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

Type	Codec
Video	VP8, H.264
Audio	AAC, Opus

Type

Codec

Video

VP8, H.264

Audio

AAC, Opus

Encoders

Type	Codec	Codec of Configuration
Video	VP8	vp8
	H.264	h264 (Automatic Codec Selection)
		h264_openh264
		h264_nvenc
		h264_qsv
		h264_beamr (Enterprise Only)
Audio	AAC	aac
	Opus	opus
Image	JPEG	jpeg
	PNG	png

Type

Codec

Codec of Configuration

Video

VP8

vp8

H.264

h264 (Automatic Codec Selection)

h264_openh264

h264_nvenc

h264_qsv

h264_beamr (Enterprise Only)

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>
    <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

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:

Property Description

Property	Description
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

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.

Presets	openh264	h264_nvenc	h264_qsv	vp8
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

Presets

openh264

h264_nvenc

h264_qsv

vp8

slower

QP( 10-39)

No Support

best

slow

QP (16-45)

No Support

best

medium

QP (24-51)

No Support

good

fast

QP (32-51)

No Support

realtime

faster

QP (40-51)

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:

Property Description

Property	Description
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

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:

Property Description

Property	Description
Codec	Specifies the `jpeg` or `png` codec to use
Width	Width of resolution
Height	Height of resolution
Framerate	Frames per second

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.

Bypass without transcoding

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 transcoding

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

Elements	Condition	Description
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

Elements

Condition

Description

Codec (Optional)

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)

Compare ratio of video resolution

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

Matching elements in audio

Elements	Condition	Description
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

Elements

Condition

Description

Codec (Optional)

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 the original with transcoding

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 while 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>

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 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

<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> 
		</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:

Protocol	Supported Codec
WebRTC	VP8, H.264, Opus
LLHLS	H.264, AAC

Protocol

Supported Codec

WebRTC

VP8, H.264, Opus

LLHLS

H.264, 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 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.