H.264, H.265 or AV1 — which codec for the smallest file at the same quality?

AV1 is the most efficient (royalty-free, ~30% smaller than H.265) but encoding is slow. H.265 (HEVC) saves ~30–50% over H.264 and is supported by every modern phone and desktop. H.264 remains the safest baseline for legacy compatibility. Rule of thumb: archives → AV1, daily use → H.265, broadest reach → H.264.

CRF, two-pass or constant bitrate — which encoding mode?

CRF (Constant Rate Factor) is the best default for offline files: ffmpeg picks the bitrate frame-by-frame to maintain perceived quality. Two-pass is only better when you must hit an exact final size (DVD targets). Constant bitrate is for streaming with a fixed channel. For "smallest at quality X" always use CRF.

Why is the audio out of sync after converting?

Common causes: (1) variable framerate source rendered as constant (use -vsync vfr to preserve VFR); (2) different audio sample rates not resampled (add -ar 48000 ); (3) container limitations (MP4 with variable framerate misbehaves — prefer MKV during editing, encode to MP4 only at the end). Always run ffprobe on both source and output to compare timing.

Can I convert without re-encoding to save time?

Yes if you only change the container: ffmpeg -i in.mkv -c copy out.mp4 . This remuxes the stream without re-encoding, takes seconds even for hours of footage. Limitations: codec must be supported by the target container (e.g. you cannot put H.264 in WebM, only VP8/VP9/AV1). To shrink size you must re-encode.

H.264, H.265 or AV1 — which codec for the smallest file at the same quality?

AV1 is the most efficient (royalty-free, ~30% smaller than H.265) but encoding is slow. H.265 (HEVC) saves ~30–50% over H.264 and is supported by every modern phone and desktop. H.264 remains the safest baseline for legacy compatibility. Rule of thumb: archives → AV1, daily use → H.265, broadest reach → H.264.

CRF, two-pass or constant bitrate — which encoding mode?

CRF (Constant Rate Factor) is the best default for offline files: ffmpeg picks the bitrate frame-by-frame to maintain perceived quality. Two-pass is only better when you must hit an exact final size (DVD targets). Constant bitrate is for streaming with a fixed channel. For "smallest at quality X" always use CRF.

Why is the audio out of sync after converting?

Common causes: (1) variable framerate source rendered as constant (use -vsync vfr to preserve VFR); (2) different audio sample rates not resampled (add -ar 48000 ); (3) container limitations (MP4 with variable framerate misbehaves — prefer MKV during editing, encode to MP4 only at the end). Always run ffprobe on both source and output to compare timing.

Can I convert without re-encoding to save time?

Yes if you only change the container: ffmpeg -i in.mkv -c copy out.mp4 . This remuxes the stream without re-encoding, takes seconds even for hours of footage. Limitations: codec must be supported by the target container (e.g. you cannot put H.264 in WebM, only VP8/VP9/AV1). To shrink size you must re-encode.

Extract Audio from Video with FFmpeg: MP3, AAC, FLAC

Extracting Audio with FFmpeg

FFmpeg is the standard tool for extracting audio from video. The most efficient operation: direct extraction without re-encoding (-c:a copy) — instant and lossless.

Direct Extraction (No Re-encoding)

ffmpeg -i video.mp4 -vn -c:a copy audio.aac
ffmpeg -i video.mkv -vn -c:a copy audio.ac3
ffmpeg -i video.webm -vn -c:a copy audio.ogg

# Identify audio codec
ffprobe -v error -select_streams a:0 \
  -show_entries stream=codec_name -of default=nw=1 video.mp4

Extract and Convert to MP3

ffmpeg -i video.mp4 -vn -c:a libmp3lame -b:a 192k audio.mp3
ffmpeg -i video.mp4 -vn -c:a libmp3lame -b:a 320k audio_hq.mp3
ffmpeg -i video.mp4 -vn -c:a libmp3lame -q:a 2 audio_vbr.mp3

Extract and Convert to AAC

ffmpeg -i video.mp4 -vn -c:a aac -b:a 128k audio.aac
ffmpeg -i video.mp4 -vn -c:a aac -b:a 256k audio.m4a

