fisiere FLAC ascultate prin DAC sau CD-audio

[steliani]

....nu cred ca doresc sa platesc inca o data ca sa aflu...

dar unul dintre albumele lui Pava, adica Pavarotti 24 Greatest HD Tracks are la proprietatile track file compresie 0%. La Qbuz Buddha Bar XVI are compresie 23%. Dar la francezi este disponibil numai flac postat ca si CD quality 44/16.

Ideea este ca nu sunt flac fan. Nu am fost de acord cu afirmatia ca flac inseamna neaparat compresie. Si repet ca eu, personal nu am sesizat diferente intre formate pe sistemul meu.

Dar daca vorbim despre ce se aude mai bine...XRCD rip este in top. Se vede clar ca nu formatul conteaza ci masterizarea.

Dar nici asta nu se aude rau...recomand.

http://www.hdtracks.com/eric-clapton-friends-the-breeze-an-appreciation-of-jj-cale

[bibi]

Nu am fost de acord cu afirmatia ca flac inseamna neaparat compresie.

 

Cu tot respectul, dar aici mă tem că avem o divergență majoră de opinii.

[steliani]

bibi a scris:

Cu tot respectul, dar aici mă tem că avem o divergență majoră de opinii.

Nicio problema, iti respect pdv; dar eu cred in ceea ce vad si ce stiu ca am realizat pana acum.

Uite o conversie in AIFF, FLAC, WAV facuta cu dBpoweramp pt Ultimate Santana - Into The Night

toate bune

[hersorin]

Si eu sunt un adept al PC-ului cu dac . Din punctul meu de vedere exista diferenta FLAC - WAV (chiar acelasi flac transformat in WAV) . Am pus aceasta problema cu mult timp in urma dar se lasa cu diferite contre .

Cred ca la decompresie se face diferenta chiar daca PC-ul e ultimul i7 cu maxim de performanta.

Am facut si pentru mine demonstratia in ABX si am recunoscut 4 din 5 . Muzica din formatul Wav a fost mai plina mai corpolenta .

Nu ma intereseaza polemici cu altii dar cei care aud diferente FLAC-WAV acestea exista .

[FATHER]

Din punctul meu de vedere exista diferenta FLAC - WAV (chiar acelasi flac transformat in WAV) . Am pus aceasta problema cu mult timp in urma dar se lasa cu diferite contre .

Nu ma intereseaza polemici cu altii dar cei care aud diferente FLAC-WAV acestea exista .

 

Sustin cele de mai sus. 100% .

[andreic]

Si eu folosesc combinatia PC+Dac si am facut un test in felul urmator: la intrarile digitale are dac-ului am conectat un cd player dedicat Revox si iesirea digitala a pc-ului. Am copiat initial cd-ul pe calc in format wav si am redat simultan aceeasi piesa pe cd si pe pc. Comutand intre cele doua intrari de n ori nu am reusit sa sesizez absolut nicio diferenta intre cele doua.

Dac Cambridge audio magic

Amp: Roksan kandy l3

Boxe: Opera seconda.

Nu am facut testul si fu fisierul codat in flac, asa ca prefer sa nu dau cu presupusul.

[dinica]

Am intalnit deseori oameni care auzeau, judecau emotional sunetul, incat am ajuns sa privesc cu rezerva afirmatiile ce se bazeaza exclusiv pe experienta emotionala. Pe http://www.whatsbestforum.com/ gasiti discutii purtate intre ingineri de studio si pasionati cu o singura concluzie, nu exista diferenta intre flac si wav. Pot aparea diferente in situatia in care se foloseste un soft prost. Ca sa evitati acest lucru incercati http://www.jriver.com/ si va garantez ca diferentele vor disparea.

[Andi]

Are cineva experienta cu un program de upmixing-upconversion a fisierelor wav stereo in wav dts 5.1?

[Phaenius]

Si eu folosesc combinatia PC+Dac si am facut un test in felul urmator: la intrarile digitale are dac-ului am conectat un cd player dedicat Revox si iesirea digitala a pc-ului. Am copiat initial cd-ul pe calc in format wav si am redat simultan aceeasi piesa pe cd si pe pc. Comutand intre cele doua intrari de n ori nu am reusit sa sesizez absolut nicio diferenta intre cele doua.

Dac Cambridge audio magic

Amp: Roksan kandy l3

Boxe: Opera seconda.

Nu am facut testul si fu fisierul codat in flac, asa ca prefer sa nu dau cu presupusul.

 

Cum comuta Cambridge-ul sursa, digital sau analog ? Daca e digital, s-ar putea sa ai un delay destul de mare intre comutatie, care sa-si stearga ultima impresie. De asta cel mai bine e un comutator mecanic daca se poate, care sa comute instant. Asa vezi mai bine diferentele daca exista. Si eu zic ca nu sint diferente sesizabile pentru majoritatea oamenilor.

 

Nu ma intereseaza polemici cu altii dar cei care aud diferente FLAC-WAV acestea exista .

 

Si eu intreb, fara dorinta de a starni o polemica, de unde vin acele diferente, atat timp cat FLAC-ul contine exact aceeasi informatie ca WAV-ul, doar ca e comprimat fara pierderi.

 

Am intalnit deseori oameni care auzeau, judecau emotional sunetul, incat am ajuns sa privesc cu rezerva afirmatiile ce se bazeaza exclusiv pe experienta emotionala. Pe http://www.whatsbestforum.com/ gasiti discutii purtate intre ingineri de studio si pasionati cu o singura concluzie, nu exista diferenta intre flac si wav. Pot aparea diferente in situatia in care se foloseste un soft prost. Ca sa evitati acest lucru incercati http://www.jriver.com/ si va garantez ca diferentele vor disparea.

 

Eu nu stiu ce poate face un soft prost, atat timp cat unui FLAC nu trebuie sa-i faci decat un singur lucru cand il redai: sa-l decomprimi. E ca si cum ai arhiva cu RAR un fisier si apoi ar conta cu ce il decomprimi, Winrar, 7ZIP, Winace, etc.

[Mariutim]

Am intalnit deseori oameni care auzeau, judecau emotional sunetul, incat am ajuns sa privesc cu rezerva afirmatiile ce se bazeaza exclusiv pe experienta emotionala. Pe http://www.whatsbestforum.com/ gasiti discutii purtate intre ingineri de studio si pasionati cu o singura concluzie, nu exista diferenta intre flac si wav. Pot aparea diferente in situatia in care se foloseste un soft prost. Ca sa evitati acest lucru incercati http://www.jriver.com/ si va garantez ca diferentele vor disparea.

