Speex FAQ - Revision history

Saoshyant at 04:48, 4 February 2008

2008-02-04T04:48:49Z

← Older revision		Revision as of 04:48, 4 February 2008
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	There is not (yet) a complete description of the algorithm (except for the source code itself). However, several aspects are documented either in the [http://speex.org/docs/ manual] or in the paper [http://people.xiph.org/~jm/papers/aes120_speex_vorbis.pdf Improved Noise Weighting in CELP Coding of Speech — Applying the Vorbis Psychoacoustic Model To Speex].		There is not (yet) a complete description of the algorithm (except for the source code itself). However, several aspects are documented either in the [http://speex.org/docs/ manual] or in the paper [http://people.xiph.org/~jm/papers/aes120_speex_vorbis.pdf Improved Noise Weighting in CELP Coding of Speech — Applying the Vorbis Psychoacoustic Model To Speex].
		+
		+	[[Category:Speex]]

Saoshyant: /* Using libspeex */ added question regarding 1.0 vs 1.2

2007-10-26T02:42:20Z

Using libspeex: added question regarding 1.0 vs 1.2

← Older revision		Revision as of 02:42, 26 October 2007
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	Yes, but you must resample the signal to a supported sample rate before encoding. Speex is optimised for 8 kHz and 16 kHz and it can (your mileage may vary) encode 32 kHz files as well. Anything else is unsupported and tends to be heavily sub-optimal. You might as well use Vorbis instead. Note that Speex includes a resampler module as of version 1.2beta2. Refer to the section called Resampler in the Speex documentation.		Yes, but you must resample the signal to a supported sample rate before encoding. Speex is optimised for 8 kHz and 16 kHz and it can (your mileage may vary) encode 32 kHz files as well. Anything else is unsupported and tends to be heavily sub-optimal. You might as well use Vorbis instead. Note that Speex includes a resampler module as of version 1.2beta2. Refer to the section called Resampler in the Speex documentation.
		+
		+	=== Should I use version 1.0.x or 1.2? ===
		+	While currently 1.2 is marked as beta, the author believes 1.2 is vastly superior and more stable than the 1.0.x series regarding compression and API performance. As such, it is recommended to use version 1.2, even though it is marked as beta.

	== Technical ==		== Technical ==

Jmspeex: reverted most of Saoshyant's changes

2007-10-15T12:30:09Z

reverted most of Saoshyant's changes

Tgrand: Added note about resampler

2007-10-14T16:14:31Z

Added note about resampler

← Older revision		Revision as of 16:14, 14 October 2007
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	=== Can I use Speex for 22.05 kHz, 44.1 kHz or 48 kHz signals? ===		=== Can I use Speex for 22.05 kHz, 44.1 kHz or 48 kHz signals? ===

-	Yes~~... as long as~~ you ~~don't care about how it sounds~~. ~~Seriously,~~ Speex is optimised for 8 kHz and 16 kHz and it can (your mileage may vary) encode 32 kHz files as well. Anything else is unsupported and tends to be heavily sub-optimal. You might as well use Vorbis instead.	+	Yes, but you must resample the signal to a supported sample rate before encoding. Speex is optimised for 8 kHz and 16 kHz and it can (your mileage may vary) encode 32 kHz files as well. Anything else is unsupported and tends to be heavily sub-optimal. You might as well use Vorbis instead. Note that Speex includes a resampler module as of version 1.2beta2. Refer to the section called Resampler in the Speex documentation.

