如何使用ffmpeg将wav文件转换为AIF文件？
我需要制作16位中的一个，一个24位，一个32位。
我还需要制作不同的采样率。例如，一个在176,400 kHz，一个在44,100 kHz。
我知道ffmpeg -i input-file.wav output-file.aif会转换文件，但我不确定其余的文件。
https://www.ffmpeg.org/general.html#Audio-Codecs说ffmpeg支持AIFF，但没有关于AIFF编码的文档：https://en.wikipedia.org/wiki/Audio_Interchange_File_Format#AIFF-C_common_compression_types
答案 0 :(得分：1)
查看libavformat/aiff.h显示：
static const AVCodecTag ff_codec_aiff_tags[] = {
{ AV_CODEC_ID_PCM_S16BE, MKTAG('N','O','N','E') },
{ AV_CODEC_ID_PCM_S8, MKTAG('N','O','N','E') },
{ AV_CODEC_ID_PCM_U8, MKTAG('r','a','w',' ') },
{ AV_CODEC_ID_PCM_S24BE, MKTAG('N','O','N','E') },
{ AV_CODEC_ID_PCM_S32BE, MKTAG('N','O','N','E') },
{ AV_CODEC_ID_PCM_F32BE, MKTAG('f','l','3','2') },
{ AV_CODEC_ID_PCM_F64BE, MKTAG('f','l','6','4') },
{ AV_CODEC_ID_PCM_ALAW, MKTAG('a','l','a','w') },
{ AV_CODEC_ID_PCM_MULAW, MKTAG('u','l','a','w') },
{ AV_CODEC_ID_PCM_S24BE, MKTAG('i','n','2','4') },
{ AV_CODEC_ID_PCM_S32BE, MKTAG('i','n','3','2') },
{ AV_CODEC_ID_MACE3, MKTAG('M','A','C','3') },
{ AV_CODEC_ID_MACE6, MKTAG('M','A','C','6') },
{ AV_CODEC_ID_GSM, MKTAG('G','S','M',' ') },
{ AV_CODEC_ID_ADPCM_G722, MKTAG('G','7','2','2') },
{ AV_CODEC_ID_ADPCM_G726LE, MKTAG('G','7','2','6') },
{ AV_CODEC_ID_PCM_S16BE, MKTAG('t','w','o','s') },
{ AV_CODEC_ID_PCM_S16LE, MKTAG('s','o','w','t') },
{ AV_CODEC_ID_ADPCM_IMA_QT, MKTAG('i','m','a','4') },
{ AV_CODEC_ID_QDMC, MKTAG('Q','D','M','C') },
{ AV_CODEC_ID_QDM2, MKTAG('Q','D','M','2') },
{ AV_CODEC_ID_QCELP, MKTAG('Q','c','l','p') },
{ AV_CODEC_ID_SDX2_DPCM, MKTAG('S','D','X','2') },
{ AV_CODEC_ID_ADPCM_IMA_WS, MKTAG('A','D','P','4') },
{ AV_CODEC_ID_NONE, 0 },
};
其中一些仅在FFmpeg中具有解码支持。请参阅ffmpeg -codecs或ffmpeg -encoders或您提供的第一个链接。
答案 1 :(得分：1)
