On Jun 18, 2010, at 11:38 AM, alsa-devel-request@alsa-project.org wrote:
Send Alsa-devel mailing list submissions to alsa-devel@alsa-project.org
To subscribe or unsubscribe via the World Wide Web, visit http://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request@alsa-project.org
You can reach the person managing the list at alsa-devel-owner@alsa-project.org
When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..."
Today's Topics:
- Re: [PATCH 2/5] ALSA: usb-audio: unify UAC macros and struct names (Daniel Mack)
- Re: Timing Info (Clemens Ladisch)
- Re: [PATCH] ASoC: pandora: fix CLKX polarity (Mark Brown)
- [PATCH] ALSA: hda - add ideapad model for Conexant 5051 codec (Herton Ronaldo Krzesinski)
- Re: [PATCH] ALSA: hda - add ideapad model for Conexant 5051 codec (Takashi Iwai)
- Re: snd_mixer_selem_get_playback_volume return value outside allowed range (Raymond Yau)
- Re: [PATCH] eukrea-tlv320: add support for our i.MX25 board (Mark Brown)
- SOC_DOUBLE_R_SX_TLV (was Update on TLV320AIC3204 Driver) (Stuart Longland)
Message: 1 Date: Thu, 17 Jun 2010 17:26:26 +0200 From: Daniel Mack daniel@caiaq.de Subject: Re: [alsa-devel] [PATCH 2/5] ALSA: usb-audio: unify UAC macros and struct names To: Alex Lee alexlee188@gmail.com Cc: Takashi Iwai tiwai@suse.de, "alsa-devel@alsa-project.org" alsa-devel@alsa-project.org, "clemens@ladisch.de" clemens@ladisch.de Message-ID: 20100617152626.GL17833@buzzloop.caiaq.de Content-Type: text/plain; charset=us-ascii
(hint, hint: don't top-post :))
On Thu, Jun 17, 2010 at 08:21:22AM -0700, Alex Lee wrote:
It will not help existing codes, I agree. But it may make writing new UAC2 specific code easier?
Not as long as most of the code stays hybrid for both versions. If there were versions of these enum values for both UAC1 and UAC2, you would need to choose one of them in the parser codes. Which one you go for doesn't actually make any difference as they're both the same, but the code would read as if it was specific to any version.
I'd say this would add more to the confusion, and not ease any.
Daniel
Message: 2 Date: Thu, 17 Jun 2010 17:36:59 +0200 From: Clemens Ladisch clemens@ladisch.de Subject: Re: [alsa-devel] Timing Info To: Paul Dugas paul@dugasenterprises.com Cc: alsa-devel@alsa-project.org Message-ID: 4C1A411B.2060301@ladisch.de Content-Type: text/plain; charset=us-ascii
Paul Dugas wrote:
On Sun, Jun 13, 2010 at 10:10 AM, Clemens Ladisch clemens@ladisch.de wrote:
Set SND_PCM_TSTAMP_ENABLE to generate a timestamp whenever the hardware position is updated. Call snd_pcm_status to read the position and the corresponding timestamp.
I've re-read this a number of times and have been fiddling with code that dumps the status at various points but I'm still not grasping the concept here. Can comeone point me to an explanation of the delay, avail, and avail_max values?
The avail value tells you how many frames are available, i.e., can be written, in the output buffer. The largest value that this had since the last snd_pcm_status call is avail_max.
The delay value is the time that will elapse before the next frame that you write into the buffer will actually be output by the hardware. This is not only the time until the hardware reads this frame out of the buffer (which could be computed from avail) but also any processing after that.
HTH Clemens
Message: 3 Date: Thu, 17 Jun 2010 16:50:01 +0100 From: Mark Brown broonie@opensource.wolfsonmicro.com Subject: Re: [alsa-devel] [PATCH] ASoC: pandora: fix CLKX polarity To: Grazvydas Ignotas notasas@gmail.com, alsa-devel@alsa-project.org Cc: lrg@slimlogic.co.uk Message-ID: u6g5khs6pbrq63fngu2ncjvv.1276789801383@email.android.com Content-Type: text/plain; charset=utf-8
Acked-by: Mark Brown broonie@opensource.wolfsonmicro.com
Probably best to add a CC to stable too. Sorry about the rubbish formatting, on my phone and it can't do any better.
