Some codecs operate specifically by eliminating audio that falls outside a given frequency range. There is a correlation between the sample rate and the maxium sound frequency that can be represented by a waveform represented by a codec. At a theoretical level, the maximum frequency a codec can represent is the sample rate divided by two; this frequency is called the Nyquist frequency. In reality, the maximum is slightly lower, but it's close. The audio frequency bandwidth comes into play especially vividly when a codec is designed or configured to represent human speech rather than a broad range of sounds.
Human speech generally resides within the audio frequency range of Hz to 18 kHz. However, the vast majority of human vocalizations exist in the range Hz to 8 kHz, and you can capture enough of human vocalizations in the frequency range Hz to 3 kHz to still be understandable. For that reason, speech-specific codecs often begin by dropping sound that falls outside a set range.
That range is the audio frequency bandwidth. This reduces the amount of data that needs to be encoded from the outset. Below we take a brief look at each of these codecs, looking at their basic capabilities and their primary use cases. Designed to be able to provide more compression with higher audio fidelity than MP3, AAC has become a popular choice, and is the standard format for audio in many types of media, including Blu-Ray discs and HDTV, as well as being the format used for songs purchased from online vendors including iTunes.
AAC has a number of profiles that define methods of compressing audio for specific use cases, including everything from high quality surround sound to low-fidelity audio for speech-only use.
As a patent-encumbered format, AAC support is somewhat less predictable. For example, Firefox only supports AAC if support is provided by the operating system or an external library. Due to patent issues, Firefox does not directly support AAC. Instead, Firefox relies upon a platform's native support for AAC.
This capability was introduced on each platform in different Firefox releases:. In addition, AAC is not available in Chromium builds. After initially being a closed format, Apple opened it up under an Apache license. Cross-platform and browser support for ALAC is not very strong, making it a less than ideal choice for general usage. Otherwise, FLAC is likely a better choice, if you must use a lossless codec. Keep in mind, however, that lossless codecs require substantially more bandwidth and storage capacity, and may not be a good choice outside very specific use cases.
The Adaptive Multi-Rate audio codec is optimized for encoding human speech efficiently. It was standardized in as part of the 3GPP audio standard used for both GSM and UMTS cellular telephony, and uses a multi-rate narrowband algorithm to encode audio frequencies at a telephony-grade quality level at around 7.
In addition to being used for real-time telephony, AMR audio may be used for voicemail and other short audio recordings. AMR audio which is stored in files may be typed. As a speech-specific codec, AMR is essentially useless for any other content, including audio containing only singing voices. Additionally, because AMR is designed to minimize capacity requirements, it only captures the portion of the full audio frequency bandwidth of human speech which is absolutely necessary to understand what's being said, so quality is reduced accordingly.
However, if you need high-fidelity reproduction of human speech—or even low-quality music reproduction—you need to choose another format. It provides good compression rates with no loss of audio fidelity; that is, the decompressed audio is identical to the original. Because the compression algorithm is specifically designed for audio, it gives better results than would be achieved using a general-purpose compression algorithm. FLAC is a great choice for smaller audio effects files where pristine quality and tonal accuracy are desired, as well as for archival of music.
The G. It supports voice-grade audio covering frequencies from to Hz. It is used extensively for telephone traffic and voicemail, and it is the highest quality audio encoding which can be transmitted through the public telephone network.
There are two flavors of G. There is no substantial quality difference between the two laws, and it is simple to transcode audio from one to the other. Nevertheless, it is important to specify which law is in use in any replay application or file format. This codec is required to be supported by all WebRTC solutions because it is simple, easy to implement, widely-used, and broadly compatible across all modern computing platforms. Its audio coding bandwidth is limited to the 50 Hz to 7, Hz range, which covers most of the frequency range of typical human vocalization.
This makes it ill-suited for handling any audio that might range outside the human speech range, such as music. The patents behind MP3 have expired, removing many or most licensing concerns around using MP3 files in your projects. That makes them a good choice for many projects. For patent reasons, Firefox did not directly support MP3 prior to version 71; instead, platform-native libraries were used to support MP3.
The Opus audio format was created by the Xiph. It's a good general-purpose audio codec that can efficiently handle both low-complexity audio such as speech as well as music and other high-complexity sounds. Opus supports multiple compression algorithms, and can even use more than one algorithm in the same audio file, since the encoder can choose the bit rate, audio bandwidth, algorithm, and other details of the compression settings for each frame of audio.
Get it Free for 1 Year Pricing. What are codecs? The following is a list of Codecs that are in common use today:. Audio codecs:. GSM — 13 Kbps full rate , 20ms frame size. ITU G. Speex — 2. LPC10 — 2. Video codecs:. It defines how the video, audio and other data is stored within the container. If you want to change the way the data is being read, you can swap out the container without changing the file type.
The process of changing the container format of a file for delivery to a different target platform is called transmuxing. One of the biggest sources of confusion about video is not realising that the file format aka container format is completely different from the video format the video codec. Examples of file formats i.
The flash container could also hold video encoded with vp6 instead of H. On the web,. Like the. WEBP image file,. WEBM was created by Google as an efficient means of disseminating media to a large audience. WEBM video files are relatively small in size, and as such are not as high in terms of quality as some of the other file types on this list. They are low in file size but also relatively low in quality. They also have lossy compression, meaning their quality will degrade after being edited numerous times.
MPV files are best used when video will be recorded once and never edited. OGG files are an open-source alternative to. MPG files, and are used for high-quality videos to be streamed via the internet. OGG files are used for streaming, they are higher in quality than. WEBM files — meaning they will take longer to be delivered to the end-user.
Due to. OGG files being open sourced, they can be used in a variety of applications, including GPS receivers and media players both desktop and portable.
M4V are similar to. MPG files in that they can contain audio and video, or can simply be solely audio files. M4P, and. M4V are used for streaming video via the internet. They are generally higher in quality than. WEBM files, but tend to be larger in file size.
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