Dev

ABR and Transcoding

OvenMediaEngine has a built-in live transcoder. The live transcoder can decode the incoming live source and re-encode it with the set codec or adjust the quality to encode at multiple bitrates.

Supported Video, Audio and Image Codecs

Decoders

TypeCodec

Video

VP8, H.264, H.265

Audio

AAC, Opus

Encoders

TypeCodecCodec of Configuration

Video

VP8

vp8

H.264

h264 (Automatic Codec Selection)

x264

h264_x264

OpenH264

h264_openh264

NVIDIA Hardware

h264_nvenc

Intel Hardware

h264_qsv

Xilinx Hardware

h264_xma

NetInt Hardware

h264_nilogan

H.265

h265 (Automatic Codec Selection)

NVIDIA Hardware

h265_nvenc

Intel Hardware

h265_qsv

Xilinx Hardware

h265_xma

NetInt Hardware

h265_nilogan

Audio

AAC

aac

Opus

opus

Image

JPEG

jpeg

PNG

png

OutputProfiles

OutputProfile

The <OutputProfile> setting allows incoming streams to be re-encoded via the <Encodes> setting to create a new output stream. The name of the new output stream is determined by the rules set in <OutputStreamName>, and the newly created stream can be used according to the streaming URL format.

<OutputProfiles>
    <!-- 
    Common setting for decoders. Decodes is optional.

    <Decodes>
	Number of threads for the decoder.
	<ThreadCount>2</ThreadCount>

	To reduce resource usage, only keyframes are decoded.
	It is only activated when image encoding and video bypass are enabled.
	<KeyframeOnlyIfNeed>false</KeyframeOnlyIfNeed>
    </Decodes>
    -->

    <OutputProfile>
        <Name>bypass_stream</Name>
        <OutputStreamName>${OriginStreamName}_bypass</OutputStreamName>
        <Encodes>
            <Video>
                <Bypass>true</Bypass>
            </Video>
            <Audio>
                <Bypass>true</Bypass>
            </Audio>            
            <Audio>
                <Codec>opus</Codec>
                <Bitrate>128000</Bitrate>
                <Samplerate>48000</Samplerate>
                <Channel>2</Channel>
            </Audio>
        </Encodes>
    </OutputProfile>
</OutputProfiles>

According to the above setting, if the incoming stream name is stream, the output stream becomes stream_bypassand the stream URL can be used as follows.

  • WebRTC ws://192.168.0.1:3333/app/stream_bypass

  • LLHLS http://192.168.0.1:8080/app/stream_bypass/llhls.m3u8

  • HLS http://192.168.0.1:8080/app/stream_bypass/ts:playlist.m3u8

Encodes

Video

You can set the video profile as below:

<Encodes>
    <Video>
        <Name>h264_hd</Name>
        <Codec>h264</Codec>
        <Width>1280</Width>
        <Height>720</Height>
        <Bitrate>2000000</Bitrate>
        <Framerate>30.0</Framerate>
        <KeyFrameInterval>30</KeyFrameInterval>
        <BFrames>0</BFrames>
        <Preset>fast</Preset>
        <ThreadCount>4</ThreadCount>
    </Video>
</Encodes>

The meaning of each property is as follows:

PropertyDescription

Codec*

Specifies the vp8 or h264 codec to use

Bitrate*

Bit per second

Name

Encode name for Renditions

Width

Width of resolution

Height

Height of resolution

Framerate

Frames per second

KeyFrameInterval

Number of frames between two keyframes (0~600) default is framerate (i.e. 1 second)

BFrames

Number of B-frame (0~16) default is 0

Profile

H264 only encoding profile (baseline, main, high)

Preset

Presets of encoding quality and performance

ThreadCount

Number of threads in encoding

* required

Table of presets

A table in which presets provided for each codec library are mapped to OvenMediaEngine presets. Slow presets are of good quality and use a lot of resources, whereas Fast presets have lower quality and better performance. It can be set according to your own system environment and service purpose.

