lib/gems/baw-audio-tools/lib/baw_audio_tools/audio_ffmpeg.rb
# frozen_string_literal: true
module BawAudioTools
# Used to manipulate the ffmpeg command line tool
class AudioFfmpeg
ERROR_FRAME_SIZE_1 = 'Could not find codec parameters for stream [0-9]+ \\(Audio\\: [a-zA-Z0-9]+\\, [0-9]+ channels\\, [a-zA-Z0-9]+\\)\\: unspecified frame size'
ERROR_FRAME_SIZE_2 = 'Failed to read frame size: Could not seek to [0-9]+\\.'
ERROR_END_OF_FILE = 'End of file'
# @param [String] ffmpeg_executable
# @param [String] ffprobe_executable
# @param [Logger] logger
# @param [String] temp_dir
def initialize(ffmpeg_executable, ffprobe_executable, logger, temp_dir)
@ffmpeg_executable = ffmpeg_executable
@ffprobe_executable = ffprobe_executable
@logger = logger
@temp_dir = temp_dir
@class_name = self.class.name
end
def info_command(source)
"#{@ffprobe_executable} -hide_banner -sexagesimal -print_format default -show_error -show_streams -show_format \"#{source}\""
end
def integrity_command(source)
"#{@ffmpeg_executable} -hide_banner -loglevel repeat+verbose -nostdin -i \"#{source}\" -codec copy -f null -"
end
def modify_command(source, source_info, target, start_offset = nil, end_offset = nil, channel = nil, sample_rate = nil)
raise Exceptions::FileNotFoundError, "Source does not exist: #{source}" unless File.exist? source
raise Exceptions::FileAlreadyExistsError, "Target exists: #{target}" if File.exist? target
raise ArgumentError, "Source and Target are the same file: #{target}" if source == target
cmd_offsets = arg_offsets(start_offset, end_offset)
cmd_sample_rate = if sample_rate.blank? && source_info.include?(:sample_rate)
arg_sample_rate(source_info[:sample_rate])
else
arg_sample_rate(sample_rate)
end
cmd_channel = arg_channel(channel)
codec_info = codec_calc(target)
# AT 2020: added -nostdin and -y to force ffmpeg to complete in race conditions, and to never look for console input.
# Both are speculative fixes. Races shouldn't happen but we think there's a bug with the unique job system.
# AT 2020: moved command offsets so that they occur before -i. This apparently changes performance profile for cut+seek
# from O(n) to O(1), which makes a big difference on longer files!
# For a 2 hour file cut speed drop to < 3 seconds, from 8.25 seconds.
# See: https://github.com/QutEcoacoustics/baw-server/pull/498
audio_cmd = "#{@ffmpeg_executable} -hide_banner #{cmd_offsets} -i \"#{source}\" #{cmd_sample_rate} #{cmd_channel} #{codec_info[:codec]} \"#{codec_info[:target]}\" -nostdin -y"
cmd = ''
if codec_info[:target] == codec_info[:old_target]
cmd = audio_cmd
else
partial_cmd = "\"#{codec_info[:target]}\" \"#{codec_info[:old_target]}\""
separator_move = '; mv'
cmd = "#{audio_cmd} #{separator_move} #{partial_cmd}"
end
cmd
end
def check_for_errors(execute_msg)
#stdout = execute_msg[:stdout]
stderr = execute_msg[:stderr]
if !stderr.blank? && /#{ERROR_FRAME_SIZE_1}/.match(stderr)
raise Exceptions::FileCorruptError,
"Ffmpeg could not get frame size (msg type 1).\n\t#{execute_msg[:execute_msg]}"
end
if !stderr.blank? && /#{ERROR_FRAME_SIZE_2}/.match(stderr)
raise Exceptions::FileCorruptError,
"Ffmpeg could not get frame size (msg type 2).\n\t#{execute_msg[:execute_msg]}"
end
if !stderr.blank? && /#{ERROR_END_OF_FILE}/i.match(stderr)
raise Exceptions::FileCorruptError, "Ffmpeg encountered unexpected end of file.\n\t#{execute_msg[:execute_msg]}"
end
end
def check_integrity_output(execute_msg)
#stdout = execute_msg[:stdout]
stderr = execute_msg[:stderr]
result = {
errors: [],
warnings: [],
info: {
read: {
packets: 0, bytes: 0, frames: 0, samples: 0
},
write: {
packets: 0, bytes: 0, frames: 0, samples: 0
}
}
}
return result if stderr.blank?
