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

[alsa-devel] [PATCH] ASoC: rt5645: Rename HP control to Headphone
by Nicolas Boichat 19 May '15

19 May '15

Use the standard name "Headphone" instead of "HP" for better userspace integration. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- sound/soc/codecs/rt5645.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index be4d741..bc925f0 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -464,9 +464,9 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Channel Switch", RT5645_HP_VOL, + SOC_DOUBLE("Headphone Channel Switch", RT5645_HP_VOL, RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), - SOC_DOUBLE_TLV("HP Playback Volume", RT5645_HP_VOL, + SOC_DOUBLE_TLV("Headphone Playback Volume", RT5645_HP_VOL, RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), /* OUTPUT Control */ -- 2.2.0.rc0.207.ga3a616c

4 4

[alsa-devel] [PATCH] ASoC: rt5645: fix kernel hang when call rt5645_set_jack_detect()
by John Lin 19 May '15

19 May '15

rt5645_set_jack_detect() is usually called from snd_soc_dai_link.init() and it calls snd_soc_jack_report() from rt5645_irq_detection() if jack is inserted. snd_soc_jack_report() results in kernel hang if it is called from a context which cannot sleep. This patch makes sure snd_soc_jack_report() is called from workqueue. It can fix the kernel hang issue. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 14b12c5..aaede45 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2876,6 +2876,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } static int rt5645_irq_detection(struct rt5645_priv *rt5645); +static irqreturn_t rt5645_irq(int irq, void *data); int rt5645_set_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, @@ -2895,7 +2896,7 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); } - rt5645_irq_detection(rt5645); + rt5645_irq(0, rt5645); return 0; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix NULL dereference from CA0132 DSP loader
by Takashi Iwai 19 May '15

19 May '15

The CA0132 DSP loader leads to NULL deference since the recent transition to HDA core code, as it unconditionally accesses hdac_stream->substream->runtime. For DSP loading, the substream shouldn't be assigned. This patch addresses the NULL dereference above in addition to assure the substream is cleared while DSP loading. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98151 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_stream.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 52a894fe478f..4c15d0accc9e 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -139,9 +139,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_reset); int snd_hdac_stream_setup(struct hdac_stream *azx_dev) { struct hdac_bus *bus = azx_dev->bus; - struct snd_pcm_runtime *runtime = azx_dev->substream->runtime; + struct snd_pcm_runtime *runtime; unsigned int val; + if (azx_dev->substream) + runtime = azx_dev->substream->runtime; + else + runtime = NULL; /* make sure the run bit is zero for SD */ snd_hdac_stream_clear(azx_dev); /* program the stream_tag */ @@ -189,14 +193,15 @@ int snd_hdac_stream_setup(struct hdac_stream *azx_dev) * we ignore it; currently set the threshold statically to * 64 frames */ - if (runtime->period_size > 64) + if (runtime && runtime->period_size > 64) azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64); else azx_dev->delay_negative_threshold = 0; /* wallclk has 24Mhz clock source */ - azx_dev->period_wallclk = (((runtime->period_size * 24000) / + if (runtime) + azx_dev->period_wallclk = (((runtime->period_size * 24000) / runtime->rate) * 1000); return 0; @@ -611,6 +616,7 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format, if (err < 0) goto err_alloc; + azx_dev->substream = NULL; azx_dev->bufsize = byte_size; azx_dev->period_bytes = byte_size; azx_dev->format_val = format; -- 2.4.1

1 0

[alsa-devel] [PATCH 0/8] ASoC: Intel: refactor common SST IPC handling
by Vinod Koul 19 May '15

19 May '15

This series refactors the common SST IPC code to remove some hardcoded assumption which are no longer valid for future platforms like Skylake, These are made configurable and right configuration applied for respective drivers. The changes are mailbox sizes and dsp busy checks. Subhransu S. Prusty (8): ASoC: Intel: Create an ops to check for DSP busy ASoC: Intel: Move the busy check to ops for Baytrail ASoC: Intel: Move the busy check to ops for HSW ASoC: Intel: Remove the direct register reference from common ipc ASoC: Intel: Allow to configure max size for mailbox data ASoC: Intel: Initialize max mailbox size for baytrail ASoC: Intel: Initialize max mailbox size for haswell ASoC: Intel: Allocate for the mailbox with max size sound/soc/intel/baytrail/sst-baytrail-ipc.c | 11 +++++++++++ sound/soc/intel/common/sst-ipc.c | 20 +++++++++++++++++--- sound/soc/intel/common/sst-ipc.h | 7 +++++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 12 ++++++++++++ 4 files changed, 45 insertions(+), 5 deletions(-) -- 1.9.1