Presetsopenh264h264_nvench264_qsvvp8

slower

QP( 10-39)

p7

No Support

best

slow

QP (16-45)

p6

No Support

best

medium

QP (24-51)

p5

No Support

good

fast

QP (32-51)

p4

No Support

realtime

faster

QP (40-51)

p3

No Support

realtime

References

  • https://trac.ffmpeg.org/wiki/Encode/VP8

  • https://docs.nvidia.com/video-technologies/video-codec-sdk/nvenc-preset-migration-guide/

Audio

You can set the audio profile as below:

<Encodes>
    <Audio>
        <Name>opus_128</Name>
        <Codec>opus</Codec>
        <Bitrate>128000</Bitrate>
        <Samplerate>48000</Samplerate>
        <Channel>2</Channel>
    </Audio>
</Encodes>

The meaning of each property is as follows:

PropertyDescription

Codec*

Specifies the opus or aac codec to use

Bitrate*

Bits per second

Name

Encode name for Renditions

Samplerate

Samples per second

Channel

The number of audio channels

* required

It is possible to have an audio only output profile by specifying the Audio profile and omitting a Video one.

Image

You can set the Image profile as below:

<Encodes>
    <Image>
        <Codec>jpeg</Codec>
        <Width>1280</Width>
        <Height>720</Height>
        <Framerate>1</Framerate>
    </Image>
</Encodes>

The meaning of each property is as follows:

PropertyDescription

Codec

Specifies the jpeg or png codec to use

Width

Width of resolution

Height

Height of resolution

Framerate

Frames per second

The image encoding profile is only used by thumbnail publishers. and, bypass option is not supported.

Passthrough (Bypass)

You can configure Video and Audio to bypass transcoding as follows:

<Video>
    <Bypass>true</Bypass>
</Video>
<Audio>
    <Bypass>true</Bypass>
</Audio>

You need to consider codec compatibility with some browsers. For example, chrome only supports OPUS codec for audio to play WebRTC stream. If you set to bypass incoming audio, it can't play on chrome.

WebRTC doesn't support AAC, so if video bypasses transcoding, audio must be encoded in OPUS.

<Encodes>
    <Video>
        <Bypass>true</Bypass>
    </Video>
    <Audio>
        <Codec>opus</Codec>
        <Bitrate>128000</Bitrate>
        <Samplerate>48000</Samplerate>
        <Channel>2</Channel>
    </Audio>
</Encodes>

Conditional Encoding

If the codec or quality of the input stream is the same as the profile to be encoded into the output stream. there is no need to perform re-transcoding while unnecessarily consuming a lot of system resources. If the quality of the input track matches all the conditions of BypassIfMatch, it will be Pass-through without encoding

Matching elements in video

ElementsConditionDescription

Codec (Optional)

eq

Compare video codecs

Width (Optional)

eq, lte, gte

Compare horizontal pixel of video resolution

Height (Optional)

eq, lte, gte

Compare vertical pixel of video resolution

SAR (Optional)

eq

Compare ratio of video resolution

* eq: equal to / lte: less than or equal to / gte: greater than or equal to

Matching elements in audio

ElementsConditionDescription

Codec (Optional)

eq

Compare audio codecs

Samplerate (Optional)

eq, lte, gte

Compare sampling rate of audio

Channel (Optional)

eq, lte, gte

Compare number of channels in audio

* eq: equal to / lte: less than or equal to / gte: greater than or equal to

To support WebRTC and LLHLS, AAC and Opus codecs must be supported at the same time. Use the settings below to reduce unnecessary audio encoding.

<Encodes>
	<Video>
                <Bypass>true</Bypass>	
	</Video>
	<Audio>
		<Name>cond_audio_aac</Name>
		<Codec>aac</Codec>
		<Bitrate>128000</Bitrate>
		<Samplerate>48000</Samplerate>
		<Channel>2</Channel>
		<BypassIfMatch>
			<Codec>eq</Codec>
			<Samplerate>lte</Samplerate>
			<Channel>eq</Channel>			
		</BypassIfMatch>
	</Audio>		
	<Audio>
		<Name>cond_audio_opus</Name>
		<Codec>opus</Codec>
		<Bitrate>128000</Bitrate>
		<Samplerate>48000</Samplerate>
		<Channel>2</Channel>
		<BypassIfMatch>
			<Codec>eq</Codec>
			<Samplerate>lte</Samplerate>
			<Channel>eq</Channel>	
		</BypassIfMatch>	
	</Audio>
</Encodes>

If a video track with a lower quality than the encoding option is input, unnecessary upscaling can be prevented. SAR (Storage Aspect Ratio) is the ratio of original pixels. In the example below, even if the width and height of the original video are smaller than or equal to the width and height set in the encoding option, if the ratio is different, it means that encoding is performed without bypassing.

