March 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [Re: Crash when snd_pcm_open is called from a thread with "pulse" as playback]
by Nicolas.Castagne＠imag.fr 10 Mar '09

10 Mar '09

Hi folks, I'd need the help of some alsa developpers regarding the bug report : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4426 We've made a few progress on it, but we are stuck again. Basically, it seems that calling snd_config_update_free_global() from a pthread causes the program to crash when the thread terminates. Questions are: Should config_update_free_global NOT be called from a thread ? In other words, is the following code valid (though dumb :=)) ? Or is there some reason for it to crash ? ********************************** #include <pthread.h> #include <alsa/asoundlib.h> void * threadWork(void *arg) { snd_pcm_t *playback_handle; if (snd_pcm_open (&playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0) < 0) fprintf (stderr, "error\n"; snd_pcm_close(playback_handle); snd_config_update_free_global(); return NULL; } int main() { pthread_attr_t attr; pthread_attr_init(&attr); pthread_t threadid; pthread_create( &threadid, &attr, threadWork, NULL); return EXIT_SUCCESS; } ********************************** I thank you much, Nicolas ---------------------------- Message original ---------------------------- Objet: Crash when snd_pcm_open is called from a thread with "pulse" as playback De: castagne(a)imag.fr Date: Mer 4 mars 2009 8:20 À: alsa-devel(a)alsa-project.org -------------------------------------------------------------------------- Hi folks, I hope I post on the correct mail list... I've read somewhere that I should report bugs on a list, and not only on the Tracker. That's the aim of this message. On all my Fedora 10 machines, whenever snd_pcm_open has been called from a pthread using "pulse" as playback then the program crashes when the thread terminates. Crash in __nptl_deallocate_tsd However, AFAIK, one should be able to call snd_pcm_open from a thread. The full bug report is there, with a very simple sample code : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4426 I think this bug relates directly to alsa. I hope this report is useful. Best- Nicolas

3 5

[alsa-devel] [PATCH] S3C: Fix S3C2412_IISMOD_SDF_{MSB, LSB} definitions
by Ben Dooks 09 Mar '09

09 Mar '09

The definitions of S3C2412_IISMOD_SDF_MSB and S3C2412_IISMOD_SDF_LSB are incorrect, being the same S3C2412_IISMOD_SDF_IIS which is the only correct one in this series. Signed-off-by: Ben Dooks <ben-linux(a)fluff.org> --- arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h index 25d4058..e5988b4 100644 --- a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h +++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h @@ -44,8 +44,8 @@ #define S3C2412_IISMOD_LR_LLOW (0 << 7) #define S3C2412_IISMOD_LR_RLOW (1 << 7) #define S3C2412_IISMOD_SDF_IIS (0 << 5) -#define S3C2412_IISMOD_SDF_MSB (0 << 5) -#define S3C2412_IISMOD_SDF_LSB (0 << 5) +#define S3C2412_IISMOD_SDF_MSB (1 << 5) +#define S3C2412_IISMOD_SDF_LSB (2 << 5) #define S3C2412_IISMOD_SDF_MASK (3 << 5) #define S3C2412_IISMOD_RCLK_256FS (0 << 3) #define S3C2412_IISMOD_RCLK_512FS (1 << 3) -- 1.5.6.5

2 1

[alsa-devel] Crash when snd_pcm_open is called from a thread with "pulse" as playback
by Nicolas.Castagne＠imag.fr 09 Mar '09

09 Mar '09

Hi folks, I hope I post on the correct mail list... I've read somewhere that I should report bugs on a list, and not only on the Tracker. That's the aim of this message. On all my Fedora 10 machines, whenever snd_pcm_open has been called from a pthread using "pulse" as playback then the program crashes when the thread terminates. Crash in __nptl_deallocate_tsd However, AFAIK, one should be able to call snd_pcm_open from a thread. The full bug report is there, with a very simple sample code : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4426 I think this bug relates directly to alsa. I hope this report is useful. Best- Nicolas

2 2

[alsa-devel] [PATCH 1/1] echomixer, fix vmixer GUI
by Giuliano Pochini 09 Mar '09

