Comment on page
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.
Type | Codec |
|---|---|
Video | VP8, H.264 |
Audio | AAC, Opus |
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 |
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.WebRTCws://192.168.0.1:3333/app/stream_bypassLLHLShttp://192.168.0.1:8080/app/stream_bypass/llhls.m3u8
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 |
|---|---|
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 |
References
- https://trac.ffmpeg.org/wiki/Encode/VP8
- https://docs.nvidia.com/video-technologies/video-codec-sdk/nvenc-preset-migration-guide/
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 |
|---|---|
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.
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 |
|---|---|
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.
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>
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
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 |
* eq: equal to / lte: less than or equal to / gte: greater than or equal to
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 |
* 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>
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>
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>
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>.m3u8The 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>
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 |
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.