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[alsa-devel] Alsa: Get Playback Volume fails if set by outside source
by Ashwin Chandra 16 May '08

16 May '08

Hello, I am having an issue with the Alsa mixer. Currenty I have a function that gets the volume via snd_mixer_selem_get_playback_volume. This function works fine to get the last-set volume if my application is setting the volume. However if I change the volume using an outside source, such as alsamixer or gnome-volume-control and then I run the above function, the old volume that my application had set last is returned. Now I edited amixer's source code to put a while(1) around the same functions (the snd_mixer_selem_get_playback_volume), and it too does not get the correct volume if set from an outside source. Alsamixer on the other hand seems to work find, but I am not quite sure how they do it. I thought maybe I had to reinitialize the alsa interface (ie. snd_mixer_close...and then open, attach, register, etc), but it doesn't seem that alsamixer does that. Does anyone have any suggestions about what I can do to get the correct volume if set by an outside source?? Thanks!

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[alsa-devel] pcm output timing question
by Paul Adolph 15 May '08

15 May '08

ALSA newbie warning... I have a funny application that does av sync by using the timing of a callback to set the audio timebase. In other words, I need to invoke that callback "as close as possible" to the time a previously-sent buffer of samples is presented, i.e. exits the speakers. The application uses the clock time of the arrival of this callback and does some math to figure out whether to repeat or skip video frames. I do not like this arrangement, but unfortunately I am stuck with it. My job is to hook this app to ALSA audio output. My current design uses a producer-consumer model with a thread that writes fixed-size application buffers to the ALSA ring buffer, driven by a message queue fed by the application. This is a little redundant I think, but I do not own the buffers sent by the app so this is what I came up with. The application needs a regular callback with the timing described above each time one of their buffers is played, and it is assumed they are played in sequence. I tried setting the ALSA callback to trigger when there are an application-buffer-size amount of frames available, using snd_pcm_sw_params_set_avail_min(). My idea was that if I primed the ring buffer to be full before it starts transfers with snd_pcm_sw_params_set_start_threshold(), I would get the ALSA callback each time that number of frames exited the ring buffer. But this did not work: I seem to be getting callbacks much more frequently that I wanted, probably on ALSA period boundaries. So this means I do not understand how snd_pcm_sw_params_set_start_threshold() is supposed to work I guess. Will somebody please comment on the viability of and/or problems with my current design, or suggest an alternative way to configure ALSA that will give me the output timing I need? Currently I am testing on an x86 ubuntu box, kernel 2.6.22-14, ALSA 1.0.14, gcc 4.1.3, glibc 2.6.1, but will also cross-compile for several embedded platforms in the future. Thanks in advance.

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[alsa-devel] Oh, BTW... (OPL3 PM issue)
by Andreas Mohr 15 May '08

15 May '08

Hi, unfortunately I forgot to mention this: when I tested OPL3 functionality of my driver, I of course also tested this to do proper power management, and... well... it didn't! The result of doing an S2D was that the formerly perfectly playing emulated MIDI synth ended up with distorted (sort of Micky-Mouse'ish) voices directly after resume. I really don't think that it's my driver's power management which is at fault here, and I'd strongly guesstimate that ALSA does its own generic OPL3 power management handling for standard OPL3 hardware registration. So??? (aka "what the...!?") Thanks, Andreas Mohr

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[alsa-devel] [PATCH asoc-dev (take #2)] ASoC: Convert N810 machine driver to use gpiolib
by Jarkko Nikula 15 May '08

15 May '08

Use gpiolib since it is now available for OMAPs. Change also references to HW version RX44 to product name N810. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/n810.c | 25 ++++++++++++++----------- 1 files changed, 14 insertions(+), 11 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c2afffb..c32487b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -30,15 +30,15 @@ #include <asm/mach-types.h> #include <asm/arch/hardware.h> -#include <asm/arch/gpio.h> +#include <linux/gpio.h> #include <asm/arch/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" #include "../codecs/tlv320aic3x.h" -#define RX44_HEADSET_AMP_GPIO 10 -#define RX44_SPEAKER_AMP_GPIO 101 +#define N810_HEADSET_AMP_GPIO 10 +#define N810_SPEAKER_AMP_GPIO 101 static struct clk *sys_clkout2; static struct clk *sys_clkout2_src; @@ -154,9 +154,9 @@ static int n810_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 0); return 0; } @@ -165,9 +165,9 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1); + gpio_set_value(N810_HEADSET_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0); + gpio_set_value(N810_HEADSET_AMP_GPIO, 0); return 0; } @@ -303,12 +303,12 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0) + if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) BUG(); - if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0) + if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) BUG(); - omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0); - omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); + gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); return 0; err2: @@ -323,6 +323,9 @@ err1: static void __exit n810_soc_exit(void) { + gpio_free(N810_SPEAKER_AMP_GPIO); + gpio_free(N810_HEADSET_AMP_GPIO); + platform_device_unregister(n810_snd_device); } -- 1.5.5.1