stderr.each_line do |line|
match_result = /\A\[(.+?) @ 0x.+?\] (.*)/.match(line)
item = nil
item = check_integrity_item(id: match_result[1], description: match_result[2]) unless match_result.blank?
result[:warnings].push(item) unless item.blank?
if /: End of file$/ =~ line
result[:warnings].push(id: 'end of file', description: 'End of file')
end
read_packets_match = /Total: (\d+) packets \((\d+) bytes\) demuxed/.match(line)
unless read_packets_match.blank?
result[:info][:read][:packets] = read_packets_match[1]
result[:info][:read][:bytes] = read_packets_match[2]
end
read_frames_match = /(\d+) frames decoded \((\d+) samples\);/.match(line)
unless read_frames_match.blank?
result[:info][:read][:frames] = read_frames_match[1]
result[:info][:read][:samples] = read_frames_match[2]
end
write_packets_match = /Total: (\d+) packets \((\d+) bytes\) muxed/.match(line)
unless write_packets_match.blank?
result[:info][:write][:packets] = write_packets_match[1]
result[:info][:write][:bytes] = write_packets_match[2]
end
write_frames_match = /(\d+) frames encoded \((\d+) samples\);/.match(line)
unless write_frames_match.blank?
result[:info][:write][:frames] = write_frames_match[1]
result[:info][:write][:samples] = write_frames_match[2]
end
end
result
end
def check_integrity_item(hash)
return nil if hash.blank?
return nil if hash[:id].starts_with?('graph')
return nil if hash[:id].starts_with?('audio format')
return nil if hash[:id].starts_with?('auto-inserted')
return nil if hash[:description].starts_with?('parser not found for')
return nil if hash[:description].starts_with?('max_analyze_duration')
if hash[:description].starts_with?('Application provided invalid, non monotonically increasing dts to muxer in stream')
return nil
end
hash
end
def find_remove_warning(mod_stderr, match_regex)
match_info = mod_stderr.match(match_regex)
mod_stderr = mod_stderr.gsub(match_regex, '')
unless match_info.blank?
@logger.warn(@class_name) {
"Found and removed '#{match_info}' in ffmpeg output."
}
end
mod_stderr
end
def has_regex?(string, regex)
!!(string =~ regex)
end
# returns the duration in seconds (and fractions if present)
def parse_duration(duration_string)
duration_match = /(?<hour>\d+):(?<minute>\d+):(?<second>[\d+.]+)/i.match(duration_string)
duration = 0
if !duration_match.nil? && duration_match.size == 4
duration = (duration_match[:hour].to_f * 60 * 60) + (duration_match[:minute].to_f * 60) + duration_match[:second].to_f
end
duration
end
def parse_ffprobe_output(source, execute_msg)
# ffprobe std err contains info (separate on first equals(=))
result = {}
ffprobe_current_block_name = ''
execute_msg[:stdout].strip.split(/\r?\n|\r/).each do |line|
line.strip!
if line[0] == '['
# this chomp reverse stuff is due to the lack of a proper 'trim'
ffprobe_current_block_name = line.chomp(']').reverse.chomp('[').reverse
else
# some lines won't output key=value data, especially when FLAC metadata
# is output as `\t\tkey\t:\tvalue` (where \t denotes a tab)
# When encountering such lines, just skip for now.
index = line.index('=')
unless index.nil?