<Encodes>
	<Video>                                                                                                    
		<Name>prevent_upscaling_video</Name>
		<Codec>h264</Codec>
		<Bitrate>2048000</Bitrate>
		<Width>1280</Width>
		<Height>720</Height>
		<Framerate>30</Framerate>
		<BypassIfMatch>
			<Codec>eq</Codec>
			<Width>lte</Width>
			<Height>lte</Height>
			<SAR>eq</SAR>
		</BypassIfMatch>
	</Video>
</Encodes>

Keep Original and Change Bitrate

If you want to transcode with the same quality as the original. See the sample below for possible parameters that OME supports to keep original. If you remove the Width, Height, Framerate, Samplerate, and Channel parameters. then, It is transcoded with the same options as the original.

<Encodes>
    <Video>
        <Codec>vp8</Codec>
        <Bitrate>2000000</Bitrate>
    </Video>
     <Audio>
        <Codec>opus</Codec>
        <Bitrate>128000</Bitrate>
    </Audio>  
    <Image>
        <Codec>jpeg</Codec>
    </Image>
</Encodes>

Rescaling with Keep the Aspect Ratio

To change the video resolution when transcoding, use the values of width and height in the Video encode option. If you don't know the resolution of the original, it will be difficult to keep the aspect ratio after transcoding. Please use the following methods to solve these problems. For example, if you input only the Width value in the Video encoding option, the Height value is automatically generated according to the ratio of the original video.

<Encodes>
    <Video>
        <Codec>h264</Codec>
        <Bitrate>2000000</Bitrate>
        <Width>1280</Width>
        <!-- Height is automatically calculated as the original video ratio -->
        <Framerate>30.0</Framerate>
    </Video>
    <Video>
        <Codec>h264</Codec>
        <Bitrate>2000000</Bitrate>
        <!-- Width is automatically calculated as the original video ratio -->
        <Height>720</Height>
        <Framerate>30.0</Framerate>
    </Video>    
</Encodes>

Decoding Thread Configuration

The software decoder uses 2 threads by default. If the CPU speed is too low for decoding, increasing the thread count can improve performance.

<OutputProfiles>
    <!-- 
    Common setting for decoders. Decodes is optional.
    -->
    <Decodes>
	<!-- Number of threads for the decoder.-->
	<ThreadCount>2</ThreadCount>
    </Decodes>

    <OutputProfile>
    ....
    </OutputProfile>
</OutputProfiles>

Keyframe Decoding Only

If only thumbnails are used without video encoding, decoding all frames wastes CPU resources. By decoding only keyframes for thumbnails, resource usage can be reduced. However, the frame rate of the output image may not be accurate. Use the <KeyFrameOnlyIfNeed> option. Even if this value is set to true, if video encoding is enabled (Bypass is not included), all frames will be decoded. 

<OutputProfiles>
    <!-- 
    Common setting for decoders. Decodes is optional.
    -->
    <Decodes>
	<!-- 
	To reduce resource usage, only keyframes are decoded.
	It is only activated when image encoding and video bypass are enabled. 
	-->
        <KeyframeOnlyIfNeed>false</KeyframeOnlyIfNeed>
    </Decodes>

    <OutputProfile>
       <Encodes>
           <Video>
                <Bypass>true</Bypass>	
  	   </Video>
           <Image>
               <Codec>jpeg</Codec>
               <Width>1280</Width>
               <Height>720</Height>
               <Framerate>1</Framerate>
          </Image>
       </Encodes>
    </OutputProfile>
</OutputProfiles>

Adaptive Bitrate Streaming (ABR)

From version 0.14.0, OvenMediaEngine can encode same source with multiple bitrates renditions and deliver it to the player.

As shown in the example configuration below, you can provide ABR by adding <Playlists> to <OutputProfile>. There can be multiple playlists, and each playlist can be accessed with <FileName>.

