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[alsa-devel] How can OSS applications support multiple streams?
by Timur Tabi 08 Nov '07

08 Nov '07

I've written an ALSA driver that supports only one playback stream, because that's what our hardware supports. I have a customer that has an OSS application that tries to open 32 simultaneous playback streams, which obviously doesn't work. What do I (or the customer) need to do to get 32 streams to work? Is this what a sound server is for? If so, which one should I use? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] Any OSS changes from kernel 2.6.21 to 2.6.23? Something broke.
by Timur Tabi 08 Nov '07

08 Nov '07

I have a customer who says that mplayer (using OSS) does not work when he tries a kernel based on 2.6.23-rc4, but it works when he uses a kernel based on 2.6.21. Everything else is the same (including the version of alsa-lib), so I presume something broke in the OSS emulation in the kernel. Unfortunately, our git repositories for these trees are not compatible, so I can't use git-bisect to narrow the problem down. Is it possible to back-level the ALSA support in the kernel itself? Can I just download ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.XX.tar.bz2 and follow the instructions in the INSTALL file? If you update the kernel, are you supposed to use the latest alsa-lib as well? For a given kernel, how do I know what the right version of alsa-lib is? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] Fwd: [PATCH] ehci-hcd: complete iso urbs ASAP for number_of_packets != (n * 8) also
by Karsten Wiese 08 Nov '07