current_key = line[0, index].strip
current_value = line[index + 1, line.length].strip
result["#{ffprobe_current_block_name} #{current_key}"] = current_value
end
end
end
unless File.exist?(source)
raise Exceptions::AudioFileNotFoundError, "Could not locate #{source}\n\t#{execute_msg[:execute_msg]}"
end
actual_stream_codec_type = result['STREAM codec_type']
expected_stream_codec_type = 'audio'
if actual_stream_codec_type != expected_stream_codec_type
msg = "Not an audio file #{source} ('#{actual_stream_codec_type}' is not '#{expected_stream_codec_type}'): #{result.to_json}\n\t#{execute_msg[:execute_msg]}"
raise Exceptions::NotAnAudioFileError, msg
end
result
end
def arg_channel(channel)
cmd_arg = ''
unless channel.blank?
channel_number = channel.to_i
cmd_arg = if channel_number < 1
# mix down to mono
' -ac 1 '
else
# select the channel (0 index based)
" -map_channel 0.0.#{channel_number - 1} "
end
end
cmd_arg
end
def arg_sample_rate(sample_rate)
cmd_arg = ''
unless sample_rate.blank?
# -ar Set the audio sampling frequency (default = 44100 Hz).
# -ab Set the audio bitrate in bit/s (default = 64k).
cmd_arg = " -ar #{sample_rate} "
end
cmd_arg
end
def arg_offsets(start_offset, end_offset)
cmd_arg = ''
# start offset
# -ss Seek to given time position in seconds. hh:mm:ss[.xxx] syntax is also supported.
unless start_offset.blank?
start_offset_formatted = Time.at(start_offset.to_f).utc.strftime('%H:%M:%S.%3N')
cmd_arg = " -ss #{start_offset_formatted}"
end
# end offset
# -t Restrict the transcoded/captured video sequence to the duration specified in seconds. hh:mm:ss[.xxx] syntax is also supported.
unless end_offset.blank?
#end_offset_formatted = Time.at(modify_parameters[:end_offset]).utc.strftime('%H:%M:%S.%3N')
end_offset_raw = end_offset.to_f
end_offset_time = Time.at(end_offset_raw).utc
if start_offset.blank?
# if start offset was not included, include audio from the start of the file.
cmd_arg += " -t #{end_offset_time.strftime('%H:%M:%S.%3N')}"
else
start_offset_raw = start_offset.to_f
#start_offset_time = Time.at(start_offset_raw).utc
cmd_arg += " -t #{Time.at(end_offset_raw - start_offset_raw).utc.strftime('%H:%M:%S.%3N')}"
end
end
cmd_arg
end
def codec_calc(target)
# high quality codec settings
# https://trac.ffmpeg.org/wiki/GuidelinesHighQualityAudio
# http://trac.ffmpeg.org/wiki/TheoraVorbisEncodingGuide
# http://en.wikipedia.org/wiki/Vorbis#Technical_details
# http://wiki.hydrogenaudio.org/index.php?title=Recommended_Ogg_Vorbis#Recommended_Encoder_Settings
# -aq 6 will be approx 192kbit/s
codec_high_vorbis = 'libvorbis -aq 6'
# pcm signed 16-bit little endian - compatible with CDDA
codec_high_wav = 'pcm_s16le'
# http://lame.cvs.sourceforge.net/viewvc/lame/lame/USAGE
# 0 = slowest algorithms, but potentially highest quality
# 9 = faster algorithms, very poor quality
# http://trac.ffmpeg.org/wiki/Encoding%20VBR%20(Variable%20Bit%20Rate)%20mp3%20audio
# -aq 2 recommended, but still cuts off at 10khz at 22.05khz
# using CBR: -b:a 192k
codec_high_mp3 = 'libmp3lame -aq 0'
codec_high_wavpack = 'wavpack'