	== Technical ==		== Technical ==

Jmspeex at 14:56, 10 October 2007

2007-10-10T14:56:22Z

← Older revision		Revision as of 14:56, 10 October 2007
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	No. I have written Speex while pursuing my Ph.D. at the University of Sherbrooke (2002-2005) in mobile robotics. Although I did my master with the Sherbrooke speech coding group (in speech enhancement, not coding), I was no longer associated with them when developing Speex. Furthermore, Speex does not make use of any code or proprietary technology developed in the Sherbrooke speech coding group.		No. I have written Speex while pursuing my Ph.D. at the University of Sherbrooke (2002-2005) in mobile robotics. Although I did my master with the Sherbrooke speech coding group (in speech enhancement, not coding), I was no longer associated with them when developing Speex. Furthermore, Speex does not make use of any code or proprietary technology developed in the Sherbrooke speech coding group.
		+
		+	=== When will version X of Speex be released ===
		+
		+	When it's ready, which will be sooner if you help.
		+
		+	=== How can I help if I don't know about speech processing? ===
		+
		+	There's always "semi-technical" work to do. The documentation can be improved, and so can this FAQ.

	== License ==		== License ==

Jmspeex: license

2007-10-10T14:52:16Z

license

← Older revision		Revision as of 14:52, 10 October 2007
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	Yes. As long as you comply with the license. This basically means you have to keep the copyright notice and you can't use our name to promote your product without authorization.		Yes. As long as you comply with the license. This basically means you have to keep the copyright notice and you can't use our name to promote your product without authorization.
		+
		+	=== Do I have to release source code if I use Speex in a proprietary application? ===
		+
		+	No. The BSD license does '''not''' require you to release any source code. It is however '''appreciated''' (but not required) if you contribute back useful changes you make to Speex. This is generally also in your interest because it means you get maintenance of that code for free (i.e. no need to merge again in newer versions).

	== Using libspeex ==		== Using libspeex ==

Saoshyant: minor clean up

2007-10-05T13:06:39Z

minor clean up

Jmspeex: sampling rates

2007-10-05T12:04:34Z

sampling rates

← Older revision		Revision as of 12:04, 5 October 2007
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	...and many others		...and many others
		+
		+	=== Can I use Speex for 22.05 kHz, 44.1 kHz or 48 kHz signals? ===
		+
		+	Yes... as long as you don't care about how it sounds. Seriously, Speex is optimised for 8 kHz and 16 kHz and it can (your mileage may vary) encode 32 kHz files as well. Anything else is unsupported and tends to be heavily sub-optimal. You might as well use Vorbis instead.

	== Technical ==		== Technical ==

Jmspeex at 11:45, 5 October 2007

2007-10-05T11:45:44Z

← Older revision		Revision as of 11:45, 5 October 2007
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	There are many possible causes for that. One of them is errors in the way the bits are manipulated. Another possible cause is the use of the same encoder or decoder state for more than one audio stream (channel), which produces strange effects with the filter memories. If the input speech has an amplitude close to +-32767, it is possible that at decoding, the amplitude be a bit higher than that, causing clipping when saving as 16-bit PCM.		There are many possible causes for that. One of them is errors in the way the bits are manipulated. Another possible cause is the use of the same encoder or decoder state for more than one audio stream (channel), which produces strange effects with the filter memories. If the input speech has an amplitude close to +-32767, it is possible that at decoding, the amplitude be a bit higher than that, causing clipping when saving as 16-bit PCM.

-	=== Can Speex run on fixed-point processors or DSPs ===	+	=== Can Speex run on fixed-point processors or DSPs? ===

	Yes. You can compile Speex for fixed-point CPUs by passing the --enable-fixed-point option to the configure script or defining FIXED_POINT.		Yes. You can compile Speex for fixed-point CPUs by passing the --enable-fixed-point option to the configure script or defining FIXED_POINT.
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	=== What architectures are supported? ===		=== What architectures are supported? ===

-	Speex is designed to run on pretty much ~~and~~ CPU that can do 16x16 multiplications. That includes:	+	Speex is designed to run on pretty much any CPU that can do 16x16 multiplications. That includes:
	* x86		* x86
	* PowerPC		* PowerPC

