February 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] Add a quirk for Plantronics DA45
by denk 01 Mar '16

01 Mar '16

Plantronics DA45 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x4" and "cannot get freq at ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and avoids those error messages. --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2585c17..5b03296 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1123,6 +1123,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ --- Signed-off-by: Dennis Kadioglu <denk(a)post.com>

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[alsa-devel] [PATCH 1/1] alsabat: truncate sample frames for faster FFT analysis
by han.lu＠intel.com 01 Mar '16

01 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Truncate the sample frames to powers of 2, since the FFTW algorithm runs especially fast in this case, and other sizes may be computed by means of a slow, general-purpose algorithm. In my test environment applying the patch, a sound clip of 33072 frames is cut off to 32768 frames before analysis, and the time cost is reduced from 6.128s to 0.224s. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/analyze.c b/bat/analyze.c index 60e2d1c..5cfdac3 100644 --- a/bat/analyze.c +++ b/bat/analyze.c @@ -256,6 +256,20 @@ static int reorder_data(struct bat *bat) return 0; } +/* truncate sample frames for faster FFT analysis process */ +static int truncate_frames(struct bat *bat) +{ + int shift = SHIFT_MAX; + + for (; shift > SHIFT_MIN; shift--) + if (bat->frames & (1 << shift)) { + bat->frames = 1 << shift; + return 0; + } + + return -EINVAL; +} + int analyze_capture(struct bat *bat) { int err = 0; @@ -263,6 +277,13 @@ int analyze_capture(struct bat *bat) int c; struct analyze a; + err = truncate_frames(bat); + if (err < 0) { + fprintf(bat->err, _("Invalid frame number for analysis: %d\n"), + bat->frames); + return err; + } + fprintf(bat->log, _("\nBAT analysis: signal has %d frames at %d Hz,"), bat->frames, bat->rate); fprintf(bat->log, _(" %d channels, %d bytes per sample.\n"), diff --git a/bat/common.h b/bat/common.h index b8f6670..ff03fc1 100644 --- a/bat/common.h +++ b/bat/common.h @@ -68,6 +68,12 @@ #define FOUND_DC (1<<1) #define FOUND_WRONG_PEAK (1<<0) +/* Truncate sample frames to (1 << N), for faster FFT analysis process. The + * valid range of N is (SHIFT_MIN, SHIFT_MAX). When N increases, the analysis + * will be more time-consuming, and the result will be more accurate. */ +#define SHIFT_MAX (sizeof(int) * 8 - 2) +#define SHIFT_MIN 8 + struct wav_header { unsigned int magic; /* 'RIFF' */ unsigned int length; /* file len */ -- 2.5.0

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[alsa-devel] [PATCH] teac ud-501/ud-503 usb delay
by Guillaume Fougnies 01 Mar '16

01 Mar '16

Hello, This is a small patch against sound/usb/quirks.c to correct "Teac Corp." products delay. Tested on Teac UD-501 and UD-503 products, could not harm other Teac products. It corrects the "clock source 41 is not valid, cannot use" error for both devices. And more critically on the new UD-503, a complete freeze of the teac usb interface when switching between different rate/format. (Freeze only solved by a power switch of the device...) The patch matches the problem of the "Playback Design" products. I just refactored a bit the code to include Teac products properly. Regards, Guillaume --- sound/usb/quirks.c | 58 +++++++++++++++++++++++++++++++----------------------- 1 file changed, 33 insertions(+), 25 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 23ea6d8..f3faaef 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1202,40 +1202,48 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) void snd_usb_set_interface_quirk(struct usb_device *dev) { /* - * "Playback Design" products need a 50ms delay after setting the + * Products needing a 50ms delay after setting the * USB interface. */ - if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) - mdelay(50); + switch (le16_to_cpu(dev->descriptor.idVendor)) { + case 0x23ba: /* Playback Design */ + case 0x0644: /* TEAC Corp. */ + mdelay(50); + } } void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size) { - /* - * "Playback Design" products need a 20ms delay after each - * class compliant request - */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); - - /* Marantz/Denon devices with USB DAC functionality need a delay - * after each class compliant request - */ - if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), - le16_to_cpu(dev->descriptor.idProduct))) - && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + if ((requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { + switch (le16_to_cpu(dev->descriptor.idVendor)) { + /* + * "Playback Design"/"TEAC Corp." products need a 20ms delay after each + * class compliant request + */ + case 0x23ba: /* "Playback Design" */ + case 0x0644: /* TEAC Corp. */ + mdelay(20); + return; + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + case 0x1686: + /* restrict to 0x00dd */ + if (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) + mdelay(1); + return; + } - /* Zoom R16/24 needs a tiny delay here, otherwise requests like - * get/set frequency return as failed despite actually succeeding. - */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && - (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(1); + /* Marantz/Denon devices with USB DAC functionality need a delay + * after each class compliant request + */ + if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct)))) + mdelay(20); + } } /* -- 2.6.4