The method to access the playlist set through LLHLS is as follows.

http[s]://<domain>[:port]/<app>/<stream>/<FileName>.m3u8

The method to access the playlist set through HLS is as follows.

http[s]://<domain>[:port]/<app>/<stream>/<FileName>.m3u8?format=ts

The method to access the Playlist set through WebRTC is as follows.

ws[s]://<domain>[:port]/<app>/<stream>/<FileName>

Note that <FileName> must never contain the playlist and chunklist keywords. This is a reserved word used inside the system.

To set up <Rendition>, you need to add <Name> to the elements of <Encodes>. Connect the set <Name> into <Rendition><Video> or <Rendition><Audio>.

In the example below, three quality renditions are provided and the URL to play the abr playlist as LLHLS is https://domain:port/app/stream/abr.m3u8 and The WebRTC playback URL is wss://domain:port/app/stream/abr

TS files used in HLS must have A/V pre-muxed, so the EnableTsPackaging option must be set in the Playlist.

<OutputProfile>
	<Name>bypass_stream</Name>
	<OutputStreamName>${OriginStreamName}</OutputStreamName>
	<!--LLHLS URL : https://domain/app/stream/abr.m3u8 --> 
	<Playlist>
		<Name>For LLHLS</Name>
		<FileName>abr</FileName>
		<Options> <!-- Optinal -->
			<!-- 
			Automatically switch rendition in WebRTC ABR 
			[Default] : true
			-->
			<WebRtcAutoAbr>true</WebRtcAutoAbr> 
			<EnableTsPackaging>true</EnableTsPackaging>
		</Options>
		<Rendition>
			<Name>Bypass</Name>
			<Video>bypass_video</Video>
			<Audio>bypass_audio</Audio>
		</Rendition>
		<Rendition>
			<Name>FHD</Name>
			<Video>video_1280</Video>
			<Audio>bypass_audio</Audio>
		</Rendition>
		<Rendition>
			<Name>HD</Name>
			<Video>video_720</Video>
			<Audio>bypass_audio</Audio>
		</Rendition>
	</Playlist>
	<!--LLHLS URL : https://domain/app/stream/llhls.m3u8 --> 
	<Playlist>
		<Name>Change Default</Name>
		<FileName>llhls</FileName>
		<Rendition>
			<Name>HD</Name>
			<Video>video_720</Video>
			<Audio>bypass_audio</Audio>
		</Rendition>
	</Playlist> 
	<Encodes>
		<Audio>
			<Name>bypass_audio</Name>
			<Bypass>true</Bypass>
		</Audio>
		<Video>
			<Name>bypass_video</Name>
			<Bypass>true</Bypass>
		</Video>
		<Audio>
			<Codec>opus</Codec>
			<Bitrate>128000</Bitrate>
			<Samplerate>48000</Samplerate>
			<Channel>2</Channel>
		</Audio>
		<Video>
			<Name>video_1280</Name>
			<Codec>h264</Codec>
			<Bitrate>5024000</Bitrate>
			<Framerate>30</Framerate>
			<Width>1920</Width>
			<Height>1280</Height>
			<Preset>faster</Preset>
		</Video>
		<Video>
			<Name>video_720</Name>
			<Codec>h264</Codec>
			<Bitrate>2024000</Bitrate>
			<Framerate>30</Framerate>
			<Width>1280</Width>
			<Height>720</Height>
			<Preset>faster</Preset>
		</Video>
	</Encodes>
</OutputProfile>

Supported codecs by streaming protocol

Even if you set up multiple codecs, there is a codec that matches each streaming protocol supported by OME, so it can automatically select and stream codecs that match the protocol. However, if you don't set a codec that matches the streaming protocol you want to use, it won't be streamed.

The following is a list of codecs that match each streaming protocol:

ProtocolSupported Codec

WebRTC

VP8, H.264, Opus

LLHLS

H.264, H.265, AAC

Therefore, you set it up as shown in the table. If you want to stream using LLHLS, you need to set up H.264, H.265 and AAC, and if you want to stream using WebRTC, you need to set up Opus.

Also, if you are going to use WebRTC on all platforms, you need to configure both VP8 and H.264. This is because different codecs are supported for each browser, for example, VP8 only, H264 only, or both.

However, don't worry. If you set the codecs correctly, OME automatically sends the stream of codecs requested by the browser.

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