October 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Constant noise on HDA ALC275
by Julien Danjou 08 Oct '12

08 Oct '12

Hi there, I've an HDA based chipset in a new Sony Vaio Z (Ivy Bridge). As sound as the system starts and the sound module are loaded, there's a constant noise (shhhhhhht) in the speakers. The noise change a bit if I toggle the mute button, but it's still there. This noise stops as sound as I plug something in the jack port. I've tried to play with HDA Analyzer¹, but I've no clue what to look for, so any hint appreciated. Some information: 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04) % aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC275 Analog [ALC275 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 % lsmod | grep snd snd_hda_codec_hdmi 30783 1 snd_hda_codec_realtek 50906 1 snd_hda_intel 26504 3 snd_hda_codec 83533 3 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel snd_hwdep 13186 1 snd_hda_codec snd_pcm 64080 3 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 12969 2 snd_pcm,snd_hda_intel snd_seq 45130 0 snd_seq_device 13176 1 snd_seq snd_timer 22917 2 snd_pcm,snd_seq snd 53077 15 snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 13026 1 snd /proc/asound/card0/codec#0 ========================== /proc/asound/card0/codec#3 ========================== ¹ http://www.alsa-project.org/main/index.php/HDA_Analyzer -- Julien Danjou # Free Software hacker & freelance # http://julien.danjou.info

2 16

Re: [alsa-devel] 3.6.0 regression, distorted sound failure while using jack and hd-audio
by Daniel Mack 08 Oct '12

08 Oct '12

(cc alsa-devel) On 07.10.2012 21:34, Helge Hafting wrote: > My sound setup works fine with 3.5.4, but have serious problems with 3.6.0 > > It is sound through jack that breaks, simply using "aplay" directly with the > alsa device works. > > When I use jack, my setup works fine with 3.5.4. But boot into 3.6.0, > and I get badly distorted sound as if using an extremely overdone reverb effect. > And jack reports tons of xruns, several per second. > > This can be worked around by going from 256 to 512 frames/period in the jack > settings. But that also double the latency, which makes keyboard playing less > pleasant. > > It is not a case of 256 frames/period being (almost) too extreme for the machine. With kernel 3.5.4, > I can use 128 frames/period and get mostly ok sound, with only an occational > xrun (and a correseponding pop in the sound). So I use 256 because that's solid. > But with 3.6.0, not even 256 is enough. And it is not an occational pop, but > awful distortion all the time. > > I don't really know what has changed from 3.5.4 to 3.6.0. The hd-audio driver? > Something in alsa? Or the scheduling? > > Machine: Dell precision M6600 laptop, 8GB, 2.2GHz quad-core i7. > Usually at 800MHz to save power, this is enough for making sound. > Kernels: 64-bit 3.5.4 or 3.6.0 > Distro: funtoo > > Audio according to lspci: > 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 04) > > Please tell if there is anything I should try. I can test other kernel versions > or try patches. > > Helge Hafting > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

2 1

[alsa-devel] [PATCH] ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
by David Henningsson 08 Oct '12

08 Oct '12

In case there is one "Headphone Jack" and one "Dock Headphone Jack", one of them will get an index, even though that is not needed. This patch fixes that issue. BugLink: https://bugs.launchpad.net/bugs/1060729 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 49 +++++++++++++++++++++++++++++---------- 1 file changed, 37 insertions(+), 12 deletions(-) Alsa-info at: https://launchpadlibrarian.net/118211803/AlsaInfo diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index a98e25e..4ec6dc8 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -498,6 +498,38 @@ static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, return channel_sfx[i]; } +static const char *check_output_pfx(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + int attr = snd_hda_get_input_pin_attr(def_conf); + + /* check the location */ + switch (attr) { + case INPUT_PIN_ATTR_DOCK: + return "Dock "; + case INPUT_PIN_ATTR_FRONT: + return "Front "; + } + return ""; +} + +static int get_hp_label_index(struct hda_codec *codec, hda_nid_t nid, + const hda_nid_t *pins, int num_pins) +{ + int i, j, idx = 0; + + const char *pfx = check_output_pfx(codec, nid); + + i = find_idx_in_nid_list(nid, pins, num_pins); + if (i < 0) + return -1; + for (j = 0; j < i; j++) + if (pfx == check_output_pfx(codec, pins[j])) + idx++; + + return idx; +} + static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, const struct auto_pin_cfg *cfg, const char *name, char *label, int maxlen, @@ -505,20 +537,13 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, { unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); int attr = snd_hda_get_input_pin_attr(def_conf); - const char *pfx = "", *sfx = ""; + const char *pfx, *sfx = ""; /* handle as a speaker if it's a fixed line-out */ if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) name = "Speaker"; - /* check the location */ - switch (attr) { - case INPUT_PIN_ATTR_DOCK: - pfx = "Dock "; - break; - case INPUT_PIN_ATTR_FRONT: - pfx = "Front "; - break; - } + pfx = check_output_pfx(codec, nid); + if (cfg) { /* try to give a unique suffix if needed */ sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs, @@ -528,8 +553,8 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, indexp); if (!sfx) { /* don't add channel suffix for Headphone controls */ - int idx = find_idx_in_nid_list(nid, cfg->hp_pins, - cfg->hp_outs); + int idx = get_hp_label_index(codec, nid, cfg->hp_pins, + cfg->hp_outs); if (idx >= 0) *indexp = idx; sfx = ""; -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: aloop - add locking to timer access
by Omair Mohammed Abdullah 08 Oct '12