08 Nov '07

Hi I just sent attached patch to usb-devel. It makes special workaround for iso usb communication using ehci-hcd highspeed needless. Comments, Bug/success reports welcome! kind regards Karsten Upstream (as of 2.6.23) ehci-hcd only completes iso urbs, if the last frame (= 8 uframes) they overlapped with has elapsed. That can be as late as when the following urb emits its interrupt on completion. soundcard drivers tend to work around by only transfering iso urbs with number_of_packets = (n * 8). Patch lets iso urbs complete asap by scanning itd always up to the elapsed uframe. An itd's last scanned uframe is stored in the new struct ehci_itd member uframe_scanned. Itds stay kept in the hcd's schedule until the complete frame they cover has elapsed, like without patch. To make this possible, ehci->periodic_sched is changed based on the number of active itds instead of the number of active iso urbs. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- drivers/usb/host/ehci-sched.c | 108 ++++++++++++++++++++--------------------- drivers/usb/host/ehci.h | 1 + 2 files changed, 54 insertions(+), 55 deletions(-) diff --git a/drivers/usb/host/ehci-sched.c b/drivers/usb/host/ehci-sched.c index e682f23..c456b16 100644 --- a/drivers/usb/host/ehci-sched.c +++ b/drivers/usb/host/ehci-sched.c @@ -1481,6 +1481,8 @@ itd_link (struct ehci_hcd *ehci, unsigned frame, struct ehci_itd *itd) itd->frame = frame; wmb (); ehci->periodic[frame] = cpu_to_hc32(ehci, itd->itd_dma | Q_TYPE_ITD); + if (unlikely (!ehci->periodic_sched++)) + enable_periodic (ehci); } /* fit urb's itds into the selected schedule slot; activate as needed */ @@ -1553,30 +1555,30 @@ itd_link_urb ( urb->hcpriv = NULL; timer_action (ehci, TIMER_IO_WATCHDOG); - if (unlikely (!ehci->periodic_sched++)) - return enable_periodic (ehci); return 0; } #define ISO_ERRS (EHCI_ISOC_BUF_ERR | EHCI_ISOC_BABBLE | EHCI_ISOC_XACTERR) -static unsigned -itd_complete ( - struct ehci_hcd *ehci, - struct ehci_itd *itd -) { +static bool +itd_scan ( + struct ehci_hcd *ehci, + struct ehci_itd *itd, + unsigned uframe_after +) +{ struct urb *urb = itd->urb; struct usb_iso_packet_descriptor *desc; u32 t; unsigned uframe; int urb_index = -1; struct ehci_iso_stream *stream = itd->stream; - struct usb_device *dev; /* for each uframe with a packet */ - for (uframe = 0; uframe < 8; uframe++) { + for (uframe = itd->uframe_scanned; uframe < uframe_after; uframe++) { if (likely (itd->index[uframe] == -1)) continue; + urb_index = itd->index[uframe]; desc = &urb->iso_frame_desc [urb_index]; @@ -1603,45 +1605,53 @@ itd_complete ( desc->status = 0; desc->actual_length = EHCI_ITD_LENGTH (t); } + + /* handle completion now? */ + if (urb_index + 1 == urb->number_of_packets) { + struct usb_device *dev; + + usb_put_urb (urb); + /* ASSERT: it's really the last itd for this urb + struct ehci_itd *stritd; + list_for_each_entry (stritd, &stream->td_list, itd_list) + BUG_ON (stritd != itd && stritd->urb == urb); + */ + itd->urb = NULL; + + /* give urb back to the driver ... can be out-of-order */ + dev = urb->dev; + ehci_urb_done (ehci, urb); + urb = NULL; + + ehci_to_hcd(ehci)->self.bandwidth_isoc_reqs--; + + if (unlikely (list_empty (&stream->td_list))) { + ehci_to_hcd(ehci)->self.bandwidth_allocated + -= stream->bandwidth; + ehci_vdbg (ehci, + "deschedule devp %s ep%d%s-iso\n", + dev->devpath, stream->bEndpointAddress & 0x0f, + (stream->bEndpointAddress & USB_DIR_IN) ? "in" : "out"); + } + iso_stream_put (ehci, stream); + } } - usb_put_urb (urb); - itd->urb = NULL; + itd->uframe_scanned = uframe_after; + if (uframe_after < 8) + return false; + itd->stream = NULL; + itd->uframe_scanned = 0; list_move (&itd->itd_list, &stream->free_list); iso_stream_put (ehci, stream); - /* handle completion now? */ - if (likely ((urb_index + 1) != urb->number_of_packets)) - return 0; - - /* ASSERT: it's really the last itd for this urb - list_for_each_entry (itd, &stream->td_list, itd_list) - BUG_ON (itd->urb == urb); - */ - - /* give urb back to the driver ... can be out-of-order */ - dev = urb->dev; - ehci_urb_done (ehci, urb); - urb = NULL; - /* defer stopping schedule; completion can submit */ ehci->periodic_sched--; if (unlikely (!ehci->periodic_sched)) (void) disable_periodic (ehci); - ehci_to_hcd(ehci)->self.bandwidth_isoc_reqs--; - if (unlikely (list_empty (&stream->td_list))) { - ehci_to_hcd(ehci)->self.bandwidth_allocated - -= stream->bandwidth; - ehci_vdbg (ehci, - "deschedule devp %s ep%d%s-iso\n", - dev->devpath, stream->bEndpointAddress & 0x0f, - (stream->bEndpointAddress & USB_DIR_IN) ? "in" : "out"); - } - iso_stream_put (ehci, stream); - - return 1; + return true; } /*-------------------------------------------------------------------------*/ @@ -2156,30 +2166,18 @@ restart: q = q.fstn->fstn_next; break; case Q_TYPE_ITD: - /* skip itds for later in the frame */ rmb (); - for (uf = live ? uframes : 8; uf < 8; uf++) { - if (0 == (q.itd->hw_transaction [uf] - & ITD_ACTIVE(ehci))) - continue; + modified = itd_scan(ehci, q.itd, uf = live ? uframes : 8); + if (!modified) { q_p = &q.itd->itd_next; hw_p = &q.itd->hw_next; - type = Q_NEXT_TYPE(ehci, - q.itd->hw_next); - q = *q_p; - break; + } else { + *q_p = q.itd->itd_next; + *hw_p = q.itd->hw_next; } - if (uf != 8) - break; - - /* this one's ready ... HC won't cache the - * pointer for much longer, if at all. - */ - *q_p = q.itd->itd_next; - *hw_p = q.itd->hw_next; type = Q_NEXT_TYPE(ehci, q.itd->hw_next); + wmb(); - modified = itd_complete (ehci, q.itd); q = *q_p; break; case Q_TYPE_SITD: diff --git a/drivers/usb/host/ehci.h b/drivers/usb/host/ehci.h index 951d69f..df35964 100644 --- a/drivers/usb/host/ehci.h +++ b/drivers/usb/host/ehci.h @@ -581,6 +581,7 @@ struct ehci_itd { struct urb *urb; struct ehci_iso_stream *stream; /* endpoint's queue */ struct list_head itd_list; /* list of stream's itds */ + unsigned uframe_scanned; /* any/all hw_transactions here may be used by that urb */ unsigned frame; /* where scheduled */ -- 1.5.3.3

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[alsa-devel] Audio quality issues with dmix plugin
by Pharaoh . 08 Nov '07

08 Nov '07

Hi, When I play a 44Khz file without using any ALSA plugin like #aplay -Dhw:0,0 44k.wav the sound quality is awesome but when I play the same file using dmix, clicks are introduced. #aplay -Dplug:dmix 44k.wav changing period size and buffer size doesn't help. I am testing my ALSA driver on an OMAP850 based board. Regards, Pharaoh.