您可以使用-ar选项来设置采样率，并使用-sample_fmt来“设置位深”。
如果您不指定 -acodec，它将默认为多路复用器的默认音频编解码器，对于 pcm_s16be 文件为 aiff，并且该编解码器不支持您的所有位深度想要。
# for each wav file in the directory, create an aiff file permutation across each bit depth and sample rate.
for FILE in *.wav; do
# bit depths
for BIT_DEPTH in 16 24 32; do
# sampling rates
for SAMPLING_RATE in 44100 176400; do
# sample_fmt is determined by the codec, which is determined by the muxer.
# So check the muxer used for the file you're trying to convert to:
#
# > ffmpeg -h muxer=aiff
# Muxer aiff [Audio IFF]:
# Common extensions: aif,aiff,afc,aifc.
# Mime type: audio/aiff.
# Default video codec: png.
# Default audio codec: pcm_s16be.
#
# Now check for the specific encoder names for that codec:
#
# > ffmpeg -codecs | grep -P 'pcm_s\d\dbe'
# DEAI.S pcm_s16be PCM signed 16-bit big-endian
# DEAI.S pcm_s16be_planar PCM signed 16-bit big-endian planar
# DEAI.S pcm_s24be PCM signed 24-bit big-endian
# DEAI.S pcm_s32be PCM signed 32-bit big-endian
# DEAI.S pcm_s64be PCM signed 64-bit big-endian
# and make sure you use only the appropriate arguments for each.
SAMPLE_FMT="s${BIT_DEPTH}"
if [[ "$BIT_DEPTH" -eq "24" ]]; then
# pcm_s24be codec only supports s32 sample_fmt (check with `ffmpeg -h encoder=pcm_s24be`)
SAMPLE_FMT="s32"
fi
ffmpeg -i "$FILE" -ar "$SAMPLING_RATE" -sample_fmt "$SAMPLE_FMT" -y -acodec "pcm_s${BIT_DEPTH}be" "${FILE%.wav}.$BIT_DEPTH.$SAMPLING_RATE.aiff"
done
done
done
然后，您可以从使用 find *.aiff -exec ffprobe -v error -select_streams a:0 -show_entries stream {} \; 创建的 aiff 文件中获取详细信息，以确保它们具有正确的采样率和位深度。
└─[$] <> ffmpeg -h
ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
built with Apple clang version 12.0.0 (clang-1200.0.32.29)
configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1_9 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox
libavutil 56. 51.100 / 56. 51.100
libavcodec 58. 91.100 / 58. 91.100
libavformat 58. 45.100 / 58. 45.100
libavdevice 58. 10.100 / 58. 10.100
libavfilter 7. 85.100 / 7. 85.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 7.100 / 5. 7.100
libswresample 3. 7.100 / 3. 7.100
libpostproc 55. 7.100 / 55. 7.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...
Getting help:
-h -- print basic options
-h long -- print more options
-h full -- print all options (including all format and codec specific options, very long)
-h type=name -- print all options for the named decoder/encoder/demuxer/muxer/filter/bsf/protocol
See man ffmpeg for detailed description of the options.
Print help / information / capabilities:
-L show license
-h topic show help
-? topic show help
-help topic show help
--help topic show help
-version show version
-buildconf show build configuration
-formats show available formats
-muxers show available muxers
-demuxers show available demuxers
-devices show available devices
-codecs show available codecs
-decoders show available decoders
-encoders show available encoders
-bsfs show available bit stream filters
-protocols show available protocols
-filters show available filters
-pix_fmts show available pixel formats
-layouts show standard channel layouts
-sample_fmts show available audio sample formats
-colors show available color names
-sources device list sources of the input device
-sinks device list sinks of the output device
-hwaccels show available HW acceleration methods
Global options (affect whole program instead of just one file):
-loglevel loglevel set logging level
-v loglevel set logging level
-report generate a report
-max_alloc bytes set maximum size of a single allocated block
-y overwrite output files
-n never overwrite output files
-ignore_unknown Ignore unknown stream types
-filter_threads number of non-complex filter threads
-filter_complex_threads number of threads for -filter_complex
-stats print progress report during encoding
-max_error_rate maximum error rate ratio of errors (0.0: no errors, 1.0: 100% errors) above which ffmpeg returns an error instead of success.
-bits_per_raw_sample number set the number of bits per raw sample
-vol volume change audio volume (256=normal)
Per-file main options:
-f fmt force format
-c codec codec name
-codec codec codec name
-pre preset preset name
-map_metadata outfile[,metadata]:infile[,metadata] set metadata information of outfile from infile
-t duration record or transcode "duration" seconds of audio/video
-to time_stop record or transcode stop time
-fs limit_size set the limit file size in bytes
-ss time_off set the start time offset
-sseof time_off set the start time offset relative to EOF
-seek_timestamp enable/disable seeking by timestamp with -ss
-timestamp time set the recording timestamp ('now' to set the current time)
-metadata string=string add metadata
-program title=string:st=number... add program with specified streams
-target type specify target file type ("vcd", "svcd", "dvd", "dv" or "dv50" with optional prefixes "pal-", "ntsc-" or "film-")
-apad audio pad
-frames number set the number of frames to output
-filter filter_graph set stream filtergraph
-filter_script filename read stream filtergraph description from a file
-reinit_filter reinit filtergraph on input parameter changes
-discard discard
-disposition disposition
Video options:
-vframes number set the number of video frames to output
-r rate set frame rate (Hz value, fraction or abbreviation)
-s size set frame size (WxH or abbreviation)
-aspect aspect set aspect ratio (4:3, 16:9 or 1.3333, 1.7777)
-bits_per_raw_sample number set the number of bits per raw sample
-vn disable video
-vcodec codec force video codec ('copy' to copy stream)
-timecode hh:mm:ss[:;.]ff set initial TimeCode value.
-pass n select the pass number (1 to 3)
-vf filter_graph set video filters
-ab bitrate audio bitrate (please use -b:a)
-b bitrate video bitrate (please use -b:v)
-dn disable data
Audio options:
-aframes number set the number of audio frames to output
-aq quality set audio quality (codec-specific)
-ar rate set audio sampling rate (in Hz)
-ac channels set number of audio channels
-an disable audio
-acodec codec force audio codec ('copy' to copy stream)
-vol volume change audio volume (256=normal)
-af filter_graph set audio filters
Subtitle options:
-s size set frame size (WxH or abbreviation)
-sn disable subtitle
-scodec codec force subtitle codec ('copy' to copy stream)
-stag fourcc/tag force subtitle tag/fourcc
-fix_sub_duration fix subtitles duration
-canvas_size size set canvas size (WxH or abbreviation)
-spre preset set the subtitle options to the indicated preset