Enum AudioEncoder
- Namespace
- VisioForge.Core.Types.FFMPEGEXE
- Assembly
- VisioForge.Core.dll
Specifies the audio encoder to be used by the FFMPEG executable for audio encoding operations.
public enum AudioEncoderFields
Auto = 0-
Automatically selects the audio encoder based on the output format.
When set to Auto, FFMPEG will choose the most appropriate encoder based on the output container format specified. This is the recommended option when you want FFMPEG to handle encoder selection automatically.
AAC = 1-
AAC (Advanced Audio Coding) encoder for high-quality compressed audio.
AAC is a lossy compression format that provides better sound quality than MP3 at similar bit rates. Widely supported by modern devices and streaming platforms. Recommended for general-purpose audio encoding.
AC3 = 2-
AC-3 (Dolby Digital) encoder for surround sound audio.
AC-3 is the standard audio format for DVDs and digital television (ATSC). Supports up to 5.1 surround sound channels. Commonly used in home theater systems and broadcast applications.
ADPCM_G722 = 3-
G.722 ADPCM encoder for wideband speech coding.
G.722 is a 7 kHz wideband speech codec operating at 48, 56, and 64 kbit/s. Commonly used in VoIP applications and professional audio conferencing systems for improved voice quality compared to narrowband codecs.
ADPCM_G726 = 4-
G.726 ADPCM encoder for speech compression.
G.726 is an ITU-T standard for speech compression using ADPCM at rates of 16, 24, 32, and 40 kbit/s. Commonly used in digital telephony and voice storage applications.
ADPCM_MS = 5-
Microsoft ADPCM encoder for Windows audio compression.
Microsoft's implementation of ADPCM provides 4:1 compression ratio. Commonly used in older Windows multimedia files and games. Provides reasonable quality for speech and simple audio content.
ALAC = 6-
ALAC (Apple Lossless Audio Codec) for lossless audio compression.
ALAC provides lossless compression with file sizes typically 40-60% of the original. Developed by Apple and commonly used in iTunes and Apple devices. Ideal for archiving audio while maintaining perfect quality.
AMR_NB = 7-
AMR-NB (Adaptive Multi-Rate NarrowBand) encoder for speech compression.
AMR-NB is optimized for speech coding at 8 kHz sampling rate. Widely used in GSM and UMTS mobile phone networks. Provides good speech quality at very low bitrates (4.75 to 12.2 kbit/s).
AMR_WB = 8-
AMR-WB (Adaptive Multi-Rate WideBand) encoder for wideband speech.
AMR-WB provides improved speech quality over AMR-NB with 16 kHz sampling rate. Also known as HD Voice in mobile networks. Operates at bitrates from 6.60 to 23.85 kbit/s.
EAC3 = 9-
E-AC-3 (Enhanced AC-3, Dolby Digital Plus) encoder for advanced surround sound.
E-AC-3 is an enhanced version of AC-3 supporting up to 7.1 channels and higher bitrates. Used in Blu-ray discs, streaming services, and digital broadcasting. Provides better efficiency and quality than standard AC-3.
FLAC = 10-
FLAC (Free Lossless Audio Codec) for open-source lossless compression.
FLAC is an open-source lossless codec that typically achieves 50-70% compression. Widely supported and ideal for audio archiving and high-quality music distribution. Provides perfect audio reproduction with no quality loss.
G723_1 = 11-
G.723.1 encoder for low bitrate speech compression.
G.723.1 is optimized for speech at very low bitrates (5.3 and 6.3 kbit/s). Commonly used in VoIP applications where bandwidth is limited. Provides acceptable speech quality for telephony applications.
MP2 = 12-
MP2 (MPEG-1 Audio Layer II) encoder for broadcast audio.
MP2 is widely used in broadcasting and digital television. Provides good quality at bitrates of 128-384 kbit/s. Still used in DAB radio and some TV standards despite being older than MP3.
MP3 = 13-
MP3 (MPEG-1 Audio Layer III) encoder for universal audio compression.
MP3 is the most widely supported lossy audio format. Provides good quality at 128-320 kbit/s for music. Despite newer formats offering better efficiency, MP3 remains popular due to universal compatibility.
OPUS = 14-
Opus encoder for versatile audio compression.
Opus is a modern codec that excels at both speech and music compression. Provides excellent quality at low bitrates and low latency. Ideal for real-time applications like VoIP, streaming, and video conferencing.
PCM_ALAW = 15-
PCM A-law (G.711 A-law) encoder for telephony audio.
A-law is a companding algorithm used in European telephone networks. Provides 8-bit PCM encoding with logarithmic compression. Standard for digital telephony in Europe and international circuits.
PCM_F32BE = 16-
PCM 32-bit floating point big-endian encoder.
Uncompressed 32-bit floating point audio in big-endian byte order. Provides high dynamic range and precision. Used in professional audio applications where quality is paramount and file size is not a concern.
PCM_F32LE = 17-
PCM 32-bit floating point little-endian encoder.
Uncompressed 32-bit floating point audio in little-endian byte order. Common on x86 systems. Provides professional-quality audio with high dynamic range, suitable for audio processing and mastering.
PCM_F64BE = 18-
PCM 64-bit floating point big-endian encoder.
Uncompressed 64-bit floating point audio in big-endian byte order. Provides extreme precision for scientific and high-end audio applications. Rarely needed except for specialized audio analysis or processing.
PCM_F64LE = 19-
PCM 64-bit floating point little-endian encoder.
Uncompressed 64-bit floating point audio in little-endian byte order. Provides maximum precision for audio processing. Used in scientific applications and high-end audio production where precision is critical.
PCM_MULAW = 20-
PCM μ-law (G.711 μ-law) encoder for telephony audio.
