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

[alsa-devel] [PATCH] ASoC: cs42l52: Fix off-by-one boundary checking for cs42l52_get_clk()
by Axel Lin 16 Dec '12

16 Dec '12

The caller uses the return value of cs42l52_get_clk as array index, the valid array index is 0 .. ARRAY_SIZE(clk_map_table) - 1. Thus we need to ensure ret is less than ARRAY_SIZE(clk_map_table). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..66817f6 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -749,7 +749,7 @@ static int cs42l52_get_clk(int mclk, int rate) } } } - if (ret > ARRAY_SIZE(clk_map_table)) + if (ret >= ARRAY_SIZE(clk_map_table)) return -EINVAL; return ret; } -- 1.7.9.5

1 1

[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 15 Dec '12

15 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,7 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation 1.7.8.3

2 1

[alsa-devel] [PATCH] ASoC: Prevent pop_wait overwrite
by Misael Lopez Cruz 15 Dec '12

15 Dec '12

pop_wait is used to determine if a deferred playback close needs to be cancelled when the a PCM is open or if after the power-down delay expires it needs to run. pop_wait is associated with the CODEC DAI, so the CODEC DAI must be unique. This holds true for most CODECs, except for the dummy CODEC and its DAI. In DAI links with non-unique dummy CODECs (e.g. front-ends), pop_wait can be overwritten by another DAI link using also a dummy CODEC. Failure to cancel a deferred close can cause mute due to the DAPM STOP event sent in the deferred work. One scenario where pop_wait is overwritten and causing mute is below (where hw:0,0 and hw:0,1 are two front-ends with default pmdown_time = 5 secs): aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1 sleep 1 aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 & aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE Since CODECs may not be unique, pop_wait is moved to the PCM runtime structure. Creating separate dummy CODECs for each DAI link can also solve the problem, but at this point it's only pop_wait variable in the CODEC DAI that has negative effects by not being unique. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> --- include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/soc-compress.c | 2 +- sound/soc/soc-pcm.c | 12 ++++++------ 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 628db7b..3953cea 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -242,7 +242,6 @@ struct snd_soc_dai { unsigned int symmetric_rates:1; struct snd_pcm_runtime *runtime; unsigned int active; - unsigned char pop_wait:1; unsigned char probed:1; struct snd_soc_dapm_widget *playback_widget; diff --git a/include/sound/soc.h b/include/sound/soc.h index 91244a0..769e27c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1039,6 +1039,7 @@ struct snd_soc_pcm_runtime { struct snd_soc_dpcm_runtime dpcm[2]; long pmdown_time; + unsigned char pop_wait:1; /* runtime devices */ struct snd_pcm *pcm; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 967d0e1..5fbfb06 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -113,7 +113,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } else - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } else { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5c3ca2a..d7711fc 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -334,11 +334,11 @@ static void close_delayed_work(struct work_struct *work) dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); + rtd->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; + if (rtd->pop_wait == 1) { + rtd->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } @@ -408,7 +408,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } @@ -480,8 +480,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) /* cancel any delayed stream shutdown that is pending */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; + rtd->pop_wait) { + rtd->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] configure: do not detect incorrect cross-compiler
by Olivier Blin 15 Dec '12

15 Dec '12

On Ubuntu 11.04, configuring with --build=x86_64-unknown-linux-gnu --host=x86_64-linux-gnu finds a wrong cross-compiler: checking for cross-compiler... x86_64-x86_64-pc-linux-gnu-gcc This happens because of a dash vs underscore inconsistency in configure.in: host=x86_64-pc-linux-gnu host_cpu=x86_64 host_os=linux-gnu which ${host_cpu}-${host_os}-gcc >/dev/null 2>&1 && echo ${host_cpu}-${host-os}-gcc This bug has been introduced in the initial --with-host support from 2002, commit eb267ade29c9a49c07b1c33dc9bf7a6790217400 This configure command is about "cross-compiling for i586", where the system compiler is used, which just -m32 additional options. The --build value comes from config.guess. --- configure.in | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.in b/configure.in index e397dad..4af917b 100644 --- a/configure.in +++ b/configure.in @@ -31,7 +31,7 @@ then which ${program_prefix}gcc >/dev/null 2>&1 && CC=${program_prefix}gcc which ${host_cpu}-${host_os}-gcc >/dev/null 2>&1 \ - && CC=${host_cpu}-${host-os}-gcc + && CC=${host_cpu}-${host_os}-gcc which ${host_cpu}-${host_vendor}-${host_os}-gcc >/dev/null 2>&1 \ && CC=${host_cpu}-${host_vendor}-${host_os}-gcc -- 1.8.0.2