Extract to FLAC (Lossless)

ffmpeg -i video.mkv -vn -c:a flac audio.flac

Extract to WAV

ffmpeg -i video.mp4 -vn -c:a pcm_s16le -ar 44100 audio.wav
ffmpeg -i video.mp4 -vn -c:a pcm_s24le audio_24bit.wav

Extract a Time Segment

ffmpeg -i video.mp4 -ss 00:01:30 -to 00:03:00 -vn -c:a libmp3lame -b:a 192k segment.mp3
ffmpeg -i video.mp4 -ss 90 -t 60 -vn -c:a copy segment.aac

Multiple Audio Tracks (MKV)

ffprobe -v error -select_streams a \
  -show_entries stream=index,codec_name -of csv video.mkv

ffmpeg -i video.mkv -map 0:a:0 -c:a flac track_1.flac
ffmpeg -i video.mkv -map 0:a:1 -c:a flac track_2.flac

Batch Processing

for f in *.mp4; do
  ffmpeg -i "$f" -vn -c:a libmp3lame -b:a 192k "${f%.mp4}.mp3"
done

for f in *.mkv; do
  ffmpeg -i "$f" -vn -c:a flac "${f%.mkv}.flac"
done

Recommended Formats

Use	Format	Command
Podcast / streaming	MP3 192 kbps	`-c:a libmp3lame -b:a 192k`
Music platforms	AAC 256 kbps	`-c:a aac -b:a 256k`
Professional editing	WAV 24-bit	`-c:a pcm_s24le`
Lossless archive	FLAC	`-c:a flac`

Conclusion

FFmpeg extracts audio from any video with a single command. Use -c:a copy for maximum efficiency. For distribution, MP3 at 192 kbps offers the best quality/compatibility balance.

Advanced Use Cases

Beyond casual conversion, modern audio formats cover highly specific professional scenarios. Music production: professional studios work with WAV or FLAC during mastering to avoid cumulative loss between passes — lossless format guarantees every modification is bit-perfect relative to the previous state. Streaming platform distribution: Spotify accepts WAV/FLAC at upload but re-encodes to Ogg Vorbis 320 kbps for Premium and AAC 128 kbps for free users; Apple Music uses AAC 256 kbps VBR; Tidal and Qobuz distribute FLAC for audiophiles. Professional podcasting: Apple Podcasts recommends AAC 128 kbps mono for voice (optimal efficiency); Spotify accepts MP3 192 kbps; master files are saved in WAV for future remasters. Audiobooks: Audible uses AAX (DRM variant of AAC) with variable bitrate by quality. Video games: many engines (Unity, Unreal) prefer OGG Vorbis for its royalty-free status and perfect looping support.

Best Practices and Professional Tips

Archival workflow: always keep a master FLAC or WAV of every important audio file. Lossy formats (MP3, AAC, OGG) are derivatives — you can always generate MP3 from FLAC, but never recover quality lost in the reverse direction. Bitrate selection: for music, MP3 320 kbps or AAC 256 kbps are indistinguishable from lossless on casual equipment; for podcast 96-128 kbps mono is enough. Sample rate: 44.1 kHz is the CD standard for music; 48 kHz for video; 96/192 kHz only adds value in professional production with studio monitoring. Metadata and cover art: always use ID3v2.4 with UTF-8 to ensure special characters (accents, ñ, Chinese symbols) display correctly in any player. Embed cover art at 600×600 px JPEG progressive — universally recognized standard format.

Compatibility and Technical Considerations

KaijuConverter processes audio files on dedicated cloud infrastructure with FFmpeg 6.x as the main engine and the best available codec libraries (LAME 3.100 for MP3, libfdk-aac for AAC, FLAC reference encoder, libopus for Opus). Processing is streaming — we don't wait to receive the full file before starting to process, significantly reducing perceived latency. We support files up to 200 MB with sample rates from 8 kHz to 192 kHz, depths of 16/24/32-bit (including float for professional editing), and configurations from mono to surround 7.1 when the destination format allows. Privacy: files are processed in isolated Docker containers, encrypted in transit with TLS 1.3, and automatically deleted with multi-pass overwrite after 2 hours. DRM: protected files (Apple Music DRM, Audible AAX) cannot be converted — DRM blocks access to the original stream. For those cases, you must obtain a legal DRM-free version before conversion.

Related conversions

Common video conversions that pair well with this guide:

Extract Audio from Video with FFmpeg: MP3, AAC, FLAC, and WAV