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[alsa-devel] [PATCH] ASoC: dapm: Fix ctl value accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

snd_soc_dapm_dai_link_get() and _put() access the associated ctl values as value.integer.value[]. However, this is an enum ctl, and it has to be accessed via value.enumerated.item[]. The former is long while the latter is unsigned int, so they don't align. Fixes: c66150824b8a ('ASoC: dapm: add code to configure dai link parameters') Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d3707987900..581175a51ecf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3573,7 +3573,7 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = w->params_select; + ucontrol->value.enumerated.item[0] = w->params_select; return 0; } @@ -3587,13 +3587,13 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, if (w->power) return -EBUSY; - if (ucontrol->value.integer.value[0] == w->params_select) + if (ucontrol->value.enumerated.item[0] == w->params_select) return 0; - if (ucontrol->value.integer.value[0] >= w->num_params) + if (ucontrol->value.enumerated.item[0] >= w->num_params) return -EINVAL; - w->params_select = ucontrol->value.integer.value[0]; + w->params_select = ucontrol->value.enumerated.item[0]; return 0; } -- 2.7.2

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[alsa-devel] [PATCH 0/2] ASoC: omap: Fix enum ctl accesses
by Takashi Iwai 01 Mar '16

01 Mar '16

Hi, here is a patchset to correct the wrong type accesses in omap drivers for enum ctls. I didn't put Cc to stable for them, supposing these are only for 32bit systems. If not, feel free to put Cc to stable. thanks, Takashi === Takashi Iwai (2): ASoC: omap: n810: Fix enum ctl accesses in a wrong type ASoC: omap: rx51: Fix enum ctl accesses in a wrong type sound/soc/omap/n810.c | 18 +++++++++--------- sound/soc/omap/rx51.c | 18 +++++++++--------- 2 files changed, 18 insertions(+), 18 deletions(-) -- 2.7.2

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[alsa-devel] [PATCH] ASoC: intel: mfld: Fix enum ctl accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

"Playback Switch" and "Lineout Mux" ctls in medfld machine driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I didn't put Cc to stable, as I assume medfiled is 32bit only. If it also supports 64bit, put Cc to stable. sound/soc/intel/boards/mfld_machine.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c index 49c09a0add79..34f46c72a0e2 100644 --- a/sound/soc/intel/boards/mfld_machine.c +++ b/sound/soc/intel/boards/mfld_machine.c @@ -94,7 +94,7 @@ static const struct soc_enum lo_enum = static int headset_get_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = hs_switch; + ucontrol->value.enumerated.item[0] = hs_switch; return 0; } @@ -104,12 +104,12 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_context *dapm = &card->dapm; - if (ucontrol->value.integer.value[0] == hs_switch) + if (ucontrol->value.enumerated.item[0] == hs_switch) return 0; snd_soc_dapm_mutex_lock(dapm); - if (ucontrol->value.integer.value[0]) { + if (ucontrol->value.enumerated.item[0]) { pr_debug("hs_set HS path\n"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); @@ -123,7 +123,7 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, snd_soc_dapm_mutex_unlock(dapm); - hs_switch = ucontrol->value.integer.value[0]; + hs_switch = ucontrol->value.enumerated.item[0]; return 0; } @@ -148,7 +148,7 @@ static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm) static int lo_get_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = lo_dac; + ucontrol->value.enumerated.item[0] = lo_dac; return 0; } @@ -158,7 +158,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_context *dapm = &card->dapm; - if (ucontrol->value.integer.value[0] == lo_dac) + if (ucontrol->value.enumerated.item[0] == lo_dac) return 0; snd_soc_dapm_mutex_lock(dapm); @@ -168,7 +168,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, */ lo_enable_out_pins(dapm); - switch (ucontrol->value.integer.value[0]) { + switch (ucontrol->value.enumerated.item[0]) { case 0: pr_debug("set vibra path\n"); snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT"); @@ -202,7 +202,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, snd_soc_dapm_mutex_unlock(dapm); - lo_dac = ucontrol->value.integer.value[0]; + lo_dac = ucontrol->value.enumerated.item[0]; return 0; } -- 2.7.2

