August 2011 - Alsa-devel - mailman.alsa-project.org

[PATCH] alsa-plugins: Pulse: only underrun if no more data has been written
by David Henningsson 19 Aug '11

19 Aug '11

If more data has already been written after the underrun, the underrun will automatically end and therefore we should not report it or restart the stream. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.in | 7 +++++++ pulse/pcm_pulse.c | 28 ++++++++++++++++++++++++++-- 2 files changed, 33 insertions(+), 2 deletions(-) diff --git a/configure.in b/configure.in index ccf59ba..f6886b1 100644 --- a/configure.in +++ b/configure.in @@ -31,8 +31,14 @@ AC_ARG_ENABLE([pulseaudio], if test "x$enable_pulseaudio" != "xno"; then PKG_CHECK_MODULES(pulseaudio, [libpulse >= 0.9.11], [HAVE_PULSE=yes], [HAVE_PULSE=no]) + PKG_CHECK_MODULES(pulseaudio_099, [libpulse >= 0.99.1], + [HAVE_PULSE_UNDERRUN_INDEX=yes], [HAVE_PULSE_UNDERRUN_INDEX=no]) + fi AM_CONDITIONAL(HAVE_PULSE, test x$HAVE_PULSE = xyes) +if test "$HAVE_PULSE_UNDERRUN_INDEX" = "yes"; then + AC_DEFINE([HAVE_PULSE_UNDERRUN_INDEX], 1, Pulseaudio underrun index available) +fi AC_ARG_ENABLE([samplerate], AS_HELP_STRING([--disable-samplerate], [Disable building of samplerate plugin])) @@ -185,6 +191,7 @@ echo "Pulseaudio plugin: $HAVE_PULSE" if test "$HAVE_PULSE" = "yes"; then echo " pulseaudio_CFLAGS: $pulseaudio_CFLAGS" echo " pulseaudio_LIBS: $pulseaudio_LIBS" + echo " pulseaudio underrun index: $HAVE_PULSE_UNDERRUN_INDEX" fi echo "Samplerate plugin: $HAVE_SAMPLERATE" if test "$HAVE_SAMPLERATE" = "yes"; then diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index d6c6792..61f1c0c 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -26,6 +26,10 @@ #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + #include "pulse.h" typedef struct snd_pcm_pulse { @@ -39,9 +43,10 @@ typedef struct snd_pcm_pulse { size_t last_size; size_t ptr; int underrun; - int handle_underrun; + int handle_underrun; /* can be 0=never, 1=always or 2,3=only if more data has not been written */ size_t offset; + int64_t written; pa_stream *stream; @@ -459,6 +464,7 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, } /* Make sure the buffer pointer is in sync */ + pcm->written += writebytes; pcm->last_size -= writebytes; ret = update_ptr(pcm); if (ret < 0) @@ -594,7 +600,12 @@ static void stream_underrun_cb(pa_stream * p, void *userdata) if (!pcm->p) return; - pcm->underrun = 1; + if (pcm->handle_underrun == 1 +#ifdef HAVE_PULSE_UNDERRUN_INDEX + || pcm->written <= pa_stream_get_underflow_index(p) +#endif + ) + pcm->underrun = 1; } static void stream_latency_cb(pa_stream *p, void *userdata) { @@ -691,6 +702,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) goto finish; } + pcm->written = 0; pa_stream_set_state_callback(pcm->stream, stream_state_cb, pcm); pa_stream_set_latency_update_callback(pcm->stream, stream_latency_cb, pcm); @@ -983,7 +995,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) const char *server = NULL; const char *device = NULL; const char *fallback_name = NULL; +#ifdef HAVE_PULSE_UNDERRUN_INDEX + int handle_underrun = 2; +#else int handle_underrun = 0; +#endif int err; snd_pcm_pulse_t *pcm; @@ -1017,6 +1033,14 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) handle_underrun = err; continue; } + if (strcmp(id, "handle_underrun_detect") == 0) { + if ((err = snd_config_get_bool(n)) < 0) { + SNDERR("Invalid value for %s", id); + return -EINVAL; + } + handle_underrun = err ? handle_underrun | 2 : handle_underrun & ~2; + continue; + } if (strcmp(id, "fallback") == 0) { if (snd_config_get_string(n, &fallback_name) < 0) { SNDERR("Invalid value for %s", id); -- 1.7.5.4 --------------040503050706040000000002--

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[alsa-devel] Getting played tsms in ALSA
by Irfan Shaikh 19 Aug '11

19 Aug '11

Hello all At setup time, when we create alsa handle and get the pcm_status object from it. /******************************/ snd_pcm_status *status; snd_pcm_status_alloca(&status); At run time: snd_timestamp_t timestamp = {0}; snd_pcm_status(pcm_handle, status); snd_pcm_status_get_tstamp(status, &timestamp); print timestamp.tv_sec and timestamp.tv_usec /******************************/ Here when we start first instance, timestamp.tv_sec goes incrementing as we play as expected. But we when shutdown (destroy alsa handle, re-init) , the timestamps increase from the same point. i.e if we stopped playback at time x secs, the next instance starts showing the time from x secs. The time reported by snd_pcm_status_get_tstamp() resets after we reboot the board. Another observation is that when we have two instances. We start the first instance , pause it after x secs. And then start the next instance, the time given by ALSA again starts at x secs. I think the snd_pcm_get_tstamp should be handle based but it seems it's not the case. How can we reset the timestamp values ? Regards SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