Nu exista diferente intre flac si wav.Softurile pot modifica marimea fisierului,insa nu e auzibil .Am facut teste Flac si Wav redate si de pe Pc si de pe disc.Nu sunt diferente audio,sau eu nu le aud.Dar nu toti oamenii aud la fel.

[Mariutim]

Sa analizam din nou ce este formatul Flac si cum functioneaza.

C (/ˈflæk/; Free Lossless Audio Codec) is an audio coding format for lossless compression of digital audio, and is also the name of the reference codec implementation. Digital audio compressed by FLAC's algorithm can typically be reduced to 50–60% of its original size[4] and decompressed to an identical copy of the original audio data.

FLAC is an open format with royalty-free licensing and a reference implementation which is free software. FLAC has support for metadata tagging, album cover art, and fast seeking.

Though FLAC cannot store floating-point data, and playback support in portable audio devices and dedicated audio systems is limited compared to lossy formats such as MP3 or uncompressed PCM, FLAC is supported by more hardware devices than competing lossless compressed formats such as Wav pack.

istory

Development was started in 2000 by Josh Coalson.[5] The bit-stream format was frozen when FLAC entered beta stage with the release of version 0.5 of the reference implementation on 15 January 2001. Version 1.0 was released on 20 July 2001.[5]

On 29 January 2003, the Xiph.Org Foundation and the FLAC project announced the incorporation of FLAC under the Xiph.org banner. Xiph.org is behind other free compression formats such as Vorbis, Theora and Speex.[5][6][7]

Version 1.3.0 was released on 26 May 2013. Development was moved to the Xiph.org git repository.[8]

Composition

The FLAC project consists of:

The stream formats

A simple container format for the stream, also called FLAC (or Native FLAC)

libFLAC, a library of reference encoders and decoders, and a metadata interface

libFLAC++, an object wrapper around libFLAC

flac, a command-line program based on libFLAC to encode and decode FLAC streams

metaflac, a command-line metadata editor for .flac files and for applying ReplayGain

Input plugins for various music players (Winamp, XMMS, foobar2000, musikCube, and many more)

With Xiph.org incorporation, the Ogg container format, suitable for streaming (also called Ogg FLAC)

The specification of the stream format can be implemented by anyone without prior permission (Xiph.org reserves the right to set the FLAC specification and certify compliance), and neither the FLAC format nor any of the implemented encoding / decoding methods are covered by any patent. The reference implementation is free software. The source code for libFLAC and libFLAC++ is available under the BSD license, and the sources for flac, metaflac, and the plugins are available under the GNU General Public License.

In its stated goals, the FLAC project encourages its developers not to implement copy prevention features (DRM) of any kind.[9]

Design

Audio sources encoded to FLAC are typically reduced to 50–60% of their original size.[4] FLAC supports only fixed-point samples, not floating-point. It can handle any PCM bit resolution from 4 to 32 bits per sample, any sampling rate from 1 Hz to 655,350 Hz in 1 Hz increments,[10] and any number of channels from 1 to 8.[11] Channels can be grouped in cases, for example stereo and 5.1 channel surround, to take advantage of interchannel correlations to increase compression. FLAC uses CRC checksums for identifying corrupted frames when used in a streaming protocol, and also has a complete MD5 hash of the raw PCM audio stored in its STREAMINFO metadata header. FLAC allows for a Rice parameter between 0 and 16. FLAC supports ReplayGain.

FLAC uses linear prediction to convert the audio samples to a series of small, uncorrelated numbers (known as the residual), which are stored efficiently using Golomb-Rice coding. It also uses run-length encoding for blocks of identical samples, such as silent passages. For tagging, FLAC uses the same system as Vorbis comments.[10] The libFLAC API is organized into streams, seekable streams, and files (listed in the order of increasing abstraction from the base FLAC bitstream). Most FLAC applications will generally restrict themselves to encoding/decoding using libFLAC at the file level interface.

Compression levels

libFLAC uses a compression level parameter that varies from 0 (fastest) to 8 (smallest). The compressed files are always perfect "lossless" representations of the original data. Although the compression process involves a tradeoff between speed and size, the decoding process is always quite fast, and not very dependent on the level of compression.[12][13]

Comparison to other formats

FLAC is specifically designed for efficient packing of audio data, unlike general-purpose lossless algorithms such as DEFLATE, which is used in ZIP and gzip. While ZIP may reduce the size of a CD-quality audio file by 10–20%, FLAC is able to reduce the size of audio data by 40–50% by taking advantage of the characteristics of audio.

The technical strengths of FLAC compared to other lossless formats lie in its ability to be streamed and decoded quickly, independently of compression level. In a comparison of compressed audio formats, FFmpeg's FLAC implementation was noted to have the fastest and most efficient embedded decoder of any modern lossless audio format.[14]

Since FLAC is a lossless scheme, it is suitable as an archive format for owners of CDs and other media who wish to preserve their audio collections. If the original media is lost, damaged, or worn out, a FLAC copy of the audio tracks ensures that an exact duplicate of the original data can be recovered at any time. An exact restoration from a lossy archive (e.g., MP3) of the same data is impossible. FLAC being lossless means it is highly suitable for transcode e.g. to MP3, without the normally associated transcoding quality loss. A CUE file can optionally be created when ripping a CD. If a CD is read and ripped perfectly to FLAC files, the CUE file allows later burning of an audio CD that is identical in audio data to the original CD, including track order, pregaps, and CD-Text. However, additional data present on some audio CDs such as lyrics and CD+G graphics are beyond the scope of a CUE file and most ripping software, and that data will not be archived.[citation needed]

Adoption and implementations

See also: list of hardware and software that supports FLAC

The reference implementation of FLAC is implemented as the libFLAC core encoder & decoder library, with the main distributable program flac being the reference implementation of the libFLAC API. This codec API is also available in C++ as libFLAC++. The reference implementation of FLAC compiles on many platforms, including most Unix (such as Solaris and Mac OS X) and Unix-like (including GNU/Linux, BSD), Microsoft Windows, BeOS, and OS/2 operating systems. There are build systems for autoconf/automake, MSVC, Watcom C, and Xcode. There is currently no multicore support in libFLAC.

Though FLAC playback support in portable audio devices and dedicated audio systems is limited compared to formats such as MP3[15] or uncompressed PCM, FLAC is supported by more hardware devices than competing lossless compressed formats such as WavPack.[4] FLAC support is included by default in Android devices.

In 2014, several aftermarket mobile electronics companies have introduced multimedia solutions that include support for FLAC. These include the NEX series from Pioneer Electronics and the VX404 and NX404 from Clarion.