μ-law (mu-law) is a companding algorithm used in North American and Japanese telephone networks. Provides 8-bit PCM encoding with logarithmic compression. Standard for digital telephony in North America.
PCM_S16BE = 21-
PCM signed 16-bit big-endian encoder.
Standard CD-quality uncompressed audio in big-endian byte order. Provides 96 dB dynamic range. Common format for audio interchange on big-endian systems and professional audio equipment.
PCM_S16BE_Planar = 22-
PCM signed 16-bit big-endian planar encoder.
Uncompressed 16-bit audio with planar channel layout in big-endian order. Each channel is stored separately rather than interleaved. Useful for audio processing applications that operate on individual channels.
PCM_S16LE = 23-
PCM signed 16-bit little-endian encoder.
Standard CD-quality uncompressed audio in little-endian byte order. The most common uncompressed format on Windows and x86 systems. Provides excellent quality for most audio applications.
PCM_S16LE_Planar = 24-
PCM signed 16-bit little-endian planar encoder.
Uncompressed 16-bit audio with planar channel layout in little-endian order. Channels are stored separately, facilitating multi-channel audio processing and analysis operations.
PCM_S24BE = 25-
PCM signed 24-bit big-endian encoder.
High-quality uncompressed audio with 144 dB dynamic range in big-endian order. Common in professional audio production and mastering. Provides superior quality compared to 16-bit while using less space than 32-bit.
PCM_S24daud = 26-
PCM D-Cinema audio signed 24-bit encoder.
Specialized 24-bit PCM format used in Digital Cinema Package (DCP) files. Conforms to digital cinema standards for theatrical presentation. Required for professional cinema audio tracks.
PCM_S24LE = 27-
PCM signed 24-bit little-endian encoder.
High-quality uncompressed audio with 144 dB dynamic range in little-endian order. Standard for professional audio on Windows and x86 systems. Ideal balance between quality and file size for high-end audio production.
PCM_S24LE_Planar = 28-
PCM signed 24-bit little-endian planar encoder.
High-quality 24-bit audio with planar channel layout in little-endian order. Facilitates professional multi-channel audio processing while maintaining high dynamic range and precision.
PCM_S32BE = 29-
PCM signed 32-bit big-endian encoder.
Uncompressed 32-bit integer audio in big-endian byte order. Provides very high dynamic range for professional applications. Common in high-end audio workstations and mastering systems.
PCM_S32LE = 30-
PCM signed 32-bit little-endian encoder.
Uncompressed 32-bit integer audio in little-endian byte order. Provides extensive headroom for audio processing. Standard format for high-end audio production on x86 systems.
PCM_S32LE_Planar = 31-
PCM signed 32-bit little-endian planar encoder.
High-precision 32-bit audio with planar channel layout in little-endian order. Ideal for complex multi-channel audio processing requiring maximum precision and dynamic range.
PCM_S8 = 32-
PCM signed 8-bit encoder.
Basic uncompressed 8-bit audio format. Provides limited dynamic range (48 dB) but small file sizes. Suitable for simple audio, voice recordings, or retro-style audio applications.
PCM_S8_Planar = 33-
PCM signed 8-bit planar encoder.
Basic 8-bit audio with planar channel layout. Each channel stored separately. Used in specialized applications requiring channel-wise processing of low-quality audio data.
PCM_U16BE = 34-
PCM unsigned 16-bit big-endian encoder.
Uncompressed 16-bit audio using unsigned integers in big-endian order. Less common than signed formats. Used in some legacy systems and specific audio hardware interfaces.
PCM_U16LE = 35-
PCM unsigned 16-bit little-endian encoder.
Uncompressed 16-bit audio using unsigned integers in little-endian order. Sometimes used in Windows multimedia files and certain audio APIs that expect unsigned sample data.
PCM_U24BE = 36-
PCM unsigned 24-bit big-endian encoder.
Uncompressed 24-bit audio using unsigned integers in big-endian order. Rare format used in specialized audio equipment or legacy systems requiring unsigned sample representation.
PCM_U24LE = 37-
PCM unsigned 24-bit little-endian encoder.
Uncompressed 24-bit audio using unsigned integers in little-endian order. Uncommon format, occasionally used in specific audio processing pipelines or hardware interfaces.
PCM_U32BE = 38-
PCM unsigned 32-bit big-endian encoder.
Uncompressed 32-bit audio using unsigned integers in big-endian order. Very rare format, used only in specialized applications requiring unsigned 32-bit sample representation.
PCM_U32LE = 39-
PCM unsigned 32-bit little-endian encoder.
Uncompressed 32-bit audio using unsigned integers in little-endian order. Rarely used format, primarily for compatibility with specific audio processing systems or hardware requirements.
PCM_U8 = 40-
PCM unsigned 8-bit encoder.
Basic uncompressed 8-bit audio using unsigned values (0-255). Common in older WAV files and legacy audio systems. Provides limited quality but maximum compatibility with older software.
Speex = 41-
Speex encoder for speech compression.
Speex is designed specifically for speech compression with features like variable bitrate and voice activity detection. Although superseded by Opus for new applications, it remains in use for compatibility.
Vorbis = 42-
Vorbis encoder for open-source audio compression.
Vorbis is an open-source lossy audio codec that provides better quality than MP3 at equivalent bitrates. Commonly used in OGG containers and game audio. Free from patent restrictions.
WavPack = 43-
WavPack encoder for hybrid lossless/lossy compression.
WavPack provides both lossless and lossy compression modes with the unique ability to create hybrid files. Supports high-resolution audio and provides good compression ratios. Ideal for audio archiving with flexibility.
Remarks
This enumeration provides a comprehensive list of audio encoders supported by FFMPEG. The choice of encoder affects the output file format, compression efficiency, quality, and compatibility with different playback devices and software.