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[alsa-devel] [PATCH] ASoC: cs42l51: Fix enum ctl accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

"PCM channel mixer" ctl in cs42l51 codec driver is enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- like other patchsets, I didn't put Cc to stable. If this driver is used for 64bit archs, feel free to put Cc to stable. sound/soc/codecs/cs42l51.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index b3951524339f..35488f14e237 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -60,15 +60,15 @@ static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol, switch (value) { default: case 0: - ucontrol->value.integer.value[0] = 0; + ucontrol->value.enumerated.item[0] = 0; break; /* same value : (L+R)/2 and (R+L)/2 */ case 1: case 2: - ucontrol->value.integer.value[0] = 1; + ucontrol->value.enumerated.item[0] = 1; break; case 3: - ucontrol->value.integer.value[0] = 2; + ucontrol->value.enumerated.item[0] = 2; break; } @@ -85,7 +85,7 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); unsigned char val; - switch (ucontrol->value.integer.value[0]) { + switch (ucontrol->value.enumerated.item[0]) { default: case 0: val = CHAN_MIX_NORMAL; -- 2.7.2

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[alsa-devel] [PATCH] ASoC: da732x: Fix enum ctl accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

"DAC1 High Pass Filter Mode" & co in da732x codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- like other patchsets, I didn't put Cc to stable. If this driver is used for 64bit archs, feel free to put Cc to stable. sound/soc/codecs/da732x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 1d5a89c5164b..461506a4ca6a 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -334,7 +334,7 @@ static int da732x_hpf_set(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; unsigned int reg = enum_ctrl->reg; - unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.enumerated.item[0]; unsigned int bits; switch (sel) { @@ -368,13 +368,13 @@ static int da732x_hpf_get(struct snd_kcontrol *kcontrol, switch (val) { case DA732X_HPF_VOICE_EN: - ucontrol->value.integer.value[0] = DA732X_HPF_VOICE; + ucontrol->value.enumerated.item[0] = DA732X_HPF_VOICE; break; case DA732X_HPF_MUSIC_EN: - ucontrol->value.integer.value[0] = DA732X_HPF_MUSIC; + ucontrol->value.enumerated.item[0] = DA732X_HPF_MUSIC; break; default: - ucontrol->value.integer.value[0] = DA732X_HPF_DISABLED; + ucontrol->value.enumerated.item[0] = DA732X_HPF_DISABLED; break; } -- 2.7.2

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[alsa-devel] [PATCH 0/5] ASoC: Fix enum ctl accesses in various codec drivers
by Takashi Iwai 01 Mar '16

01 Mar '16

Hi, here are the rest of buggy drivers to be fixed for the wrong ctl access types to enum ctls. Again, put Cc to stable appropriately if needed. thanks, Takashi === Takashi Iwai (5): ASoC: ab8500: Fix enum ctl accesses in a wrong type ASoC: max98088: Fix enum ctl accesses in a wrong type ASoC: max98095: Fix enum ctl accesses in a wrong type ASoC: tlv320dac33: Fix enum ctl accesses in a wrong type ASoC: wl1273: Fix enum ctl accesses in a wrong type sound/soc/codecs/ab8500-codec.c | 8 ++++---- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 4 ++-- sound/soc/codecs/tlv320dac33.c | 9 ++++----- sound/soc/codecs/wl1273.c | 13 ++++++------- 5 files changed, 17 insertions(+), 19 deletions(-) -- 2.7.2

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[alsa-devel] [PATCH v2 0/6] ALSA core X32 ABI fixes
by Takashi Iwai 29 Feb '16

29 Feb '16

Hi, this is the revised patchset for fixing the incompatible ioctls in ALSA core with X32 ABI. Too bad that we have to deal in this way, but it's life... Takashi === Takashi Iwai (6): ALSA: ctl: Fix ioctls for X32 ABI ALSA: pcm: Fix ioctls for X32 ABI ALSA: rawmidi: Use comapt_put_timespec() ALSA: rawmidi: Fix ioctls X32 ABI ALSA: timer: Fix broken compat timer user status ioctl ALSA: timer: Fix ioctls for X32 ABI sound/core/control_compat.c | 90 ++++++++++++++++++---- sound/core/pcm_compat.c | 177 +++++++++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 56 +++++++++++++- sound/core/timer_compat.c | 18 ++++- 4 files changed, 320 insertions(+), 21 deletions(-) -- 2.7.2

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