The European Broadcasting Union (EBU) has adopted the FLAC format for the distribution of high quality audio over its Euroradio network.[16] The Android operating system has supported native FLAC playback since version 3.1.[17][18]

The Pono music player and music service[19] and the Qobuz music streaming service[20] both use the FLAC format. Since January 2014 gog.com offers video game soundtracks in FLAC format as bonus.[

Preluat de pe Wikipedia.

[Mariutim]

Si acum WAV.

Waveform Audio File Format (WAVE, or more commonly known as WAV due to its filename extension)[3][6][7][8] (rarely, Audio for Windows[9]) is a Microsoft and IBM audio file format standard for storing an audio bitstream on PCs. It is an application of the Resource Interchange File Format (RIFF) bitstream format method for storing data in "chunks", and thus is also close to the 8SVX and the AIFF format used on Amiga and Macintosh computers, respectively. It is the main format used on Windows systems for raw and typically uncompressed audio. The usual bitstream encoding is the linear pulse-code modulation (LPCM) format.

Description

Both WAVs and AIFFs are compatible with Windows, Macintosh, and Linux operating systems. The format takes into account some differences of the Intel CPU such as little-endian byte order. The RIFF format acts as a "wrapper" for various audio coding formats.

Though a WAV file can contain compressed audio, the most common WAV audio format is uncompressed audio in the linear pulse code modulation (LPCM) format. LPCM is also the standard audio coding format for audio CDs, which store two-channel LPCM audio sampled 44,100 times per second with 16 bits per sample. Since LPCM is uncompressed and retains all of the samples of an audio track, professional users or audio experts may use the WAV format with LPCM audio for maximum audio quality. WAV files can also be edited and manipulated with relative ease using software.

The WAV format supports compressed audio, using, on Windows, the Audio Compression Manager. Any ACM codec can be used to compress a WAV file. The user interface (UI) for Audio Compression Manager may be accessed through various programs that use it, including Sound Recorder in some versions of Windows.

Beginning with Windows 2000, a WAVE_FORMAT_EXTENSIBLE header was defined which specifies multiple audio channel data along with speaker positions, eliminates ambiguity regarding sample types and container sizes in the standard WAV format and supports defining custom extensions to the format chunk.[4][5][10]

There are some inconsistencies in the WAV format: for example, 8-bit data is unsigned while 16-bit data is signed, and many chunks duplicate information found in other chunks.

Specification

The WAV file is an instance of a Resource Interchange File Format (RIFF) defined by IBM and Microsoft.[11]

RIFF

A RIFF file is a tagged file format. It has a specific container format (a chunk) that includes a four character tag (FOURCC) and the size (number of bytes) of the chunk. The tag specifies how the data within the chunk should be interpreted, and there are several standard FOURCC tags. Tags consisting of all capital letters are reserved tags. The outermost chunk of a RIFF file has a RIFF form tag; the first four bytes of chunk data are a FOURCC that specify the form type and are followed by a sequence of subchunks. In the case of a WAV file, those four bytes are the FOURCC WAVE. The remainder of the RIFF data is a sequence of chunks describing the audio information.

The advantage of a tagged file format is that the format can be extended later without confusing existing file readers.[12] The rule for a RIFF (or WAV) reader is that it should ignore any tagged chunk that it does not recognize.[13] The reader won't be able to use the new information, but the reader should not be confused.

The specification for RIFF files includes the definition of an INFO chunk. The chunk may include information such as the title of the work, the author, the creation date, and copyright information. Although the INFO chunk was defined in version 1.0, the chunk was not referenced in the formal specification of a WAV file. If the chunk were present in the file, then a reader should know how to interpret it, but many readers had trouble. Some readers would abort when they encountered the chunk, some readers would process the chunk if it were the first chunk in the RIFF form,[14] and other readers would process it if it followed all of the expected waveform data. Consequently, the safest thing to do from an interchange standpoint was to omit the INFO chunk and other extensions and send a lowest-common-denominator file. There are other INFO chunk placement problems.

RIFF files were expected to be used in international environments, so there is CSET chunk to specify the country code, language, dialect, and code page for the strings in a RIFF file.[15] For example, specifying an appropriate CSET chunk should allow the strings in an INFO chunk (and other chunks throughout the RIFF file) to be interpreted as Cyrillic or Japanese characters.

RIFF also defines a JUNK chunk whose contents are uninteresting.[16] The chunk allows a chunk to be deleted by just changing its FOURCC. The chunk could also be used to reserve some space for future edits so the file could be modified without being rewritten. A later definition of RIFF introduced a similar PAD chunk.[17]

RIFF WAVE

The toplevel definition of a WAV file is:[18]

→ RIFF('WAVE'

// Format

[] // Fact chunk

[] // Cue points

[] // Playlist

[] // Associated data list

) // Wave data

The definition shows a toplevel RIFF form with the WAVE tag. It is followed by a mandatory format chunk that describes the format of the sample data that follows. The format chunk includes information such as the sample encoding, number of bits per channel, the number of channels, the sample rate. The WAV specification includes some optional features. The optional fact chunk reports the number of samples for some compressed coding schemes. The cue point (cue ) chunk identifies some significant sample numbers in the wave file. The playlist chunk allows the samples to be played out of order or repeated rather than just from beginning to end. The associated data list allows labels and notes (labl and note) to be attached to cue points; text annotation (ltxt) may be given for a group of samples (e.g., caption information). Finally, the mandatory wave data chunk contains the actual samples (in the specified format).

Note that the WAV file definition does not show where an INFO chunk should be placed. It is also silent about the placement of a CSET chunk (which specifies the character set used).

The RIFF specification attempts to be a formal specification, but its formalism lacks the precision seen in other tagged formats. For example, the RIFF specification does not clearly distinguish between a set of subchunks and an ordered sequence of subchunks. The RIFF form chunk suggests it should be a sequence container.[19] The specification suggests a LIST chunk is also a sequence: "A LIST chunk contains a list, or ordered sequence, of subchunks."[20] However, the specification does not give a formal specification of the INFO chunk; an example INFO LIST chunk ignores the chunk sequence implied in the INFO description.[21] The LIST chunk definition for does use the LIST chunk as a sequence container with good formal semantics.

The WAV specification allows for not only a single, contiguous, array of audio samples, but also discrete blocks of samples and silence that are played in order. Most WAV files use a single array of data. The specification for the sample data is confused:[22]

The contains the waveform data. It is defined as follows:

→ { | }

→ data( )

→ LIST( 'wavl' { | // Wave samples

}... ) // Silence

→ slnt( ) // Count of silent samples

These productions are confused. Apparently (undefined) and (defined but not referenced) should be identical. Even if that problem is fixed, the productions then allow a to contain a recursive (which implies data interpretation problems). The specification should have been something like:

→ { | }

→ data( ... )

→ LIST( 'wavl' { | // Wave samples

}... ) // Silence

→ slnt( ) // Count of silent samples

to avoid the recursion.

