- The machine I used for encoding the test files is a PII-333 with 256 megs of RAM, running Windows NT 4.0 SP5. Unfortunately, Windows is the lowest-common-denominator platform for all the encoders. Apple Lossless was tested on a newer machine (P4-2.4GHz Windows 2000); only the overall encoding and decoding times are shown, and the times are scaled to the PII-333 by multiplying by the ratio of flac times on the PII to P4.
-
- The input corpus currently consists entirely of CD music tracks. In the future it may include more kinds of input (like speech, other sample rates/resolutions, etc). There are 14 tracks whose genres range from rock to pop to death metal to classical to chant.
-
- The first table is a summary of results on all input tracks. The remaining tables show the results of the encoders on each track. The summary table has more modes, whereas the individual tables have just the interesting ones.
-
- In the summary table, entries are sorted by average compression ratio, which is the average of the ratios for each track; this keeps long tracks from having more influence than short ones. In the individual tables, this is the same as the straight compression ratio, which is compressed size / uncompressed size.
-
+ The machine I used for encoding the test files is a PII-333 with 256 megs of RAM, running Windows NT 4.0 SP5. Unfortunately, Windows is the lowest-common-denominator platform for all the encoders. Apple Lossless was tested on a newer machine (P4-2.4GHz Windows 2000); only the overall encoding and decoding times are shown, and the times are scaled to the PII-333 by multiplying by the ratio of flac times on the PII to P4.
+
+ The input corpus currently consists entirely of CD music tracks. In the future it may include more kinds of input (like speech, other sample rates/resolutions, etc). There are 14 tracks whose genres range from rock to pop to death metal to classical to chant.
+
+ The first table is a summary of results on all input tracks. The remaining tables show the results of the encoders on each track. The summary table has more modes, whereas the individual tables have just the interesting ones.
+
+ In the summary table, entries are sorted by average compression ratio, which is the average of the ratios for each track; this keeps long tracks from having more influence than short ones. In the individual tables, this is the same as the straight compression ratio, which is compressed size / uncompressed size.
+
Some interesting things to note:
- flac -5 is right in the middle with respect to compression, relatively fast on the encoding range, and one of the fastest decoding. This is about what you would expect; FLAC is designed to put most of the processing on the encoding side, which is only done once, whereas the adaptive codecs take as long to decode as encode. FLAC is more suited in this way for playback on low-power devices and is one of the reasons it is the only lossless codec with any kind of hardware support.
@@ -440,8 +440,8 @@
- RKAU also has a tendency to get bigger in the 'high' mode.
- Another ironic fact is that the encoders that are patented or cost money turn out to be the worst by most measures. SPS is so archane and crippled that I gave up trying to put together results for it after one track.
- This is a summary table with just the most 'economic' modes (the ones that give the most compression for the least amount of encode/decode time) for each codec.
-
+ This is a summary table with just the most 'economic' modes (the ones that give the most compression for the least amount of encode/decode time) for each codec.
+
diff --git a/doc/html/developers.html b/doc/html/developers.html
index 5d2b7ab3..1734c58b 100644
--- a/doc/html/developers.html
+++ b/doc/html/developers.html
@@ -53,8 +53,8 @@
- FLAC is an open source project and we are happy to enlist the help of anyone who wants to contribute. The preferred method for transmitting improvements is patch files (in "diff -c" format) sent to the developer mailing list, but zipped up sources are OK. Make sure to read the FLAC goals first; there are some thing the we don't want added to FLAC, like copy protection and lossy compression.
-
+ FLAC is an open source project and we are happy to enlist the help of anyone who wants to contribute. The preferred method for transmitting improvements is patch files (in "diff -c" format) sent to the developer mailing list, but zipped up sources are OK. Make sure to read the FLAC goals first; there are some thing the we don't want added to FLAC, like copy protection and lossy compression.
+
High priority items are:
-
diff --git a/doc/html/documentation.html b/doc/html/documentation.html
index 2d022c3e..eb7b5955 100644
--- a/doc/html/documentation.html
+++ b/doc/html/documentation.html
@@ -77,8 +77,8 @@
- flac has been tuned so that the default options yield a good speed vs. compression tradeoff for many kinds of input. However, if you are looking to maximize the compression rate or speed, or want to use the full power of FLAC's metadata system, this section is for you. If not, just skip to the next section.
-
+ flac has been tuned so that the default options yield a good speed vs. compression tradeoff for many kinds of input. However, if you are looking to maximize the compression rate or speed, or want to use the full power of FLAC's metadata system, this section is for you. If not, just skip to the next section.
+
The basic structure of a FLAC stream is:
- The four byte string "fLaC"
@@ -86,50 +86,50 @@
- Zero or more other metadata blocks
- One or more audio frames
- The first four bytes are to identify the FLAC stream. The metadata that follows contains all the information about the stream except for the audio data itself. After the metadata comes the encoded audio data.
-
- METADATA
-
- FLAC defines several types of metadata blocks (see the format page for the complete list). Metadata blocks can be any length and new ones can be defined. A decoder is allowed to skip any metadata types it does not understand. Only one is mandatory: the STREAMINFO block. This block has information like the sample rate, number of channels, etc., and data that can help the decoder manage its buffers, like the minimum and maximum data rate and minimum and maximum block size. Also included in the STREAMINFO block is the MD5 signature of the unencoded audio data. This is useful for checking an entire stream for transmission errors.
-
- Other blocks allow for padding, seek tables, tags, cuesheets, and application-specific data. You can see flac options below for adding PADDING blocks or specifying seek points. FLAC does not require seek points for seeking but they can speed up seeks, or be used for cueing in editing applications.
-
- Also, if you have a need of a custom metadata block, you can define your own and request an ID here. Then you can reserve a PADDING block of the correct size when encoding, and overwrite the padding block with your APPLICATION block after encoding. The resulting stream will be FLAC compatible; decoders that are aware of your metadata can use it and the rest will safely ignore it.
-
- AUDIO DATA
-
- After the metadata comes the encoded audio data. Audio data and metadata are not interleaved. Like most audio codecs, FLAC splits the unencoded audio data into blocks, and encodes each block separately. The encoded block is packed into a frame and appended to the stream. The reference encoder uses a single block size for the whole stream but the FLAC format does not require it.
-
- BLOCKING
-
- The block size is an important parameter to encoding. If it is too small, the frame overhead will lower the compression. If it is too large, the modeling stage of the compressor will not be able to generate an efficient model. Understanding FLAC's modeling will help you to improve compression for some kinds of input by varying the block size. In the most general case, using linear prediction on 44.1kHz audio, the optimal block size will be between 2-6 ksamples. flac defaults to a block size of 4608 in this case. Using the fast fixed predictors, a smaller block size is usually preferable because of the smaller frame header.