08 Oct '12

When the loopback timer handler is running, calling del_timer() (for STOP trigger) will not wait for the handler to complete before deactivating the timer. The timer gets rescheduled in the handler as usual. Then a subsequent START trigger will try to start the timer using add_timer() with a timer pending leading to a kernel panic. Serialize the calls to add_timer() and del_timer() using a spin lock to avoid this. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/drivers/aloop.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 5a34355..0fe6d64 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -120,6 +120,7 @@ struct loopback_pcm { unsigned int last_drift; unsigned long last_jiffies; struct timer_list timer; + spinlock_t timer_lock; }; static struct platform_device *devices[SNDRV_CARDS]; @@ -170,6 +171,7 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) unsigned long tick; unsigned int rate_shift = get_rate_shift(dpcm); + spin_lock(&dpcm->timer_lock); if (rate_shift != dpcm->pcm_rate_shift) { dpcm->pcm_rate_shift = rate_shift; dpcm->period_size_frac = frac_pos(dpcm, dpcm->pcm_period_size); @@ -182,12 +184,15 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps; dpcm->timer.expires = jiffies + tick; add_timer(&dpcm->timer); + spin_unlock(&dpcm->timer_lock); } static inline void loopback_timer_stop(struct loopback_pcm *dpcm) { + spin_lock(&dpcm->timer_lock); del_timer(&dpcm->timer); dpcm->timer.expires = 0; + spin_unlock(&dpcm->timer_lock); } #define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK) @@ -667,6 +672,7 @@ static int loopback_open(struct snd_pcm_substream *substream) dpcm->substream = substream; setup_timer(&dpcm->timer, loopback_timer_function, (unsigned long)dpcm); + spin_lock_init(&dpcm->timer_lock); cable = loopback->cables[substream->number][dev]; if (!cable) { -- 1.7.4.1

3 3

[alsa-devel] [PATCH] hda-emu: Add check for unnecessary indices of kcontrols to testsuite
by David Henningsson 08 Oct '12

08 Oct '12

--- tester/runner.py | 29 +++++++++++++++++++++++------ 1 file changed, 23 insertions(+), 6 deletions(-) diff --git a/tester/runner.py b/tester/runner.py index 30118ce..1e7ecf9 100644 --- a/tester/runner.py +++ b/tester/runner.py @@ -25,7 +25,9 @@ class ControlInfo(): self.runner = runner carr = list_info.split(":") self.numid = carr[1] - self.name = carr[2] + self.name = carr[2][1:] + self.index, _ = tuple(carr[3].split(" ")) + self.index = int(self.index) def add_info(self, get_info): minmax_regex = re.compile("MIN/MAX: (\d+)/(\d+),\s+VAL:(( \\[\d+\\])+)") @@ -222,13 +224,28 @@ class HdaEmuRunner(): if (int(a) != int(b)): self.add_error("Tried to set " + c.name + " to " + str([int(x) for x in values]) + ", but got " + str(c.values) + " instead", "Error") - def run_kcontrol_test(self): - minmax_regex = re.compile("MIN/MAX: (\d+)/(\d+),\s+VAL:(( \\[\d+\\])+)") - val_regex = re.compile(" \\[(\d+)\\]") + def check_kcontrol_indices(self, cinfo): + d = {} + for c in cinfo: + if not c.name in d: + d[c.name] = [] + d[c.name].append(c.index) + + for k, v in d.iteritems(): + j = 0 + for i in sorted(v): + if i > j: + self.add_error("Unneccesary index on control '" + k + "' (" + str(i) + ")", "Error") +# Commented out - gives false positives for 'Playback Channel Map' +# elif i < j: +# self.add_error("Duplicate control '" + k + "' (" + str(i) + ")", "Error") + j = i+1 + def run_kcontrol_test(self): controls = self.run_command("list") - for control in controls: - c = ControlInfo(self, control) + cinfo = [ControlInfo(self, control) for control in controls] + self.check_kcontrol_indices(cinfo) + for c in cinfo: c.add_info(self.run_command("get " + c.numid)) if "minval" in c.__dict__: minval = c.minval -- 1.7.9.5

