* Juli@ ICE1724 and 24 bit audio @ 2009-02-09 20:09 Demian Martin 2009-02-09 20:26 ` Pavel Hofman 0 siblings, 1 reply; 12+ messages in thread From: Demian Martin @ 2009-02-09 20:09 UTC (permalink / raw) To: alsa-devel; +Cc: pavel.hofman, pavel.hofman I have the driver working fine for sample rates from 44.1K to 196K however it seems to be truncating the data to 16 bits. I wasn't sure until I started testing with the HRx 176.4K 24 bit files that have an embedded HDCD flag in the LSB. The files work OK on Windoze systems (with a lot of low level settings tweaked) and they play fine but the flag isn't detected by a system that can detect them. Further looking at the data stream with a scope it seems the last 8 bits aren't changing. Is there anything I can do to control the driver to confirm this problem or change the playback settings to make it work? Demian Martin Product Design Services 784 Cary Drive San Leandro, CA 94577 209 613 6990 ^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: Juli@ ICE1724 and 24 bit audio 2009-02-09 20:09 Juli@ ICE1724 and 24 bit audio Demian Martin @ 2009-02-09 20:26 ` Pavel Hofman 2009-02-09 21:29 ` Demian Martin 0 siblings, 1 reply; 12+ messages in thread From: Pavel Hofman @ 2009-02-09 20:26 UTC (permalink / raw) To: Demian Martin; +Cc: alsa-devel, pavel.hofman Demian Martin napsal(a): > I have the driver working fine for sample rates from 44.1K to 196K however > it seems to be truncating the data to 16 bits. I wasn't sure until I started > testing with the HRx 176.4K 24 bit files that have an embedded HDCD flag in > the LSB. The files work OK on Windoze systems (with a lot of low level > settings tweaked) and they play fine but the flag isn't detected by a system > that can detect them. Further looking at the data stream with a scope it > seems the last 8 bits aren't changing. Is there anything I can do to control > the driver to confirm this problem or change the playback settings to make > it work? > > How do you play the files? Do you use the hw or plug:hw device? IIRC, the standard-setup dmix resampler is 16-bit only. I did tests with SPDIF OUT/IN and found it bit-perfect, for 24bit too. Regular ICE1724 cards do not output 176.4kHz SPDIF, only 88.2kHz. But Juli is not affected by that bug. Regards, Pavel. ^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: Juli@ ICE1724 and 24 bit audio 2009-02-09 20:26 ` Pavel Hofman @ 2009-02-09 21:29 ` Demian Martin 2009-02-09 21:38 ` Vedran Miletić 0 siblings, 1 reply; 12+ messages in thread From: Demian Martin @ 2009-02-09 21:29 UTC (permalink / raw) To: 'Pavel Hofman'; +Cc: alsa-devel Pavel: Thanks for all your help and work on this project. I'm using the Juli@ card because it supports 176.4KHz. The only other candidates are much more expensive and bring their own problems. I'm using Xine set for dolby/dts passthrough which seems to send the audio data at its native rate. The resampling is bypassed (Dolby Digital would be trashed otherwise). I will try playing the files from the command line when I am next in front of the system, tomorrow. This is the Asound.conf: Asound.conf (/etc) pcm.asym_spdif { type asym playback.pcm "plughw:0,1" capture.pcm "plughw:0" } pcm.!default asym_spdif Demian Martin PDS > -----Original Message----- > From: Pavel Hofman [mailto:pavel.hofman@insite.cz] > Sent: Monday, February 09, 2009 12:27 PM > To: Demian Martin > Cc: alsa-devel@alsa-project.org; pavel.hofman@insite.cz > Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio > > Demian Martin napsal(a): > > I have the driver working fine for sample rates from 44.1K to 196K > however > > it seems to be truncating the data to 16 bits. I wasn't sure until I > started > > testing with the HRx 176.4K 24 bit files that have an embedded HDCD flag > in > > the LSB. The files work OK on Windoze systems (with a lot of low level > > settings tweaked) and they play fine but the flag isn't detected by a > system > > that can detect them. Further looking at the data stream with a scope it > > seems the last 8 bits aren't changing. Is there anything I can do to > control > > the driver to confirm this problem or change the playback settings to > make > > it