09 Mar '09

This patch fixes several bugs in the vmixer GUI and removes the "misc" button from the main window when the card has no global controls. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- alsa-tools-1.0.19/echomixer/echomixer.c__orig 2009-02-27 22:51:47.000000000 +0100 +++ alsa-tools-1.0.19/echomixer/echomixer.c 2009-03-01 16:29:42.000000000 +0100 @@ -129,38 +129,8 @@ struct mixel { snd_ctl_t *ctlhandle; -#if __GNUC__ == 3 // gcc 2.x doesn't like unnamed unions inside structures - -struct mixerControl_s { - union { // Currently selected channels - int vchannel; - int input; - }; - union { // Number of channels - int vchannels; - int inputs; - }; - int output, outputs; - int id; - GtkWidget *window; - GtkWidget *volume[ECHO_MAXAUDIOOUTPUTS]; - GtkWidget *label[ECHO_MAXAUDIOOUTPUTS]; - GtkObject *adj[ECHO_MAXAUDIOOUTPUTS]; - GtkWidget *outsel[ECHO_MAXAUDIOOUTPUTS]; - union { - GtkWidget *inpsel[ECHO_MAXAUDIOINPUTS]; - GtkWidget *vchsel[ECHO_MAXAUDIOOUTPUTS]; - }; - struct mixel mixer[ECHO_MAXAUDIOOUTPUTS][ECHO_MAXAUDIOOUTPUTS]; -} mixerControl, vmixerControl; - -#else - struct mixerControl_s { - int vchannel; - int input; - int vchannels; - int inputs; + int input, inputs; int output, outputs; int id; GtkWidget *window; @@ -173,9 +143,6 @@ struct mixerControl_s { struct mixel mixer[ECHO_MAXAUDIOOUTPUTS][ECHO_MAXAUDIOOUTPUTS]; } mixerControl, vmixerControl; -#endif - - struct VolumeControl_s { int input, output; // Currently selected channels int inputs, outputs; @@ -677,8 +644,8 @@ gint DrawMixer(gpointer unused) { gdk_draw_rectangle(Mixpixmap, gc, TRUE, XCELLTOT*(mixerControl.output+1), YCELLTOT*mixerControl.input, XCELLTOT, Mixheight); if (vmixerId) { gdk_gc_set_foreground(gc, &Hilight2); - gdk_draw_rectangle(Mixpixmap, gc, TRUE, 0, YCELLTOT*(GMixerSection.VmixerFirst+vmixerControl.vchannel), XCELLTOT*(vmixerControl.output+1), YCELLTOT); - gdk_draw_rectangle(Mixpixmap, gc, TRUE, XCELLTOT*(vmixerControl.output+1), YCELLTOT*(GMixerSection.VmixerFirst+vmixerControl.vchannel), XCELLTOT, Mixheight); + gdk_draw_rectangle(Mixpixmap, gc, TRUE, 0, YCELLTOT*(GMixerSection.VmixerFirst+vmixerControl.input), XCELLTOT*(vmixerControl.output+1), YCELLTOT); + gdk_draw_rectangle(Mixpixmap, gc, TRUE, XCELLTOT*(vmixerControl.output+1), YCELLTOT*(GMixerSection.VmixerFirst+vmixerControl.input), XCELLTOT, Mixheight); } // Draw the grid @@ -720,7 +687,7 @@ gint DrawMixer(gpointer unused) { // Draw vchannels levels and peaks (Vmixer cards only) if (vmixerId) { - for (i=0; i<vmixerControl.vchannels; i++) + for (i=0; i<vmixerControl.inputs; i++) DrawBar(0, i+GMixerSection.VmixerFirst, VirLevel[i], VirPeak[i], DONT_DRAW); } @@ -739,7 +706,7 @@ gint DrawMixer(gpointer unused) { // Draw vmixer elements (Vmixer cards only) if (vmixerId) { for (o=0; o<GMixerSection.Outputs; o++) - for (i=0; i<vmixerControl.vchannels; i++) { + for (i=0; i<vmixerControl.inputs; i++) { dB=Add_dB(vmixerControl.mixer[o][i].Gain, VirLevel[i]); DrawBar(o+1, i+GMixerSection.VmixerFirst, dB, DONT_DRAW, vmixerControl.mixer[o][i].Gain); } @@ -945,7 +912,7 @@ static gint Gmixer_button_press(GtkWidge } else if (GMixerRow>=GMixerSection.VmixerFirst && GMixerRow<=GMixerSection.VmixerLast) { if (GMixerColumn!=vmixerControl.output) gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(vmixerControl.outsel[GMixerColumn]), TRUE); - if (GMixerRow!=vmixerControl.vchannel) + if (GMixerRow!=vmixerControl.input) gtk_widget_grab_focus(GTK_WIDGET(vmixerControl.volume[GMixerRow-GMixerSection.VmixerFirst])); } @@ -967,13 +934,13 @@ static gint Gmixer_button_press(GtkWidge return TRUE; // See the note above - if (GMixerRow<GMixerSection.Inputs) { + if (GMixerRow<GMixerSection.VmixerFirst) { if (GMixerRow!