Grazvydas Ignotas notasas@gmail.com wrote:
After mass production started it was found that several boards exhibit noise problems during sound playback. After some investigation it was determined that CLKX polarity is set incorrectly, and even if most boards can tolerate the wrong setting, there are some that don't.
Fix polarity setup in the board file. As the clock settings for input and output now match, merge in and out functions and structures to simplify code.
Signed-off-by: Grazvydas Ignotas notasas@gmail.com
sound/soc/omap/omap3pandora.c | 36 ++++++++---------------------------- 1 files changed, 8 insertions(+), 28 deletions(-)
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 87ce842..9eecac1 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -43,12 +43,14 @@
static struct regulator *omap3pandora_dac_reg;
-static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params, unsigned int fmt)
+static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
{ struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS;int ret;
/* Set codec DAI configuration */
@@ -91,24 +93,6 @@ static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, return 0; }
-static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- return omap3pandora_cmn_hw_params(substream, params,
SND_SOC_DAIFMT_I2S |SND_SOC_DAIFMT_IB_NF |SND_SOC_DAIFMT_CBS_CFS);-}
-static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- return omap3pandora_cmn_hw_params(substream, params,
SND_SOC_DAIFMT_I2S |SND_SOC_DAIFMT_NB_NF |SND_SOC_DAIFMT_CBS_CFS);-}
static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -231,12 +215,8 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) return snd_soc_dapm_sync(codec); }
-static struct snd_soc_ops omap3pandora_out_ops = {
- .hw_params = omap3pandora_out_hw_params,
-};
-static struct snd_soc_ops omap3pandora_in_ops = {
- .hw_params = omap3pandora_in_hw_params,
+static struct snd_soc_ops omap3pandora_ops = {
- .hw_params = omap3pandora_hw_params,
};
/* Digital audio interface glue - connects codec <--> CPU */ @@ -246,14 +226,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap3pandora_out_ops,
.ops = &omap3pandora_ops,
.ops = &omap3pandora_in_ops,
.ops = &omap3pandora_ops,};
1.6.3.3
Message: 4 Date: Thu, 17 Jun 2010 14:15:06 -0300 From: Herton Ronaldo Krzesinski herton@mandriva.com.br Subject: [alsa-devel] [PATCH] ALSA: hda - add ideapad model for Conexant 5051 codec To: alsa-devel@alsa-project.org Cc: Takashi Iwai tiwai@suse.de Message-ID: 201006171415.07101.herton@mandriva.com.br Content-Type: Text/Plain; charset="us-ascii"
Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b, which isn't muted when headphone is plugged in. This adds additional support to the extra subwoofer via new ideapad model.