work? > > > > > > How do you play the files? Do you use the hw or plug:hw device? IIRC, > the standard-setup dmix resampler is 16-bit only. > > I did tests with SPDIF OUT/IN and found it bit-perfect, for 24bit too. > > Regular ICE1724 cards do not output 176.4kHz SPDIF, only 88.2kHz. But > Juli is not affected by that bug. > > Regards, > > > Pavel. ^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: Juli@ ICE1724 and 24 bit audio 2009-02-09 21:29 ` Demian Martin @ 2009-02-09 21:38 ` Vedran Miletić 2009-02-09 21:54 ` Demian Martin 0 siblings, 1 reply; 12+ messages in thread From: Vedran Miletić @ 2009-02-09 21:38 UTC (permalink / raw) To: Demian Martin; +Cc: alsa-devel, Pavel Hofman The plughw part is what makese it resample to 16-bit. You should just use hw. On Mon, Feb 9, 2009 at 10:29 PM, Demian Martin <demianm_1@yahoo.com> wrote: > Pavel: > Thanks for all your help and work on this project. > > I'm using the Juli@ card because it supports 176.4KHz. The only other > candidates are much more expensive and bring their own problems. > > I'm using Xine set for dolby/dts passthrough which seems to send the audio > data at its native rate. The resampling is bypassed (Dolby Digital would be > trashed otherwise). > > I will try playing the files from the command line when I am next in front > of the system, tomorrow. > > This is the Asound.conf: > Asound.conf (/etc) > > pcm.asym_spdif { > > type asym > > playback.pcm "plughw:0,1" > > capture.pcm "plughw:0" > > } > > > pcm.!default asym_spdif > > Demian Martin > PDS > >> -----Original Message----- >> From: Pavel Hofman [mailto:pavel.hofman@insite.cz] >> Sent: Monday, February 09, 2009 12:27 PM >> To: Demian Martin >> Cc: alsa-devel@alsa-project.org; pavel.hofman@insite.cz >> Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio >> >> Demian Martin napsal(a): >> > I have the driver working fine for sample rates from 44.1K to 196K >> however >> > it seems to be truncating the data to 16 bits. I wasn't sure until I >> started >> > testing with the HRx 176.4K 24 bit files that have an embedded HDCD flag >> in >> > the LSB. The files work OK on Windoze systems (with a lot of low level >> > settings tweaked) and they play fine but the flag isn't detected by a >> system >> > that can detect them. Further looking at the data stream with a scope it >> > seems the last 8 bits aren't changing. Is there anything I can do to >> control >> > the driver to confirm this problem or change the playback settings to >> make >> > it work? >> > >> > >> >> How do you play the files? Do you use the hw or plug:hw device? IIRC, >> the standard-setup dmix resampler is 16-bit only. >> >> I did tests with SPDIF OUT/IN and found it bit-perfect, for 24bit too. >> >> Regular ICE1724 cards do not output 176.4kHz SPDIF, only 88.2kHz. But >> Juli is not affected by that bug. >> >> Regards, >> >> >> Pavel. > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel@alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -- Vedran Miletić _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel ^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: Juli@ ICE1724 and 24 bit audio 2009-02-09 21:38 ` Vedran Miletić @ 2009-02-09 21:54 ` Demian Martin 2009-02-10 11:33 ` Vedran Miletić 0 siblings, 1 reply; 12+ messages in thread From: Demian Martin @ 2009-02-09 21:54 UTC (permalink / raw) To: 'Vedran Miletić'; +Cc: alsa-devel, 'Pavel Hofman' Thanks, I'll try that. I guess this e-mail will serve as more documentation on that feature (bug). I had not seen that factoid anywhere. Demian Martin PDS > -----Original Message----- > From: Vedran Miletić [mailto:rivanvx@gmail.com] > Sent: Monday, February 09, 2009 1:39 PM > To: Demian Martin > Cc: Pavel Hofman; alsa-devel@alsa-project.org > Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio > > The plughw part is what makese it resample to 16-bit. You should just use > hw. > > On Mon, Feb 9, 2009 at 10:29 PM, Demian Martin <demianm_1@yahoo.com> > wrote: > > Pavel: > > Thanks for all your help and work on this project. > > > > I'm using the Juli@ card because it supports 176.4KHz. The only other > > candidates are much more expensive and bring their own problems. > > > > I'm