Jmspeex: architectures

2007-10-05T11:44:18Z

architectures

← Older revision		Revision as of 11:44, 5 October 2007
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	There are many possible causes for that. One of them is errors in the way the bits are manipulated. Another possible cause is the use of the same encoder or decoder state for more than one audio stream (channel), which produces strange effects with the filter memories. If the input speech has an amplitude close to +-32767, it is possible that at decoding, the amplitude be a bit higher than that, causing clipping when saving as 16-bit PCM.		There are many possible causes for that. One of them is errors in the way the bits are manipulated. Another possible cause is the use of the same encoder or decoder state for more than one audio stream (channel), which produces strange effects with the filter memories. If the input speech has an amplitude close to +-32767, it is possible that at decoding, the amplitude be a bit higher than that, causing clipping when saving as 16-bit PCM.
		+
		+	=== Can Speex run on fixed-point processors or DSPs ===
		+
		+	Yes. You can compile Speex for fixed-point CPUs by passing the --enable-fixed-point option to the configure script or defining FIXED_POINT.
		+
		+	=== What architectures are supported? ===
		+
		+	Speex is designed to run on pretty much and CPU that can do 16x16 multiplications. That includes:
		+	* x86
		+	* PowerPC
		+	* ARM
		+	* Blackfin
		+	* TI C5x and C6x
		+	* dsPIC (unsupported, unofficial port)
		+	* Cell (in progress)
		+
		+	...and many others

	== Technical ==		== Technical ==

← Older revision		Revision as of 12:30, 15 October 2007
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	== General ==		== General ==

-	=== Vorbis is ~~free software~~ and patent-free; why do we need Speex? ===	+	=== Vorbis is open source and patent-free; why do we need Speex? ===

-	Vorbis is a great project but its goals are not the same as Speex. Vorbis is mostly aimed at compressing music and audio in general, while Speex targets ~~human~~ speech only. For that reason Speex can achieve much better results than Vorbis on speech, typically 2-4 times higher compression at equal quality.	+	Vorbis is a great project but its goals are not the same as Speex. Vorbis is mostly aimed at compressing music and audio in general, while Speex targets speech only. For that reason Speex can achieve much better results than Vorbis on speech, typically 2-4 times higher compression at equal quality.

-	=== Isn't there ~~a free software~~ implementation of the GSM-FR codec? Why is Speex necessary? ===	+	=== Isn't there an open source implementation of the GSM-FR codec? Why is Speex necessary? ===

	First of all, it's not clear whether GSM-FR is covered by a Philips patent (see http://kbs.cs.tu-berlin.de/~jutta/toast.html). Also, GSM-FR offers mediocre quality at a relatively high bit-rate, while Speex can offer equivalent quality at almost half the bit-rate. Last but not least, Speex offers a wide range of bit-rates and sampling rates, while GSM-FR is limited to 8 kHz speech at 13 kbps.		First of all, it's not clear whether GSM-FR is covered by a Philips patent (see http://kbs.cs.tu-berlin.de/~jutta/toast.html). Also, GSM-FR offers mediocre quality at a relatively high bit-rate, while Speex can offer equivalent quality at almost half the bit-rate. Last but not least, Speex offers a wide range of bit-rates and sampling rates, while GSM-FR is limited to 8 kHz speech at 13 kbps.
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	=== Ogg, Speex, Vorbis, what's the difference? ===		=== Ogg, Speex, Vorbis, what's the difference? ===

-	Ogg is a container format for holding multimedia data. Vorbis is an audio codec that uses Ogg to store its bit-streams as files, hence the name Ogg Vorbis. Speex also uses the Ogg format to store its bit-streams as files, so technically they would be "Ogg Speex" files, ~~but you should perceive~~ them as Speex files. One difference with Vorbis however, is that Speex is less tied with Ogg. Actually, if you just do Voice over IP (VoIP), you don't need Ogg at all.	+	Ogg is a container format for holding multimedia data. Vorbis is an audio codec that uses Ogg to store its bit-streams as files, hence the name Ogg Vorbis. Speex also uses the Ogg format to store its bit-streams as files, so technically they would be "Ogg Speex" files. However, most people refer to them as Speex files. One difference with Vorbis however, is that Speex is less tied with Ogg. Actually, if you just do Voice over IP (VoIP), you don't need Ogg at all.

