This article is a draft and may contain incomplete or incorrect information.
AC3 (Dolby Digital) AC3 (Dolby Digital) EAC3 (Dolby Digital Plus) The codec to encode the audio track. Codec 'Without reencoding' copies the audio stream from the input file into output without re-encoding if possible. Dolby Digital Plus (E-AC-3) provides up to twice the efficiency of Dolby Digital while adding new features like 7.1-ch audio, support for descriptive video services, and support for Dolby Atmos. Dolby Digital Plus is widely used by streaming and broadcast services to deliver surround sound audio at lower bitrates.
Dieser Artikel ist ein Entwurf und enthält möglicherweise unvollständige oder falsche Information.
Encoding vs Passthrough
HandBrake supports two modes of operation for handling audio tracks from thesource.
It can either:
Re-Encode the track into a new format. For example, AC3 or DTS audio from aDVD or BluRay into AAC for playback on mobile devices.
Passthru copies the audio track from the source to the newly encoded file.
Advanced Audio Coding (AAC) is one of the most popular formats availablein HandBrake and is widly supported by media players and mobile devices.
On Linux and Windows, HandBrake uses the FFmpeg AAC encoder.
On Mac, HandBrake uses Apple’s CoreAudio AAC encoder. It’s known to bemuch better quality and is the default option for Mac users.
HE-AAC ( Mac Only)
High Efficiency AAC encoder. This encoder is optimized for low-bitrateapplications which may be useful for streaming.
Available on Mac only.
- HandBrake also offers the MP3 audio codec, for which it uses the verywell-regarded lame encoder.
- HandBrake offers Vorbis audio encoding as well, for which it uses theaoTuV branch that offers quality tweaks.
AC3 and E-AC3:
HandBrake can encode to AC3 using FFmpeg.
This is not the same as Passthru. You can encode any audio track to AC3.
- Available in MKV files.
Flac 16 and 24bit (FFmpeg)
Loss-less audio compression.
Available in MKV files.
It is not currently possible to passthru the compression level to theffmpeg encoder which is currently defaulted to “5”
HandBrake supports passthru for a limited number of popular source formats.These are: AC3, E-AC3, DTS, DTS-HD, TrueHD, MP3, MP2, AAC or flac (Passthru)
This copies the audio bit-for-bit from the source so offers no loss in audioquality. These audio tracks can be quite large so can impact the final size ofthe output file quite substantially.
In the audio encoders list, there is an option called “Auto Passthru”. Whenencoding, this will automatically passthru any supported format to the sourcefile. If the source format is not supported, or not supported in the outputcontainer, the fallback encoder will be used. The default is AAC.
In this article, I will discuss and compare the key features of some popular audio codecs. In particular, I will focus on the aspects that are most relevant from a user’s perspective (and not from an algorithm perspective).
If you’ve been wondering what exactly is the difference between MP3, AAC, AC3, WAV, WMA, and Opus, then this article is for you.
This article assumes that you have an understanding of the common terms related to digital audio like frequency, sampling rate, and channels. If not, do check my article on the basics of digital audio here.
A note on “bit rate” before we start. The bit rate of a codec refers to the (average) number of bits it uses to encode one second of raw audio. Bit rate is expressed in units of kbps (kilobits per second). One kilobit is 1,000 bits. Most often, the bit rate is a tunable parameter during encoding.
Bit rate is important because it has a direct impact on the size of the encoded file (in storage applications) and the network bandwidth needed for audio transmission (in streaming and broadcast applications). The lower the bit rate, the lower the size of the encoded audio and the network bandwidth requirement.
Let’s now jump right in and take a look at the characteristics of some popular codecs.
MP3 is the most well-known audio codec by far and almost synonymous with digital audio in the popular imagination. MP3 ushered in the digital audio revolution in the ‘90s.
MP3 was standardized by ISO as MPEG-1 Audio Layer III in 1993.
MPEG-2 (Part 3) included an extension to MP3 that supported lower sampling rates (16, 22.05, 24 kHz) and channels up to 5.1. But these sampling rates discard a good amount of human audible frequencies (i.e., above 8, 11.025, and 12 kHz respectively). As a result, they are not useful for all kinds of audio.
Over now to some facts about MP3.