# The flac reference implementation may not keep the .wav metadata
# https://xiph.org/flac/faq.html#general__no_wave_metadata
# The flag --keep-foreign-metadata can be included to store the WAVE non-audio data as well
# Note that when a flac file is decoded to WAVE/AIFF the flag must be used again to keep metadata
# The ffmpeg implementation of flac encoding / decoding can handle some metadata formats, this is sufficient until
# there is something concrete that requires using additional ffmpeg flags or using the reference flac implementation
# instead.
# The relevant ffmpeg flag is `-map_metadata`. "By default, global metadata is copied from the first input file,
# per-stream and per-chapter metadata is copied along with streams/chapters."
codec_high_flac = 'flac'
# output file. extension used to determine filetype.
old_target = target
# set the right codec if we know it
extension = File.extname(target).upcase!.reverse.chomp('.').reverse
case extension
when 'WAV'
codec = codec_high_wav
when 'MP3'
codec = codec_high_mp3
when 'OGG'
codec = codec_high_vorbis
when 'OGA'
codec = codec_high_vorbis
target = "#{target.chomp(File.extname(target))}.ogg"
when 'WEBM'
codec = codec_high_vorbis
when 'WEBMA'
codec = codec_high_vorbis
target = "#{target.chomp(File.extname(target))}.webm"
when 'WV'
codec = codec_high_wavpack
when 'FLAC'
codec = codec_high_flac
else
# don't specify codec for any other extension
# Alternative: Use the 'copy' special value to specify that the raw codec data must be copied as is.
codec = ''
end
# -acodec Force audio codec to codec.
{
codec: codec.blank? ? '' : " -acodec #{codec}",
target: target,
old_target: old_target
}
end
# mime type to ffmpeg string identifier conversions
def get_mime_type(ffmpeg_info)
#[:info][:ffmpeg]['STREAM codec_type']+'/'+file_info[:info][:ffmpeg]['STREAM codec_name']
case ffmpeg_info['FORMAT format_long_name']
when 'WAV / WAVE (Waveform Audio)'
# :codec_name => 'pcm_s16le',
# :codec_long_name => 'PCM signed 16-bit little-endian',
'audio/wav'
when 'MP2/3 (MPEG audio layer 2/3)', 'MP3 (MPEG audio layer 3)'
# :codec_name => 'mp3',
# :codec_long_name => 'MP3 (MPEG audio layer 3)',
'audio/mp3'
when 'Matroska / WebM', 'WebM'
# :codec_name => 'vorbis',
# :codec_long_name => 'Vorbis',
# :format_name => 'matroska,webm',
'audio/webm'
when 'Ogg'
# :codec_name => 'vorbis',
# :codec_long_name => 'Vorbis',
'audio/ogg'
when 'ASF (Advanced / Active Streaming Format)'
# :codec_name => 'wmav2',
# :codec_long_name => 'Windows Media Audio 2',
'audio/asf'
when 'WavPack', 'raw WavPack'
# :codec_name => 'wavpack',
# :codec_long_name => 'WavPack',
'audio/wavpack'
when 'QuickTime / MOV', 'MP4 (MPEG-4 Part 14)', 'PSP MP4 (MPEG-4 Part 14)', 'iPod H.264 MP4 (MPEG-4 Part 14)'
# codec_name=alac
# codec_long_name=ALAC (Apple Lossless Audio Codec)
# format_name=mov,mp4,m4a,3gp,3g2,mj2
'audio/mp4'
when 'AAC (Advanced Audio Coding)', 'AAC LATM (Advanced Audio Coding LATM syntax)',
'ADTS AAC (Advanced Audio Coding)', 'raw ADTS AAC (Advanced Audio Coding)'
# codec_name=aac
# codec_long_name=AAC (Advanced Audio Coding)
# format_name=aac
'audio/aac'
when 'FLAC (Free Lossless Audio Codec)', 'raw FLAC', 'flac'
# codec_name=flac
# codec_long_name=raw FLAC
# format_name=flac
'audio/x-flac'
else
'application/octet-stream'
end
end
end
end