WAV files can contain embedded IFF "lists", which can contain several "sub-chunks".[23][24][25]

Metadata

As a derivative of RIFF, WAV files can be tagged with metadata in the INFO chunk. In addition, WAV files can embed Extensible Metadata Platform (XMP) data. Applications may not handle this extra information or may expect to see it in a particular place. Although the RIFF specification requires that applications ignore chunks they do not recognize, some applications are confused by additional chunks.[citation needed]

Popularity

Uncompressed WAV files are large, so file sharing of WAV files over the Internet is uncommon. However, it is a commonly used file type, suitable for retaining first generation archived files of high quality, for use on a system where disk space is not a constraint, or in applications such as audio editing, where the time involved in compressing and uncompressing data is a concern.

More frequently, the smaller file sizes of compressed but lossy formats such as MP3 are used to store and transfer audio. Their small file sizes allow faster Internet transmission, as well as lower consumption of space on memory media. There are also lossless-compression formats such as FLAC.

The usage of the WAV format has more to do with its familiarity and simple structure. Because of this, it continues to enjoy widespread use with a variety of software applications, often functioning as a 'lowest common denominator' when it comes to exchanging sound files among different programs.

Use by broadcasters

In spite of their large size, uncompressed WAV files are sometimes used by some radio broadcasters, especially those that have adopted a tapeless system. BBC Radio in the UK uses 44.1 kHz 16-bit two-channel WAV audio as standard in their VCS system. [note 1]

UK Commercial radio company Global Radio uses 44.1 kHz 16-bit two-channel WAV files in the Genesys playout system, and throughout their broadcast chain.

The ABC "D-Cart" system, which was developed by the Australian broadcaster, uses 48 kHz 16-bit two-channel WAV files, which is identical to that of Digital Audio Tape.

The Digital Radio Mondiale consortium uses WAV files as an informal standard for transmitter simulation and receiver testing.

Limitations

The WAV format is limited to files that are less than 4 GB, because of its use of a 32-bit unsigned integer to record the file size header (some programs limit the file size to 2 GB).[26] Although this is equivalent to about 6.8 hours of CD-quality audio (44.1 kHz, 16-bit stereo), it is sometimes necessary to exceed this limit, especially when greater sampling rates or bit resolutions are required. The W64 format was therefore created for use in Sound Forge. Its 64-bit header allows for much longer recording times. The RF64 format specified by the European Broadcasting Union has also been created to solve this problem.

Non-audio data

Since the sampling rate of a WAV file can vary from 1 Hz to 4.3 GHz, and the number of channels can be as high as 65535, .wav files have also been used for non-audio data. LTspice, for instance, can store multiple circuit trace waveforms in separate channels, at any appropriate sampling rate, with the full-scale range representing ±1 V or A rather than a sound pressure.[27]

A number of software-defined radio (SDR) computer programs are able to capture large chunks of the RF spectrum and store the data as a WAV file. Typically these files have a high sampling rate such as 2 MHz or more, and they have two channels, although they are not considered stereo as the data in each channel are the I and Q components of an IQ signal. An SDR device with a bandwidth of 20 MHz could be tuned to a radio frequency of 98 MHz, where it would capture all radio stations in the entire FM radio band at once. This data could then be saved to a WAV file with a sampling rate of 40 MHz. The same or another SDR program can later read this WAV file and "tune" in to any of the radio stations captured in the recording, even listening to multiple stations at the same time.

Audio CDs

Audio CDs do not use the WAV file format, using instead Red Book audio. The commonality is that both audio CDs and WAV files encode the audio as PCM. WAV is a file format for a computer to use that cannot be understood by most CD players directly. To record WAV files to an Audio CD the file headers must be stripped and the remaining PCM data written directly to the disc as individual tracks with zero-padding added to match the CD's sector size. In order for a WAV file to be able to be burned to a CD, it should be in the 44100 Hz, 16-bit stereo format.

WAV file audio coding formats compared

This article needs attention from an expert in Professional sound production. Please add a reason or a talk parameter to this template to explain the issue with the article. WikiProject Professional sound production (or its Portal) may be able to help recruit an expert. (October 2009)

Main article: Audio compression (data)

Further information: Comparison of audio coding formats

Audio in WAV files can be encoded in a variety of audio coding formats, such as GSM or MP3, to reduce the file size.

This is a reference to compare the monophonic (not stereophonic) audio quality and compression bitrates of audio coding formats available for WAV files including PCM, ADPCM, Microsoft GSM 06.10, CELP, SBC, Truespeech and MPEG Layer-3.

Format Bitrate [kbit/s][28] 1 Minute = [KiB][29] Sample

11,025 Hz 16 bit PCM 176.4 1292 11k16bitpcm.wav

8,000 Hz 16 bit PCM 128 938 8k16bitpcm.wav

11,025 Hz 8 bit PCM 88.2 646 11k8bitpcm.wav

11,025 Hz µ-Law 88.2 646 11kulaw.wav

8,000 Hz 8 bit PCM 64 469 8k8bitpcm.wav

8,000 Hz µ-Law 64 469 8kulaw.wav

11,025 Hz 4 bit ADPCM 44.1 323 11kadpcm.wav

8,000 Hz 4 bit ADPCM 32 234 8kadpcm.wav

11,025 Hz GSM 06.10 18 132 11kgsm.wav

8,000 Hz MP3 16 kbit/s 16 117 8kmp316.wav

8,000 Hz GSM 06.10 13 103 8kgsm.wav

8,000 Hz Lernout & Hauspie SBC 12 kbit/s 12 88 8ksbc12.wav

8,000 Hz DSP Group Truespeech 9 66 8ktruespeech.wav

8,000 Hz MP3 8 kbit/s 8 60 8kmp38.wav

8,000 Hz Lernout & Hauspie CELP 4.8 35 8kcelp.wav

The above are WAV files; even those that use MP3 compression have the ".wav" extension.

See also

Comparison of audio coding formats

Audio Compression Manager

Broadcast Wave Format (BWF)

RF64, an extended file format for audio (multichannel file format enabling file sizes to exceed 4 gigabytes)

Notes

A phased migration to 48 kHz sample rate has been announced. See http://guidelines.gateway.bbc.co.uk/dq/radio/delivery.shtml[dead link]. BBC English Regions already uses 48 kHz.