-
- INTER-CHANNEL DECORRELATION
-
- In the case of stereo input, once the data is blocked it is optionally passed through an inter-channel decorrelation stage. The left and right channels are converted to center and side channels through the following transformation: mid = (left + right) / 2, side = left - right. This is a lossless process, unlike joint stereo. For normal CD audio this can result in significant extra compression. flac has two options for this: -m always compresses both the left-right and mid-side versions of the block and takes the smallest frame, and -M, which adaptively switches between left-right and mid-side.
-
- MODELING
-
- In the next stage, the encoder tries to approximate the signal with a function in such a way that when the approximation is subracted, the result (called the residual, residue, or error) requires fewer bits-per-sample to encode. The function's parameters also have to be transmitted so they should not be so complex as to eat up the savings. FLAC has two methods of forming approximations: 1) fitting a simple polynomial to the signal; and 2) general linear predictive coding (LPC). I will not go into the details here, only some generalities that involve the encoding options.
-
- First, fixed polynomial prediction (specified with -l 0) is much faster, but less accurate than LPC. The higher the maximum LPC order, the slower, but more accurate, the model will be. However, there are diminishing returns with increasing orders. Also, at some point (usually around order 9) the part of the encoder that guesses what is the best order to use will start to get it wrong and the compression will actually decrease slightly; at that point you will have to you will have to use the exhaustive search option -e to overcome this, which is significantly slower.
-
- Second, the parameters for the fixed predictors can be transmitted in 3 bits whereas the parameters for the LPC model depend on the bits-per-sample and LPC order. This means the frame header length varies depending on the method and order you choose and can affect the optimal block size.
-
- RESIDUAL CODING
-
- Once the model is generated, the encoder subracts the approximation from the original signal to get the residual (error) signal. The error signal is then losslessly coded. To do this, FLAC takes advantage of the fact that the error signal generally has a Laplacian (two-sided geometric) distribution, and that there are a set of special Huffman codes called Rice codes that can be used to efficiently encode these kind of signals quickly and without needing a dictionary.
-
- Rice coding involves finding a single parameter that matches a signal's distribution, then using that parameter to generate the codes. As the distribution changes, the optimal parameter changes, so FLAC supports a method that allows the parameter to change as needed. The residual can be broken into several contexts or partitions, each with it's own Rice parameter. flac allows you to specify how the partitioning is done with the -r option. The residual can be broken into 2^ n partitions, by using the option -r n,n. The parameter n is called the partition order. Furthermore, the encoder can be made to search through m to n partition orders, taking the best one, by specifying -r m,n. Generally, the choice of n does not affect encoding speed but m,n does. The larger the difference between m and n, the more time it will take the encoder to search for the best order. The block size will also affect the optimal order.
-
- FRAMING
-
- An audio frame is preceded by a frame header and trailed by a frame footer. The header starts with a sync code, and contains the minimum information necessary for a decoder to play the stream, like sample rate, bits per sample, etc. It also contains the block or sample number and an 8-bit CRC of the frame header. The sync code, frame header CRC, and block/sample number allow resynchronization and seeking even in the absence of seek points. The frame footer contains a 16-bit CRC of the entire encoded frame for error detection. If the reference decoder detects a CRC error it will generate a silent block.
-
- MISCELLANEOUS
-
- As a convenience, the reference decoder knows how to skip ID3v1 and ID3v2 tags. Note however that the FLAC specification does not require compliant implementations to support ID3 in any form and their use is discouraged.
-
+ The first four bytes are to identify the FLAC stream. The metadata that follows contains all the information about the stream except for the audio data itself. After the metadata comes the encoded audio data.
+
+ METADATA
+
+ FLAC defines several types of metadata blocks (see the format page for the complete list). Metadata blocks can be any length and new ones can be defined. A decoder is allowed to skip any metadata types it does not understand. Only one is mandatory: the STREAMINFO block. This block has information like the sample rate, number of channels, etc., and data that can help the decoder manage its buffers, like the minimum and maximum data rate and minimum and maximum block size. Also included in the STREAMINFO block is the MD5 signature of the unencoded audio data. This is useful for checking an entire stream for transmission errors.
+
+ Other blocks allow for padding, seek tables, tags, cuesheets, and application-specific data. You can see flac options below for adding PADDING blocks or specifying seek points. FLAC does not require seek points for seeking but they can speed up seeks, or be used for cueing in editing applications.
+
+ Also, if you have a need of a custom metadata block, you can define your own and request an ID here. Then you can reserve a PADDING block of the correct size when encoding, and overwrite the padding block with your APPLICATION block after encoding. The resulting stream will be FLAC compatible; decoders that are aware of your metadata can use it and the rest will safely ignore it.
+
+ AUDIO DATA
+
+ After the metadata comes the encoded audio data. Audio data and metadata are not interleaved. Like most audio codecs, FLAC splits the unencoded audio data into blocks, and encodes each block separately. The encoded block is packed into a frame and appended to the stream. The reference encoder uses a single block size for the whole stream but the FLAC format does not require it.
+
+ BLOCKING
+
+ The block size is an important parameter to encoding. If it is too small, the frame overhead will lower the compression. If it is too large, the modeling stage of the compressor will not be able to generate an efficient model. Understanding FLAC's modeling will help you to improve compression for some kinds of input by varying the block size. In the most general case, using linear prediction on 44.1kHz audio, the optimal block size will be between 2-6 ksamples. flac defaults to a block size of 4608 in this case. Using the fast fixed predictors, a smaller block size is usually preferable because of the smaller frame header.
+
+ INTER-CHANNEL DECORRELATION
+
+ In the case of stereo input, once the data is blocked it is optionally passed through an inter-channel decorrelation stage. The left and right channels are converted to center and side channels through the following transformation: mid = (left + right) / 2, side = left - right. This is a lossless process, unlike joint stereo. For normal CD audio this can result in significant extra compression. flac has two options for this: -m always compresses both the left-right and mid-side versions of the block and takes the smallest frame, and -M, which adaptively switches between left-right and mid-side.
+
+ MODELING
+
+ In the next stage, the encoder tries to approximate the signal with a function in such a way that when the approximation is subracted, the result (called the residual, residue, or error) requires fewer bits-per-sample to encode. The function's parameters also have to be transmitted so they should not be so complex as to eat up the savings. FLAC has two methods of forming approximations: 1) fitting a simple polynomial to the signal; and 2) general linear predictive coding (LPC). I will not go into the details here, only some generalities that involve the encoding options.
+
+ First, fixed polynomial prediction (specified with -l 0) is much faster, but less accurate than LPC. The higher the maximum LPC order, the slower, but more accurate, the model will be. However, there are diminishing returns with increasing orders. Also, at some point (usually around order 9) the part of the encoder that guesses what is the best order to use will start to get it wrong and the compression will actually decrease slightly; at that point you will have to you will have to use the exhaustive search option -e to overcome this, which is significantly slower.