Signed-off-by: Herton Ronaldo Krzesinski herton@mandriva.com.br
Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 20 ++++++++++++++++++++ 2 files changed, 21 insertions(+), 0 deletions(-)
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 1d38b0d..84e81ad 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -282,6 +282,7 @@ Conexant 5051 hp HP Spartan laptop hp-dv6736 HP dv6736 hp-f700 HP Compaq Presario F700
- ideapad Lenovo IdeaPad laptop lenovo-x200 Lenovo X200 laptop toshiba Toshiba Satellite M300
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2bf2cb5..54f7419 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1632,6 +1632,11 @@ static void cxt5051_update_speaker(struct hda_codec *codec) pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
- /* on ideapad there is an aditional speaker (subwoofer) to mute */
- if (spec->ideapad)
snd_hda_codec_write(codec, 0x1b, 0,AC_VERB_SET_PIN_WIDGET_CONTROL,pinctl);}
/* turn on/off EAPD (+ mute HP) as a master switch */ @@ -1888,6 +1893,13 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, #endif }
+static struct hda_verb cxt5051_ideapad_init_verbs[] = {
- /* Subwoofer */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- { } /* end */
+};
/* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1917,6 +1929,7 @@ enum { CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini Dock 250410 */ CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_TOSHIBA, /* Toshiba M300 & co */
- CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ CXT5051_MODELS
};
@@ -1927,6 +1940,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LENOVO_X200] = "lenovo-x200", [CXT5051_F700] = "hp-700", [CXT5051_TOSHIBA] = "toshiba",
- [CXT5051_IDEAPAD] = "ideapad",
};
static struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1938,6 +1952,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200),
- SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), {}
};
@@ -1999,6 +2014,11 @@ static int patch_cxt5051(struct hda_codec *codec) spec->mixers[0] = cxt5051_toshiba_mixers; spec->auto_mic = AUTO_MIC_PORTB; break;
case CXT5051_IDEAPAD:
spec->init_verbs[spec->num_init_verbs++] =cxt5051_ideapad_init_verbs;spec->ideapad = 1;break;}
return 0;
-- 1.7.1
Message: 5 Date: Thu, 17 Jun 2010 20:39:20 +0200 From: Takashi Iwai tiwai@suse.de Subject: Re: [alsa-devel] [PATCH] ALSA: hda - add ideapad model for Conexant 5051 codec To: Herton Ronaldo Krzesinski herton@mandriva.com.br Cc: alsa-devel@alsa-project.org Message-ID: s5hk4pxld0n.wl%tiwai@suse.de Content-Type: text/plain; charset=US-ASCII
At Thu, 17 Jun 2010 14:15:06 -0300, Herton Ronaldo Krzesinski wrote:
Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b, which isn't muted when headphone is plugged in. This adds additional support to the extra subwoofer via new ideapad model.
Signed-off-by: Herton Ronaldo Krzesinski herton@mandriva.com.br
Thanks, applied now.
Takashi
Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 20 ++++++++++++++++++++ 2 files changed, 21 insertions(+), 0 deletions(-)
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 1d38b0d..84e81ad 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -282,6 +282,7 @@ Conexant 5051 hp HP Spartan laptop hp-dv6736 HP dv6736 hp-f700 HP Compaq Presario F700
- ideapad Lenovo IdeaPad laptop lenovo-x200 Lenovo X200 laptop toshiba Toshiba Satellite M300
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2bf2cb5..54f7419 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1632,6 +1632,11 @@ static void cxt5051_update_speaker(struct hda_codec *codec) pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
- /* on ideapad there is an aditional speaker (subwoofer) to mute */
- if (spec->ideapad)
snd_hda_codec_write(codec, 0x1b, 0,AC_VERB_SET_PIN_WIDGET_CONTROL,pinctl);}
/* turn on/off EAPD (+ mute HP) as a master switch */ @@ -1888,6 +1893,13 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, #endif }
+static struct hda_verb cxt5051_ideapad_init_verbs[] = {
- /* Subwoofer */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- { } /* end */
+};
/* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1917,6 +1929,7 @@ enum { CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini Dock 250410 */ CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_TOSHIBA, /* Toshiba M300 & co */
- CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ CXT5051_MODELS
};
@@ -1927,6 +1940,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LENOVO_X200] = "lenovo-x200", [CXT5051_F700] = "hp-700", [CXT5051_TOSHIBA] = "toshiba",
- [CXT5051_IDEAPAD] = "ideapad",
};
static struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1938,6 +1952,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200),
- SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), {}
};
@@ -1999,6 +2014,11 @@ static int patch_cxt5051(struct hda_codec *codec) spec->mixers[0] = cxt5051_toshiba_mixers; spec->auto_mic = AUTO_MIC_PORTB; break;
case CXT5051_IDEAPAD:
spec->init_verbs[spec->num_init_verbs++] =cxt5051_ideapad_init_verbs;spec->ideapad = 1;break;}
return 0;
-- 1.7.1
Message: 6 Date: Fri, 18 Jun 2010 07:39:07 +0800 From: Raymond Yau superquad.vortex2@gmail.com Subject: Re: [alsa-devel] snd_mixer_selem_get_playback_volume return value outside allowed range To: ALSA Development Mailing List alsa-devel@alsa-project.org Message-ID: AANLkTin7y2JzkaCYb4p69uDfWOo4cTa85in0Q3l7RwbD@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1
2010/6/17 Clemens Ladisch clemens@ladisch.de
Raymond Yau wrote:
Should snd_mixer_selem_get_playback_volume() perform range check ?