=mixerControl.input) gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(mixerControl.inpsel[GMixerRow]), TRUE); if (GMixerColumn!=mixerControl.output) gtk_widget_grab_focus(GTK_WIDGET(mixerControl.volume[GMixerColumn])); } else if (GMixerRow>=GMixerSection.VmixerFirst && GMixerRow<=GMixerSection.VmixerLast) { - if (GMixerRow!=vmixerControl.vchannel) + if (GMixerRow!=vmixerControl.input+GMixerSection.VmixerFirst) gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(vmixerControl.vchsel[GMixerRow-GMixerSection.VmixerFirst]), TRUE); if (GMixerColumn!=vmixerControl.output) gtk_widget_grab_focus(GTK_WIDGET(vmixerControl.volume[GMixerColumn])); @@ -1035,11 +1002,11 @@ static gint Gmixer_motion_notify(GtkWidg gtk_adjustment_set_value(GTK_ADJUSTMENT(mixerControl.adj[mixerControl.output]), (gfloat)val); #endif } else if (GMixerRow>=GMixerSection.VmixerFirst && GMixerRow<=GMixerSection.VmixerLast) { - val=INVERT(vmixerControl.mixer[vmixerControl.output][vmixerControl.vchannel].Gain); + val=INVERT(vmixerControl.mixer[vmixerControl.output][vmixerControl.input].Gain); val+=y-mouseY; mouseY=y; #ifdef REVERSE - gtk_adjustment_set_value(GTK_ADJUSTMENT(vmixerControl.adj[vmixerControl.vchannel]), (gfloat)val); + gtk_adjustment_set_value(GTK_ADJUSTMENT(vmixerControl.adj[vmixerControl.input]), (gfloat)val); #else gtk_adjustment_set_value(GTK_ADJUSTMENT(vmixerControl.adj[vmixerControl.output]), (gfloat)val); #endif @@ -1259,7 +1226,7 @@ void Vmixer_volume_changed(GtkWidget *wi v=channel; o=vmixerControl.output; #else - v=vmixerControl.vchannel; + v=vmixerControl.input; o=channel; #endif @@ -1281,8 +1248,8 @@ void Vmixer_volume_changed(GtkWidget *wi void Vmixer_volume_clicked(GtkWidget *widget, gpointer ch) { #ifdef REVERSE - vmixerControl.vchannel=(int)(long)ch; - UI_DEBUG(("Vmixer_volume_clicked vch=%d\n",vmixerControl.vchannel)); + vmixerControl.input=(int)(long)ch; + UI_DEBUG(("Vmixer_volume_clicked vch=%d\n",vmixerControl.input)); #else vmixerControl.output=(int)(long)ch; UI_DEBUG(("Vmixer_volume_clicked out=%d\n",vmixerControl.output)); @@ -1306,7 +1273,7 @@ void Vmixer_output_selector_clicked(GtkW snd_ctl_elem_id_alloca(&id); snd_ctl_elem_value_alloca(&control); snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); - for (c=vmixerControl.vchannels-1; c>=0; c--) { + for (c=vmixerControl.inputs-1; c>=0; c--) { val=INVERT(vmixerControl.mixer[vmixerControl.output][c].Gain); gtk_adjustment_set_value(GTK_ADJUSTMENT(vmixerControl.adj[c]), (gfloat)val); } @@ -1319,16 +1286,16 @@ void Vmixer_vchannel_selector_clicked(Gt snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *control; - if (vmixerControl.vchannel==(int)(long)ch) + if (vmixerControl.input==(int)(long)ch) return; - vmixerControl.vchannel=(int)(long)ch; + vmixerControl.input=(int)(long)ch; - UI_DEBUG(("Vmixer_selector_clicked vch=%d\n",vmixerControl.vchannel)); + UI_DEBUG(("Vmixer_selector_clicked vch=%d\n",vmixerControl.input)); snd_ctl_elem_id_alloca(&id); snd_ctl_elem_value_alloca(&control); snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); for (c=vmixerControl.outputs-1; c>=0; c--) { - val=INVERT(vmixerControl.mixer[c][vmixerControl.vchannel].Gain); + val=INVERT(vmixerControl.mixer[c][vmixerControl.input].Gain); gtk_adjustment_set_value(GTK_ADJUSTMENT(vmixerControl.adj[c]), (gfloat)val); } } @@ -1722,7 +1689,7 @@ int OpenControls(const char *card, const vmixerId=vmixerControl.id=numid; CTLID_DEBUG(("First Vmixer id=%d\n", vmixerId)); vmixerControl.outputs=snd_ctl_elem_info_get_dimension(info, 0); - vmixerControl.vchannels=snd_ctl_elem_info_get_dimension(info, 1); + vmixerControl.inputs=snd_ctl_elem_info_get_dimension(info, 1); } } else if (!strcmp("PCM Playback Volume", snd_ctl_elem_id_get_name(id))) { pcmoutId=pcmoutControl.id=numid; @@ -1808,7 +1775,7 @@ int OpenControls(const char *card, const #ifndef REAL vmixerId=1000; -vmixerControl.vchannels=12; +vmixerControl.inputs=12; vmixerControl.outputs=mixerControl.outputs=nLOut=10; metersStreams=3; metersNumber=16; @@ -1828,8 +1795,8 @@ printf("nIn=%d fdIn=%d nLOut=%d nPOut=%d printf("** Warning - Vmixer cards without LineOut volume control are not supported !