1 9

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for MS LifeCam Studio
by Takashi Iwai 19 May '15

19 May '15

Microsoft LifeCam Studio (045e:0772) needs a similar quirk for suppressing the wrong sample rate inquiry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7c5a70139278..46facfc9aec1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1117,6 +1117,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) switch (chip->usb_id) { case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema */ case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ + case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ return true; } -- 2.4.1

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Add headset-mode* model options for ALC269 & co
by Takashi Iwai 19 May '15

19 May '15

Modern machines tend to have only one headset jack nowadays, and they often need these quirks. Let's allow them applicable via model option for ease of debugging. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio-Models.txt | 2 ++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 4 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 5a3163cac6c3..d92b85d38adc 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -29,6 +29,8 @@ ALC269/270/275/276/28x/29x alc271-dmic Enable ALC271X digital mic workaround inv-dmic Inverted internal mic workaround headset-mic Indicates a combined headset (headphone+mic) jack + headset-mode More comprehensive headset support for ALC269 & co + headset-mode-no-hp-mic Headset mode support without headphone mic lenovo-dock Enables docking station I/O for some Lenovos dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6e1c8c34e841..4d11fbfca5e2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5242,6 +5242,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC269_FIXUP_HEADSET_MIC, .name = "headset-mic"}, + {.id = ALC269_FIXUP_HEADSET_MODE, .name = "headset-mode"}, + {.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, -- 2.4.1

1 1

[alsa-devel] [RFC PATCH] ALSA: pcm: Modify double acknowledged interrupts check condition
by Koro Chen 19 May '15

19 May '15

Currently in snd_pcm_update_hw_ptr0 during interrupt, we consider there were double acknowledged interrupts when: 1. HW reported pointer is smaller than expected, and 2. Time from last update time (hdelta) is over half a buffer time. However, when HW reported pointer is only a few bytes smaller than expected, and when hdelta is just a little larger than half a buffer time (e.g. ping-pong buffer), it wrongly treats this IRQ as double acknowledged. The condition #2 uses jiffies, but jiffies is not high resolution since it is integer. We should consider jiffies inaccuracy. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ac6b33f..7d45645 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -339,7 +339,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (delta > new_hw_ptr) { /* check for double acknowledged interrupts */ hdelta = curr_jiffies - runtime->hw_ptr_jiffies; - if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { + if (hdelta > runtime->hw_ptr_buffer_jiffies/2 + 1) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) { hw_base = 0; -- 1.8.1.1.dirty

2 1

[alsa-devel] Intel HDA / ca0132: support for Alienware 15 Creative Sound Core3D-EX
by Gabriele Martino 19 May '15

19 May '15

Hi, the new Alienware 15 laptop (2015, not the M15x) has a Creative Sound Core3D-EX Audio which seems to work with ca0132. I'm using a 4.0.0 kernel with gentoo patchset, but with the default configuration: - the line out/headphone jack doesn't work, it only outputs some white noise - alsamixer shows only "Master" and "PCM" channel, no "Front", "Line Out" or similar - alsamixer has a toggle switch for the "HP/Speaker" but no headphone/line out switch - the jack detection is enabled on the "line in" jack instead of the "line out" (if I plug something on the "line in", the speakers are muted) - if I play too much with these switches, the speakers stop working until suspend/reboot To fix the first issue I run "hdajacksensetest" to get the headphone jack pin: hdajacksensetest -c 1 -a Pin 0x0b (Internal Speaker): present = Yes Pin 0x0c (Not connected): present = No Pin 0x0d (Not connected): present = No Pin 0x0e (Not connected): present = No Pin 0x0f (Not connected): present = Yes Pin 0x10 (Not connected): present = No Pin 0x11 (Black Line In, Left side): present = No Pin 0x12 (Internal Mic, Mobile-In): present = No Pin 0x13 (Not connected): present = No Pin 0x18 (Not connected): present = No and replaced the pin number: --- /usr/src/linux-4.0.0-gentoo/sound/pci/hda/patch_ca0132.c.old 2015-04-13 00:12:50.000000000 +0200 +++ /usr/src/linux-4.0.0-gentoo/sound/pci/hda/patch_ca0132.c 2015-04-25 01:43:09.607531550 +0200 @@ -4625,7 +4625,7 @@ spec->num_outputs = 2; spec->out_pins[0] = 0x0b; /* speaker out */ - spec->out_pins[1] = 0x10; /* headphone out */ + spec->out_pins[1] = 0x0f; /* default: 0x10 - headphone out -0x0f on Alienware 15 2015 */ spec->shared_out_nid = 0x2; spec->num_inputs = 3; Now the "HP/Speaker Auto Detect" is ignored on both jacks, but I can switch between headphones and speakers manually with the "HP/Speaker" item. This is a dirty hack (and probably won't work on different cards), is there a better way to fix this and get at least the jack detection working? Regards, Gabriele