2 1

[alsa-devel] [PATCH] usb-audio: ignore-quirk for HP Wireless Audio
by Eldad Zack 15 Dec '12

15 Dec '12

As Joe Cooper <swelljoe(a)gmail.com> reported, "On most HP Envy laptops the snd-usb-audio module causes the system to become unresponsive and Gnome Shell 3 to crash.". See also: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/057729.h… Add a quirk to ignore this device (for now) to solve the instability issue and allow other USB audio devices to be used. Reported-by: Joe Cooper <swelljoe(a)gmail.com> Tested-by: Isaac Smith <hunternet93(a)gmail.com> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/quirks-table.h | 36 ++++++++++++++++++++++++++++++++++++ 1 files changed, 36 insertions(+), 0 deletions(-) Applies against mainline 3.7.0 and sound-3.8. diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 49f9af9..e3c59f0 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -99,6 +99,42 @@ }, /* + * HP Wireless Audio + * When not ignored, causes instability issues for some users, forcing them to + * blacklist the entire module. + */ +{ + USB_DEVICE(0x0424, 0xb832), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Standard Microsystems Corp.", + .product_name = "HP Wireless Audio", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + /* Mixer */ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* HID Device, .ifnum = 3 */ + { + .ifnum = -1, + } + } + } +}, + +/* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface * class matches do not take effect without an explicit ID match. */ -- 1.7.8.6

2 1

[alsa-devel] [PATCH v2 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pin
by mengdong.lin＠intel.com 14 Dec '12

14 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI. So this patch - allow generic hdmi patch to pick up and apply fixups for different codecs. - add fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override the connection list entries with proper value. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..42fdf14 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,7 @@ #include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" static bool static_hdmi_pcm; @@ -79,6 +80,7 @@ struct hdmi_spec_per_pin { }; struct hdmi_spec { + struct hda_gen_spec gen; int num_cvts; struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; @@ -1687,6 +1689,54 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + unsigned int vendor_param; + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + /* enable DP1.2 mode */ + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + snd_hda_override_conn_list(codec, 0x06, 3, list); + hdmi_read_pin_conn(codec, 1); +} + + +/* available models for fixup */ +enum { + INTEL_HASWELL, +}; + +static const struct hda_model_fixup hdmi_models[] = { + {.id = INTEL_HASWELL, .name = "Haswell"}, + {} +}; + +static const struct snd_pci_quirk hdmi_fixup_tbl[] = { + SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), + {} /* terminator */ +}; + +static const struct hda_fixup hdmi_fixups[] = { + [INTEL_HASWELL] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_connect_list, + }, +}; + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,6 +1746,11 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + snd_hda_gen_init(&spec->gen); + + snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); -- 1.7.9.5

2 2

[alsa-devel] [PATCH 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pin
by mengdong.lin＠intel.com 14 Dec '12

14 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI. So this patch - allow generic hdmi patch to pick up and apply fixups for different codecs. - add fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override invalid connection list entries reported by hardware. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..2bf1758 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,7 @@ #include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" static bool static_hdmi_pcm; @@ -79,6 +80,7 @@ struct hdmi_spec_per_pin { }; struct hdmi_spec { + struct hda_gen_spec gen; int num_cvts; struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; @@ -1687,6 +1689,69 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_enable_dp12_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + unsigned int vendor_param; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); +} + +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + unsigned int vendor_param; + + if (action != HDA_FIXUP_ACT_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + snd_hda_override_conn_list(codec, 0x06, 3, list); + hdmi_read_pin_conn(codec, 1); +} + + +/* available models for fixup */ +enum { + INTEL_HASWELL, + INTEL_HASWELL_CONNECT_LIST, +}; + +static const struct hda_model_fixup hdmi_models[] = { + {.id = INTEL_HASWELL, .name = "Haswell"}, + {} +}; + +static const struct snd_pci_quirk hdmi_fixup_tbl[] = { + SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), + {} /* terminator */ +}; + +static const struct hda_fixup hdmi_fixups[] = { + [INTEL_HASWELL] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_enable_dp12_mode, + .chained = true, + .chain_id = INTEL_HASWELL_CONNECT_LIST, + }, + [INTEL_HASWELL_CONNECT_LIST] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_connect_list, + }, +}; + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,11 +1761,19 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + snd_hda_gen_init(&spec->gen); + + snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); return -EINVAL; } + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + codec->patch_ops = generic_hdmi_patch_ops; generic_hdmi_init_per_pins(codec); -- 1.7.9.5