\n"); if (vmixerId) { - if (vmixerControl.vchannels!=nPOut || vmixerControl.outputs!=nLOut) { - printf("** Error - vmixer/channels mismatch: vmp=%d npo=%d vmo=%d nlo=%d !!\n", vmixerControl.vchannels, nPOut, vmixerControl.outputs, nLOut); + if (vmixerControl.inputs!=nPOut || vmixerControl.outputs!=nLOut) { + printf("** Error - vmixer/channels mismatch: vmp=%d npo=%d vmo=%d nlo=%d !!\n", vmixerControl.inputs, nPOut, vmixerControl.outputs, nLOut); return(1); } } @@ -1850,7 +1817,7 @@ printf("nIn=%d fdIn=%d nLOut=%d nPOut=%d if (vmixerId) for (o=0, numid=vmixerId; o<vmixerControl.outputs; o++) { - for (i=0; i<vmixerControl.vchannels; i++) { + for (i=0; i<vmixerControl.inputs; i++) { vmixerControl.mixer[o][i].id=numid++; } } @@ -1931,10 +1898,10 @@ printf("components = %s\n", snd_ctl_card Mixerw_geom.st=NOPOS; Vmixerw_geom.st=NOPOS; VUwindow=GMwindow=0; - GMixerSection.Inputs=fdIn+2; // The correct value is set by Digital_mode_activate() - GMixerSection.Outputs=fdOut+2; + GMixerSection.Inputs=nIn; // The correct value is set by Digital_mode_activate() + GMixerSection.Outputs=nLOut; GMixerSection.VmixerFirst=nIn; - GMixerSection.VmixerLast=nIn+vmixerControl.vchannels-1; + GMixerSection.VmixerLast=nIn+vmixerControl.inputs-1; GMixerSection.LineOut=GMixerSection.VmixerLast+1; // Read current mixer setting. @@ -2523,7 +2490,7 @@ printf("components = %s\n", snd_ctl_card gtk_widget_show(hbox); gtk_container_add(GTK_CONTAINER(frame), hbox); - for (i=0; i<vmixerControl.vchannels; i++) { + for (i=0; i<vmixerControl.inputs; i++) { vbox=gtk_vbox_new(FALSE, 0); gtk_widget_show(vbox); gtk_container_add(GTK_CONTAINER(hbox), vbox); @@ -2582,7 +2549,7 @@ printf("components = %s\n", snd_ctl_card gtk_container_add(GTK_CONTAINER(frame), vbsel); bgroup=0; - for (i=0; i<vmixerControl.vchannels; i++) { + for (i=0; i<vmixerControl.inputs; i++) { sprintf(str, "V%d", i); if (i) bgroup=gtk_radio_button_group(GTK_RADIO_BUTTON(vmixerControl.vchsel[i-1])); @@ -2626,7 +2593,7 @@ printf("components = %s\n", snd_ctl_card gtk_box_pack_start(GTK_BOX(vbox), vmixerControl.label[i], FALSE, FALSE, 0); } gtk_widget_set_usize(GTK_WIDGET(vmixerControl.volume[0]), 0, 170); // Set minimum y size - vmixerControl.vchannel=-1; + vmixerControl.input=-1; Vmixer_vchannel_selector_clicked(0, 0); #endif } @@ -2693,13 +2660,15 @@ printf("components = %s\n", snd_ctl_card gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(button), TRUE); // Misc controls button - button=gtk_toggle_button_new_with_label("Misc"); - gtk_widget_show(button); - gtk_box_pack_start(GTK_BOX(hbox), button, TRUE, TRUE, 1); - gtk_signal_connect(GTK_OBJECT(button), "toggled", ToggleWindow, (gpointer)Miscwindow); - Miscw_geom.toggler=button; - if (Miscw_geom.st==1) - gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(button), TRUE); + if (p4InId || p4OutId || phantomId || (dmodeId && ndmodes>1) || (clocksrcId && nclocksrc>1) || (spdifmodeId && nspdifmodes>1)) { + button=gtk_toggle_button_new_with_label("Misc"); + gtk_widget_show(button); + gtk_box_pack_start(GTK_BOX(hbox), button, TRUE, TRUE, 1); + gtk_signal_connect(GTK_OBJECT(button), "toggled", ToggleWindow, (gpointer)Miscwindow); + Miscw_geom.toggler=button; + if (Miscw_geom.st==1) + gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(button), TRUE); + } if (mixerId) { // Graphical mixer button -- Giuliano.