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[alsa-devel] [PATCH 1/2] ucm: fix seg fault in execute_cset()
by Lu Guanqun 19 Aug '11

19 Aug '11

When there's no space in 'cset', 'pos' gets NULL, dereferencing 'pos' gets a seg fault. We need to be more robust. Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- src/ucm/main.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 8e9a85d..9ecbe43 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -196,7 +196,8 @@ static int execute_cset(snd_ctl_t *ctl, char *cset) goto __fail; err = 0; __fail: - *pos = ' '; + if (pos != NULL) + *pos = ' '; if (id != NULL) free(id);

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[alsa-devel] M-Audio FTU remaining mixer controls and external clocking support.
by Grant Diffey 19 Aug '11

19 Aug '11

Daniel, I remember you saying you'd added the remaining controls for the ftu and had worked out how to enable external clock support? Was that right or am I just remembering optimistically, Additionally have you had any thoughts on clemens outline of how the api between the pcm and usb stuff should work? Grant.

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[alsa-devel] Audio:suspend to ram issue
by Rajeev kumar 19 Aug '11

19 Aug '11

Hi Mark I am not sure about the behavior in case of audio suspend/resume. I tested the audio driver with aplay and then suspended to ram (using echo mem > /sys/power/state). After pressing wake to resume, the audio does not play unless aplay is closed and re-run. Do we expect the audio suspend/resume to work? In the platform trigger part I am doing nothing and in case of cpu(i2s) , just disabling the i2s. Best Regards Rajeev

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Re: [alsa-devel] hda-intel and runtime power-management
by Takashi Iwai 19 Aug '11

19 Aug '11

At Mon, 15 Aug 2011 16:04:46 -0500, Pierre-Louis Bossart wrote: > > > Another approach would be to set the stream to mute when paused and let > > the codec do the amp power check which will power it down if needed. > > What do you think? > > PulseAudio and other equivalent layers don't pause the output. They close > the ALSA sink after a timeout and reopen it when the sink/Hal state change. > You may want to avoid working your tail off with a seemingly smart way of > using the paused state, if in the end it's not that useful... Indeed, there are only few apps that use the pause operation, so it won't a big win in comparison to the added code complexity. thanks, Takashi

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Re: [alsa-devel] [PATCH 1/1] Sound, echoaudio: Place echoaudio drivers in submenu
by Takashi Iwai 19 Aug '11

19 Aug '11

At Thu, 18 Aug 2011 10:11:12 -0700, John Daiker wrote: > > Put all the echoaudio boards into their own Kconfig submneu. They used to be > listed individually. Because they are indeed individual drivers :) I don't mind much to include such a patch, but OTOH, people also don't like to add yet-another-config item. Also default=y isn't always preferred... thanks, Takashi > > Signed-off-by: John Daiker <daikerjohn(a)gmail.com> > --- > sound/pci/Kconfig | 25 +++++++++++++++++++++++++ > 1 files changed, 25 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig > index 8816804..9344274 100644 > --- a/sound/pci/Kconfig > +++ b/sound/pci/Kconfig > @@ -312,8 +312,18 @@ config SND_CTXFI > To compile this driver as a module, choose M here: the module > will be called snd-ctxfi. > > +config SND_ECHOAUDIO > + tristate "Echoaudio Support" > + default y > + select SND_PCM > + help > + Support for EchoAudio based devices. > + > +if SND_ECHOAUDIO > + > config SND_DARLA20 > tristate "(Echoaudio) Darla20" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -324,6 +334,7 @@ config SND_DARLA20 > > config SND_GINA20 > tristate "(Echoaudio) Gina20" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -334,6 +345,7 @@ config SND_GINA20 > > config SND_LAYLA20 > tristate "(Echoaudio) Layla20" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_RAWMIDI > select SND_PCM > @@ -345,6 +357,7 @@ config SND_LAYLA20 > > config SND_DARLA24 > tristate "(Echoaudio) Darla24" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -355,6 +368,7 @@ config SND_DARLA24 > > config SND_GINA24 > tristate "(Echoaudio) Gina24" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -365,6 +379,7 @@ config SND_GINA24 > > config SND_LAYLA24 > tristate "(Echoaudio) Layla24" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_RAWMIDI > select SND_PCM > @@ -376,6 +391,7 @@ config SND_LAYLA24 > > config SND_MONA > tristate "(Echoaudio) Mona" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_RAWMIDI > select SND_PCM > @@ -387,6 +403,7 @@ config SND_MONA > > config SND_MIA > tristate "(Echoaudio) Mia" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_RAWMIDI > select SND_PCM > @@ -398,6 +415,7 @@ config SND_MIA > > config SND_ECHO3G > tristate "(Echoaudio) 3G cards" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_RAWMIDI > select SND_PCM > @@ -409,6 +427,7 @@ config SND_ECHO3G > > config SND_INDIGO > tristate "(Echoaudio) Indigo" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -419,6 +438,7 @@ config SND_INDIGO > > config SND_INDIGOIO > tristate "(Echoaudio) Indigo IO" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -429,6 +449,7 @@ config SND_INDIGOIO > > config SND_INDIGODJ > tristate "(Echoaudio) Indigo DJ" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -439,6 +460,7 @@ config SND_INDIGODJ > > config SND_INDIGOIOX > tristate "(Echoaudio) Indigo IOx" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -449,6 +471,7 @@ config SND_INDIGOIOX > > config SND_INDIGODJX > tristate "(Echoaudio) Indigo DJx" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -457,6 +480,8 @@ config SND_INDIGODJX > To compile this driver as a module, choose M here: the module > will be called snd-indigodjx > > +endif > + > config SND_EMU10K1 > tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" > select FW_LOADER > -- > 1.7.4.1 >