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Re: [alsa-devel] ALSA:aloop -- inconsistent {in-softirq-W} -> {softirq-on-W} usage
by Takashi Iwai 14 Dec '12

14 Dec '12

At Thu, 13 Dec 2012 21:36:05 +0100, Andreas Fenkart wrote: > > > Hi, > > Seen this error when stopping playback to loopback. > > # aplay -Dhw:0 -fcd /dev/urandom > Playing raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo > ^CAborted by signa[ 65.562992] > [ 65.565493] ================================= > [ 65.570044] [ INFO: inconsistent lock state ] > [ 65.574601] 3.7.0-rc8+ #303 Not tainted > [ 65.578605] --------------------------------- > [ 65.583156] inconsistent {IN-SOFTIRQ-W} -> {SOFTIRQ-ON-W} usage. > [ 65.589436] aplay/156 [HC0[0]:SC0[0]:HE1:SE1] takes: > [ 65.594620] (&(&dpcm->timer_lock)->rlock){+.?...}, at: [<c02e90cd>] loopback_close+0x29/0x98 > [ 65.603604] {IN-SOFTIRQ-W} state was registered at: > [ 65.608699] [<c005d8ef>] mark_irqflags+0xa3/0x11c > [ 65.613820] [<c005dca1>] __lock_acquire+0x339/0x77c > [ 65.619112] [<c005e575>] lock_acquire+0x65/0xac > [ 65.624038] [<c03e0c17>] _raw_spin_lock+0x1f/0x2c > [ 65.629155] [<c02e8cdd>] loopback_timer_start+0x3d/0x108 > [ 65.634899] [<c02e9455>] loopback_timer_function+0x21/0x3c > [ 65.640821] [<c0035f3f>] call_timer_fn+0x63/0xfc > [ 65.645849] [<c0036127>] run_timer_softirq+0x14f/0x190 > [ 65.651410] [<c00311d1>] __do_softirq+0xb1/0x174 > [ 65.656426] [<c0031529>] irq_exit+0x6d/0x70 > [ 65.660989] [<c000dde9>] handle_IRQ+0x2d/0x60 > [ 65.665743] [<c000848d>] omap3_intc_handle_irq+0x59/0x60 > [ 65.671484] [<c000cfff>] __irq_svc+0x3f/0x64 > [ 65.676138] [<c000de86>] default_idle+0x22/0x24 > [ 65.681063] [<c000df89>] cpu_idle+0x89/0x94 > [ 65.685626] [<c05cf5bb>] start_kernel+0x25b/0x264 > [ 65.690743] irq event stamp: 74155 > [ 65.694296] hardirqs last enabled at (74155): [<c03df9b5>] __mutex_unlock_slowpath+0x85/0xec > [ 65.703215] hardirqs last disabled at (74154): [<c03df96b>] __mutex_unlock_slowpath+0x3b/0xec > [ 65.712132] softirqs last enabled at (72724): [<c0031215>] __do_softirq+0xf5/0x174 > [ 65.720141] softirqs last disabled at (72717): [<c0031529>] irq_exit+0x6d/0x70 > [ 65.727697] > [ 65.727697] other info that might help us debug this: > [ 65.734516] Possible unsafe locking scenario: > [ 65.734516] > [ 65.740697] CPU0 > [ 65.743250] ---- > [ 65.745802] lock(&(&dpcm->timer_lock)->rlock); > [ 65.750639] <Interrupt> > [ 65.753373] lock(&(&dpcm->timer_lock)->rlock); > [ 65.758392] > [ 65.758392] *** DEADLOCK *** > [ 65.758392] > [ 65.764581] 1 lock held by aplay/156: > [ 65.768404] #0: (&pcm->open_mutex){+.+.+.}, at: [<c02e0c49>] snd_pcm_release+0x1d/0x7c > [ 65.776907] > [ 65.776907] stack backtrace: > [ 65.781487] [<c0011a15>] (unwind_backtrace+0x1/0xa0) from [<c005d2c9>] (print_usage_bug+0x149/0x184) > [ 65.791039] [<c005d2c9>] (print_usage_bug+0x149/0x184) from [<c005d4c5>] (mark_lock_irq+0x1c1/0x1e0) > [ 65.800590] [<c005d4c5>] (mark_lock_irq+0x1c1/0x1e0) from [<c005d5f7>] (mark_lock+0x113/0x368) > [ 65.809597] [<c005d5f7>] (mark_lock+0x113/0x368) from [<c005d915>] (mark_irqflags+0xc9/0x11c) > [ 65.818514] [<c005d915>] (mark_irqflags+0xc9/0x11c) from [<c005dca1>] (__lock_acquire+0x339/0x77c) > [ 65.827884] [<c005dca1>] (__lock_acquire+0x339/0x77c) from [<c005e575>] (lock_acquire+0x65/0xac) > [ 65.846083] [<c03e0c17>] (_raw_spin_lock+0x1f/0x2c) from [<c02e90cd>] (loopback_close+0x29/0x98) > [ 65.855273] [<c02e90cd>] (loopback_close+0x29/0x98) from [<c02e0bf1>] (snd_pcm_release_substream+0x39/0x74) > [ 65.865458] [<c02e0bf1>] (snd_pcm_release_substream+0x39/0x74) from [<c02e0c4f>] (snd_pcm_release+0x23/0x7c) > [ 65.875748] [<c02e0c4f>] (snd_pcm_release+0x23/0x7c) from [<c00b4dad>] (__fput+0x61/0x170) > [ 65.884409] [<c00b4dad>] (__fput+0x61/0x170) from [<c0040485>] (task_work_run+0x5d/0x8c) > [ 65.892879] [<c0040485>] (task_work_run+0x5d/0x8c) from [<c001017f>] (do_work_pending+0x67/0x7c) > [ 65.902069] [<c001017f>] (do_work_pending+0x67/0x7c) from [<c000d569>] (work_pending+0x9/0x18) > > This seems to be a real deadlock, since loopback_timer_stop is > triggered from ioctl, while loopback_timer_start can be called > from softirq. Could you check the latest Linus tree? The spinlocks of aloop have been already fixed there. Takashi