No, it's the responsibility of the control implementation to return valid values (and to check that values that are being set are valid).
Regards, Clemens
If it is the responsibility of the driver/external plugin to keep the value inside the allowed range
your patch actually help those buggy driver/external plugin hide the bug only
Message: 7 Date: Fri, 18 Jun 2010 01:56:37 +0100 From: Mark Brown broonie@opensource.wolfsonmicro.com Subject: Re: [alsa-devel] [PATCH] eukrea-tlv320: add support for our i.MX25 board To: Eric B?nard eric@eukrea.com Cc: alsa-devel@alsa-project.org, s.hauer@pengutronix.de, lrg@slimlogic.co.uk Message-ID: 20100618005637.GA3333@opensource.wolfsonmicro.com Content-Type: text/plain; charset=iso-8859-1
On Thu, Jun 17, 2010 at 03:44:01PM +0200, Eric B?nard wrote:
- tdm slot has to be configured to get sound working on i.MX25
Signed-off-by: Eric B?nard eric@eukrea.com
Applied but I had to hand apply the Kconfig update, please check that it applied correctly.
Message: 8 Date: Fri, 18 Jun 2010 12:38:10 +1000 From: Stuart Longland redhatter@gentoo.org Subject: [alsa-devel] SOC_DOUBLE_R_SX_TLV (was Update on TLV320AIC3204 Driver) To: Mark Brown broonie@opensource.wolfsonmicro.com Cc: Takashi Iwai tiwai@suse.de, alsa-devel@alsa-project.org, Eric B??nard eric@eukrea.com, Liam Girdwood lrg@slimlogic.co.uk Message-ID: 20100618023810.GS7759@www.longlandclan.yi.org Content-Type: text/plain; charset=us-ascii
On Tue, Jun 15, 2010 at 03:11:10PM +1000, Stuart Longland wrote:
When a read is performed, this structure points to this function as the means for reading the register (defined in sound/soc-core.c):
/**
- snd_soc_get_volsw_2r_sx - double with tlv and variable data size
- mixer get callback
- @kcontrol: mixer control
- @uinfo: control element information
- Returns 0 for success.
*/ int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int mask = (1<<mc->shift)-1; int min = mc->min; int val = snd_soc_read(codec, mc->reg) & mask; int valr = snd_soc_read(codec, mc->rreg) & mask;
ucontrol->value.integer.value[0] = ((val & 0xff)-min); ucontrol->value.integer.value[1] = ((valr & 0xff)-min); return 0;} EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx);
I've figured it out now... and there'll be a patch to fix it.
The mute <--> gain interaction was pure coincidence as it turns out. There's no interaction between the two, just that when I toggle the mute, the gain gets re-read, and hence it *appeared* to interact.
Notice the value calculated for the channel is:
output = ( regval & 0xff ) - minimum
In my case, my minimum is -6. Suppose I set my driver gain to -1dB, in hexadecimal, 0x3f. Since no sign extension is done, (regval & 0xff) = 63. Adding 6 to this (double negation) you get 69... which is greater than the maximum gain of 29. Hence the scale shoots up to maximum.
The fix, is to change the above so that they mask the result; thus wrapping around the zero point like so:
output = ( ( regval & 0xff ) - minimum ) & mask
This fixes the problem that I observe, and shouldn't break the other user of the control. A patch to fix this issue is coming. -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.'
I haven't lost my mind... ...it's backed up on a tape somewhere.
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
End of Alsa-devel Digest, Vol 40, Issue 80