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[alsa-devel] [PATCH] Prefer kernel-versioned firmware directory.
by linux＠audioscience.com 19 Aug '11

19 Aug '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Different kernel versions may have different driver versions installed, which in turn require different firmware versions. If /lib/firmware/$(uname -r) exists, use it in preference to the generic /lib/firmware. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- configure.in | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/configure.in b/configure.in index c278987..ac9ee52 100644 --- a/configure.in +++ b/configure.in @@ -33,7 +33,9 @@ AM_CONDITIONAL(BUILD_FW, test "$buildfw" = "yes") AC_MSG_CHECKING(firmware installation directory) # where to put the firmware. If none of these is right, can specify where # --with-hotplug-dir -if test -d /lib/firmware ; then +if test -d /lib/firmware/$(uname -r) ; then + detected_fwdir="/lib/firmware/$(uname -r)" +elif test -d /lib/firmware ; then detected_fwdir="/lib/firmware" elif test -d /lib/hotplug/firmware ; then detected_fwdir="/lib/hotplug/firmware" -- 1.7.0.4

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Re: [alsa-devel] Lost dB volumes on USB headset between 2.6.38 and 3.0
by Alex Wolfson 19 Aug '11

19 Aug '11

> 2011/8/18 Takashi Iwai<tiwai at suse.de <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>>: > >/ At Tue, 16 Aug 2011 13:31:52 +0100, > />/ Colin Guthrie wrote: > />>/ > />>/ > diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c > />>/ > index c04d7c7..cdd19d7 100644 > />>/ > --- a/sound/usb/mixer.c > />>/ > +++ b/sound/usb/mixer.c > />>/ > @@ -152,6 +152,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p, > />>/ > if (p&& p->dB) { > />>/ > cval->dBmin = p->dB->min; > />>/ > cval->dBmax = p->dB->max; > />>/ > + cval->initialized = 1; > />>/ > } > />>/ > } > />>/ > > />>/ > @@ -1092,7 +1093,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, > />>/ > " Switch" : " Volume"); > />>/ > if (control == UAC_FU_VOLUME) { > />>/ > check_mapped_dB(map, cval); > />>/ > - if (cval->dBmin< cval->dBmax) { > />>/ > + if (cval->dBmin< cval->dBmax || !cval->initialized) { > />>/ > kctl->tlv.c = mixer_vol_tlv; > />>/ > kctl->vd[0].access |= > />>/ > SNDRV_CTL_ELEM_ACCESS_TLV_READ | > />>/ > />>/ > />>/ > />>/ Not had a chance to test this yet but I will when I get a moment. > />/ > />/ Any chance to test the patch? > />/ > / > The patch seem fix the no dB scale problem , however the broken dB > test which you added seem redundant as cval->dBmin and cval->dBmax are > still zero because error in get_min_max() at build_feature_ctl() and > it seem that the valid dBmin and dBMax are calculated by get_min_max() > at mixer_ctl_feature_info() on my usb logitech web cam > > > LSA endpoint.c:450: 2:3:1: add audio endpoint 0x86 > ALSA mixer.c:1182: [5] FU [Mic Capture Switch] ch = 1, val = 0/1/1 > ALSA mixer.c:804: 5:2: cannot get min/max values for control 2 (id 5) > ALSA mixer.c:1182: [5] FU [Mic Capture Volume] ch = 1, val = 0/1/1 > ALSA mixer.c:804: 5:2: cannot get min/max values for control 2 (id 5) > usbcore: registered new interface driver snd-usb-audio Fixed dB scale problem for my USB Logitech headset as well (046d:0a0b)

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Re: [alsa-devel] [PATCH] ASoC: sta32x: Move resource allocation and release to the corresponding callback functions
by Mark Brown 18 Aug '11

18 Aug '11

On Thu, Aug 18, 2011 at 03:31:04PM +0800, Axel Lin wrote: > This patch includes below small fixes: > > 1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove(). > 2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(), > as we will call regulator_bulk_free() in sta32x_remove(). > 3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove. > The i2c core will set the clientdata to NULL. Applied, thanks.

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