+
+ Second, the parameters for the fixed predictors can be transmitted in 3 bits whereas the parameters for the LPC model depend on the bits-per-sample and LPC order. This means the frame header length varies depending on the method and order you choose and can affect the optimal block size.
+
+ RESIDUAL CODING
+
+ Once the model is generated, the encoder subracts the approximation from the original signal to get the residual (error) signal. The error signal is then losslessly coded. To do this, FLAC takes advantage of the fact that the error signal generally has a Laplacian (two-sided geometric) distribution, and that there are a set of special Huffman codes called Rice codes that can be used to efficiently encode these kind of signals quickly and without needing a dictionary.
+
+ Rice coding involves finding a single parameter that matches a signal's distribution, then using that parameter to generate the codes. As the distribution changes, the optimal parameter changes, so FLAC supports a method that allows the parameter to change as needed. The residual can be broken into several contexts or partitions, each with it's own Rice parameter. flac allows you to specify how the partitioning is done with the -r option. The residual can be broken into 2^ n partitions, by using the option -r n,n. The parameter n is called the partition order. Furthermore, the encoder can be made to search through m to n partition orders, taking the best one, by specifying -r m,n. Generally, the choice of n does not affect encoding speed but m,n does. The larger the difference between m and n, the more time it will take the encoder to search for the best order. The block size will also affect the optimal order.
+
+ FRAMING
+
+ An audio frame is preceded by a frame header and trailed by a frame footer. The header starts with a sync code, and contains the minimum information necessary for a decoder to play the stream, like sample rate, bits per sample, etc. It also contains the block or sample number and an 8-bit CRC of the frame header. The sync code, frame header CRC, and block/sample number allow resynchronization and seeking even in the absence of seek points. The frame footer contains a 16-bit CRC of the entire encoded frame for error detection. If the reference decoder detects a CRC error it will generate a silent block.
+
+ MISCELLANEOUS
+
+ As a convenience, the reference decoder knows how to skip ID3v1 and ID3v2 tags. Note however that the FLAC specification does not require compliant implementations to support ID3 in any form and their use is discouraged.
+
flac has a verify option -V that verifies the output while encoding. With this option, a decoder is run in parallel to the encoder and its output is compared against the original input. If a difference is found flac will stop with an error.
- flac is the command-line file encoder/decoder. The encoder currently supports as input RIFF WAVE, AIFF, FLAC or Ogg FLAC format, or raw interleaved samples. The decoder currently can output to RIFF WAVE or AIFF format, or raw interleaved samples. flac only supports linear PCM samples (in other words, no A-LAW, uLAW, etc.), and the input must be between 4 and 24 bits per sample. This is not a limitation of the FLAC format, just the reference encoder/decoder.
-
- flac assumes that files ending in ".wav" or that have the RIFF WAVE header present are WAVE files, files ending in ".aif" or ".aiff" or have the AIFF header present are AIFF files, and files ending in ".flac" or have the FLAC header present are FLAC files. This assumption may be overridden with a command-line option. It also assumes that files ending in ".ogg" of have the Ogg FLAC header present are Ogg FLAC files. Other than this, flac makes no assumptions about file extensions, though the convention is that FLAC files have the extension ".flac" (or ".fla" on ancient "8.3" file systems like FAT-16).
-
- Before going into the full command-line description, a few other things help to sort it out: 1) flac encodes by default, so you must use -d to decode; 2) the options -0 .. -8 (or --fast and --best) that control the compression level actually are just synonyms for different groups of specific encoding options (described later) and you can get the same effect by using the same options; 3) flac behaves similarly to gzip in the way it handles input and output files.
-
+ flac is the command-line file encoder/decoder. The encoder currently supports as input RIFF WAVE, AIFF, FLAC or Ogg FLAC format, or raw interleaved samples. The decoder currently can output to RIFF WAVE or AIFF format, or raw interleaved samples. flac only supports linear PCM samples (in other words, no A-LAW, uLAW, etc.), and the input must be between 4 and 24 bits per sample. This is not a limitation of the FLAC format, just the reference encoder/decoder.
+
+ flac assumes that files ending in ".wav" or that have the RIFF WAVE header present are WAVE files, files ending in ".aif" or ".aiff" or have the AIFF header present are AIFF files, and files ending in ".flac" or have the FLAC header present are FLAC files. This assumption may be overridden with a command-line option. It also assumes that files ending in ".ogg" of have the Ogg FLAC header present are Ogg FLAC files. Other than this, flac makes no assumptions about file extensions, though the convention is that FLAC files have the extension ".flac" (or ".fla" on ancient "8.3" file systems like FAT-16).
+
+ Before going into the full command-line description, a few other things help to sort it out: 1) flac encodes by default, so you must use -d to decode; 2) the options -0 .. -8 (or --fast and --best) that control the compression level actually are just synonyms for different groups of specific encoding options (described later) and you can get the same effect by using the same options; 3) flac behaves similarly to gzip in the way it handles input and output files.
+
flac will be invoked one of four ways, depending on whether you are encoding, decoding, testing, or analyzing:
- In any case, if no inputfile is specified, stdin is assumed. If only one inputfile is specified, it may be "-" for stdin. When stdin is used as input, flac will write to stdout. Otherwise flac will perform the desired operation on each input file to similarly named output files (meaning for encoding, the extension will be replaced with ".flac", or appended with ".flac" if the input file has no extension, and for decoding, the extension will be ".wav" for WAVE output and ".raw" for raw output). The original file is not deleted unless --delete-input-file is specified.
-
+ In any case, if no inputfile is specified, stdin is assumed. If only one inputfile is specified, it may be "-" for stdin. When stdin is used as input, flac will write to stdout. Otherwise flac will perform the desired operation on each input file to similarly named output files (meaning for encoding, the extension will be replaced with ".flac", or appended with ".flac" if the input file has no extension, and for decoding, the extension will be ".wav" for WAVE output and ".raw" for raw output). The original file is not deleted unless --delete-input-file is specified.
+
If you are encoding/decoding from stdin to a file, you should use the -o option like so:
-
@@ -184,22 +184,22 @@
flac -d [options] > outputfile
- since the former allows flac to seek backwards to write the STREAMINFO or RIFF WAVE header contents when necessary.
-
- Also, you can force output data to go to stdout using -c.
-
+ since the former allows flac to seek backwards to write the STREAMINFO or RIFF WAVE header contents when necessary.
+
+ Also, you can force output data to go to stdout using -c.
+
To encode or decode files that start with a dash, use -- to signal the end of options, to keep the filenames themselves from being treated as options:
-
flac -V -- -01-filename.wav
- The encoding options affect the compression ratio and encoding speed. The format options are used to tell flac the arrangement of samples if the input file (or output file when decoding) is a raw file. If it is a RIFF WAVE or AIFF file the format options are not needed since they are read from the AIFF/WAVE header.
-
- In test mode, flac acts just like in decode mode, except no output file is written. Both decode and test modes detect errors in the stream, but they also detect when the MD5 signature of the decoded audio does not match the stored MD5 signature, even when the bitstream is valid.