References

Microsoft Corporation (June 1998). "WAVE and AVI Codec Registries - RFC 2361". IETF. Retrieved 2009-12-06.

http://filext.com/file-extension/WAV

IBM Corporation and Microsoft Corporation (August 1991), Multimedia Programming Interface and Data Specifications 1.0 (TXT), retrieved 2009-12-06

P. Kabal (2006-06-19). "Audio File Format Specifications - WAVE or RIFF WAVE sound file". McGill University. Retrieved 2010-03-16.

"Multiple Channel Audio Data and WAVE Files". Microsoft Corporation. 2007-03-07. Retrieved 2010-03-16.

IBM Corporation and Microsoft Corporation (August 1991). "Multimedia Programming Interface and Data Specifications 1.0". Retrieved 2009-12-06.

Library of Congress (2008-09-12). "WAVE Audio File Format". Retrieved 2009-12-06.

Microsoft Corporation (June 20, 1999). "Waveform Audio File Format, MIME Sub-type Registration - INTERNET-DRAFT". IETF. Retrieved 2009-12-06.

"Information about the Multimedia file types that Windows Media Player supports". Microsoft Help and Support. Microsoft Corporation. 12 May 2008. Retrieved 29 May 2009. "Windows uses the Wave Form Audio (WAV) file format to store sounds as waveforms. One minute of Pulse Code Modulation (PCM)-encoded sound can occupy as little as 644 kilobytes (KB) or as much as 27 megabytes (MB) of storage."

EBU (July 2009), EBU Tech 3306 - MBWF / RF64: An Extended File Format for Audio (PDF), retrieved 2010-01-19

IBM; Microsoft (August 1991), Multimedia Programming Interface and Data Specifications 1.0

IBM & Microsoft 1991, p. 1-1, "The main advantage of RIFF is its extensibility; file formats based on RIFF can be future-proofed, as format changes can be ignored by existing applications."

IBM & Microsoft 1991, PDF p. 56, "Programs must expect (and ignore) any unknown chunks encountered, as with all RIFF forms."

IBM & Microsoft 1991, PDF p. 60 shows an example WAV file with an INFO chunk in this position.

IBM & Microsoft 1991, pp. 2-17 to 2-18

IBM & Microsoft 1991, pp. 2–18

Microsoft Multimedia Standards Update, New Multimedia Data Types and Data Techniques, Revision 3.0, April 15, 1994, page 6.

IBM & Microsoft 1991, PDF p. 56

IBM & Microsoft 1991, PDF p. 56 specifies sequencing information in the RIFF form of a WAV file consistent with the formalism: "However, must always occur before , and both of these chunks are mandatory in a WAVE file."

IBM & Microsoft 1991, PDF p. 23

IBM & Microsoft 1991, PDF p. 21, INAM appears before ICOP

Specification from IBM & Microsoft 1991 which also describes how the production syntax is interpreted.

"WAVE File Format". archive.org. 1999-11-15. Archived from the original on 1999-11-15. Retrieved 2010-03-16.

"WAVE PCM soundfile format". archive.org. 2003-01-20. Retrieved 2010-03-16.

"The WAVE File Format". Retrieved 2010-03-16.

1 GB = 1024 MB; 1 MB = 1024 KB; 1 KB = 1024 B

http://ltspice.linear.com/software/scad3.pdf#page=98

1 kbit = 1000 bit

1 KiB (kibibyte) = 1024 B (bytes)

External links

WAVE file format specifications - from McGill University, (Last update: 2011-01-03)

A summary of the WAVE file format

WAVE_FORMAT_EXTENSIBLE Specification from Microsoft (Updated on March 7, 2007)

More information on WAVE_FORMAT_EXTENSIBLE - University of Bath

WAVE File Format - technical details (1999)

WAV & BWF Metadata Guide

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Multimedia compression and container formats

Video

compression

ISO/IEC

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Part 2

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Lagarith

OMS Video

Pixlet

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Animation

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RealVideo

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SheerVideo

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Audio

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ITU-T

G.711

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SD2

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Image

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IEC, ISO,

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GXF

IFF

M2TS

Matroska

WebM

MXF

Ogg

QuickTime File Format

RatDVD

RealMedia

REDCODE

RIFF

WAV

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VOB, IFO and BUP

See Compression methods for methods and Compression software for codecs

Categories:

Container formats

Digital audio

Microsoft Windows multimedia technology

Preluat de pe Wikipedia.

[Mariutim]

Acum Red Book(compact disc digital audio).Are similaritati cu formatul WAV insa si diferente,cea mai mare ca nu poate fi redat de PC-uri decat transformat in WAV.

Compact Disc Digital Audio (CDDA or CD-DA) is the standard format for audio compact discs. The standard is defined in the Red Book, one of a series of "Rainbow Books" (named for their binding colors) that contain the technical specifications for all CD formats.

Standard

The Red Book specifies the physical parameters and properties of the CD, the optical "stylus" parameters, deviations and error rate, modulation system (eight-to-fourteen modulation, EFM) and error correction facility (cross-interleaved Reed–Solomon coding, CIRC), and the eight subcode channels. These parameters are common to all compact discs and used by all logical formats, such as CD-ROM. The standard also specifies the form of digital audio encoding: 2-channel signed 16-bit Linear PCM sampled at 44,100 Hz. Although rarely used, the specification allows for discs to be mastered with a form of emphasis.

The first edition of the Red Book was released in 1980 by Philips and Sony;[1][2] it was adopted by the Digital Audio Disc Committee and ratified as IEC 60908 (published in 1987).[3] The second edition of IEC 60908 was published in 1999[4] and it cancels and replaces the first edition, amendment 1 (1992) and the corrigendum to amendment 1. The IEC 60908 however does not contain all the information for extensions that is available in the Red Book, such as the details for CD-Text, CD+G and CD+EG.[5][6]

The standard is not freely available and must be licensed. It is available from Philips and the IEC. As of 2013, Philips outsources licensing of the standard to Adminius, which charges US$100 for the Red Book, plus US$50 each for the Subcode Channels R-W and CD Text Mode annexes.[7] As of 2013, the IEC 60908 document is available as a PDF download for US$372.[8]

Basic specifications

The basic specifications state that:

Maximum playing time is 79.8 minutes[9]

Minimum duration for a track is 4 seconds (including 2-second pause)

Maximum number of tracks is 99

Maximum number of index points (subdivisions of a track) is 99 with no maximum time limit

International Standard Recording Code (ISRC) should be included

Audio format

The audio contained in a CD-DA consists of two-channel signed 16-bit Linear PCM sampled at 44,100 Hz.

Sample rate

For more details on this topic, see 44.1 kHz.

The sampling rate is adapted from that attained when recording digital audio on a PAL (or NTSC) videotape with a PCM adaptor, an earlier way of storing digital audio.[10] An audio CD can represent frequencies up to 22.05 kHz, the Nyquist frequency of the 44.1 kHz sample rate.