2 20

[alsa-devel] [PATCH 0/4] firewire-lib: purge restriction of synchronization for non-blocking mode
by Takashi Sakamoto 19 May '15

19 May '15

This patchset allows non-blocking streams to use timestamp synchronization. In full duplex synchronization mode, current implementation pass the value of 'syt' field of CIP header in incoming packets to outgoing packet processing, while the number of data blocks is not passed. Therefore, this mode is allowed just to blocking streams, because of a lack of validator for the number which actual devices transfers. As long as the number is suspicious, it's not reused to outgoing packet processing. To allow the synchronization mode to non-blocking streams, this patchset: * adds a validator for the number of data blocks in incoming packets * adds a flag still to support illegal models (i.e. Behringer FCA202) * pass the evaluated number from incoming packet processing to outgoing * remove the restriction Takashi Sakamoto (4): ALSA: firewire-lib: add buffer-over-run protection at receiving more data blocks than expected ALSA: firewire-lib: simplify function to calculate the number of data blocks ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets ALSA: firewire-lib: remove restriction for non-blocking mode sound/firewire/amdtp.c | 108 ++++++++++++++++++++++++-------------- sound/firewire/amdtp.h | 4 ++ sound/firewire/oxfw/oxfw-stream.c | 10 +++- 3 files changed, 81 insertions(+), 41 deletions(-) -- 2.1.4

2 8

[alsa-devel] [PATCH RESEND] sound: ppc: keywest: drop using attach adapter
by Wolfram Sang 18 May '15

18 May '15

As we now have deferred probing, we can use a custom mechanism and finally get rid of this legacy interface from the i2c core. Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> Tested-by: Dan DeVoto <dand1972(a)yahoo.com> Tested-by: Mark Elliott <txlitebeer(a)gmail.com> --- Changes since last time: Added two test tags from the Debian PPC list. I'd love to have that in 4.2, so I can get rid of that legacy interface. I can take it via I2C but ALSA looks like the proper path to me. sound/ppc/keywest.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 0d1c27e911b8dd..d7627bae08362e 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -52,7 +52,7 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) return -EINVAL; if (strncmp(adapter->name, "mac-io", 6)) - return 0; /* ignored */ + return -EINVAL; /* ignored */ memset(&info, 0, sizeof(struct i2c_board_info)); strlcpy(info.type, "keywest", I2C_NAME_SIZE); @@ -100,7 +100,6 @@ static struct i2c_driver keywest_driver = { .driver = { .name = "PMac Keywest Audio", }, - .attach_adapter = keywest_attach_adapter, .probe = keywest_probe, .remove = keywest_remove, .id_table = keywest_i2c_id, @@ -132,16 +131,32 @@ int snd_pmac_tumbler_post_init(void) /* exported */ int snd_pmac_keywest_init(struct pmac_keywest *i2c) { - int err; + struct i2c_adapter *adap; + int err, i = 0; if (keywest_ctx) return -EBUSY; + adap = i2c_get_adapter(0); + if (!adap) + return -EPROBE_DEFER; + keywest_ctx = i2c; if ((err = i2c_add_driver(&keywest_driver))) { snd_printk(KERN_ERR "cannot register keywest i2c driver\n"); + i2c_put_adapter(adap); return err; } - return 0; + + /* We assume Macs have consecutive I2C bus numbers starting at 0 */ + while (adap) { + err = keywest_attach_adapter(adap); + if (!err) + return 0; + i2c_put_adapter(adap); + adap = i2c_get_adapter(++i); + } + + return -ENODEV; } -- 2.1.4

2 1