2 1

[alsa-devel] [PATCH] hda-emu: Improve directory/path handling for testsuite
by David Henningsson 08 Oct '12

08 Oct '12

There were some shortcomings when running the test scripts from a random directory. Also added functionality to test an entire directory tree at the same time. --- tester/runner.py | 9 ++++++++- tester/summary.py | 28 ++++++++++++++++++++++++---- 2 files changed, 32 insertions(+), 5 deletions(-) With this improvement we can run it over the entire alsa-info database. The initial result is 619 of 5911 (10%) failing, mostly due to invalid verbs sent (no input/output-amp, invalid amp index, setting *_EN without caps, etc). diff --git a/tester/runner.py b/tester/runner.py index b585ad3..30118ce 100644 --- a/tester/runner.py +++ b/tester/runner.py @@ -44,10 +44,13 @@ class ControlInfo(): class HdaEmuRunner(): def __init__(self): - self.child_args = "../hda-emu -M -F" + import os.path + hda_emu_path = os.path.join(os.path.dirname(os.path.realpath(__file__)),"../hda-emu") + self.child_args = '"' + hda_emu_path + '" -M -F' self.alsa_info = "/proc/asound/card0/codec#0" self.errors = 0 self.warnings = 0 + self.fatals = 0 self.print_errors = False self.comm_log_enabled = False self.last_command = None @@ -81,6 +84,8 @@ class HdaEmuRunner(): self.warnings += 1 else: self.errors += 1 + if severity == "Fatal": + self.fatals += 1 if self.print_errors: if not self.last_command_printed: if self.last_command: @@ -89,6 +94,8 @@ class HdaEmuRunner(): print "Encountered during initial parsing:" self.last_command_printed = True print " ", message + if self.fatals > 0: + raise Exception(message) def check_stdout(self): s = os.read(self.child.stdout.fileno(), 65536) diff --git a/tester/summary.py b/tester/summary.py index 6e94c9b..0f9fbf3 100755 --- a/tester/summary.py +++ b/tester/summary.py @@ -18,6 +18,11 @@ # You should have received a copy of the GNU General Public License along # with this program. If not, see <http://www.gnu.org/licenses/>. +def defaultpath(s): + import os.path + q = os.path.dirname(os.path.realpath(__file__)) + return os.path.join(q, s) + def main(): import os import os.path @@ -26,18 +31,28 @@ def main(): import argparse parser = argparse.ArgumentParser(description='Hda-emu automated test wrapper.') parser.add_argument('--verbose', '-v', action='count') + parser.add_argument('--directory', '-d', action='store', default=defaultpath("../codecs/canonical/")) + parser.add_argument('--recursive', '-r', action="store_true", default=False) + parser_dict = parser.parse_args() verbose = parser_dict.verbose + directory = parser_dict.directory - os.chdir(os.path.dirname(os.path.realpath(__file__))) - directory = "../codecs/canonical/" - files = os.listdir(directory) + if parser_dict.recursive: + files = [os.path.join(root, filename) for root, _, filenames in os.walk(directory) for filename in filenames] + else: + files = os.listdir(directory) + files.sort() successes = 0 fails = 0 warnings = 0 errors = 0 + index = 0 for f in files: + index += 1 + if verbose > 2: + print '[{0}/{1}]: Testing {2}'.format(index, len(files), f) try: r = runner.HdaEmuRunner() r.set_alsa_info_file(os.path.join(directory, f)) @@ -51,7 +66,12 @@ def main(): print '{0} errors, {1} warnings. ({2})'.format(r.errors, r.warnings, f) else: successes += 1 - except: + except KeyboardInterrupt: + import sys + sys.exit(1) + except Exception as e: + if verbose > 0: + print 'Fatal error for {0}: {1}'.format(f, e) errors += 1 fails += 1 -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - fix indices on boost volume on Conexant
by David Henningsson 08 Oct '12

08 Oct '12

After the recent patch "ALSA: hda - use both input paths on Conexant auto parser" suddenly we can have more than one "Mic Boost", this happened on Acer Aspire One 722. Therefore we must add the possibility to put an index on this "Mic Boost" just as we do for the other "Mic Boost" earlier in the same function. BugLink: https://bugs.launchpad.net/bugs/1059523 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c03d3b8..c11ad8a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4273,7 +4273,7 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, if (get_wcaps(codec, mux) & AC_WCAP_OUT_AMP) { spec->imux_info[idx].boost = mux; - return cx_auto_add_volume(codec, label, " Boost", 0, + return cx_auto_add_volume(codec, label, " Boost", cidx, mux, HDA_OUTPUT); } return 0; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix hang caused by race during suspend.
by Dylan Reid 08 Oct '12