The selection of the sample rate was based primarily on the need to reproduce the audible frequency range of 20–20,000 Hz (20 kHz). The Nyquist–Shannon sampling theorem states that a sampling rate of more than twice the maximum frequency of the signal to be recorded is needed, resulting in a required rate of at least 40 kHz. The exact sampling rate of 44.1 kHz was inherited from a method of converting digital audio into an analog video signal for storage on U-matic video tape, which was the most affordable way to transfer data from the recording studio to the CD manufacturer at the time the CD specification was being developed. The device that converts an analog audio signal into PCM audio, which in turn is changed into an analog video signal is called a PCM adaptor. This technology could store six samples (three samples per stereo channel) in a single horizontal line. A standard NTSC video signal has 245 usable lines per field, and 59.94 fields/s, which works out to be 44,056 samples/s/stereo channel. Similarly, PAL has 294 lines and 50 fields, which gives 44,100 samples/s/stereo channel. This system could store 14-bit samples with some error correction, or 16-bit samples with almost no error correction.[citation needed]

There was a long debate over the use of 14-bit (Philips) or 16-bit (Sony) quantization, and 44,056 or 44,100 samples/s (Sony) or approximately 44,000 samples/s (Philips). When the Sony/Philips task force designed the Compact Disc, Philips had already developed a 14-bit D/A converter (DAC), but Sony insisted on 16-bit. In the end, 16 bits and 44.1 kilosamples per second prevailed. Philips found a way to produce 16-bit quality using its 14-bit DAC by using four times oversampling.[citation needed]

Pre-emphasis

Main article: Emphasis (telecommunications)

Some CDs are mastered with pre-emphasis, an artificial boost of the treble or high frequencies. This serves to improve the apparent signal to noise ratio, by filtering out some the high frequency noise on playback. On playback, the player applies a low-pass filter to restore the frequency response curve to an overall linear one. Pre-emphasis time constants are 50uS or 15uS, and a binary flag in the disc subcode instructs the player to apply de-emphasis filtering if appropriate. Playback of such discs in a computer or 'ripping' to wave files typically does not take into account the pre-emphasis, so such files play back with a distorted frequency response.

Storage capacity and playing time

The creators of the CD originally aimed at a playing time of 60 minutes with a disc diameter of 100 mm (Sony) or 115 mm (Philips).[11] Sony vice-president Norio Ohga suggested extending the capacity to 74 minutes to accommodate Wilhelm Furtwängler's recording of Ludwig van Beethoven's Symphony No. 9 from the 1951 Bayreuth Festival.[12][13] The additional 14-minute playing time subsequently required changing to a 120 mm disc. Kees Immink, Philips' chief engineer, however, denies this, claiming that the increase was motivated by technical considerations, and that even after the increase in size, the Furtwängler recording would not have fit on one of the earliest CDs.[11][14]

According to a Sunday Tribune interview,[15] the story is slightly more involved. In 1979, Philips owned PolyGram, one of the world's largest distributors of music. PolyGram had set up a large experimental CD plant in Hannover, Germany, which could produce huge numbers of CDs having a diameter of 115 mm. Sony did not yet have such a facility. If Sony had agreed on the 115-mm disc, Philips would have had a significant competitive edge in the market. The long playing time of Beethoven's Ninth Symphony imposed by Ohga was used to push Philips to accept 120 mm, so that Philips' PolyGram lost its edge on disc fabrication.[15]

The 74-minute playing time of a CD, which was longer than the 22 minutes per side[16][17] typical of long-playing (LP) vinyl albums, was often used to the CD's advantage during the early years when CDs and LPs vied for commercial sales. CDs would often be released with one or more bonus tracks, enticing consumers to buy the CD for the extra material. However, attempts to combine double LPs onto one CD occasionally resulted in the opposite situation in which the CD would actually offer fewer tracks than the equivalent LP, though bonus tracks were also added to CD re-releases of double LPs as well.

Playing times beyond 74 minutes are achieved by decreasing track pitch beyond the original Red Book standard. Most players can accommodate the more closely spaced data.[18] Christian Thielemann's live Deutsche Grammophon recording of Bruckner's Fifth with the Munich Philharmonic in 2004 clocks at 82:34.[19] The Kirov Orchestra recording of Pyotr Ilyich Tchaikovsky's The Nutcracker conducted by Valery Gergiev and released by Philips/PolyGram Records (catalogue number 462 114) on October 20, 1998, clocks at 81:14.[citation needed] The Mission of Burma compilation album Mission of Burma, released in 1988 by Rykodisc, previously held the record at 80:08.[20]

Current manufacturing processes allow an audio CD to contain up to 80 minutes (variable from one replication plant to another) without requiring the content creator to sign a waiver releasing the plant owner from responsibility if the CD produced is marginally or entirely unreadable by some playback equipment. Thus, in current practice, maximum CD playing time has crept higher by reducing minimum engineering tolerances; by and large, this has not unacceptably reduced reliability.[citation needed]

Technical specifications

Data encoding

Each audio sample is a signed 16-bit two's complement integer, with sample values ranging from −32768 to +32767. The source audio data is divided into frames, containing twelve samples each (six left and right samples, alternating), for a total of 192 bits (24 bytes) of audio data per frame.

This stream of audio frames, as a whole, is then subjected to CIRC encoding, which segments and rearranges the data and expands it with parity bits in a way that allows occasional read errors to be detected and corrected. CIRC encoding also interleaves the audio frames throughout the disc over several consecutive frames so that the information will be more resistant to burst errors. Therefore, a physical frame on the disc will actually contain information from multiple logical audio frames. This process adds 64 bits of error correction data to each frame. After this, 8 bits of subcode or subchannel data are added to each of these encoded frames, which is used for control and addressing when playing the CD.

CIRC encoding plus the subcode byte generate 33-bytes long frames, called "channel-data" frames. These frames are then modulated through eight-to-fourteen modulation (EFM), where each 8-bit word is replaced with a corresponding 14-bit word designed to reduce the number of transitions between 0 and 1. This reduces the density of physical pits on the disc and provides an additional degree of error tolerance. Three "merging" bits are added before each 14-bit word for disambiguation and synchronization. In total there are 33 × (14 + 3) = 561 bits. A 27-bit word (a 24-bit pattern plus 3 merging bits) is added to the beginning of each frame to assist with synchronization, so the reading device can locate frames easily. With this, a frame ends up containing 588 bits of "channel data" (which are decoded to only 192 bits music).

The frames of channel data are finally written to disc physically in the form of pits and lands, with each pit or land representing a series of zeroes, and with the transition points—the edge of each pit—representing 1.

