream名称。譬如：
对于RTMP流：rtmp://localhost/live/livestream
HLS配置路径：hls_path        /data/nginx/html;
那么会生成以下文件：
/data/nginx/html/live/livestream.m3u8
/data/nginx/html/live/livestream-0.ts
/data/nginx/html/live/livestream-1.ts
/data/nginx/html/live/livestream-2.ts
最后的HLS地址为：http://localhost/live/livestream.m3u8
  • hls_fragment：秒，指定ts切片的最小长度。实际上ts文件的长度由以下公式决定：
ts文件时长 = max(hls_fragment, gop_size)
hls_fragment：配置文件中的长度。譬如：5秒。
gop_size：编码器配置的gop的长度，譬如ffmpeg指定fps为20帧/秒，gop为200帧，则gop_size=gop/fps=10秒。
那么，最终ts的时长为max(5, 10) = 10秒。这也是为什么有些流配置了hls_fragment，但是ts时长仍然比这个大的原因。
  • hls_window：秒，指定HLS窗口大小，即m3u8中ts文件的时长之和，超过总时长后，丢弃第一个m3u8中的第一个切片，直到ts的总时长在这个配置项范围之内。即SRS保证下面的公式：
hls_window >= sum(m3u8中每个ts的时长)

部署分发HLS的实例，参考：Usage: HLS

HLSAudioOnly

SRS支持分发HLS纯音频流，当RTMP流没有视频，且音频为aac（可以使用转码转为aac，参考Usage: Transcode2HLS），SRS只切片音频。

若RTMP流中已经有视频和音频，需要支持纯音频HLS流，可以用转码将视频去掉，参考：转码: 禁用流。然后分发音频流。

分发纯音频流不需要特殊配置，和HLS分发一样，参考：Usage: HLS

HLS和Forward

Forward的流和普通流不做区分，若forward的流所在的VHOST配置了HLS，一样会应用HLS配置进行切片。

因此，可以对原始流进行Transcode之后，保证流符合h.264/aac的规范，然后forward到多个配置了HLS的VHOST进行切片。支持多个源站的热备。

HLS和Transcode

HLS要求RTMP流的编码为h.264+aac，否则会自动禁用HLS，会出现RTMP流能看HLS流不能看（或者看到的HLS是之前的流）。

Transcode将RTMP流转码后，可以让SRS接入任何编码的RTMP流，然后转换成HLS要求的h.264/aac编码方式。

配置Transcode时，若需要控制ts长度，需要配置ffmpeg编码的gop，譬如：

vhost hls.transcode.vhost.com {
    transcode {
        enabled     on;
        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
        engine hls {
            enabled         on;
            vfilter {
            }
            vcodec          libx264;
            vbitrate        500;
            vfps            20;
            vwidth          768;
            vheight         320;
            vthreads        2;
            vprofile        baseline;
            vpreset         superfast;
            vparams {
                g           100;
            }
            acodec          libaacplus;
            abitrate        45;
            asample_rate    44100;
            achannels       2;
            aparams {
            }
            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
        }
    }
}

该FFMPEG转码参数，指定gop时长为100/20=5秒，fps帧率（vfps=20），gop帧数（g=100）。

HLS自适应码流

SRS目前不支持HLS自适应码流，需要调研这个功能。

HLS实例

live.m3u8

#EXTM3U
#EXT-X-VERSION:3
#EXT-X-ALLOW-CACHE:NO
#EXT-X-TARGETDURATION:13
#EXT-X-MEDIA-SEQUENCE:430
#EXTINF:11.800
news-430.ts
#EXTINF:10.120
news-431.ts
#EXTINF:11.160
news-432.ts

event.m3u8

#EXTM3U
#EXT-X-VERSION:3
#EXT-X-ALLOW-CACHE:NO
#EXT-X-TARGETDURATION:13
#EXT-X-MEDIA-SEQUENCE:430
#EXT-X-PLAYLIST-TYPE:EVENT
#EXTINF:11.800
news-430.ts
#EXTINF:10.120
news-431.ts
#EXTINF:11.160
news-432.ts

vod.m3u8

#EXTM3U
#EXT-X-VERSION:3
#EXT-X-ALLOW-CACHE:NO
#EXT-X-TARGETDURATION:13
#EXT-X-MEDIA-SEQUENCE:430
#EXT-X-PLAYLIST-TYPE:VOD
#EXTINF:11.800
news-430.ts
#EXTINF:10.120
news-431.ts
#EXTINF:11.160
news-432.ts
#EXT-X-ENDLIST

loop.m3u8

#EXTM3U
#EXT-X-VERSION:3
#EXT-X-ALLOW-CACHE:NO
#EXT-X-TARGETDURATION:13
#EXT-X-MEDIA-SEQUENCE:430
#EXT-X-PLAYLIST-TYPE:VOD
#EXTINF:11.800
news-430.ts
#EXTINF:10.120
news-431.ts
#EXT-X-DISCONTINUITY
#EXTINF:11.952
news-430.ts
#EXTINF:12.640
news-431.ts
#EXTINF:11.160
news-432.ts
#EXT-X-DISCONTINUITY
#EXTINF:11.751
news-430.ts
#EXTINF:2.040
news-431.ts
#EXT-X-ENDLIST