-
- flac can also re-encode FLAC files. In other words, you can specify a FLAC or Ogg FLAC file as an input to the encoder and it will decoder it and re-encode it according to the options you specify. It will also preserve all the metadata unless you override it with other options (e.g. specifying new tags, seekpoints, cuesheet, padding, etc.).
-
+ The encoding options affect the compression ratio and encoding speed. The format options are used to tell flac the arrangement of samples if the input file (or output file when decoding) is a raw file. If it is a RIFF WAVE or AIFF file the format options are not needed since they are read from the AIFF/WAVE header.
+
+ In test mode, flac acts just like in decode mode, except no output file is written. Both decode and test modes detect errors in the stream, but they also detect when the MD5 signature of the decoded audio does not match the stored MD5 signature, even when the bitstream is valid.
+
+ flac can also re-encode FLAC files. In other words, you can specify a FLAC or Ogg FLAC file as an input to the encoder and it will decoder it and re-encode it according to the options you specify. It will also preserve all the metadata unless you override it with other options (e.g. specifying new tags, seekpoints, cuesheet, padding, etc.).
+
@@ -331,8 +331,10 @@
--skip={#|mm:ss.ss}
- Skip over the first # of samples of the input. This works for both encoding and decoding, but not testing. The alternative form mm:ss.ss can be used to specify minutes, seconds, and fractions of a second.
- Examples:
+ Skip over the first # of samples of the input. This works for both encoding and decoding, but not testing. The alternative form mm:ss.ss can be used to specify minutes, seconds, and fractions of a second.
+
+ Examples:
+
--skip=123 : skip the first 123 samples of the input
--skip=1:23.45 : skip the first 1 minute and 23.45 seconds of the input
|
@@ -343,8 +345,10 @@
--until={#|[+|-]mm:ss.ss}
- Stop at the given sample number for each input file. This works for both encoding and decoding, but not testing. The given sample number is not included in the decoded output. The alternative form mm:ss.ss can be used to specify minutes, seconds, and fractions of a second. If a + sign is at the beginning, the --until point is relative to the --skip point. If a - sign is at the beginning, the --until point is relative to end of the audio.
- Examples:
+ Stop at the given sample number for each input file. This works for both encoding and decoding, but not testing. The given sample number is not included in the decoded output. The alternative form mm:ss.ss can be used to specify minutes, seconds, and fractions of a second. If a + sign is at the beginning, the --until point is relative to the --skip point. If a - sign is at the beginning, the --until point is relative to end of the audio.
+
+ Examples:
+
--until=123 : decode only the first 123 samples of the input (samples 0-122, stopping at 123)
--until=1:23.45 : decode only the first 1 minute and 23.45 seconds of the input
--skip=1:00 --until=+1:23.45 : decode 1:00.00 to 2:23.45
@@ -358,8 +362,10 @@
--ogg
|
- When encoding, generate Ogg FLAC output instead of native FLAC. Ogg FLAC streams are FLAC streams wrapped in an Ogg transport layer. The resulting file should have an '.ogg' extension and will still be decodable by flac.
- When decoding, force the input to be treated as Ogg FLAC. This is useful when piping input from stdin or when the filename does not end in '.ogg'.
+ When encoding, generate Ogg FLAC output instead of native FLAC. Ogg FLAC streams are FLAC streams wrapped in an Ogg transport layer. The resulting file should have an '.ogg' extension and will still be decodable by flac.
+
+ When decoding, force the input to be treated as Ogg FLAC. This is useful when piping input from stdin or when the filename does not end in '.ogg'.
+
NOTE: Ogg FLAC files created prior to flac 1.1.1 used an ad-hoc mapping and do not support seeking. They should be decoded and re-encoded with flac 1.1.1 or later.
|
@@ -420,8 +426,10 @@
--cue=[#.#][-[#.#]]
- Set the beginning and ending cuepoints to decode. The optional first #.# is the track and index point at which decoding will start; the default is the beginning of the stream. The optional second #.# is the track and index point at which decoding will end; the default is the end of the stream. If the cuepoint does not exist, the closest one before it (for the start point) or after it (for the end point) will be used. If those don't exist, the start of the stream (for the start point) or end of the stream (for the end point) will be used. The cuepoints are merely translated into sample numbers then used as --skip and --until.
- Examples:
+ Set the beginning and ending cuepoints to decode. The optional first #.# is the track and index point at which decoding will start; the default is the beginning of the stream. The optional second #.# is the track and index point at which decoding will end; the default is the end of the stream. If the cuepoint does not exist, the closest one before it (for the start point) or after it (for the end point) will be used. If those don't exist, the start of the stream (for the start point) or end of the stream (for the end point) will be used. The cuepoints are merely translated into sample numbers then used as --skip and --until.
+
+ Examples:
+
--cue=- : decode the entire stream
--cue=4.1 : decode from track 4, index 1 to the end of the stream
--cue=4.1- : decode from track 4, index 1 to the end of the stream
@@ -446,13 +454,14 @@
-@@@-apply-replaygain-which-is-not-lossless[=<specification>]
|
- Applies ReplayGain values while decoding.
-
- WARNING: THIS IS NOT LOSSLESS. DECODED AUDIO WILL NOT BE IDENTICAL TO THE ORIGINAL WITH THIS OPTION.
-
- The equals sign and <specification> is optional. If omitted, the default is 0aLn1.
-
- The <specification> is a shorthand notation for describing how to apply ReplayGain. All components are optional but order is important. '[]' means 'optional'. '|' means 'or'. '{}' means required. The format is:
+ Applies ReplayGain values while decoding.
+
+ WARNING: THIS IS NOT LOSSLESS. DECODED AUDIO WILL NOT BE IDENTICAL TO THE ORIGINAL WITH THIS OPTION.
+
+ The equals sign and <specification> is optional. If omitted, the default is 0aLn1.
+
+ The <specification> is a shorthand notation for describing how to apply ReplayGain. All components are optional but order is important. '[]' means 'optional'. '|' means 'or'. '{}' means required. The format is:
+
[<preamp>][a|t][l|L][n{0|1|2|3}]
-
@@ -472,10 +481,10 @@
Specify the amount of noise shaping. ReplayGain synthesis happens in floating point; the result is dithered before converting back to integer. This quantization adds noise. Noise shaping tries to move the noise where you won't hear it as much. 0 means no noise shaping, 1 means 'low', 2 means 'medium', 3 means 'high'.
- For example, the default of 0aLn1 means 0dB preamp, use album gain, 6dB hard limit, low noise shaping.
-
- -@@@-apply-replaygain-which-is-not-lossless=3 means 3dB preamp, use album gain, no limiting, no noise shaping.
-
+ For example, the default of 0aLn1 means 0dB preamp, use album gain, 6dB hard limit, low noise shaping.