using Xine set for dolby/dts passthrough which seems to send the > audio > > data at its native rate. The resampling is bypassed (Dolby Digital would > be > > trashed otherwise). > > > > I will try playing the files from the command line when I am next in > front > > of the system, tomorrow. > > > > This is the Asound.conf: > > Asound.conf (/etc) > > > > pcm.asym_spdif { > > > > type asym > > > > playback.pcm "plughw:0,1" > > > > capture.pcm "plughw:0" > > > > } > > > > > > pcm.!default asym_spdif > > > > Demian Martin > > PDS > > > >> -----Original Message----- > >> From: Pavel Hofman [mailto:pavel.hofman@insite.cz] > >> Sent: Monday, February 09, 2009 12:27 PM > >> To: Demian Martin > >> Cc: alsa-devel@alsa-project.org; pavel.hofman@insite.cz > >> Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio > >> > >> Demian Martin napsal(a): > >> > I have the driver working fine for sample rates from 44.1K to 196K > >> however > >> > it seems to be truncating the data to 16 bits. I wasn't sure until I > >> started > >> > testing with the HRx 176.4K 24 bit files that have an embedded HDCD > flag > >> in > >> > the LSB. The files work OK on Windoze systems (with a lot of low > level > >> > settings tweaked) and they play fine but the flag isn't detected by a > >> system > >> > that can detect them. Further looking at the data stream with a scope > it > >> > seems the last 8 bits aren't changing. Is there anything I can do to > >> control > >> > the driver to confirm this problem or change the playback settings to > >> make > >> > it work? > >> > > >> > > >> > >> How do you play the files? Do you use the hw or plug:hw device? IIRC, > >> the standard-setup dmix resampler is 16-bit only. > >> > >> I did tests with SPDIF OUT/IN and found it bit-perfect, for 24bit too. > >> > >> Regular ICE1724 cards do not output 176.4kHz SPDIF, only 88.2kHz. But > >> Juli is not affected by that bug. > >> > >> Regards, > >> > >> > >> Pavel. > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel@alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > > -- > Vedran Miletić ^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: Juli@ ICE1724 and 24 bit audio 2009-02-09 21:54 ` Demian Martin @ 2009-02-10 11:33 ` Vedran Miletić 2009-02-10 15:34 ` Pavel Hofman 0 siblings, 1 reply; 12+ messages in thread From: Vedran Miletić @ 2009-02-10 11:33 UTC (permalink / raw) To: Demian Martin; +Cc: alsa-devel, Pavel Hofman Yeah, ALSA's documentation is actually quite lacking on many fronts and is mostly scattered around various wikis and mailing lists. But I guess it is as it is, complaining about it won't fix it. On 2/9/09, Demian Martin <demianm_1@yahoo.com> wrote: > Thanks, I'll try that. I guess this e-mail will serve as more documentation > on that feature (bug). I had not seen that factoid anywhere. > > Demian Martin > PDS > > >> -----Original Message----- >> From: Vedran Miletić [mailto:rivanvx@gmail.com] >> Sent: Monday, February 09, 2009 1:39 PM >> To: Demian Martin >> Cc: Pavel Hofman; alsa-devel@alsa-project.org >> Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio >> >> The plughw part is what makese it resample to 16-bit. You should just use >> hw. >> >> On Mon, Feb 9, 2009 at 10:29 PM, Demian Martin <demianm_1@yahoo.com> >> wrote: >> > Pavel: >> > Thanks for all your help and work on this project. >> > >> > I'm using the Juli@ card because it supports 176.4KHz. The only other >> > candidates are much more expensive and bring their own problems. >> > >> > I'm using Xine set for dolby/dts passthrough which seems to send the >> audio >> > data at its native rate. The resampling is bypassed (Dolby Digital would >> be >> > trashed otherwise). >> > >> > I will try playing the files from the command line when I am next in >> front >> > of the system, tomorrow. >> > >> > This is the Asound.conf: >> > Asound.conf (/etc) >> > >> > pcm.asym_spdif { >> > >> > type asym >> > >> > playback.pcm "plughw:0,1" >> > >> > capture.pcm "plughw:0" >> > >> > } >> > >> > >> > pcm.!default asym_spdif >> > >> > Demian Martin >> > PDS >> > >> >> -----Original Message----- >> >> From: Pavel Hofman [mailto:pavel.hofman@insite.cz] >> >> Sent: Monday, February 