Data structure

This image of a CD-R demonstrates some of the visible features of an audio CD, including the lead-in, program area, and lead-out. A microscopic spiral of digital information[21] begins near the disc's middle and ends near the edge. Data-free areas of the disc and silent portions of the spiral reflect light differently, sometimes allowing track boundaries to be seen

The audio data stream in an audio CD is continuous, but has three parts. The main portion, which is further divided into playable audio tracks, is the program area. This section is preceded by a lead-in track and followed by a lead-out track. The lead-in and lead-out tracks encode only silent audio, but all three sections contain subcode data streams.

The lead-in's subcode contains repeated copies of the disc's Table Of Contents (TOC), which provides an index of the start positions of the tracks in the program area and lead-out. The track positions are referenced by absolute timecode, relative to the start of the program area, in MSF format: minutes, seconds, and fractional seconds called frames. Each timecode frame is one seventy-fifth of a second, and corresponds to a block of 98 channel-data frames—ultimately, a block of 588 pairs of left and right audio samples. Timecode contained in the subchannel data allows the reading device to locate the region of the disc that corresponds to the timecode in the TOC. The TOC on discs is analogous to the partition table on hard drives. Nonstandard or corrupted TOC records are abused as a form of CD/DVD copy protection, in e.g. the key2Audio scheme.

Tracks

Main article: Track (CD) § Audio tracks

The largest entity on a CD is called a track. A CD can contain up to 99 tracks (including a data track for mixed mode discs). Each track can in turn have up to 100 indexes, though players which handle this feature are rarely found outside of pro audio, particularly radio broadcasting[citation needed]. The vast majority of songs are recorded under index 1, with the pre-gap being index 0. Sometimes hidden tracks are placed at the end of the last track of the disc, often using index 2 or 3. This is also the case with some discs offering "101 sound effects", with 100 and 101 being indexed as two and three on track 99. The index, if used, is occasionally put on the track listing as a decimal part of the track number, such as 99.2 or 99.3. (Information Society's Hack was one of very few CD releases to do this, following a release with an equally obscure CD+G feature.) The track and index structure of the CD were carried forward to the DVD format as title and chapter, respectively.

Tracks, in turn, are divided into timecode frames (or sectors), which are further subdivided into channel-data frames.

Frames and timecode frames

Main article: Track (CD) § Sector structure

The smallest entity in a CD is a channel-data frame, which consists of 33 bytes and contains six complete 16-bit stereo samples: 24 bytes for the audio (two bytes × two channels × six samples = 24 bytes), eight CIRC error-correction bytes, and one subcode byte. As described in the "Data encoding" section, after the EFM modulation the number of bits in a frame totals 588.

On a Red Book audio CD, data is addressed using the MSF scheme, with timecodes expressed in minutes, seconds and another type of frames (mm:ss:ff), where one frame corresponds to 1/75th of a second of audio: 588 pairs of left and right samples. This timecode frame is distinct from the 33-byte channel-data frame described above, and is used for time display and positioning the reading laser. When editing and extracting CD audio, this timecode frame is the smallest addressable time interval for an audio CD; thus, track boundaries only occur on these frame boundaries. Each of these structures contains 98 channel-data frames, totaling 98 × 24 = 2,352 bytes of music. The CD is played at a speed of 75 frames (or sectors) per second, thus 44,100 samples or 176,400 bytes per second.

In the 1990s, CD-ROM and related Digital Audio Extraction (DAE) technology introduced the term sector to refer to each timecode frame, with each sector being identified by a sequential integer number starting at zero, and with tracks aligned on sector boundaries. An audio CD sector corresponds to 2,352 bytes of decoded data. The Red Book does not refer to sectors, nor does it distinguish the corresponding sections of the disc's data stream except as "frames" in the MSF addressing scheme.

The following table shows the relation between tracks, timecode frames (sectors) and channel-data frames:

Track level Track N

Timecode frame or sector level Timecode frame or sector 1 (2,352 b of data) Timecode frame or sector 2 (2,352 b of data) ...

Channel-data frame level Channel-data frame 1 (24 b of data) ... Channel-data frame 98 (24 b of data) ... ...

Bit rate

The audio bit rate is 1,411.2 kbit/s (as 2 channels × 44,100 samples per second per channel × 16 bits per sample = 1,411,200 bit/s = 1,411.2 kbit/s). Likewise, in a computer, audio data coming in from a CD drive is accessed by sectors, each sector being 2,352 bytes, and with 75 sectors containing 1 second of audio, for the same bit rate of 2,352 × 75 = 176.4 KiB/s (1,411.2 kbit/s). In comparison, the bit rate of a "1×" CD-ROM is defined as 2,048 bytes per sector × 75 sectors per second = 150 KiB/s (1,228.8 kbit/s). The undecoded channel-data rate for a Red Book audio CD is 4.3218 Mbit/s, with 2.0338 Mbit/s being the rate of the undecoded audio and subcode.

Data access from computers

Unlike on a DVD or CD-ROM, there are no "files" on a Red Book audio CD; there are only the physical pits and lands, which in turn represent a single encoded data stream, which ultimately represents one continuous stream of LPCM audio data, and a parallel, smaller set of 8 subcode data streams. Computer operating systems, however, may provide access to an audio CD as if it contains files. For example, Windows represents the CD's Table of Contents as a set of Compact Disc Audio track (CDA) files, each file containing indexing information, not audio data.

In a process called ripping, digital audio extraction software can be used to read CD-DA audio data and store it in files. Common audio file formats for this purpose include WAV and AIFF, which simply preface the LPCM data with a short header; FLAC, ALAC, and Windows Media Audio Lossless, which compress the LPCM data in ways that conserve space yet allow it to be restored without any changes; and various lossy, perceptual coding formats like MP3 and AAC, which modify and compress the audio data in ways that irreversibly change the audio, but that exploit features of human hearing to make the changes difficult to discern.

Preluat de pe wikipedia.

Desi lecturarea celor trei formate poate parea plictisitoare,va asigur ca este extrem de folositoare.

[bibi]

Cred că era mai bine dacă făceai un rezumat sau îți prezentai opinia într-un mod care să poată fi urmărit ușor de către publicul larg

[johndoe21ro]

Am intalnit deseori oameni care auzeau' date=' judecau [i']emotional[/i] sunetul, incat am ajuns sa privesc cu rezerva afirmatiile ce se bazeaza exclusiv pe experienta emotionala. Pe http://www.whatsbestforum.com/ gasiti discutii purtate intre ingineri de studio si pasionati cu o singura concluzie, nu exista diferenta intre flac si wav. Pot aparea diferente in situatia in care se foloseste un soft prost. Ca sa evitati acest lucru incercati http://www.jriver.com/ si va garantez ca diferentele vor disparea.