+
+ -@@@-apply-replaygain-which-is-not-lossless=3 means 3dB preamp, use album gain, no limiting, no noise shaping.
+
flac uses the ReplayGain tags for the calculation. If a stream does not have the required tags or they can't be parsed, decoding will continue with a warning, and no ReplayGain is applied to that stream.
|
@@ -516,7 +525,8 @@
--replay-gain
- Calculate ReplayGain values and store them as FLAC tags, similar to VorbisGain. Title gains/peaks will be computed for each input file, and an album gain/peak will be computed for all files. All input files must have the same resolution, sample rate, and number of channels. Only mono and stereo files are allowed, and the sample rate must be one of 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, or 48 kHz. Also note that this option may leave a few extra bytes in a PADDING block as the exact size of the tags is not known until all files are processed.
+ Calculate ReplayGain values and store them as FLAC tags, similar to VorbisGain. Title gains/peaks will be computed for each input file, and an album gain/peak will be computed for all files. All input files must have the same resolution, sample rate, and number of channels. Only mono and stereo files are allowed, and the sample rate must be one of 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, or 48 kHz. Also note that this option may leave a few extra bytes in a PADDING block as the exact size of the tags is not known until all files are processed.
+
Note that this option cannot be used when encoding to standard output (stdout).
|
@@ -526,7 +536,8 @@
--cuesheet=FILENAME
- Import the given cuesheet file and store it in a CUESHEET metadata block. This option may only be used when encoding a single file. A seekpoint will be added for each index point in the cuesheet to the SEEKTABLE unless --no-cued-seekpoints is specified.
+ Import the given cuesheet file and store it in a CUESHEET metadata block. This option may only be used when encoding a single file. A seekpoint will be added for each index point in the cuesheet to the SEEKTABLE unless --no-cued-seekpoints is specified.
+
The cuesheet file must be of the sort written by CDRwin, CDRcue, EAC, etc. See also cuesheet syntax.
|
@@ -536,8 +547,10 @@
--picture=SPECIFICATION
- Import a picture and store it in a PICTURE metadata block. More than one --picture command can be specified. The SPECIFICATION is a string whose parts are separated by | (pipe) characters. Some parts may be left empty to invoke default values. The format of SPECIFICATION is
- [TYPE]|MIME-TYPE|[DESCRIPTION]|[WIDTHxHEIGHTxDEPTH[/COLORS]]|FILE
+ Import a picture and store it in a PICTURE metadata block. More than one --picture command can be specified. The SPECIFICATION is a string whose parts are separated by | (pipe) characters. Some parts may be left empty to invoke default values. The format of SPECIFICATION is
+
+ [TYPE]|MIME-TYPE|[DESCRIPTION]|[WIDTHxHEIGHTxDEPTH[/COLORS]]|FILE
+
TYPE is optional; it is a number from one of:
- 0: Other
@@ -562,12 +575,18 @@
- 19: Band/artist logotype
- 20: Publisher/Studio logotype
- The default is 3 (front cover). There may only be one picture each of type 1 and 2 in a file.
- MIME-TYPE is mandatory; for best compatibility with players, use pictures with MIME type image/jpeg or image/png. The MIME type can also be --> to mean that FILE is actually a URL to an image, though this use is discouraged.
- DESCRIPTION is optional; the default is an empty string.
- The next part specfies the resolution and color information. If the MIME-TYPE is image/jpeg, image/png, or image/gif, you can usually leave this empty and they can be detected from the file. Otherwise, you must specify the width in pixels, height in pixels, and color depth in bits-per-pixel. If the image has indexed colors you should also specify the number of colors used. When manually specified, it is not checked against the file for accuracy.
- FILE is the path to the picture file to be imported, or the URL if MIME type is -->
- For example, the specification |image/jpeg|||../cover.jpg will embed the JPEG file at ../cover.jpg, defaulting to type 3 (front cover) and an empty description. The resolution and color info will be retrieved from the file itself.
+ The default is 3 (front cover). There may only be one picture each of type 1 and 2 in a file.
+
+ MIME-TYPE is mandatory; for best compatibility with players, use pictures with MIME type image/jpeg or image/png. The MIME type can also be --> to mean that FILE is actually a URL to an image, though this use is discouraged.
+
+ DESCRIPTION is optional; the default is an empty string.
+
+ The next part specfies the resolution and color information. If the MIME-TYPE is image/jpeg, image/png, or image/gif, you can usually leave this empty and they can be detected from the file. Otherwise, you must specify the width in pixels, height in pixels, and color depth in bits-per-pixel. If the image has indexed colors you should also specify the number of colors used. When manually specified, it is not checked against the file for accuracy.
+
+ FILE is the path to the picture file to be imported, or the URL if MIME type is -->
+
+ For example, the specification |image/jpeg|||../cover.jpg will embed the JPEG file at ../cover.jpg, defaulting to type 3 (front cover) and an empty description. The resolution and color info will be retrieved from the file itself.
+
The specification 4|-->|CD|320x300x24/173|http://blah.blah/backcover.tiff will embed the given URL, with type 4 (back cover), description "CD", and a manually specified resolution of 320x300, 24 bits-per-pixel, and 173 colors. The file at the URL will not be fetched; the URL itself is stored in the PICTURE metadata block.
|
@@ -577,8 +596,10 @@
--sector-align
- Align encoding of multiple CD format files on sector boundaries. This option is only allowed when encoding files all of which have a 44.1kHz sample rate and 2 channels. With --sector-align, the encoder will align the resulting .flac streams so that their lengths are even multiples of a CD sector (1/75th of a second, or 588 samples). It does this by carrying over any partial sector at the end of each file to the next stream. The last stream will be padded to alignment with zeroes.
- This option will have no effect if the files are already aligned (as is the normally the case with WAVE files ripped from a CD). flac can only align a set of files given in one invocation of flac.
+ Align encoding of multiple CD format files on sector boundaries. This option is only allowed when encoding files all of which have a 44.1kHz sample rate and 2 channels. With --sector-align, the encoder will align the resulting .flac streams so that their lengths are even multiples of a CD sector (1/75th of a second, or 588 samples). It does this by carrying over any partial sector at the end of each file to the next stream. The last stream will be padded to alignment with zeroes.
+
+ This option will have no effect if the files are already aligned (as is the normally the case with WAVE files ripped from a CD). flac can only align a set of files given in one invocation of flac.
+
WARNING: The ordering of files is important! If you give a command like 'flac --sector-align *.wav' the shell may not expand the wildcard to the order you expect. To be safe you should 'echo *.wav' first to confirm the order, or be explicit like 'flac --sector-align 8.wav 9.wav 10.wav'.
|
@@ -826,7 +847,8 @@
-r [#,]#,
--rice-partition-order=[#,]#
- Set the [min,]max residual partition order. The min value defaults to 0 if unspecified.
+ Set the [min,]max residual partition order. The min value defaults to 0 if unspecified.