ISO standardized AAC in MPEG-2 and MPEG-4 as the successor to MP3. AAC is an abbreviation for Advanced Audio Coding. AAC features several improvements and new coding techniques that enable significant bit rate improvements over MP3 at the same audio quality. Another way of looking at this is that AAC gives superior audio quality than MP3 at the same bit rate.
AAC is much more versatile than MP3. It supports a broader range of sampling rates, bit rates, and numbers of channels. It also has various variants intended for different use-cases and applications. Some of the variants are AAC LC (Low Complexity), AAC Main, AAC LD (Low Delay), and HE-AAC (High Efficiency).
AC3 (Dolby Digital)
AC3 is an audio codec from Dolby. Dolby uses the market name “Dolby Digital” for AC3. Dolby Digital is almost synonymous with “surround sound” in cinema, home theaters, and broadcast.
Dolby Digital supports up to 5.1 channels – left, right, center, left surround, right surround, and a low-frequency enhancement channel. The surround channels enable producers to create a sensation of sound from any direction surrounding the listener.
Ac3 Vs Eac3 Light
Dolby upgraded AC3 with EAC3. EAC3 supports 7.1 channels. Dolby markets EAC3 as “Dolby Digital Plus”. AC3 programs will play on devices and software supporting EAC3.
Dolby technology adapts the audio to the actual number of speakers present on the playback device. So AC3 and EAC3 will play fine on Dolby-approved stereo devices.
Opus is an open-source, royalty-free audio codec designed for interactive audio applications (VoIP, video conferencing,…). It has one of the lowest latencies among audio codecs by design.
Opus is a combination of SILK (a voice codec by Skype) and CELT (an audio codec from Xiph.org). It supports three modes – SILK only, CELT only, and hybrid (a combination of SILK and CELT). The mode is selected based on the content being encoded. It even allows switching between the modes in the middle of the stream.
Ac3 Vs Eac3 Iphone
Opus has been steadily gaining in popularity after it was standardized by IETF in 2012. It is now widely supported and used in the Internet and VoIP applications. WhatsApp uses Opus as its VoIP codec.
Opus supports mono and stereo channels only.
WAV is short for Waveform Audio File Format. It is not a codec. It is an audio file format commonly used to store uncompressed audio. It belongs to the RIFF family of file formats.
The information about sample rate, number of channels, and bit depth is part of the WAV file header (or fmt sub-chunk, to be precise). It supports any value (that fits within the number of bits allocated) for these parameters.
It is popular among audio professionals because it is as close to the original recorded audio as it can get. It is widely supported in professional audio editing software and media players. It is not suited for consumer or end-user applications due to the large data sizes.
WMA (Windows Media Audio) is a family of proprietary audio codecs developed by Microsoft. WMA is a part of the Microsoft Media Foundation. It was intended to be a competitor to the more widely supported codecs that we discussed above.
The WMA family consists of four codecs – WMA, WMA Pro, WMA Lossless, and WMA Voice. While WMA can contain 48 kHz stereo at most, WMA Pro supports up to 7.1 channels and sampling rates till 96 kHz. WMA is much more common in use than WMA Pro.
WMA files can be played by most media players for Windows OS. They include Windows Media Player (of course!), VLC, Media Player Classic (MPC) family, Winamp, and RealPlayer.
WMA encoding can be carried out with Windows Media Foundation, Windows Media Player, Windows Movie Maker, RealPlayer, and Adobe Audition, among others.
This brings us to the end of this article. Thank you for reading. Let us know your opinions, comments, and suggestions below.
Mohammed Harris is a freelance writer on technology, especially multimedia and its applications.
Previously, he worked in the embedded multimedia software industry for more than 15 years, with successful stints in multimedia codec and systems engineering, project management, and technical management. He delivered several successful projects and solutions for Tier-1 customers from semiconductor, consumer electronics OEMs, and multimedia algorithm / IP licensing majors.
He has a keen interest and strong expertise in multimedia signal processing.
He holds a Bachelor’s degree in Electrical and Electronics Engineering from Birla Institute of Technology and Science (BITS), Pilani (India), and a Master’s degree in Electrical Engineering with emphasis on Communications and Signal Processing from the University of California, San Diego (USA).