	=== What's the extension for Speex files? ===		=== What's the extension for Speex files? ===
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	=== Under what license is Speex released? ===		=== Under what license is Speex released? ===

-	As of version 1.0 beta 1, Speex is released under the ~~new~~ BSD license. This license is one of the most permissive ~~free software~~ licenses.	+	As of version 1.0 beta 1, Speex is released under the revised (3-clause) BSD license. This license is one of the most permissive open source licenses.

	=== Am I allowed to use Speex in commercial software? ===		=== Am I allowed to use Speex in commercial software? ===
Line 73:		Line 73:
	For most kinds of compression, encoding is inherently slower than decoding. In the case of Speex, encoding consists of finding, for each vector of 5 to 10 samples, the entry that matches the best within a codebook consisting of 16 to 256 entries. On the other hand, at decoding all that needs to be done is look up the right entry in the codebook using the encoded index. Since a lookup is much faster than a search, the decoder works much faster than the encoder.		For most kinds of compression, encoding is inherently slower than decoding. In the case of Speex, encoding consists of finding, for each vector of 5 to 10 samples, the entry that matches the best within a codebook consisting of 16 to 256 entries. On the other hand, at decoding all that needs to be done is look up the right entry in the codebook using the encoded index. Since a lookup is much faster than a search, the decoder works much faster than the encoder.

-	=== Why is Speex so slow on my iPaq (or insert any platform without a FPU)? ===	+	=== Why is Speex so slow on my iPaq (or insert any platform without an FPU)? ===

	You probably didn't build Speex with the fixed-point option (--enable-fixed-point). Even if you did, not all modes have been ported to use fixed-point arithmetic, so the code may be slowed down by a few float operations left (e.g. in the wideband mode).		You probably didn't build Speex with the fixed-point option (--enable-fixed-point). Even if you did, not all modes have been ported to use fixed-point arithmetic, so the code may be slowed down by a few float operations left (e.g. in the wideband mode).

← Older revision		Revision as of 13:06, 5 October 2007
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	== General ==		== General ==

-	=== Vorbis is ~~open-source~~ and patent-free; why do we need Speex? ===	+	=== Vorbis is free software and patent-free; why do we need Speex? ===

-	Vorbis is a great project but its goals are not the same as Speex. Vorbis is mostly aimed at compressing music and audio in general, while Speex targets speech only. For that reason Speex can achieve much better results than Vorbis on speech, typically 2-4 times higher compression at equal quality.	+	Vorbis is a great project but its goals are not the same as Speex. Vorbis is mostly aimed at compressing music and audio in general, while Speex targets human speech only. For that reason Speex can achieve much better results than Vorbis on speech, typically 2-4 times higher compression at equal quality.

-	=== Isn't there ~~an open-source~~ implementation of the GSM-FR codec? Why is Speex necessary? ===	+	=== Isn't there a free software implementation of the GSM-FR codec? Why is Speex necessary? ===

	First of all, it's not clear whether GSM-FR is covered by a Philips patent (see http://kbs.cs.tu-berlin.de/~jutta/toast.html). Also, GSM-FR offers mediocre quality at a relatively high bit-rate, while Speex can offer equivalent quality at almost half the bit-rate. Last but not least, Speex offers a wide range of bit-rates and sampling rates, while GSM-FR is limited to 8 kHz speech at 13 kbps.		First of all, it's not clear whether GSM-FR is covered by a Philips patent (see http://kbs.cs.tu-berlin.de/~jutta/toast.html). Also, GSM-FR offers mediocre quality at a relatively high bit-rate, while Speex can offer equivalent quality at almost half the bit-rate. Last but not least, Speex offers a wide range of bit-rates and sampling rates, while GSM-FR is limited to 8 kHz speech at 13 kbps.
Line 11:		Line 11:
	=== Ogg, Speex, Vorbis, what's the difference? ===		=== Ogg, Speex, Vorbis, what's the difference? ===