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[alsa-devel] Polling issues with the PCM multi plugin
by Devin Anderson 14 Dec '12

14 Dec '12

Hi all, On the jack-devel list, we've been having a discussion about the ALSA PCM multi plugin: http://thread.gmane.org/gmane.comp.audio.jackit/26571/focus=26571 I've been having some issues with setting up my 2 Echo Layla 3Gs as a PCM multi. In particular, I've found that there are times when polling the file descriptors of the multi when poll() returns, but there isn't actually a period of data available for capture/playback, even though the poll descriptors indicate that the data should be available. After some debugging and learning about the PCM multi driver: 1.) The PCM multi driver only returns poll descriptors for one of the devices that makes up the multi 2.) When this happens, the device that's actually being polled *does* have data available 3.) When this happens, the device that's not being polled does not yet have a period's worth of data available After making sure that I had word clock enabled and that my word clock cable was working properly, I started delving into the problem some more. From what I can gather, the PCM multi device uses the function `snd_pcm_link()` to sync its devices, and, for start, stop, drain, pause, suspend, resume, reset, and prepare operations, this appears to work fine; however, I can't find any evidence that a PCM device checks to make sure that all of its linked slave devices have their poll descriptors ready with events before indicating that events are available. Questions: 1.) Should `snd_pcm_link` make I/O availability indication dependent on I/O availability of its linked slave devices? 2.) If not, how should this operation be handled? I'm sure it *can* be handled in userspace by bypassing the multi and polling the file descriptors of all sound cards directly, but duplicating this code for each application that wants to poll multiple sound cards seems wrong. Thanks, -- Devin Anderson surfacepatterns (at) gmail (dot) com blog - http://surfacepatterns.blogspot.com/ psinsights - http://psinsights.googlecode.com/ synthclone - http://synthclone.googlecode.com/

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[alsa-devel] [PATCH] ASoC: wm5110: Split input PGA controls
by Mark Brown 14 Dec '12

14 Dec '12

Though the controls are named as stereo controls in the part the main use case for the analogue inputs to the WM5102 is mono. Reflect this in the controls exposed to userspace, providing a series of mono controls rather than stereo ones. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 77 +++++++++++++++++++++++++++------------------ 1 file changed, 46 insertions(+), 31 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 794f050..55a1348 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -55,37 +55,52 @@ SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT, 1, 0), -SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1R_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2R_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3R_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), - -SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1), - -SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN4L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_IN4L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN4R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4R, + ARIZONA_IN4R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN4L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_IN4L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN4R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4R, + ARIZONA_IN4R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), -- 1.7.10.4

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