SRS如何支持HLS

SRS的HLS主要参考了nginx-rtmp的HLS实现方式，SRS没有做什么事情，都是nginx-rtmp实现的。而分发m3u8和ts文件，也是使用nginx分发的。

SRS只是读了遍ts的标准文档，把相关部分加了注释而已。譬如下面这段：

// @see: ngx_rtmp_mpegts.c
// TODO: support full mpegts feature in future.
class SrsMpegtsWriter
{
public:
    static int write_frame(int fd, SrsMpegtsFrame* frame, SrsCodecBuffer* buffer)
    {
        int ret = ERROR_SUCCESS;
        
        if (!buffer->bytes || buffer->size <= 0) {
            return ret;
        }
        
        char* last = buffer->bytes + buffer->size;
        char* pos = buffer->bytes;
        
        bool first = true;
        while (pos < last) {
            static char packet[188];
            char* p = packet;
            
            frame->cc++;
            
            // sync_byte; //8bits
            *p++ = 0x47;
            // pid; //13bits
            *p++ = (frame->pid >> 8) & 0x1f;
            // payload_unit_start_indicator; //1bit
            if (first) {
                p[-1] |= 0x40;
            }
            *p++ = frame->pid;
            
            // transport_scrambling_control; //2bits
            // adaption_field_control; //2bits, 0x01: PayloadOnly
            // continuity_counter; //4bits
            *p++ = 0x10 | (frame->cc & 0x0f);
            
            if (first) {
                first = false;
                if (frame->key) {
                    p[-1] |= 0x20; // Both Adaption and Payload
                    *p++ = 7;    // size
                    *p++ = 0x50; // random access + PCR
                    p = write_pcr(p, frame->dts - SRS_HLS_DELAY);
                }
                
                // PES header
                // packet_start_code_prefix; //24bits, '00 00 01'
                *p++ = 0x00;
                *p++ = 0x00;
                *p++ = 0x01;
                //8bits
                *p++ = frame->sid;
                
                // pts(33bits) need 5bytes.
                u_int8_t header_size = 5;
                u_int8_t flags = 0x80; // pts
                
                // dts(33bits) need 5bytes also
                if (frame->dts != frame->pts) {
                    header_size += 5;
                    flags |= 0x40; // dts
                }
                
                // 3bytes: flag fields from PES_packet_length to PES_header_data_length
                int pes_size = (last - pos) + header_size + 3;
                if (pes_size > 0xffff) {
                    /**
                    * when actual packet length > 0xffff(65535),
                    * which exceed the max u_int16_t packet length,
                    * use 0 packet length, the next unit start indicates the end of packet.
                    */
                    pes_size = 0;
                }
                
                // PES_packet_length; //16bits
                *p++ = (pes_size >> 8);
                *p++ = pes_size;
                
                // PES_scrambling_control; //2bits, '10'
                // PES_priority; //1bit
                // data_alignment_indicator; //1bit
                // copyright; //1bit
                // original_or_copy; //1bit	
                *p++ = 0x80; /* H222 */
                
                // PTS_DTS_flags; //2bits
                // ESCR_flag; //1bit
                // ES_rate_flag; //1bit
                // DSM_trick_mode_flag; //1bit
                // additional_copy_info_flag; //1bit
                // PES_CRC_flag; //1bit
                // PES_extension_flag; //1bit
                *p++ = flags;
                
                // PES_header_data_length; //8bits
                *p++ = header_size;

                // pts; // 33bits
                p = write_pts(p, flags >> 6, frame->pts + SRS_HLS_DELAY);
                
                // dts; // 33bits
                if (frame->dts != frame->pts) {
                    p = write_pts(p, 1, frame->dts + SRS_HLS_DELAY);
                }
            }
            
            int body_size = sizeof(packet) - (p - packet);
            int in_size = last - pos;
            
            if (body_size <= in_size) {
                memcpy(p, pos, body_size);
                pos += body_size;
            } else {
                p = fill_stuff(p, packet, body_size, in_size);
                memcpy(p, pos, in_size);
                pos = last;
            }
            
            // write ts packet
            if (::write(fd, packet, sizeof(packet)) != sizeof(packet)) {
                ret = ERROR_HLS_WRITE_FAILED;
                srs_error("write ts file failed. ret=%d", ret);
                return ret;
            }
        }
        
        return ret;
    }
};

Winlin 2014.11

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