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Re: [alsa-devel] Timer instability
by Takashi Iwai 09 Mar '09

09 Mar '09

At Wed, 25 Feb 2009 16:56:47 +0000, Clive Messer wrote: > > On Wednesday 25 Feb 2009 16:39:28 you wrote: > > At Wed, 25 Feb 2009 16:24:18 +0000, > > > > Clive Messer wrote: > > > On Wednesday 25 Feb 2009 15:51:33 you wrote: > > > > At Wed, 25 Feb 2009 15:11:44 +0000, > > > > > > > > Clive Messer wrote: > > > > > 36192 36191 100068 3 0 0 1 2 > > > > > 36195 36194 100090 2 0 0 1 2 > > > > > 36198 36198 100113 1 0 0 1 2 > > > > > 36207 36206 63854 2816 1600 0 1 3 > > > > > 36212 36211 63854 2815 1601 4 1 3 > > > > > 36216 36216 63854 2814 1602 4 1 3 > > > > > 36221 36220 63854 2813 1603 4 1 3 > > > > > 36225 36225 101224 4460 -44 4 1 3 > > > > > > > > This is pretty weird. Is this behavior always reproducible? > > > > > > Nope. I get different results every time I run it. > > > > > > > What shows with the patch below to alsa-lib? > > > > > > Here's one (attached) I just did with the new patch that shows the crazy > > > high numbers again. > > > hwsync: 2080(0), 6496(0) > > > 70758 70758 47142 0 4416 0 0 3 > > > hwsync: 1472(0), 6496(0) > > > 70763 70762 104573379136040 4971973988617026976 -4971973988617022560 4 1 > > >3 > > > > The problem is that hwptr goes backward here. But it didn't happen in > > this hwsync call but sometime before that. > > > > Could you try the patch below instead? This will track all sync_ptr calls. > > > > > > Takashi > > > > --- > > diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c > > index e9ce092..ad8d4a5 100644 > > --- a/src/pcm/pcm_hw.c > > +++ b/src/pcm/pcm_hw.c > > @@ -131,6 +131,10 @@ struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm) > > static int sync_ptr1(snd_pcm_hw_t *hw, unsigned int flags) > > { > > int err; > > + long old_hwptr, new_hwptr; > > + long old_applptr, new_applptr; > > + old_hwptr = *pcm->hw.ptr; > > + old_applptr = *pcm->appl.ptr; > > hw->sync_ptr->flags = flags; > > err = ioctl((hw)->fd, SNDRV_PCM_IOCTL_SYNC_PTR, (hw)->sync_ptr); > > if (err < 0) { > > @@ -138,6 +142,11 @@ static int sync_ptr1(snd_pcm_hw_t *hw, unsigned int > > flags) SYSMSG("SNDRV_PCM_IOCTL_SYNC_PTR failed"); > > return err; > > } > > + new_hwptr = *pcm->hw.ptr; > > + new_applptr = *pcm->appl.ptr; > > + printf("sync_ptr1: %ld(%ld), %ld(%ld)\n", > > + new_hwptr, new_hwptr - old_hwptr, > > + new_applptr, new_applptr - old_applptr); > > return 0; > > } > > Sorry Takashi, > > I'm not a coder so I only know that it wont compile because there isn't a pcm > variable passed into that function - it's snd_pcm_hw_t *hw. Oops, sorry, pasted a wrong version. Here it is. thanks, Takashi --- diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index e9ce092..5211d1c 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -131,6 +131,10 @@ struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm) static int sync_ptr1(snd_pcm_hw_t *hw, unsigned int flags) { int err; + long old_hwptr, new_hwptr; + long old_applptr, new_applptr; + old_hwptr = hw->sync_ptr->s.status.hw_ptr; + old_applptr = hw->sync_ptr->c.control.appl_ptr; hw->sync_ptr->flags = flags; err = ioctl((hw)->fd, SNDRV_PCM_IOCTL_SYNC_PTR, (hw)->sync_ptr); if (err < 0) { @@ -138,6 +142,11 @@ static int sync_ptr1(snd_pcm_hw_t *hw, unsigned int flags) SYSMSG("SNDRV_PCM_IOCTL_SYNC_PTR failed"); return err; } + new_hwptr = hw->sync_ptr->s.status.hw_ptr; + new_applptr = hw->sync_ptr->c.control.appl_ptr; + printf("sync_ptr1: %ld(%ld), %ld(%ld)\n", + new_hwptr, new_hwptr - old_hwptr, + new_applptr, new_applptr - old_applptr); return 0; }

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[alsa-devel] Bad page state in process 'rmmod' cs4281
by "Timo T.Hyrsylä" 09 Mar '09

09 Mar '09

Trying to make Alsa work on our embedded arm system with ixdp425 and CS4281. kernel is 2.6.20.3 and the Alsas (driver+lib+utils) latest 1.0.19 Other than the two below errors, the system seems to work ok. When removing the snd_cs4281 module, dmesg reports multiple errors about bad page <0>Bad page state in process 'rmmod' <0>page:c4007300 flags:0x00004000 mapping:00000000 mapcount:0 count:0 <0>Trying to fix it up, but a reboot is needed <0>Backtrace: <0>Bad page state in process 'rmmod' <0>page:c4007320 flags:0x00004000 mapping:00000000 mapcount:0 count:0 <0>Trying to fix it up, but a reboot is needed <0>Backtrace: I've managed to track down the error upto dma_free_coherent. snd_pcm_free > snd_pcm_lib_preallocate_free_for_all > snd_pcm_lib_preallocate_free > snd_pcm_lib_preallocate_dma_free > snd_dma_free_pages > snd_free_dev_pages > dma_free_coherent Another issue, possibly related, is that when running aplay with --mmap it will crash with: Internal error: Oops: f3 [#1] PC is at snd_pcm_mmap_data_nopage+0x10c/0x148 [snd_pcm] LR is at __handle_mm_fault+0x16c/0x8ac any helping ideas? -- Timo TH.

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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 09 Mar '09

09 Mar '09

The following changes since commit f271fa28fbaf947d9c79f188dd149176da727dd5: Takashi Iwai (1): ASoC: Fix Kconfig dependency of CONFIG_SND_S3C24XX_SOC_JIVE_WM8750 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Daniel Mack (2): ASoC: bring cs4270 feature/limitations list in sync ASoC: Add a driver for AK4104 S/PDIF transmitter Hugo Villeneuve (1): ASoC: Davinci: Fix incorrect machine type for SFFSDR board Mark Brown (3): ASoC: Re-remove hand-rolled pr_debug() macros ASoC: Fix logging severity for some S3C error messages ASoC: Remove unneeded forward reference to WM8753 SPI implementation Timur Tabi (1): ASoC: Improve pause/unpause performance in Freescale 8610 drivers sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4104.c | 363 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4104.h | 7 + sound/soc/codecs/cs4270.c | 15 +- sound/soc/codecs/wm8753.c | 5 - sound/soc/davinci/Kconfig | 2 +- sound/soc/fsl/fsl_dma.c | 3 +- sound/soc/fsl/fsl_ssi.c | 23 ++- sound/soc/s3c24xx/neo1973_wm8753.c | 44 ++--- sound/soc/s3c24xx/s3c-i2s-v2.c | 49 ++--- sound/soc/s3c24xx/s3c2412-i2s.c | 12 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 49 ++--- sound/soc/s3c24xx/s3c24xx-pcm.c | 45 ++--- 14 files changed, 482 insertions(+), 141 deletions(-) create mode 100644 sound/soc/codecs/ak4104.c create mode 100644 sound/soc/codecs/ak4104.h