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[alsa-devel] [PATCH asoc-dev] ASoC: Convert N810 machine driver to use gpiolib
by Jarkko Nikula 15 May '08

15 May '08

Use gpiolib since it is now available for OMAPs. Change also references to HW version RX44 to product name N810. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/n810.c | 25 ++++++++++++++----------- 1 files changed, 14 insertions(+), 11 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c2afffb..2d982b1 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -30,15 +30,15 @@ #include <asm/mach-types.h> #include <asm/arch/hardware.h> -#include <asm/arch/gpio.h> +#include <asm/gpio.h> #include <asm/arch/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" #include "../codecs/tlv320aic3x.h" -#define RX44_HEADSET_AMP_GPIO 10 -#define RX44_SPEAKER_AMP_GPIO 101 +#define N810_HEADSET_AMP_GPIO 10 +#define N810_SPEAKER_AMP_GPIO 101 static struct clk *sys_clkout2; static struct clk *sys_clkout2_src; @@ -154,9 +154,9 @@ static int n810_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 0); return 0; } @@ -165,9 +165,9 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1); + gpio_set_value(N810_HEADSET_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0); + gpio_set_value(N810_HEADSET_AMP_GPIO, 0); return 0; } @@ -303,12 +303,12 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0) + if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) BUG(); - if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0) + if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) BUG(); - omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0); - omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); + gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); return 0; err2: @@ -323,6 +323,9 @@ err1: static void __exit n810_soc_exit(void) { + gpio_free(N810_SPEAKER_AMP_GPIO); + gpio_free(N810_HEADSET_AMP_GPIO); + platform_device_unregister(n810_snd_device); } -- 1.5.5.1

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[alsa-devel] pcm output timing question
by Paul Adolph 14 May '08

14 May '08

ALSA newbie warning... I have a funny application that does av sync by using the timing of a callback to set the audio timebase. In other words, I need to invoke that callback "as close as possible" to the time a previously-sent buffer of samples is presented, i.e. exits the speakers. The application uses the clock time of the arrival of this callback and does some math to figure out whether to repeat or skip video frames. I do not like this arrangement, but unfortunately I am stuck with it. My job is to hook this app to ALSA audio output. My current design uses a producer-consumer model with a thread that writes fixed-size application buffers to the ALSA ring buffer, driven by a message queue fed by the application. This is a little redundant I think, but I do not own the buffers sent by the app so this is what I came up with. The application needs a regular callback with the timing described above each time one of their buffers is played, and it is assumed they are played in sequence. I tried setting the ALSA callback to trigger when there are an application-buffer-size amount of frames available, using snd_pcm_sw_params_set_avail_min(). My idea was that if I primed the ring buffer to be full before it starts transfers with snd_pcm_sw_params_set_start_threshold(), I would get the ALSA callback each time that number of frames exited the ring buffer. But this did not work: I seem to be getting callbacks much more frequently that I wanted, probably on ALSA period boundaries. So this means I do not understand how snd_pcm_sw_params_set_start_threshold() is supposed to work I guess. Will somebody please comment on the viability of and/or problems with my current design, or suggest an alternative way to configure ALSA that will give me the output timing I need? Currently I am testing on an x86 ubuntu box, kernel 2.6.22-14, ALSA 1.0.14, gcc 4.1.3, glibc 2.6.1, but will also cross-compile for several embedded platforms in the future. Thanks in advance.

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[alsa-devel] [Via8237] - VT1618 data sheet
by Goran Sterjov 14 May '08

14 May '08

Hello. i tried out the VT1618 patch from john utz but i failed to get the "Smart 5.1" surround sound working. so i figured that i'd go ahead and experiment with the driver to make it work on my via epia en15000 motherboard. i tried requesting the data sheet from VIA ten ga-gillion times but they only responded once with a "the datasheet is a high level engineering document. visit the technical support forums for help". i tried explaining that i needed the information to further develop an ALSA driver for the linux system but it seems like i wont get my hands on this god forsaken document. Is there anyone out there who has any documentation on the vt1618..? via support is shocking. Nay, beyond shocking. in John's patch he mentions various registers but unfortunately they dont help me without the docs.. Thanks Goran

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[alsa-devel] ioctl request code for specific alsa control
by Peter Wurmsdobler 14 May '08

