September 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH RESEND 1/4] sound/onyx: remove redundant case entry
by Yaowei Bai 23 Sep '15

23 Sep '15

Case ti->tag == 0 can be included in the last return, remove it. No functional change. Signed-off-by: Yaowei Bai <bywxiaobai(a)163.com> --- sound/aoa/codecs/onyx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index a04edff..a407e1f 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -683,7 +683,6 @@ static int onyx_usable(struct codec_info_item *cii, mutex_unlock(&onyx->mutex); switch (ti->tag) { - case 0: return 1; case 1: return analog_enabled; case 2: return spdif_enabled; } -- 1.9.1

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[alsa-devel] [PATCH 1/4] sound/onyx: remove redundant case entry
by Yaowei Bai 23 Sep '15

23 Sep '15

Case ti->tag == 0 can be included in the last return, remove it. No functional change. Signed-off-by: Yaowei Bai <bywxiaobai(a)163.com> --- sound/aoa/codecs/onyx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index a04edff..a407e1f 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -683,7 +683,6 @@ static int onyx_usable(struct codec_info_item *cii, mutex_unlock(&onyx->mutex); switch (ti->tag) { - case 0: return 1; case 1: return analog_enabled; case 2: return spdif_enabled; } -- 1.9.1

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[alsa-devel] [PATCH - alsa-utils 1/1] arecord: Remove only regular files
by Alexander Volkov 23 Sep '15

23 Sep '15

arecord removes a file before writing into it. It's not appropriate in some cases. For example, if you a pass a symlink to a file, then the symlink will be removed while the user expects to record into the symlink's target. Another case is recording into the device file. Some modems provide a tty device file as a voice device. And it's not possible to write into it under root with arecord, because it removes the device file. So check the type of a file before writing into it and remove only regular files. Signed-off-by: Alexander Volkov <a.volkov(a)rusbitech.ru> diff --git a/aplay/aplay.c b/aplay/aplay.c index 459f7dd..1b2cdfc 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2929,6 +2929,7 @@ static void capture(char *orig_name) char *name = orig_name; /* current filename */ char namebuf[PATH_MAX+1]; off64_t count, rest; /* number of bytes to capture */ + struct stat statbuf; /* get number of bytes to capture */ count = calc_count(); @@ -2973,7 +2974,10 @@ static void capture(char *orig_name) } /* open a new file */ - remove(name); + if (!lstat(name, &statbuf)) { + if (S_ISREG(statbuf.st_mode)) + remove(name); + } fd = safe_open(name); if (fd < 0) { perror(name); -- 2.4.0

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[alsa-devel] [PATCH alsa-lib] build: Do not try to detect cross-compiler
by Tanu Kaskinen 23 Sep '15

23 Sep '15

From: Khem Raj <raj.khem(a)gmail.com> cross compilers are passed via path may not be a gcc based cross compiler in such cases this check fails and try's to force gcc based cross compiler detection, This code is a convenience that limits the build system Signed-off-by: Khem Raj <raj.khem(a)gmail.com> --- configure.ac | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/configure.ac b/configure.ac index a482b3e..a14e52d 100644 --- a/configure.ac +++ b/configure.ac @@ -27,20 +27,6 @@ AC_PREFIX_DEFAULT(/usr) dnl Checks for programs. -dnl try to gues cross-compiler if not set -if test "x$host" != "x$build" -a -z "`echo $CC | grep -e '-gcc'`"; -then - AC_MSG_CHECKING(for cross-compiler) - - 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 - which ${host_cpu}-${host_vendor}-${host_os}-gcc >/dev/null 2>&1 \ - && CC=${host_cpu}-${host_vendor}-${host_os}-gcc - - AC_MSG_RESULT($CC) -fi - CFLAGS="$CFLAGS -D_GNU_SOURCE" -- 2.5.1

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[alsa-devel] [PATCH] sound: oss: ad1848: Fix returned errno code in ad1848_init()
by Luis de Bethencourt 23 Sep '15

23 Sep '15

The driver is using -1 instead of the -ENOMEM defined macro to specify that a buffer allocation failed. Since the error number is propagated, the caller will get a -EPERM which is the wrong error condition. Smatch tool warning: ad1848_init() warn: returning -1 instead of -ENOMEM is sloppy Signed-off-by: Luis de Bethencourt <luisbg(a)osg.samsung.com> --- sound/oss/ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 10c8de1..6b35656 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -1992,7 +1992,7 @@ int ad1848_init (char *name, struct resource *ports, int irq, int dma_playback, portc = kmalloc(sizeof(ad1848_port_info), GFP_KERNEL); if(portc==NULL) { release_region(devc->base, 4); - return -1; + return -ENOMEM; } if ((my_dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, -- 2.5.3

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[alsa-devel] [PATCH v2- ALSA 3/3] HDA - add headset-sonicmaster and headset-multi-sonicmaster models
by Raymond Yau 23 Sep '15