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[alsa-devel] [PATCH] ASoC: Add a driver for AK4104 S/PDIF transmitter
by Daniel Mack 09 Mar '09

09 Mar '09

This adds a driver for the SPI connected AK4104 S/PDIF transmitter device. It's features are fairly simple, but as there is need to set up certain bits in the IEC958 information, this better goes into a real driver. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)sirena.org.uk> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4104.c | 367 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4104.h | 7 + 4 files changed, 380 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4104.c create mode 100644 sound/soc/codecs/ak4104.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 628a591..a1af311 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -14,6 +14,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 if I2C + select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 @@ -60,6 +61,9 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate +config SND_SOC_AK4104 + tristate + config SND_SOC_AK4535 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3664cdc..4717c3c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,6 +1,7 @@ snd-soc-ac97-objs := ac97.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o @@ -30,6 +31,7 @@ snd-soc-wm9713-objs := wm9713.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c new file mode 100644 index 0000000..8508a22 --- /dev/null +++ b/sound/soc/codecs/ak4104.c @@ -0,0 +1,367 @@ +/* + * AK4104 ALSA SoC (ASoC) driver + * + * Copyright (c) 2009 Daniel Mack <daniel(a)caiaq.de> + * + * Copyright 2007-2009 Freescale Semiconductor, Inc. This file is licensed + * under the terms of the GNU General Public License version 2. This + * program is licensed "as is" without any warranty of any kind, whether + * express or implied. + */ + +#include <linux/module.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <linux/spi/spi.h> +#include <sound/asoundef.h> + +#include "ak4104.h" + +/* AK4104 registers addresses */ +#define AK4104_REG_CONTROL1 0x00 +#define AK4104_REG_RESERVED 0x01 +#define AK4104_REG_CONTROL2 0x02 +#define AK4104_REG_TX 0x03 +#define AK4104_REG_CHN_STATUS(x) ((x) + 0x04) +#define AK4104_NUM_REGS 10 + +#define AK4104_REG_MASK 0x1f +#define AK4104_READ 0xc0 +#define AK4104_WRITE 0xe0 +#define AK4104_RESERVED_VAL 0x5b + +/* Bit masks for AK4104 registers */ +#define AK4104_CONTROL1_RSTN (1 << 0) +#define AK4104_CONTROL1_PW (1 << 1) +#define AK4104_CONTROL1_DIF0 (1 << 2) +#define AK4104_CONTROL1_DIF1 (1 << 3) + +#define AK4104_CONTROL2_SEL0 (1 << 0) +#define AK4104_CONTROL2_SEL1 (1 << 1) +#define AK4104_CONTROL2_MODE (1 << 2) + +#define AK4104_TX_TXE (1 << 0) +#define AK4104_TX_V (1 << 1) + +#define DRV_NAME "ak4104" + +struct ak4104_private { + struct snd_soc_codec codec; + u8 reg_cache[AK4104_NUM_REGS]; +}; + +static int ak4104_fill_cache(struct snd_soc_codec *codec) +{ + int i; + u8 *reg_cache = codec->reg_cache; + struct spi_device *spi = codec->control_data; + + for (i = 0; i < codec->reg_cache_size; i++) { + int ret = spi_w8r8(spi, i | AK4104_READ); + if (ret < 0) { + dev_err(&spi->dev, "SPI write failure\n"); + return ret; + } + + reg_cache[i] = ret; + } + + return 0; +} + +static unsigned int ak4104_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *reg_cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + return reg_cache[reg]; +} + +static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *cache = codec->reg_cache; + struct spi_device *spi = codec->control_data; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + reg &= AK4104_REG_MASK; + reg |= AK4104_WRITE; + + /* only write to the hardware if value has changed */ + if (cache[reg] != value) { + u8 tmp[2] = { reg, value }; + if (spi_write(spi, tmp, sizeof(tmp))) { + dev_err(&spi->dev, "SPI write failed\n"); + return -EIO; + } + + cache[reg] = value; + } + + return 0; +} + +static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int val = 0; + + val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); + if (val < 0) + return val; + + val &= ~(AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1); + + /* set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= AK4104_CONTROL1_DIF0; + break; + case SND_SOC_DAIFMT_I2S: + val |= AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1; + break; + default: + dev_err(codec->dev, "invalid dai format\n"); + return -EINVAL; + } + + /* This device can only be slave */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + return ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); +} + +static int ak4104_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int ret, val = 0; + + /* set the IEC958 bits: consumer mode, no copyright bit */ + val |= IEC958_AES0_CON_NOT_COPYRIGHT; + ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(0), val); + + val = 0; + + switch (params_rate(params)) { + case 44100: + val |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + val |= IEC958_AES3_CON_FS_48000; + break; + case 32000: + val |= IEC958_AES3_CON_FS_32000; + break; + default: + dev_err(codec->dev, "unsupported sampling rate\n"); + return -EINVAL; + } + + ret = ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val); + if (ret < 0) + return ret; + + return 0; +} + +struct snd_soc_dai ak4104_dai = { + .name = DRV_NAME, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_32000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE + }, + .ops = { + .hw_params = ak4104_hw_params, + .set_fmt = ak4104_set_dai_fmt, + } +}; + +static struct snd_soc_codec *ak4104_codec; + +static int ak4104_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct ak4104_private *ak4104; + int ret, val; + + spi->bits_per_word = 8; + spi->mode = SPI_MODE_0; + ret = spi_setup(spi); + if (ret < 0) + return ret; + + ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL); + if (!ak4104) { + dev_err(&spi->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + codec = &ak4104->codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->dev = &spi->dev; + codec->name = DRV_NAME; + codec->owner = THIS_MODULE; + codec->dai = &ak4104_dai; + codec->num_dai = 1; + codec->private_data = ak4104; + codec->control_data = spi; + codec->reg_cache = ak4104->reg_cache; + codec->reg_cache_size = AK4104_NUM_REGS; + + /* read all regs and fill the cache */ + ret = ak4104_fill_cache(codec); + if (ret < 0) { + dev_err(&spi->dev, "failed to fill register cache\n"); + return ret; + } + + /* read the 'reserved' register - according to the datasheet, it + * should contain 0x5b. Not a good way to verify the presence of + * the device, but there is no hardware ID register. */ + if (ak4104_read_reg_cache(codec, AK4104_REG_RESERVED) != + AK4104_RESERVED_VAL) { + ret = -ENODEV; + goto error_free_codec; + } + + /* set power-up and non-reset bits */ + val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); + val |= AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN; + ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); + if (ret < 0) + goto error_free_codec; + + /* enable transmitter */ + val = ak4104_read_reg_cache(codec, AK4104_REG_TX); + val |= AK4104_TX_TXE; + ret = ak4104_spi_write(codec, AK4104_REG_TX, val); + if (ret < 0) + goto error_free_codec; + + ak4104_codec = codec; + ret = snd_soc_register_dai(&ak4104_dai); + if (ret < 0) { + dev_err(&spi->dev, "failed to register DAI\n"); + goto error_free_codec; + } + + spi_set_drvdata(spi, ak4104); + dev_info(&spi->dev, "SPI device initialized\n"); + return 0; + +error_free_codec: + kfree(ak4104); + ak4104_dai.dev = NULL; + return ret; +} + +static int __devexit ak4104_spi_remove(struct spi_device *spi) +{ + int ret, val; + struct ak4104_private *ak4104 = spi_get_drvdata(spi); + + val = ak4104_read_reg_cache(&ak4104->codec, AK4104_REG_CONTROL1); + if (val < 0) + return val; + + /* clear power-up and non-reset bits */ + val &= ~(AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); + ret = ak4104_spi_write(&ak4104->codec, AK4104_REG_CONTROL1, val); + if (ret < 0) + return ret; + + ak4104_codec = NULL; + kfree(ak4104); + return 0; +} + +static int ak4104_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = ak4104_codec; + int ret; + + /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ + socdev->card->codec = codec; + + /* Register PCMs */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms\n"); + return ret; + } + + /* Register the socdev */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + snd_soc_free_pcms(socdev); + return ret; + } + + return 0; +} + +static int ak4104_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + snd_soc_free_pcms(socdev); + return 0; +}; + +struct snd_soc_codec_device soc_codec_device_ak4104 = { + .probe = ak4104_probe, + .remove = ak4104_remove +}; +EXPORT_SYMBOL_GPL(soc_codec_device_ak4104); + +static struct spi_driver ak4104_spi_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = ak4104_spi_probe, + .remove = __devexit_p(ak4104_spi_remove), +}; + +static int __init ak4104_init(void) +{ + pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n"); + return spi_register_driver(&ak4104_spi_driver); +} +module_init(ak4104_init); + +static void __exit ak4104_exit(void) +{ + spi_unregister_driver(&ak4104_spi_driver); +} +module_exit(ak4104_exit); + +MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); +MODULE_DESCRIPTION("Asahi Kasei AK4104 ALSA SoC driver"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/ak4104.h b/sound/soc/codecs/ak4104.h new file mode 100644 index 0000000..eb88fe7 --- /dev/null +++ b/sound/soc/codecs/ak4104.h @@ -0,0 +1,7 @@ +#ifndef _AK4104_H +#define _AK4104_H + +extern struct snd_soc_dai ak4104_dai; +extern struct snd_soc_codec_device soc_codec_device_ak4104; + +#endif -- 1.6.1.3

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[alsa-devel] configure script doesn't succesfully end with the latest snapshots
by Ozan Çağlayan 09 Mar '09