09, 2009 12:27 PM >> >> To: Demian Martin >> >> Cc: alsa-devel@alsa-project.org; pavel.hofman@insite.cz >> >> Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio >> >> >> >> Demian Martin napsal(a): >> >> > I have the driver working fine for sample rates from 44.1K to 196K >> >> however >> >> > it seems to be truncating the data to 16 bits. I wasn't sure until I >> >> started >> >> > testing with the HRx 176.4K 24 bit files that have an embedded HDCD >> flag >> >> in >> >> > the LSB. The files work OK on Windoze systems (with a lot of low >> level >> >> > settings tweaked) and they play fine but the flag isn't detected by a >> >> system >> >> > that can detect them. Further looking at the data stream with a scope >> it >> >> > seems the last 8 bits aren't changing. Is there anything I can do to >> >> control >> >> > the driver to confirm this problem or change the playback settings to >> >> make >> >> > it work? >> >> > >> >> > >> >> >> >> How do you play the files? Do you use the hw or plug:hw device? IIRC, >> >> the standard-setup dmix resampler is 16-bit only. >> >> >> >> I did tests with SPDIF OUT/IN and found it bit-perfect, for 24bit too. >> >> >> >> Regular ICE1724 cards do not output 176.4kHz SPDIF, only 88.2kHz. But >> >> Juli is not affected by that bug. >> >> >> >> Regards, >> >> >> >> >> >> Pavel. >> > >> > _______________________________________________ >> > Alsa-devel mailing list >> > Alsa-devel@alsa-project.org >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > >> >> >> >> -- >> Vedran Miletić > > -- Vedran Miletić _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel ^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: Juli@ ICE1724 and 24 bit audio 2009-02-10 11:33 ` Vedran Miletić @ 2009-02-10 15:34 ` Pavel Hofman 2009-02-10 16:21 ` Demian Martin ` (2 more replies) 0 siblings, 3 replies; 12+ messages in thread From: Pavel Hofman @ 2009-02-10 15:34 UTC (permalink / raw) To: Vedran Miletić; +Cc: Demian Martin, alsa-devel Vedran Miletić wrote: > Yeah, ALSA's documentation is actually quite lacking on many fronts > and is mostly scattered around various wikis and mailing lists. > > But I guess it is as it is, complaining about it won't fix it. > > On 2/9/09, Demian Martin <demianm_1@yahoo.com> wrote: >> Thanks, I'll try that. I guess this e-mail will serve as more documentation >> on that feature (bug). I had not seen that factoid anywhere. >> >> Demian Martin >> PDS >> >> >>> -----Original Message----- >>> From: Vedran Miletić [mailto:rivanvx@gmail.com] >>> Sent: Monday, February 09, 2009 1:39 PM >>> To: Demian Martin >>> Cc: Pavel Hofman; alsa-devel@alsa-project.org >>> Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio >>> >>> The plughw part is what makese it resample to 16-bit. You should just use >>> hw. >>> >>> On Mon, Feb 9, 2009 at 10:29 PM, Demian Martin <demianm_1@yahoo.com> >>> wrote: >>>> Pavel: >>>> Thanks for all your help and work on this project. >>>> >>>> I'm using the Juli@ card because it supports 176.4KHz. The only other >>>> candidates are much more expensive and bring their own problems. >>>> >>>> I'm using Xine set for dolby/dts passthrough which seems to send the >>> audio >>>> data at its native rate. The resampling is bypassed (Dolby Digital would >>> be >>>> trashed otherwise). >>>> >>>> I will try playing the files from the command line when I am next in >>> front >>>> of the system, tomorrow. >>>> >>>> This is the Asound.conf: >>>> Asound.conf (/etc) >>>> >>>> pcm.asym_spdif { >>>> >>>> type asym >>>> >>>> playback.pcm "plughw:0,1" >>>> >>>> capture.pcm "plughw:0" >>>> >>>> } >>>> >>>> >>>> pcm.!default asym_spdif >>>> >>>> Demian Martin >>>> PDS >>>> >>>>> -----Original Message----- >>>>> From: Pavel Hofman [mailto:pavel.hofman@insite.cz] >>>>> Sent: Monday, February 09, 2009 12:27 PM >>>>> To: Demian Martin >>>>> Cc: alsa-devel@alsa-project.org; pavel.hofman@insite.cz >>>>> Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio >>>>> >>>>> Demian Martin napsal(a): >>>>>> I have the driver working fine for sample rates from 44.1K to 196K >>>>> however >>>>>> it seems to be truncating the data to 16 bits. I wasn't sure until I >>>>> started >>>>>> testing with the HRx 176.4K 24 bit files that have an embedded HDCD >>> flag >>>>> in >>>>>> the LSB. The files work OK on Windoze systems (with a lot of low >>> level >>>>>> settings tweaked) and they play fine but the flag isn't detected by a >>>>> system >>>>>> that can detect them. Further looking at the data stream with a scope >>> it >>>>>> seems the last 8 bits aren't changing. Is there anything I can do to >>>>> control >>>>>> the driver to confirm this problem or change the playback settings to >>>>> make >>>>>> it work? >>>>>> >>>>>> >>>>> How do you play the files? Do you use the hw or plug:hw device? IIRC, >>>>> the standard-setup dmix resampler is 16-bit only. >>>>> >>>>> I did tests with SPDIF OUT/IN and found it bit-perfect, for 24bit too. >>>>> >>>>> Regular ICE1724 cards do not output 176.4kHz SPDIF, only 88.2kHz. But >>>>> Juli is not affected by that bug. >>>>> >>>>> Regards, >>>>> >>>>> >>>>> Pavel. >>>> _______________________________________________ >>>> Alsa-devel mailing list >>>> Alsa-devel@alsa-project.org >>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >>> >>> >>> -- >>> Vedran Miletić >> What makes you guys think the plug plugin trims the data down to 16bits? Sure if the rate is changed (the source code suggests only the low-quality linear rate algorithm supports above-16bit format - operation convert, the other resampling algorithms convert using the operation convert_s16). Ice1724 supports all the general rates natively, the rate conversion in the plug plugin should not kick in for common audio formats. The only case I can think of would be switching Juli to external SPDIF-IN clock - the hw.rate_min = hw.rate_max = actual_rate and the automatic rate conversion could happen. I did not use the hw device in my tests since all the tested tracks would have to be 32-bit for the hw device to accept the format when played via my favorite aplay. My 2 cents guess is xine does the conversion. Regards, Pavel. _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel ^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: Juli@ ICE1724 and 24 bit audio 2009-02-10 15:34 ` Pavel Hofman @ 2009-02-10 16:21 ` Demian Martin 2009-02-10 21:45 ` Demian Martin 2009-02-11 1:06 ` Demian Martin 2 siblings, 0 replies; 12+ messages in thread From: Demian Martin @ 2009-02-10 16:21 UTC (permalink / raw) To: 'Pavel Hofman', 'Vedran Miletić'; +Cc: alsa-devel Now I'm more confused. I'll try the experiments today and report on what I find. If plughw with aplay works and xine doesn't that will explain a lot. I should have some good experimental answers in a few hours. Demian Martin Product Design Services -----Original Message----- From: Pavel Hofman [mailto:pavel.hofman@insite.cz] Sent: Tuesday, February 10, 2009 7:35 AM To: Vedran Miletić Cc: Demian Martin; alsa-devel@alsa-project.org Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio Vedran Miletić wrote: > Yeah, ALSA's documentation is actually quite lacking on many fronts > and is mostly scattered around various wikis and mailing lists. > > But I guess it is as it is, complaining about it won't fix it. > > On 2/9/09, Demian Martin <demianm_1@yahoo.com> wrote: >> Thanks, I'll try that. I guess this e-mail will serve as more documentation >> on that feature (bug). I had not seen that factoid anywhere. >> >> Demian Martin >> PDS >> >> What makes you guys think the plug plugin trims the data down to 16bits? Sure if the rate is changed (the source code suggests only the low-quality linear rate algorithm supports above-16bit format - operation convert, the other resampling algorithms convert using the operation convert_s16). Ice1724 supports all the general rates natively, the rate conversion in the plug plugin should not kick in for common audio formats. The only case I can think of would be switching Juli to external SPDIF-IN clock - the hw.rate_min = hw.rate_max = actual_rate and the automatic rate conversion could happen. I did not use the hw device in my tests since all the tested tracks would have to be 32-bit for the hw device to accept the format when played via my favorite aplay. My 2 cents guess is xine does the