23 Sep '15

add headset only model for ALC668 with Sonic Master Subwoofer add 625ms delay after headset is plugged for correct detection of headset type for ALC668 add headset multi model with Sonice Master Subwoofer update HD-Audio-Models.txt Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ec099d4..8baddac 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -55,7 +55,12 @@ ALC66x/67x/892 asus-mode7 ASUS asus-mode8 ASUS inv-dmic Inverted internal mic workaround + +ALC668 +====== dell-headset-multi Headset jack, which can also be used as mic-in + headset-sonicmaster Headset jack and SonicMaster Subwoofer + headset-multi-sonicmaster Headset, HP or Mic and SonicMaster Subwoofer ALC680 ====== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a75b561..91de00c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4029,7 +4029,13 @@ static void alc_update_headset_mode(struct hda_codec *codec) PIN_VREFHIZ); } spec->current_headset_mode = new_headset_mode; - + if (codec->core.vendor_id == 0x10ec0668) { + if (spec->gen.hs_mic_use_hp_sense && spec->gen.auto_mic) { + if (new_headset_mode == ALC_HEADSET_MODE_HEADPHONE) + msleep(625); + snd_hda_gen_mic_autoswitch(codec, NULL); + }; + } snd_hda_gen_update_outputs(codec); } @@ -6071,6 +6077,42 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec, } } +static void alc668_fixup_asus_headset_only(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl hs_mic_pin[] = { + { 0x1b, 0x03a1113c }, /* headset mic */ + { } + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, hs_mic_pin); + spec->gen.hs_mic_use_hp_sense = 1; + alc_fixup_disable_aamix(codec, fix, action); + break; + } +} + +static void alc668_fixup_sonicmaster(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl bass_pin[] = { + { 0x1a, 0x04110011 }, /* external subwoofer */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, bass_pin); + break; + case HDA_FIXUP_ACT_BUILD: + alc_fixup_bass_chmap(codec, fix, action); + break; + } +} + static struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -6130,6 +6172,9 @@ enum { ALC668_FIXUP_AUTO_MUTE, ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, + ALC668_FIXUP_HEADSET_ONLY, + ALC668_FIXUP_HEADSET_SONICMASTER, + ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, }; static const struct hda_fixup alc662_fixups[] = { @@ -6370,6 +6415,24 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, + [ALC668_FIXUP_HEADSET_ONLY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_asus_headset_only, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MODE, + }, + [ALC668_FIXUP_HEADSET_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_ONLY, + }, + [ALC668_FIXUP_HEADSET_MULTI_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_DISABLE_AAMIX, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6476,6 +6539,9 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, + {.id = ALC668_FIXUP_HEADSET_SONICMASTER, .name = "headset-sonicmaster"}, + {.id = ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, + .name = "headset-multi-sonicmaster"}, {} }; -- 2.5.0

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[alsa-devel] [PATCH v2- ALSA 2/3] HDA - create headset mic jack which use headphone pin sense
by Raymond Yau 23 Sep '15

23 Sep '15

- use __snd_hda_jack_add_kctl to create headset mic jack which use headphone pin sense - set headset mic jack as slave of headphone jack - allow non detectable headset mic pin and hs_mic_use_hp_sense pass check auto mic availability Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 24f9111..e8e4e34 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4662,7 +4662,9 @@ static int check_auto_mic_availability(struct hda_codec *codec) if (!spec->line_in_auto_switch && cfg->inputs[i].type != AUTO_PIN_MIC) return 0; /* only mic is allowed */ - if (!is_jack_detectable(codec, nid)) + if (!is_jack_detectable(codec, nid) && + !(spec->hs_mic_use_hp_sense && + cfg->inputs[i].is_headset_mic)) return 0; /* no unsol support */ break; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 56e4139..96f8214 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -236,6 +236,7 @@ struct hda_gen_spec { unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + unsigned int hs_mic_use_hp_sense:1; /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d81d65f..c29f05e 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -480,6 +480,10 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; + struct hda_jack_tbl *jack; + struct hda_gen_spec *spec = codec->spec; + hda_nid_t hp_pin = auto_cfg_hp_pins(cfg)[0]; + hda_nid_t hs_mic_pin = 0; int i, err; for (i = 0; i < cfg->num_inputs; i++) { @@ -493,6 +497,13 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, "Headphone Mic"); } else + if (cfg->inputs[i].is_headset_mic && + !is_jack_detectable(codec, cfg->inputs[i].pin) && + spec->hs_mic_use_hp_sense) { + hs_mic_pin = cfg->inputs[i].pin; + err = __snd_hda_jack_add_kctl(codec, cfg->inputs[i].pin, + "Headset Mic", false, hp_pin); + } else err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, NULL); if (err < 0) @@ -510,6 +521,11 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, *p, cfg, NULL); if (err < 0) return err; + if (*p == hp_pin && spec->hs_mic_use_hp_sense && hs_mic_pin) { + jack = snd_hda_jack_tbl_get(codec, *p); + if (jack) + jack->slave_nid = hs_mic_pin; + } } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ -- 2.5.0