14 May '08

Hello, The driver for our audio hardware provides 6 snd_kcontrol_new_t structures, with their proper callbacks, members and name. One is called pmic_control_pb_vol (.name = "Master Playback Volume") in the driver, but appears somehow as "Master" in amixer and can be successfully written to using the OSS interface and ioctl as: int mixerFd = open("/dev/mixer", O_RDWR); if (mixerFd >= 0) { unsigned int leftright; ioctl(mixerFd, SOUND_MIXER_WRITE_VOLUME, &leftright); close(mixerFd); } It is a big enigma for me that the ioctl with the request code used above arrives automagically in the particular audio driver and calls pmic_pb_volume_put() defined in pmic_control_pb_vol. Another control is called pmic_control_op_sw (.name = "Master Output Playback Volume") and appears in amixer as "Master Output" (in fact it is not a volume control but a switch). It is possible to set the output device using amixer. My questions are: What is the ioctl request code for a particular snd_kcontrol_new_t defined in the driver? Alternatively, how does the OSS layer in ALSA map the different ALSA controls to the predefined and fixed OSS controls? The hard way (and what I am currently trying as last resort) is to write to all of them and see what happens. Many thanks for your help, peter

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[alsa-devel] From snd_kcontrol_new_t to amixer
by Peter Wurmsdobler 14 May '08

14 May '08

Hello, I am puzzled by an observation that runs against my common sense. The pmic-alsa-mixer.c provides 6 (six) snd_kcontrol_new_t controls with their callback functions, name and other members, then registers themusing snd_ctl_add(...). However, amixer shows only 5 (five), and all with different names. What is the process of generating these names if not from the kernel structures? By looking through the mixer related code in alsa-utils and alsa-libs, I see that somehow the library loads the mixer controls and possibly retrieves all the information provided by them, including their names. Either some piece of code shortens the names defined in the driver, or it maps them to some generic controls. Any suggestions? The system I am working on uses a Freescale imx27 in conjunction with an MC13783 power management IC, also responsible for audio (it provides a voice codec and a stereo dac). The kernel used is a 2.6.19.2 as part of ltib from Freescale. amixer tells me: mx27# amixer Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined cvolume Playback channels: Mono Capture channels: Mono Limits: Playback 0 - 100 Capture 0 - 100 Mono: Playback 0 [0%] Capture 0 [0%] Simple mixer control 'Master Balance',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 50 [50%] Simple mixer control 'Master Input',0 Capabilities: cvolume Capture channels: Mono Limits: Capture 0 - 7 Mono: Capture 1 [14%] Simple mixer control 'Master Monoconfig',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 3 Mono: Playback 0 [0%] Simple mixer control 'Master Output',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 12 [80%] snd_kcontrol_new_t pmic_control_pb_vol __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .index = 0x00, .info = pmic_pb_volume_info, .get = pmic_pb_volume_get, .put = pmic_pb_volume_put, .private_value = 0xffab1, }; snd_kcontrol_new_t pmic_control_pb_bal __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Balance Playback Volume", .index = 0x00, .info = pmic_pb_balance_info, .get = pmic_pb_balance_get, .put = pmic_pb_balance_put, .private_value = 0xffab2, }; snd_kcontrol_new_t pmic_control_pb_mono __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Monoconfig Playback Volume", .index = 0x00, .info = pmic_pb_monoconfig_info, .get = pmic_pb_monoconfig_get, .put = pmic_pb_monoconfig_put, .private_value = 0xffab2, }; snd_kcontrol_new_t pmic_control_op_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Output Playback Volume", .index = 0x00, .info = pmic_mixer_output_info, .get = pmic_mixer_output_get, .put = pmic_mixer_output_put, .private_value = 0xffab4, }; snd_kcontrol_new_t pmic_control_cap_vol __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Capture Volume", .index = 0x00, .info = pmic_cap_volume_info, .get = pmic_cap_volume_get, .put = pmic_cap_volume_put, .private_value = 0xffab5, }; snd_kcontrol_new_t pmic_control_ip_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Input Capture Volume", .index = 0x00, .info = pmic_cap_input_info, .get = pmic_cap_input_get, .put = pmic_cap_input_put, .private_value = 0xffab5, }; int mxc_alsa_create_ctl(snd_card_t * card, void *p_value) { int err = 0; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_op_sw, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_pb_vol, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_pb_mono, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_pb_bal, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_cap_vol, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_ip_sw, p_value))) < 0) return err; return 0; } Cheers, peter

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Re: [alsa-devel] Problem with aloop
by Benjamin van den Hout 14 May '08

14 May '08

Gustavo da Silva Serra wrote: I had some problems until I could figure it out how to configure alsa > properly in /etc/asound.conf. Maybe this could be your problem? What is > your alsa version? I heard that aloop didn't work well until 1.0.15, or > such (alsa guys: is this true?). > My alsa is 1.0.14 but I'm using the 'HG' version of the alsa-drivers and alsa-kernel packages so that basically gets me the latest-and-greatest drivers from the source repository if I understand correctly. I saw some patches coming by in February and they are already in there so I presume I'm working with the most current source. I'm intrigued by the asound.conf, is there any specific configuration you needed to make the aloop device work properly? I'm using the distro-configured alsa (ubuntu 7.10) and there is no such file somehwere in /etc. Even on my debian server (running Etch), I can't find it. Kind regards, Benjamin

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