-	Ogg is a container format for holding multimedia data. Vorbis is an audio codec that uses Ogg to store its bit-streams as files, hence the name Ogg Vorbis. Speex also uses the Ogg format to store its bit-streams as files, so technically they would be "Ogg/Speex" files ~~(I prefer to call~~ them ~~just~~ Speex files). One difference with Vorbis however, is that Speex is less tied with Ogg. Actually, if you just do Voice over IP (VoIP), you don't need Ogg at all.	+	Ogg is a container format for holding multimedia data. Vorbis is an audio codec that uses Ogg to store its bit-streams as files, hence the name Ogg Vorbis. Speex also uses the Ogg format to store its bit-streams as files, so technically they would be "Ogg Speex" files, but you should perceive them as Speex files. One difference with Vorbis however, is that Speex is less tied with Ogg. Actually, if you just do Voice over IP (VoIP), you don't need Ogg at all.

	=== What's the extension for Speex files? ===		=== What's the extension for Speex files? ===
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	=== Under what license is Speex released? ===		=== Under what license is Speex released? ===

-	As of version 1.0 beta 1, Speex is released under ~~Xiph's version of~~ the ~~revised (3-clause)~~ BSD license. This license is one of the most permissive ~~open-source~~ licenses.	+	As of version 1.0 beta 1, Speex is released under the new BSD license. This license is one of the most permissive free software licenses.

	=== Am I allowed to use Speex in commercial software? ===		=== Am I allowed to use Speex in commercial software? ===
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	For most kinds of compression, encoding is inherently slower than decoding. In the case of Speex, encoding consists of finding, for each vector of 5 to 10 samples, the entry that matches the best within a codebook consisting of 16 to 256 entries. On the other hand, at decoding all that needs to be done is look up the right entry in the codebook using the encoded index. Since a lookup is much faster than a search, the decoder works much faster than the encoder.		For most kinds of compression, encoding is inherently slower than decoding. In the case of Speex, encoding consists of finding, for each vector of 5 to 10 samples, the entry that matches the best within a codebook consisting of 16 to 256 entries. On the other hand, at decoding all that needs to be done is look up the right entry in the codebook using the encoded index. Since a lookup is much faster than a search, the decoder works much faster than the encoder.

-	=== Why is Speex so slow on my iPaq (or insert any platform without an FPU)? ===	+	=== Why is Speex so slow on my iPaq (or insert any platform without a FPU)? ===

	You probably didn't build Speex with the fixed-point option (--enable-fixed-point). Even if you did, not all modes have been ported to use fixed-point arithmetic, so the code may be slowed down by a few float operations left (e.g. in the wideband mode).		You probably didn't build Speex with the fixed-point option (--enable-fixed-point). Even if you did, not all modes have been ported to use fixed-point arithmetic, so the code may be slowed down by a few float operations left (e.g. in the wideband mode).
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	=== Where can I get information about how Speex works? ===		=== Where can I get information about how Speex works? ===

-	There is not (yet) a complete description of the algorithm (except for the source code itself). However, several aspects are documented either in the [http://~~www.~~speex.org/docs/ manual] or in the paper [http://people.xiph.org/~jm/papers/aes120_speex_vorbis.pdf Improved Noise Weighting in CELP Coding of Speech - Applying the Vorbis Psychoacoustic Model To Speex].	+	There is not (yet) a complete description of the algorithm (except for the source code itself). However, several aspects are documented either in the [http://speex.org/docs/ manual] or in the paper [http://people.xiph.org/~jm/papers/aes120_speex_vorbis.pdf Improved Noise Weighting in CELP Coding of Speech — Applying the Vorbis Psychoacoustic Model To Speex].