+
By default the encoder uses a single Rice parameter for the subframe's entire residual. With this option, the residual is iteratively partitioned into 2^min# .. 2^max# pieces, each with its own Rice parameter. Higher values of max# yield diminishing returns. The most bang for the buck is usually with -r 2,2 (more for higher block sizes). This usually shaves off about 1.5%. The technique tends to peak out about when blocksize/(2^n)=128. Use -r 0,16 to force the highest degree of optimization.
|
@@ -964,8 +986,8 @@
- metaflac is the command-line .flac file metadata editor. You can use it to list the contents of metadata blocks, edit, delete or insert blocks, and manage padding.
-
+ metaflac is the command-line .flac file metadata editor. You can use it to list the contents of metadata blocks, edit, delete or insert blocks, and manage padding.
+
metaflac takes a set of "options" (though some are not optional) and a set of FLAC files to operate on. There are three kinds of "options":
-
@@ -978,14 +1000,14 @@
Global options, which affect all the operations.
- All of these are described in the tables below. At least one shorthand or major operation must be supplied. You can use multiple shorthand operations to do more than one thing to a file or set of files. Most of the common things to do to metadata have shorthand operations. As an example, here is how to show the MD5 signatures for a set of three FLAC files:
-
- metaflac --show-md5sum file1.flac file2.flac file3.flac
-
- Another example; this removes all DESCRIPTION and COMMENT tags in a set of FLAC files, and uses the --preserve-modtime global option to keep the FLAC file modification times the same (usually when files are edited the modification time is set to the current time):
-
- metaflac --preserve-modtime --remove-tag=DESCRIPTION --remove-tag=COMMENT file1.flac file2.flac file3.flac
-
+ All of these are described in the tables below. At least one shorthand or major operation must be supplied. You can use multiple shorthand operations to do more than one thing to a file or set of files. Most of the common things to do to metadata have shorthand operations. As an example, here is how to show the MD5 signatures for a set of three FLAC files:
+
+ metaflac --show-md5sum file1.flac file2.flac file3.flac
+
+ Another example; this removes all DESCRIPTION and COMMENT tags in a set of FLAC files, and uses the --preserve-modtime global option to keep the FLAC file modification times the same (usually when files are edited the modification time is set to the current time):
+
+ metaflac --preserve-modtime --remove-tag=DESCRIPTION --remove-tag=COMMENT file1.flac file2.flac file3.flac
+
@@ -1328,10 +1350,12 @@
--list
- List the contents of one or more metadata blocks to stdout. By default, all metadata blocks are listed in text format. Use the following options to change this behavior:
+ List the contents of one or more metadata blocks to stdout. By default, all metadata blocks are listed in text format. Use the following options to change this behavior:
+
--block-number=#[,#[...]]
- An optional comma-separated list of block numbers to display. The first block, the STREAMINFO block, is block 0.
+ An optional comma-separated list of block numbers to display. The first block, the STREAMINFO block, is block 0.
+
--block-type=type[,type[...]]
--except-block-type=type[,type[...]]
@@ -1348,7 +1372,8 @@
|
- NOTE: if both --block-number and --[except-]block-type are specified, the result is the logical AND of both arguments.
+ NOTE: if both --block-number and --[except-]block-type are specified, the result is the logical AND of both arguments.
+
--application-data-format=hexdump|text
If the application block you are displaying contains binary data but your --data-format=text, you can display a hex dump of the application data contents instead using --application-data-format=hexdump.
@@ -1360,12 +1385,14 @@
--remove
|
- Remove one or more metadata blocks from the metadata. Unless --dont-use-padding is specified, the blocks will be replaced with padding. You may not remove the STREAMINFO block.
+ Remove one or more metadata blocks from the metadata. Unless --dont-use-padding is specified, the blocks will be replaced with padding. You may not remove the STREAMINFO block.
+
--block-number=#[,#[...]]
--block-type=type[,type[...]]
--except-block-type=type[,type[...]]
- See --list above for usage.
+ See --list above for usage.
+
NOTE: if both --block-number and --[except-]block-type are specified, the result is the logical AND of both arguments.
|
@@ -1438,8 +1465,8 @@
- Bug tracking is done on the Sourceforge project page here. If you submit a bug, make sure and provide an email contact or use the Monitor feature.
-
+ Bug tracking is done on the Sourceforge project page here. If you submit a bug, make sure and provide an email contact or use the Monitor feature.
+
The following are major known bugs in the current (1.1.3) release:
-
diff --git a/doc/html/download.html b/doc/html/download.html
index 672eb522..b9d3ebae 100644
--- a/doc/html/download.html
+++ b/doc/html/download.html
@@ -115,10 +115,10 @@
- NOTE: these extras are not part of the FLAC project. Most (except those marked [$]) are freely available but distributed under their authors' own terms.
-
- NOTE: make sure to check out the links page for a large list of open-source software supporting FLAC.
-
+ NOTE: these extras are not part of the FLAC project. Most (except those marked [$]) are freely available but distributed under their authors' own terms.
+
+ NOTE: make sure to check out the links page for a large list of open-source software supporting FLAC.
+
GUI encoding/decoding front-ends:
-
diff --git a/doc/html/format.html b/doc/html/format.html
index 26659e37..98531148 100644
--- a/doc/html/format.html
+++ b/doc/html/format.html
@@ -135,8 +135,8 @@
- Acknowledgments
-
+ Acknowledgments
+
FLAC owes much to the many people who have advanced the audio compression field so freely. For instance:
- Scope
-
- It is a known fact that no algorithm can losslessly compress all possible input, so most compressors restrict themselves to a useful domain and try to work as well as possible within that domain. FLAC's domain is audio data. Though it can losslessly code any input, only certain kinds of input will get smaller. FLAC exploits the fact that audio data typically has a high degree of sample-to-sample correlation.
-
- Within the audio domain, there are many possible subdomains. For example: low bitrate speech, high-bitrate multi-channel music, etc. FLAC itself does not target a specific subdomain but many of the default parameters of the reference encoder are tuned to CD-quality music data (i.e. 44.1kHz, 2 channel, 16 bits per sample). The effect of the encoding parameters on different kinds of audio data will be examined later.
-
- Architecture
-
+ Scope
+
+ It is a known fact that no algorithm can losslessly compress all possible input, so most compressors restrict themselves to a useful domain and try to work as well as possible within that domain. FLAC's domain is audio data. Though it can losslessly code any input, only certain kinds of input will get smaller. FLAC exploits the fact that audio data typically has a high degree of sample-to-sample correlation.
+
+ Within the audio domain, there are many possible subdomains. For example: low bitrate speech, high-bitrate multi-channel music, etc. FLAC itself does not target a specific subdomain but many of the default parameters of the reference encoder are tuned to CD-quality music data (i.e. 44.1kHz, 2 channel, 16 bits per sample). The effect of the encoding parameters on different kinds of audio data will be examined later.