08 Oct '12

There was a race condition when the system suspends while hda_power_work is running in the work queue. If system suspend (snd_hda_suspend) happens after the work queue releases power_lock but before it calls hda_call_codec_suspend, codec_suspend runs with power_on=0, causing the codec to power up for register reads, and hanging when it calls cancel_delayed_work_sync from the running work queue. The call chain from the work queue will look like this: hda_power_work <<- power_on = 1, unlock, then power_on cleard by suspend hda_call_codec_suspend hda_set_power_state snd_hda_codec_read codec_exec_verb snd_hda_power_up snd_hda_power_save __snd_hda_power_up cancel_delayed_work_sync <<-- cancelling executing wq Fix this by waiting for the work queue to finish before starting suspend if suspend is not happening on the work queue. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 960800b..c0ab72c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3612,7 +3612,7 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {} * call suspend and power-down; used both from PM and power-save * this function returns the power state in the end */ -static unsigned int hda_call_codec_suspend(struct hda_codec *codec) +static unsigned int hda_call_codec_suspend(struct hda_codec *codec, bool in_wq) { unsigned int state; @@ -3620,7 +3620,9 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); - cancel_delayed_work(&codec->power_work); + /* Cancel delayed work if we aren't currently running from it. */ + if (!in_wq) + cancel_delayed_work_sync(&codec->power_work); spin_lock(&codec->power_lock); snd_hda_update_power_acct(codec); trace_hda_power_down(codec); @@ -4478,7 +4480,7 @@ static void hda_power_work(struct work_struct *work) } spin_unlock(&codec->power_lock); - state = hda_call_codec_suspend(codec); + state = hda_call_codec_suspend(codec, true); codec->pm_down_notified = 0; if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { codec->pm_down_notified = 1; @@ -5127,7 +5129,7 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) - hda_call_codec_suspend(codec); + hda_call_codec_suspend(codec, false); } return 0; } -- 1.7.12.146.g16d26b1

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix detection of ALC271X codec
by Herton Ronaldo Krzesinski 08 Oct '12

08 Oct '12

In commit af741c1 ("ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup"), alc_auto_parse_customize_define was moved after detection of ALC271X. The problem is that detection of ALC271X relies on spec->cdefine.platform_type, and it's set on alc_auto_parse_customize_define. Move the alc_auto_parse_customize_define and its required fixup setup before the block doing the ALC271X and other codec setup. BugLink: https://bugs.launchpad.net/bugs/1006690 Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) It looks safe to move the functions before the block "if (codec->vendor_id == 0x10ec0269) {", as nothing on the fixups seems to depend on what's done there. But if wanted we could just move the detection/codec rename of ALC271X below too. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4c4b01..3b187e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6273,6 +6273,12 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; + alc_pick_fixup(codec, alc269_fixup_models, + alc269_fixup_tbl, alc269_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + + alc_auto_parse_customize_define(codec); + if (codec->vendor_id == 0x10ec0269) { spec->codec_variant = ALC269_TYPE_ALC269VA; switch (alc_get_coef0(codec) & 0x00f0) { @@ -6300,12 +6306,6 @@ static int patch_alc269(struct hda_codec *codec) alc269_fill_coef(codec); } - alc_pick_fixup(codec, alc269_fixup_models, - alc269_fixup_tbl, alc269_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - - alc_auto_parse_customize_define(codec); - /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) -- 1.7.10.4

3 3

[alsa-devel] [PATCH 1/1] ALSA: CA0132 - Ensure SCP packet is little-endian format
by Ian Minett 08 Oct '12

08 Oct '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi Takashi, Great - good to hear the previous DSP loader patches are now living in the hda-ca0132-dsp branch. This patch is based on that branch, and addresses the SCP packet endianness. >> The firmware does need to be reloaded when the chip loses power, so it >> doesn't survive S3. >OK, we need suspend/resume hooks, too. Do we need to do anything specific to handle this? The firmware is already restored upon S3 by the current caching code. Thanks, - Ian --- The DSP firmware expects the SCP data to be in little-endian format regardless of the host architecture. Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2fd3121..4d7cda4 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -28,6 +28,7 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/firmware.h> +#include <asm/unaligned.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -701,9 +702,13 @@ static int dspio_send(struct hda_codec *codec, unsigned int reg, unsigned int res; int retry = 50; + /* Handling for Big Endian machine */ + unsigned int val; + put_unaligned_le32(data, &val); + /* send bits of data specified by reg to dsp */ do { - res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); + res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, val); if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) return res; } while (--retry); -- 1.7.4.1

2 1