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[alsa-devel] [PATCH v2] ASoC: pxa: pxa2xx-ac97: fix dma requestor lines
by Robert Jarzmik 22 Sep '15

22 Sep '15

PCM receive and transmit DMA requestor lines were reverted, breaking the PCM playback interface for PXA platforms using the sound/soc/ variant instead of the sound/arm variant. The commit below shows the inversion in the requestor lines. Fixes: d65a14587a9b ("ASoC: pxa: use snd_dmaengine_dai_dma_data") Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: rebase on Linus's master, not my dmaengine tip branch. --- sound/soc/pxa/pxa2xx-ac97.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1f6054650991..9e4b04e0fbd1 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -49,7 +49,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12; +static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -57,7 +57,7 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, }; -static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11; +static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, -- 2.1.4

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[alsa-devel] [PATCH v3 00/21] Update ALSA, and au0828 drivers to use Managed Media Controller API
by Shuah Khan 22 Sep '15

22 Sep '15

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API to share tuner. Please note that Managed Media Controller API and DVB and V4L2 drivers updates to use Media Controller API have been added in a prior patch series. Media Controller API is enhanced with two new interfaces to register and unregister entity_notify hooks to allow drivers to take appropriate actions when as new entities get added to the shared media device. Tested exclusion between digital, analog, and audio to ensure when tuner has an active link to DVB FE, analog, and audio will detect and honor the tuner busy conditions and vice versa. Changes since v2: http://www.spinics.net/lists/linux-media/msg91926.html 1. An important change in this patch series is made to ALSA to check for resources and hold in snd_usb_hw_params(), and release from snd_usb_hw_free(). This change fixed the lockdep warnings seen when resources were held in TRIGGER_START and released from TRIGGER_STOP which could run in IRQ context. I acknowledge Clemens Ladisch for suggesting the correct places to hold/free resources to avoid IRQ path complications. 2. With the above change, the patch series is simpler without the need to change the graph_mutex into a spinlock. 3. I split the patches up differently for easy reviews - no code bloat from v2. 4. A second important change is now the Bridge driver (au0828) owns and drives the graph creation as well as enabling and disabling tuner. It also keeps state information to avoid graph walks in enable_source and disable_source handler. I acknowledge Hans Verkuil for his suggestions and ideas for this change. History: This patch series has been updated to address comments from 3 previous versions of this series. Links to v3 version for reference are: https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89491.html https://www.mail-archive.com/linux-media@vger.kernel.org/msg89492.html https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89493.html Shuah Khan (21): Revert "[media] media: media controller entity framework enhancements for ALSA" media: Media Controller register/unregister entity_notify API media: Add ALSA Media Controller devnodes media: Media Controller enable/disable source handler API media: Media Controller fix to not let stream_count go negative media: Media Controller export non locking __media_entity_setup_link() media: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add v4l_enable/disable_media_tuner() helper functions media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: au0828 Use au8522_media_pads enum for pad defines media: au0828 fix au0828_create_media_graph() entity checks media: Change v4l-core to check for tuner availability media: dvb-frontend invoke enable/disable_source handlers media: au0828 video remove au0828_enable_analog_tuner() media: au0828 video change to use v4l_enable_media_tuner() media: au0828 change to use Managed Media Controller API media: au0828 change to register/unregister entity_notify hook media: au0828 implement enable_source and disable_source handlers media: media: dvb-frontend fix enable_source error legs sound/usb: Update ALSA driver to use Managed Media Controller API drivers/media/dvb-core/dvb_frontend.c | 142 ++----------- drivers/media/dvb-core/dvb_frontend.h | 3 + drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 46 +++- drivers/media/media-entity.c | 64 ++++-- drivers/media/usb/au0828/au0828-core.c | 305 ++++++++++++++++++++++----- drivers/media/usb/au0828/au0828-video.c | 76 ++----- drivers/media/usb/au0828/au0828.h | 8 + drivers/media/v4l2-core/v4l2-dev.c | 27 +++ drivers/media/v4l2-core/v4l2-fh.c | 1 + drivers/media/v4l2-core/v4l2-ioctl.c | 29 +++ drivers/media/v4l2-core/videobuf2-core.c | 3 + include/media/media-device.h | 41 ++++ include/media/media-entity.h | 7 +- include/media/v4l2-dev.h | 4 + include/uapi/linux/media.h | 5 + sound/usb/Makefile | 15 +- sound/usb/card.c | 5 + sound/usb/card.h | 1 + sound/usb/media.c | 178 ++++++++++++++++ sound/usb/media.h | 51 +++++ sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 +- sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 1 + 28 files changed, 790 insertions(+), 280 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

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[alsa-devel] [PATCH v3 1/2] ASoC: rt5645: Add dmi for Broadwell
by John Lin 22 Sep '15

22 Sep '15

Add DMI data for Buddy project. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index dbc1d76..e830688 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3209,6 +3209,31 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3241,7 +3266,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwell)) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); -- 1.8.1.1.439.g50a6b54

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