09 Mar '09

Latest working snapshots: alsa-driver-20090305.tar.bz2 alsa-driver-20090306.tar.bz2 Affected snapshots: alsa-driver-20090307.tar.bz2 alsa-driver-20090308.tar.bz2 Problem -------- -> 20090307 During the configuration step, configure script complains that it can't find some commands: checking whether gcc needs -traditional... no checking for current directory... /root/devel/kernel/drivers/alsa-driver/alsa-driver-20090307 ./configure: line 3863: ALSA_TOPLEVEL_INIT: command not found checking cross compile... checking for directory with kernel source... /lib/modules/2.6.25.20-114/source checking for directory with kernel build... /lib/modules/2.6.25.20-114/build .. .. checking for CONFIG_HAS_DMA... yes ./configure: line 13325: ALSA_TOPLEVEL_SELECT: command not found ./configure: line 13327: ALSA_PARSE_KCONFIG: command not found ./configure: line 13329: ALSA_TOPLEVEL_DEFINES: command not found ./configure: line 13339: ALSA_TOPLEVEL_OUTPUT: command not found configure: creating ./config.status Eventhough it complains, the configuration step ends succesfully and I can compile the drivers with make. -> 20090308 It's even worse. After complaining about the same ALSA_* stuff, it fails also on the following error: configure: creating ./config.status config.status: creating version config.status: creating Makefile.conf config.status: WARNING: 'Makefile.conf.in' seems to ignore the --datarootdir setting config.status: creating snddevices config.status: creating utils/alsa-driver.spec config.status: creating utils/buildrpm config.status: creating toplevel.config config.status: creating utils/alsasound config.status: creating utils/alsasound.posix config.status: creating include/pci_ids_compat.h config.status: creating include/i2c-id_compat.h config.status: creating include/config.h config.status: error: cannot find input file: include/config1.h.in -- Ozan Çağlayan <ozan_at_pardus.org.tr>

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[alsa-devel] SoC scenarii API
by Robert Jarzmik 08 Mar '09

08 Mar '09

Hi Liam and Mark, As mioa701 submission was stopped due to the need of a generic scenarii API, this a first attempt to design such an API in order to unblock mioa701 submission. This is oversimplified, no convenience macros are provided, just external data structures and namespace polution. I'm expecting a lot of comments from both of you. Cheers. -- Robert enum pin_change { SCEN_PIN_OFF = 0, /* Disable the pin */ SCEN_PIN_ON, /* Enable the pin */ SCEN_PIN_LEAVE /* Leave the pin in previous state */ }; struct setup_mixmux { char *mixname; /* Codec Mixer or Muxer name */ int val; /* Codec Mixer or Muxer value to enforce */ }; /** * struct soc_scenario - describes one sound usecase * @name: Scenario name, value as will be seen in alsa "SoC Mode" alsa control * @pin_setup: Pin configuration to perform on scenario activation * Table of all pins, as they should be configured. Each elements is a * pin_change value, describing how to handle a specific pin. * Table size must be the same as in snd_soc_scenario_init(). * @mixer_cleanup: Mixers/muxes to set up in phase (b) * Table of all mixers/muxes to modify, NULL terminated. * @mixer_setup: Mixers/muxers to set up in phase (c) * Table of all mixers/muxes to modify, NULL terminated * @lvol_master: Left volume aliased to "SoC Volume" * @rvol_master: Right volume aliased to "SoC Volume" * @lvol_mute: Left volume mute aliased to "SoC Volume" mute control * @rvol_mute: Right volume mute aliased to "SoC Volume" mute control * * This structure describes what a scenario change implies. The behaviour is to : * a) enable several pins, disable others, leave others in their state * (understand here snd_soc_dapm_enable_pin) * b) set up some mixers and muxers to prepare the change, and clean up. * This should be normally used to restore a "default state" if need be, before * activating the mixers/muxers into their final state. * Note: table in NULL, or always terminated by NULL pointer * Note: this cleanup may/should be shared amongst all scenarii * c) set up some mixers and muxers to the final state * This should be normally used to restore a "default state" if need be, before * activating the mixers/muxers into their final state. * Note: table in NULL, or always terminated by NULL pointer */ struct soc_scenario { const char *name; /* scenario name */ const unsigned char pin_setup[]; /* pin_change for pins */ const struct setup_mixmux mixes_cleanup[]; /* mix cleanup */ const struct setup_mixmux mixes_setup[]; /* mix setup */ const char *lvol_master; /* left volume master */ const char *rvol_master; /* right volume master */ const char *lvol_mute; /* left volume mute */ const char *rvol_mute; /* right volume mute */ }; /** * snd_soc_scenario_init - initialize soc scenarii * @codec: codec associated to the pins/mixers/muxes/volumes/mutes * @scenario: table of scenarii * @nb_scenarii: number of scenarii * @pin_names: table of pin names * @nb_pins: number of pins handled by scenario * * Initialise the soc scenarii engine. The first scenario (0) will be used. * By default, the user could leave this scenario as non intrusive (not a single * pin changed, no mixers/muxes changes, and volume master inactive). */ int snd_soc_scenario_init(struct snd_soc_codec *codec, struct soc_scenario *scenario, int nb_scenarii, char *pin_names[], int nb_pins); /** * snd_soc_scenario_init - initialize soc scenarii * @codec: codec used for the init */ void snd_soc_scenario_release(struct snd_soc_codec *code);

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