Nu exista diferente intre flac si wav.Softurile pot modifica marimea fisierului,insa nu e auzibil .Am facut teste Flac si Wav redate si de pe Pc si de pe disc.Nu sunt diferente audio,sau eu nu le aud.Dar nu toti oamenii aud la fel.

 

Cu unele concluzii trase sunt de acord si eu!

[Phaenius]

Acum Red Book(compact disc digital audio).Are similaritati cu formatul WAV insa si diferente,cea mai mare ca nu poate fi redat de PC-uri decat transformat in WAV.

 

Eu cred ca unitatile vechi puteau citi direct, fara a converti in date. Primele unitati CD-ROM foloseau CLV, erau unitati de CD playere modificate (unele aveau si butoane de play, fast forward, pause, etc. si chiar iesiri audio stereo) si nu puteau citi datele de pe CD-uri, le puteau doar rula, la single speed. Pe scurt, nu puteai face RIP, ma gandesc ca daca ar fi convertit in WAV, ar fi putut sa le si scrie pe HDD. Abia ulterior, dupa ce s-a dat o lupta crancena intre producatorii de CD-ROM-uri si casele de discuri, toate unitatile CD-ROM au putut citi muzica, nu doar rula. In clipa de fata insa, cred ca toate player-ele de PC citesc discurile audio in modul date, chiar si la rulare.

[dinica]

Eu nu stiu ce poate face un soft prost, atat timp cat unui FLAC nu trebuie sa-i faci decat un singur lucru cand il redai: sa-l decomprimi. E ca si cum ai arhiva cu RAR un fisier si apoi ar conta cu ce il decomprimi, Winrar, 7ZIP, Winace, etc.

 

Gresit. Factorul timp este foarte important in audio. Viteza trebuie sa fie atat de mare incat partea de extragere a datelor sa nu afecteze streamul audio. Ai pornit playerul, muzica trebuie sa se auda in acel moment, nu peste cateva secunde. In aceasta situatie pot exista programe care o fac precis, la timp, si programe care adauga erori. Aici ca si in cazul cititoarelor optice totul trebuie sa se petreaca la timp si fara sa existe erori si/ sau aproximari. Erorile acestea sunt jitter digital. Acesta din urma ajunge in convertorul digital - analogic si vorba aceea "caca bagi, caca iese'. Daca dupa asta mai adaugi si jitter-ul de dupa partea de conversie, situatia devine chiar dramatica.

[Phaenius]

Eu nu am gasit un player sa nu ruleze instant. In plus, piesele nu pornesc chiar instant, undele au un mic fade-in, si oricum la inceput de melodie e o mica pauza. Recomanda-mi cel mai prost player pe care il stii, sa verific. Eu zic ca toate fac aceeasi treaba.

[dinica]

Cred ca am explicat suficient de elocvent. Diferentele wav / flac pot aparea daca softul folosit induce jitter. Acesta apare daca algoritmul folosit nu este suficient de precis. Faptul ca tu poti asculta instant, nu este o garantie a calitatii. Pentru ca totul trebuie raportat la timp, este posibil ca unele softuri sa lucreze cu aproximari matematice. Nu sunt de specialitate, asa ca nu pot sa iti insir acum o intreaga demonstratie a modului in care lucreaza un asemenea soft in domeniul digital, insa poti cauta pe net.

Daca vrei detalii si demonstratii vorbeste cu Nucu de la

http://audiobyte.net/ si http://www.rockna-audio.com/

[Phaenius]

Eu vreau un soft prost, care sa dovedeasca ceea ce ai spus tu.

[hersorin]

Nu ma intereseaza polemici cu altii dar cei care aud diferente FLAC-WAV acestea exista .

Si eu intreb, fara dorinta de a starni o polemica, de unde vin acele diferente, atat timp cat FLAC-ul contine exact aceeasi informatie ca WAV-ul, doar ca e comprimat fara pierderi.

Cred ca sunt mai multe cauze care sincer nu le stiu. Poate conteaza codeci playerului (eu folosesc Foobar) , poate conteaza acea fractiune de secunda pentru dezarhivarea unui flac (unele flacuri transformate in wav isi dubleaza spatiul deci ar fi o umplere de 20-30 mb)...

Pentru mine e clar ca exista diferente audibile si este si logic.

Sunt multe lucruri care chiar nu au logica in audio si totusi unii sustin ca aud diferente (suport de cabluri , cubulete de lemn puse prin camera si alte destule uleiuri ).

[Mariutim]

Am intalnit deseori oameni care auzeau' date=' judecau [i']emotional[/i] sunetul, incat am ajuns sa privesc cu rezerva afirmatiile ce se bazeaza exclusiv pe experienta emotionala. Pe http://www.whatsbestforum.com/ gasiti discutii purtate intre ingineri de studio si pasionati cu o singura concluzie, nu exista diferenta intre flac si wav. Pot aparea diferente in situatia in care se foloseste un soft prost. Ca sa evitati acest lucru incercati http://www.jriver.com/ si va garantez ca diferentele vor disparea.

Nu exista diferente intre flac si wav.Softurile pot modifica marimea fisierului,insa nu e auzibil .Am facut teste Flac si Wav redate si de pe Pc si de pe disc.Nu sunt diferente audio,sau eu nu le aud.Dar nu toti oamenii aud la fel.

 

Cu unele concluzii trase sunt de acord si eu!

Stai linistit ca aud foarte bine.

[bibi]

offtopic

Am o rugăminte - încercați, vă rog, să nu mai citați în citat în citat .... Este adevărat că necesită un mic efort de editare (softul forumului preia citatele din citatul dorit) dar după 2-3 iterații devine destul de obositor.

Vă mulțumesc anticipat pentru înțelegere

[Mariutim]

Cred că era mai bine dacă făceai un rezumat sau îți prezentai opinia într-un mod care să poată fi urmărit ușor de către publicul larg

Cer scuze ca am consumat din spatuiul forumului.Parerea mea nu conteaza,e mai credibila declaratia unor specialisti din domeniu.Insa unii chiar au nevoie sa descopere ce inseamna cele trei formate.Puteti sterge cele trei postari,sunt dificil de citit si de inteles.

Cu respect.

[bibi]

Cer scuze ca am consumat din spatuiul forumului.

 

Nu era vorba despre consumat spațiu pe forum

Am vrut doar să spun că este greu de citit de cineva care nu este cu adevărat interesat (eu, de exemplu, am citit o parte din materialul acela cu ceva timp în urmă, dar sunt și multe informații care nu sunt neapărat necesare și diluează informația utilă, în opinia mea).