conversion. Regards, Pavel. ^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: Juli@ ICE1724 and 24 bit audio 2009-02-10 15:34 ` Pavel Hofman 2009-02-10 16:21 ` Demian Martin @ 2009-02-10 21:45 ` Demian Martin 2009-02-11 1:06 ` Demian Martin 2 siblings, 0 replies; 12+ messages in thread From: Demian Martin @ 2009-02-10 21:45 UTC (permalink / raw) To: 'Pavel Hofman', 'Vedran Miletić'; +Cc: alsa-devel Testing updates. First aplay w/ hw:0,0 gets this useless result: linuxmce@dcerouter:/home/public/data/audio/RR HDaudio$ aplay -v -Dhw:0,0 1_Mystery_Pacific.wav Playing WAVE '01_Mystery_Pacific.wav' : Signed 24 bit Little Endian in 3bytes, Rate 176400 Hz, Stereo aplay: set_params:900: Sample format non available Then w/ plughw:0,0 linuxmce@dcerouter:/home/public/data/audio/RR HDaudio$ aplay -v -Dplughw:0,0 01_Mystery_Pacific.wav Playing WAVE '01_Mystery_Pacific.wav' : Signed 24 bit Little Endian in 3bytes, Rate 176400 Hz, Stereo Plug PCM: Linear conversion PCM (S32_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S24_3LE subformat : STD channels : 2 rate : 176400 exact rate : 176400 (176400/1) msbits : 24 buffer_size : 32768 period_size : 8192 period_time : 46439 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 8192 xfer_align : 8192 start_threshold : 32768 stop_threshold : 32768 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Hardware PCM card 0 'ESI Juli@' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 176400 exact rate : 176400 (176400/1) msbits : 24 buffer_size : 32768 period_size : 8192 period_time : 46439 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 8192 xfer_align : 8192 start_threshold : 32768 stop_threshold : 32768 silence_threshold: 0 silence_size : 0 boundary : 1073741824 underrun!!! (at least 97.039 ms long) Status: state : XRUN trigger_time: 1234296814.359013515 tstamp : 1234296814.456029856 delay : 0 avail : 32768 avail_max : 32768 Aborted by signal Interrupt... I have underrun issues I need to look into as well. Perhaps some buffer setting needs to be changed? This is on a Via platform so it doesn't have a whole lot of power. Any suggestions would be appreciated. That is an area where I have gotten very lost in the past. Demian Martin Product Design Services 784 Cary Drive San Leandro, CA 94577 209 613 6990 -----Original Message----- From: Pavel Hofman [mailto:pavel.hofman@insite.cz] Sent: Tuesday, February 10, 2009 7:35 AM To: Vedran Miletić Cc: Demian Martin; alsa-devel@alsa-project.org Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio What makes you guys think the plug plugin trims the data down to 16bits? Sure if the rate is changed (the source code suggests only the low-quality linear rate algorithm supports above-16bit format - operation convert, the other resampling algorithms convert using the operation convert_s16). Ice1724 supports all the general rates natively, the rate conversion in the plug plugin should not kick in for common audio formats. The only case I can think of would be switching Juli to external SPDIF-IN clock - the hw.rate_min = hw.rate_max = actual_rate and the automatic rate conversion could happen. I did not use the hw device in my tests since all the tested tracks would have to be 32-bit for the hw device to accept the format when played via my favorite aplay. My 2 cents guess is xine does the conversion. Regards, Pavel. ^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: Juli@ ICE1724 and 24 bit audio 2009-02-10 15:34 ` Pavel Hofman 2009-02-10 16:21 ` Demian Martin 2009-02-10 21:45 ` Demian Martin @ 2009-02-11 1:06 ` Demian Martin [not found] ` <49927988.5090308@insite.cz> 2009-02-11 8:44 ` Pavel Hofman 2 siblings, 2 replies; 12+ messages in thread From: Demian Martin @ 2009-02-11 1:06 UTC (permalink / raw) To: 'Pavel Hofman', 'Vedran Miletić'; +Cc: alsa-devel I confirmed that aplay and plughw:0,1 work fine. Aplay and hw:0,1 don't work at all. And that Xine is the culprit. Now to find out how to fix/work around Xine. Demian Martin Product Design Services -----Original Message----- From: Pavel Hofman [mailto:pavel.hofman@insite.cz] Sent: Tuesday, February 10, 2009 7:35 AM To: Vedran Miletić Cc: Demian Martin; alsa-devel@alsa-project.org Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio What makes you guys think the plug plugin trims the data down to 16bits? Sure if the rate is changed (the source code suggests only the low-quality linear rate algorithm supports above-16bit format - operation convert, the other resampling algorithms convert using the operation convert_s16). Ice1724 supports all the general rates natively, the rate conversion in the plug plugin should not kick in for common audio formats. The only case I can think of would be switching Juli to external SPDIF-IN clock - the hw.rate_min = hw.rate_max = actual_rate and the automatic rate conversion could happen. I did not use the hw device in my tests since all the tested tracks would have to be 32-bit for the hw device to accept the format when played via my favorite aplay. My 2 cents guess is xine does the conversion. Regards, Pavel. ^ permalink raw reply [flat|nested] 12+ messages in thread
[parent not found: <49927988.5090308@insite.cz>]
* Re: Juli@ ICE1724 and 24 bit audio [not found] ` <49927988.5090308@insite.cz> @ 2009-02-11 8:38 ` Pavel Hofman 0 siblings, 0 replies; 12+ messages in thread From: Pavel Hofman @ 2009-02-11 8:38 UTC (permalink / raw) To: Demian Martin; +Cc: 'Vedran Miletić', alsa-devel Pavel Hofman napsal(a): > Demian Martin napsal(a): >> I confirmed that aplay and plughw:0,1 work fine. Aplay and hw:0,1 >> don't work >> at all. >> >> And that Xine is the culprit. Now to find out how to fix/work around >> Xine. >> >> Demian Martin >> Product Design Services >> >> > > The older aplay you most likely use does not support 24bit properly. > Install the latest alsa-utils from git and you will be fine. > Sorry for confusion. Your aplay handles already 24bit correctly. ICE1724 supports only 32bit natively (as most modern cards do), and hw:0 correctly does not accept any other format. That is why plughw converts your 24-bit format to 32bit - see the verbose output. For testing purposes make a 32bit file (sox -V in.wav -4 32.wav), and aplay -v -Dhw:0 32.wav will play fine. Pavel -- ----------------- inSITE, s.r.o. Rubesova 29, 326 00 Plzen Tel. +420 377 449 358 Fax +420 222 745 553 GSM: +420 603 163 973 Email: pavel.hofman@insite.cz www.educity.cz, www.jobcity.cz www.meetings.cz, www.hrzive.cz www.comben.cz, www.insite.cz ------------------------------- Navstivte www.educity.cz, server s nejvetsi nabidkou profesniho vzdelavani na ceskem internetu. ^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: Juli@ ICE1724 and 24 bit audio 2009-02-11 1:06 ` Demian Martin [not found] ` <49927988.5090308@insite.cz> @ 2009-02-11 8:44 ` Pavel Hofman 1 sibling, 0 replies; 12+ messages in thread From: Pavel Hofman @ 2009-02-11 8:44 UTC (permalink / raw) To: Demian Martin; +Cc: 'Vedran Miletić', alsa-devel Demian Martin napsal(a): > I confirmed that aplay and plughw:0,1 work fine. Aplay and hw:0,1 don't work > at all. > > And that Xine is the culprit. Now to find out how to fix/work around Xine. > > Demian Martin > Product Design Services > > The older aplay you most likely use does not support 24bit properly. Install the latest alsa-utils from git and you will be fine. Perhaps the Xine problem is caused by the passthrough mode. E.g. in mplayer the hwac3/hwdts output decoders for passthrough run in 16bit only, but they work actively with the non-audio format. You can check xine/xinelib code, the alsa output should not be that difficult to analyze. Regards, Pavel. ^ permalink raw reply [flat|nested] 12+ messages in thread
end of thread, other threads:[~2009-02-11 8:44 UTC | newest]
Thread overview: 12+ messages (download: mbox.gz follow: Atom feed
-- links below jump to the message on this page --
2009-02-09 20:09 Juli@ ICE1724 and 24 bit audio Demian Martin
2009-02-09 20:26 ` Pavel Hofman
2009-02-09 21:29 ` Demian Martin
2009-02-09 21:38 ` Vedran Miletić
2009-02-09 21:54 ` Demian Martin
2009-02-10 11:33 ` Vedran Miletić
2009-02-10 15:34 ` Pavel Hofman
2009-02-10 16:21 ` Demian Martin
2009-02-10 21:45 ` Demian Martin
2009-02-11 1:06 ` Demian Martin
[not found] ` <49927988.5090308@insite.cz>
2009-02-11 8:38 ` Pavel Hofman
2009-02-11 8:44 ` Pavel Hofman
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