+
+ Architecture
+
Similar to many audio coders, a FLAC encoder has the following stages:
-
@@ -175,10 +175,10 @@
Residual coding. If the predictor does not describe the signal exactly, the difference between the original signal and the predicted signal (called the error or residual signal) must be coded losslessy. If the predictor is effective, the residual signal will require fewer bits per sample than the original signal. FLAC currently uses only one method for encoding the residual (see the Residual coding section), but the format has reserved space for additional methods. FLAC allows the residual coding method to change from block to block, or even within the channels of a block.
- In addition, FLAC specifies a metadata system, which allows arbitrary information about the stream to be included at the beginning of the stream.
-
- Definitions
-
+ In addition, FLAC specifies a metadata system, which allows arbitrary information about the stream to be included at the beginning of the stream.
+
+ Definitions
+
Many terms like "block" and "frame" are used to mean different things in differenct encoding schemes. For example, a frame in MP3 corresponds to many samples across several channels, whereas an S/PDIF frame represents just one sample for each channel. The definitions we use for FLAC follow. Note that when we talk about blocks and subblocks we are refering to the raw unencoded audio data that is the input to the encoder, and when we talk about frames and subframes, we are refering to the FLAC-encoded data.
-
@@ -197,16 +197,16 @@
Subframe: A subframe header plus one or more encoded samples from a given channel. All subframes within a frame will contain the same number of samples.
- Blocking
-
- The size used for blocking the audio data has a direct effect on the compression ratio. If the block size is too small, the resulting large number of frames mean that excess bits will be wasted on frame headers. If the block size is too large, the characteristics of the signal may vary so much that the encoder will be unable to find a good predictor. In order to simplify encoder/decoder design, FLAC imposes a minimum block size of 16 samples, and a maximum block size of 65535 samples. This range covers the optimal size for all of the audio data FLAC supports.
-
- Currently the reference encoder uses a fixed block size, optimized on the sample rate of the input. Future versions may vary the block size depending on the characteristics of the signal.
-
- Blocked data is passed to the predictor stage one subblock (channel) at a time. Each subblock is independently coded into a subframe, and the subframes are concatenated into a frame. Because each channel is coded separately, it means that one channel of a stereo frame may be encoded as a constant subframe, and the other an LPC subframe.
-
- Interchannel Decorrelation
-
+ Blocking
+
+ The size used for blocking the audio data has a direct effect on the compression ratio. If the block size is too small, the resulting large number of frames mean that excess bits will be wasted on frame headers. If the block size is too large, the characteristics of the signal may vary so much that the encoder will be unable to find a good predictor. In order to simplify encoder/decoder design, FLAC imposes a minimum block size of 16 samples, and a maximum block size of 65535 samples. This range covers the optimal size for all of the audio data FLAC supports.
+
+ Currently the reference encoder uses a fixed block size, optimized on the sample rate of the input. Future versions may vary the block size depending on the characteristics of the signal.
+
+ Blocked data is passed to the predictor stage one subblock (channel) at a time. Each subblock is independently coded into a subframe, and the subframes are concatenated into a frame. Because each channel is coded separately, it means that one channel of a stereo frame may be encoded as a constant subframe, and the other an LPC subframe.
+
+ Interchannel Decorrelation
+
In stereo streams, in many cases there is an exploitable amount of correlation between the left and right channels. FLAC allows the frames of stereo streams to have different channel assignments, and an encoder may choose to use the best representation on a frame-by-frame basis.
-
@@ -222,10 +222,10 @@
Right-side. The right channel and side channel are coded
- Surprisingly, the left-side and right-side forms can be the most efficient in many frames, even though the raw number of bits per sample needed for the original signal is slightly more than that needed for independent or mid-side coding.
-
- Prediction
-
+ Surprisingly, the left-side and right-side forms can be the most efficient in many frames, even though the raw number of bits per sample needed for the original signal is slightly more than that needed for independent or mid-side coding.
+
+ Prediction
+
FLAC uses four methods for modeling the input signal:
-
@@ -241,18 +241,18 @@
FIR Linear prediction. For more accurate modeling (at a cost of slower encoding), FLAC supports up to 32nd order FIR linear prediction (again, for info on linear prediction, see audiopak and shorten). The reference encoder uses the Levinson-Durbin method for calculating the LPC coefficients from the autocorrelation coefficients, and the coefficients are quantized before computing the residual. Whereas encoders such as Shorten used a fixed quantization for the entire input, FLAC allows the quantized coefficient precision to vary from subframe to subframe. The FLAC reference encoder estimates the optimal precision to use based on the block size and dynamic range of the original signal.
- Residual Coding
-
- FLAC currently defines two similar methods for the coding of the error signal from the prediction stage. The error signal is coded using Rice codes in one of two ways: 1) the encoder estimates a single rice parameter based on the variance of the residual and Rice codes the entire residual using this parameter; 2) the residual is partitioned into several equal-length regions of contiguous samples, and each region is coded with its own Rice parameter based on the region's mean. (Note that the first method is a special case of the second method with one partition, except the Rice parameter is based on the residual variance instead of the mean.)
-
- The FLAC format has reserved space for other coding methods. Some possiblities for volunteers would be to explore better context-modeling of the Rice parameter, or Huffman coding. See LOCO-I and pucrunch for descriptions of several universal codes.
-
- Format
-
- This section specifies the FLAC bitstream format. FLAC has no format version information, but it does contain reserved space in several places. Future versions of the format may use this reserved space safely without breaking the format of older streams. Older decoders may choose to abort decoding or skip data encoded with newer methods. Apart from reserved patterns, in places the format specifies invalid patterns, meaning that the patterns may never appear in any valid bitstream, in any prior, present, or future versions of the format. These invalid patterns are usually used to make the synchronization mechanism more robust.
-
- All numbers used in a FLAC bitstream are integers; there are no floating-point representations. All numbers are big-endian coded. All numbers are unsigned unless otherwise specified.
-
+ Residual Coding
+
+ FLAC currently defines two similar methods for the coding of the error signal from the prediction stage. The error signal is coded using Rice codes in one of two ways: 1) the encoder estimates a single rice parameter based on the variance of the residual and Rice codes the entire residual using this parameter; 2) the residual is partitioned into several equal-length regions of contiguous samples, and each region is coded with its own Rice parameter based on the region's mean. (Note that the first method is a special case of the second method with one partition, except the Rice parameter is based on the residual variance instead of the mean.)
+
+ The FLAC format has reserved space for other coding methods. Some possiblities for volunteers would be to explore better context-modeling of the Rice parameter, or Huffman coding. See LOCO-I and pucrunch for descriptions of several universal codes.
+
+ Format
+
+ This section specifies the FLAC bitstream format. FLAC has no format version information, but it does contain reserved space in several places. Future versions of the format may use this reserved space safely without breaking the format of older streams. Older decoders may choose to abort decoding or skip data encoded with newer methods. Apart from reserved patterns, in places the format specifies invalid patterns, meaning that the patterns may never appear in any valid bitstream, in any prior, present, or future versions of the format. These invalid patterns are usually used to make the synchronization mechanism more robust.