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Re: [alsa-devel] snd_pcm_hw_params_get_xxx question
by Ben C 08 Nov '07

08 Nov '07

Takashi, You were right, it was a linking issue. I'm doing cross-compiling for a target of different processor type. I just had to make sure I'm pointing to the right location for the includes and libs. Thanks, Ben ----- Original Message ---- From: Takashi Iwai <tiwai(a)suse.de> To: Ben C <bchang_2(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Sent: Wednesday, November 7, 2007 4:24:04 AM Subject: Re: [alsa-devel] snd_pcm_hw_params_get_xxx question At Tue, 6 Nov 2007 18:21:01 -0800 (PST), Ben C wrote: > > Hello, > > I'm trying to develop an external PCM plugin. I'm calling snd_pcm_hw_params_get_xxx functions to retrieve some values. I got everything to work as expected on alsa-lib 1.0.15. However the client's setup is 1.0.13, and I don't think they want to upgrade. > > For 1.0.13, everything compiled and linked fine after some minor changes. But when I called say snd_pcm_hw_params_get_rate(hw, &val, &dir), I was expecting the return value to be a status, and val to contain "44100" which was the case for 1.0.15. However for 1.0.13, val is 0, but returned status = 44100? > > I looked in the header files, and 1.0.13 seems to be the same as 1.0.15 in regards to this function. Am I accidentally using some sort of old API that was intended for backward compatibility? If so, what should I do to use the "new" API's? Yes, it sounds like you're accessing the old API. Do you use libasound from a module? If it's used from a module, the module itself should be built once with link to libasound. For example, $(CC) -shared -o MODULE.so -lasound In such a way, the properly versioned symbol is chosen. The weak link results in the old API. This is irrelveant whether it's ALSA 1.0.13 or 1.0.15. Takashi __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] [HDA Intel/STAC92xx] Alsa 1.0.14
by Greg Jizmagian 07 Nov '07

07 Nov '07

I am trying to fix the sound on the m1330 in OS X. It currently only works out of one of the headphone jacks. I stumbled across the patch you wrote for ubuntu and hoped that I could adapt it for OS X. Or is there a way to see a post patch linux dump that would fix the problem? Any suggestions/help would be greatly appreciated. Thanks. Greg

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[alsa-devel] snd_pcm_hw_params_get_xxx question
by Ben C 07 Nov '07

07 Nov '07

Hello, I'm trying to develop an external PCM plugin. I'm calling snd_pcm_hw_params_get_xxx functions to retrieve some values. I got everything to work as expected on alsa-lib 1.0.15. However the client's setup is 1.0.13, and I don't think they want to upgrade. For 1.0.13, everything compiled and linked fine after some minor changes. But when I called say snd_pcm_hw_params_get_rate(hw, &val, &dir), I was expecting the return value to be a status, and val to contain "44100" which was the case for 1.0.15. However for 1.0.13, val is 0, but returned status = 44100? I looked in the header files, and 1.0.13 seems to be the same as 1.0.15 in regards to this function. Am I accidentally using some sort of old API that was intended for backward compatibility? If so, what should I do to use the "new" API's? Thanks, Ben __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] [PATCH] hda: STAC9228 Subsystem update
by Matthew Ranostay 07 Nov '07

07 Nov '07

Added more laptops subsystem id's that have STAC9228 DMIC support. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 43708ece3f6f pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Nov 06 11:56:17 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Nov 06 14:01:53 2007 -0500 @@ -2921,8 +2921,10 @@ static int patch_stac927x(struct hda_cod } switch (codec->subsystem_id) { - case 0x1028020A: /* STAC 9228 */ - case 0x10280209: /* STAC 9228 */ + case 0x10280242: /* STAC 9228 */ + case 0x102801f3: + case 0x1028020A: + case 0x10280209: spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; spec->dmux_nid = 0x1b;

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[alsa-devel] [PATCH] cmipci - utilize ADC48K44K bit
by Timofei V. Bondarenko 07 Nov '07

07 Nov '07

Hi. Setting the ADC48K44K greatly improves capture quality at 48k sampling rate. With this bit clear ADC does ZOH interpolation of every 22th sample at 48k. At frequencies higher than 48k there ADC performs a little better with ADC48K44K bit set. At 44.1k ADC performs a little better with this bit clear. At frequencies below 44.1k there is no difference. Regards, Tim.

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[alsa-devel] [PATCH] hda: STAC92HD71 codec mixer
by Matthew Ranostay 07 Nov '07

07 Nov '07

Added analog loopback support and missing ADC capture mixer for the STAC92HD71 codec family. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 43708ece3f6f pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Nov 06 11:56:17 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Nov 06 21:05:54 2007 -0500 @@ -428,6 +428,11 @@ static struct hda_verb stac92hd71bxx_cor { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* connect ports 0d and 0f to audio mixer */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + /* unmute dac0 input in audio mixer */ + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -512,8 +517,7 @@ static struct hda_verb stac9205_core_ini .put = stac92xx_volknob_put, \ .private_value = 127 | (knob_nid << 16), \ } - - + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -530,12 +534,19 @@ static struct snd_kcontrol_new stac92hd7 STAC_VOLKNOB(0x28), /* hardware gain controls */ - HDA_CODEC_VOLUME("Digital Mic 1 Volume", 0x18, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Digital Mic 2 Volume", 0x19, 0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Capture Volume", 0x1c, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x1c, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x1a, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + + HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), { } /* end */ };

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