+
+ All numbers used in a FLAC bitstream are integers; there are no floating-point representations. All numbers are big-endian coded. All numbers are unsigned unless otherwise specified.
+
Before the formal description of the stream, an overview might be helpful.
-
diff --git a/doc/html/index.html b/doc/html/index.html
index 49386234..646ea1cc 100644
--- a/doc/html/index.html
+++ b/doc/html/index.html
@@ -57,27 +57,27 @@
- Winamp 5.31 now includes Nullsoft FLAC plugins for encoding and decoding. The decoder is based on our reference decoder plugin. However the current encoder plugin is based on a pre-release of flake and we recommend to not use it for archival yet.
-
- The new Helios X5000 HD network media player from Neodigits supports FLAC.
-
- The Philadelphia Orchestra is making many recordings available in FLAC.
-
- A whole new batch of devices and stores support FLAC: for portables there are the iAUDIO T2 and iAUDIO F2, TrekStor's Vibez, the Onda VX737, and the AP3000 from Green Apple. For the home stereo, Slim Devices' Transporter and Ziova's CS510 and CS505. For music in FLAC format check out digital-tunes for electronic and underground, or FestivaLink.net for live shows.
-
- Bluedot's BMP-1430 portable supports FLAC.
-
- AudioReQuest's new S.Series music servers support FLAC.
-
- Cowon's A2 now supports FLAC with the latest firmware, and Olive's new Opus both plays and records to FLAC.
-
- The new Iwod G10 portable supports FLAC.
-
- Want some FLAC with your Volvo? Volvo's Digital Jukebox, developed with PhatNoise, is fully integrated with the car's audio system and available for the S60, V70, XC70, and S80. PhatNoise's PhatBox in 2002 was the first device to support FLAC natively and has gained a loyal following.
-
+ Winamp 5.31 now includes Nullsoft FLAC plugins for encoding and decoding. The decoder is based on our reference decoder plugin. However the current encoder plugin is based on a pre-release of flake and we recommend to not use it for archival yet.
+
+ The new Helios X5000 HD network media player from Neodigits supports FLAC.
+
+ The Philadelphia Orchestra is making many recordings available in FLAC.
+
+ A whole new batch of devices and stores support FLAC: for portables there are the iAUDIO T2 and iAUDIO F2, TrekStor's Vibez, the Onda VX737, and the AP3000 from Green Apple. For the home stereo, Slim Devices' Transporter and Ziova's CS510 and CS505. For music in FLAC format check out digital-tunes for electronic and underground, or FestivaLink.net for live shows.
+
+ Bluedot's BMP-1430 portable supports FLAC.
+
+ AudioReQuest's new S.Series music servers support FLAC.
+
+ Cowon's A2 now supports FLAC with the latest firmware, and Olive's new Opus both plays and records to FLAC.
+
+ The new Iwod G10 portable supports FLAC.
+
+ Want some FLAC with your Volvo? Volvo's Digital Jukebox, developed with PhatNoise, is fully integrated with the car's audio system and available for the S60, V70, XC70, and S80. PhatNoise's PhatBox in 2002 was the first device to support FLAC natively and has gained a loyal following.
+
last updated 2006-Oct-25
- FLAC stands for Free Lossless Audio Codec. Grossly oversimplified, FLAC is similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo, see supported devices) just like you would an MP3 file.
-
- FLAC is freely available and supported on most operating systems, including Windows, "unix" (Linux, *BSD, Solaris, OS X, IRIX), BeOS, OS/2, and Amiga. There are build systems for autotools, MSVC, Watcom C, and Project Builder.
-
- See the features page for a complete list of features, or the comparison page to see how FLAC compares with other lossless codecs.
-
- The FLAC project consists of:
-
+ FLAC stands for Free Lossless Audio Codec. Grossly oversimplified, FLAC is similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo, see supported devices) just like you would an MP3 file.
+
+ FLAC is freely available and supported on most operating systems, including Windows, "unix" (Linux, *BSD, Solaris, OS X, IRIX), BeOS, OS/2, and Amiga. There are build systems for autotools, MSVC, Watcom C, and Project Builder.
+
+ See the features page for a complete list of features, or the comparison page to see how FLAC compares with other lossless codecs.
+
+ The FLAC project consists of:
+
- the stream format
- reference encoders and decoders in library form
- flac, a command-line program to encode and decode FLAC files
- metaflac, a command-line metadata editor for FLAC files
- input plugins for various music players
-
-
+
+
When we say that FLAC is "Free" it means more than just that it is available at no cost. It means that the specification of the format is fully open to the public to be used for any purpose (the FLAC project reserves the right to set the FLAC specification and certify compliance), and that neither the FLAC format nor any of the implemented encoding/decoding methods are covered by any known patent. It also means that all the source code is available under open-source licenses. It is the first truly open and free lossless audio format. (For more information, see the license page.)
- The main purpose of these reviews is to give an idea of how well particular devices support FLAC. Other subjective comments here are based on our general impressions and are not meant to be thorough or authoritative. We only review devices we have tested directly ourselves.
-
- Rio Reciever: This little device is a hacker's dream. It plays audio over a network (Ethernet or HPNA) so it requires a PC to serve audio files. There are several open source clients available and since it boots its Linux distro over NFS you can write your own client. They're not made anymore but you can still find them on ebay. The main downsides: 1) small, hard-to-read LCD display; 2) FLAC support is only in third-party clients which take some work to set up.
-
+ Reviews:
+
+ The main purpose of these reviews is to give an idea of how well particular devices support FLAC. Other subjective comments here are based on our general impressions and are not meant to be thorough or authoritative. We only review devices we have tested directly ourselves.
+
+ Rio Reciever: This little device is a hacker's dream. It plays audio over a network (Ethernet or HPNA) so it requires a PC to serve audio files. There are several open source clients available and since it boots its Linux distro over NFS you can write your own client. They're not made anymore but you can still find them on ebay. The main downsides: 1) small, hard-to-read LCD display; 2) FLAC support is only in third-party clients which take some work to set up.
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Squeezebox2: A fantastic networked audio player. Has an excellent, easy-to-read vacuum fluorescent display, wired or wireless networking, optical and coax digital outs and analog out, a reputation for very high audio quality, multi-room synchronization, and a bunch of other features. The server-side software, SlimServer, is open-source, runs on Windows, Mac OS X, Linux, etc. and has an active community. FLAC support is excellent; nearly the full subset (e.g. sample rates up to 48kHz, 16- and 24-bits per sample) including all standard encoding modes are supported. Also supported are FLAC tags, automatic transcoding on the server of many audio formats to FLAC for transmission to the box, and external cuesheet support (internal cuesheet support is in the works).
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