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

[alsa-devel] limitations on number of alsa controls for a card
by Vinod Koul 02 Nov '12

02 Nov '12

Hi, Is there any limitation on maximum number of controls we can have. I had some platform side controls to represent DSP and adding them made some of codec controls to vanish. And looks like if I register these platform controls not all codec controls are visible in userpsace. Is there nay max limit I am hitting? -- ~Vinod

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[alsa-devel] Transmitting ac3 content via spdif
by Fabio Estevam 02 Nov '12

02 Nov '12

Hi, I am trying to transmit ac3 content via spdif on a mx6 (on a yet non-mainlined driver). Tried to force non-audio mode via iecset: # iecset audio 0 control "IEC958 Playback Default" (index -1) not found root@imx6qsabrelite:~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sgtl5000audio [sgtl5000-audio], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: imxhdmisoc [imx-hdmi-soc], device 0: IMX HDMI TX mxc-hdmi-soc-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Transmitting a wav file works fine though. Any suggestions as to how provide the "IEC958 Playback Default" control? Thanks, Fabio Estevam

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[alsa-devel] [PATCH RFT] ASoC: si476x: Add missing break for SNDRV_PCM_FORMAT_S8 switch case
by Axel Lin 01 Nov '12

01 Nov '12

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/si476x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 38145ba..dadf04d 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: width = SI476X_PCM_FORMAT_S8; + break; case SNDRV_PCM_FORMAT_S16_LE: width = SI476X_PCM_FORMAT_S16_LE; break; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: mxs-saif: Fix channel swap for 24-bit format
by Fabio Estevam 01 Nov '12

01 Nov '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Playing 24-bit format file leads to channel swap on mx28 and the reason is that the current driver performs one write/read to/from the SAIF_DATA register to trigger the transfer. This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register and thus is capable of storing the 16-bit left and right channels, but for the S24_LE case it can only store one channel, so in order to not lose the FIFO sync an extra read/write is needed. Reported-by: Dan Winner <DWinner(a)tc-helicon.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Tested-by: Dan Winner <DWinner(a)tc-helicon.com> --- sound/soc/mxs/mxs-saif.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aa037b2..05a875c 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -523,16 +523,22 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* - * write a data to saif data register to trigger - * the transfer + * write data to saif data register to trigger + * the transfer. + * For 24-bit format the 32-bit FIFO register stores + * only one channel, so we need to write twice. */ __raw_writel(0, saif->base + SAIF_DATA); + __raw_writel(0, saif->base + SAIF_DATA); } else { /* - * read a data from saif data register to trigger - * the receive + * read data from saif data register to trigger + * the receive. + * For 24-bit format the 32-bit FIFO register stores + * only one channel, so we need to read twice. */ __raw_readl(saif->base + SAIF_DATA); + __raw_readl(saif->base + SAIF_DATA); } master_saif->ongoing = 1; -- 1.7.9.5

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[alsa-devel] [PATCH v3 1/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 01 Nov '12

01 Nov '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v2: - Also remove 'if (np)' as dt is the only probing mechanism. Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 42 +++++++++++++++--------------------------- 2 files changed, 15 insertions(+), 43 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aa037b2..ab5bdec 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -652,10 +651,9 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; - + struct device_node *master; if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; @@ -664,31 +662,21 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

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[alsa-devel] macmini5,3 and imac27 ssid conflict
by Raymond Yau 01 Nov '12

01 Nov '12

>>> http://www.alsa-project.org/db/?f=5e7f45089ce82985c003fa2608bed82726ebbdf6 > > how many audio jacks do your macmini have ? > > > > seem only line in support jack detect, internal mic support balanced but > > ext mic does not support jack detect > > There are two audio jacks: > > 1. An analogue headset/external speakers jack. Note that in order for the external speakers to work, I need to use HDA Analyser to change the settings otherwise things remain stuck on the internal speaker. I did not try an analogue headset. https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/892697 > > 2. An analogue line-in jack. I tried this with a powered condenser microphone - I can hear a recording, but the level is too low for practical use. > > I'm not sure there is an internal microphone connected - shouting at the mac mini box does not produce an audible recording :P http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=6d… Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:1c20 (rev 05) Subsystem: 8086:7270 your macmini pci subsystem is also 8086:7270 which is same as imac27 the pin default of your macmini (init_pin_config) is replaced by the pin config of model imac27 (driver_pin_config) Codec: Cirrus Logic CS4206 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10134206 Subsystem Id: 0x106b2100 is the subsystem id in codec unique ? Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x09 0x400000f0 0x0a 0x002b4040 0x0b 0x90100130 0x0c 0x018b3020 0x0d 0x90a00110 0x0e 0x400000f0 0x0f 0x01cbe030 0x10 0x014be060 0x12 0x01ab9070 0x15 0x400000f0 /sys/class/sound/hwC0D0/driver_pin_configs: 0x09 0x012b4050 0x0a 0x90100140 0x0b 0x90100142 0x0c 0x018b3020 0x0d 0x90a00110 0x0e 0x400000f0 0x0f 0x01cbe030 0x10 0x014be060 0x12 0x01ab9070 0x15 0x400000f0 /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs:

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[alsa-devel] [PATCH] ASoC: mxs-pcm: Fix fifo size
by Fabio Estevam 31 Oct '12

31 Oct '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> According to MX28 Reference Manual the SAIF FIFO contains 4-bytes. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index f82d766..6b360d0 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -60,7 +60,7 @@ static struct snd_pcm_hardware snd_mxs_hardware = { .periods_min = 1, .periods_max = 52, .buffer_bytes_max = 64 * 1024, - .fifo_size = 32, + .fifo_size = 4, }; -- 1.7.9.5

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[alsa-devel] ALSA hda/patch_via -- Error in via_auto_fill_dac_nids
by Massimo Del Fedele 31 Oct '12

31 Oct '12

The function should scan all paths from DACs to OUTS, but like it is just takes the first available DAC. The problem arises here : if (dac) { spec->private_dac_nids[i] = dac; dac_num++; } Which correctly adds DAC to used list, but doesn't update the total number of found DACs in spec->multiout.num_dacs; following calls to is_empty_dac() will always return true for already used DACs, making parse_output_path() return always the first one. Corrected function should be like this : static int via_auto_fill_dac_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; const struct auto_pin_cfg *cfg = &spec->autocfg; int i; hda_nid_t nid; spec->multiout.dac_nids = spec->private_dac_nids; spec->multiout.num_dacs = 0; for (i = 0; i < cfg->line_outs; i++) { hda_nid_t dac = 0; nid = cfg->line_out_pins[i]; if (!nid) continue; if (parse_output_path(codec, nid, 0, 0, &spec->out_path[i])) dac = spec->out_path[i].path[0]; if (!i && parse_output_path(codec, nid, dac, 1, &spec->out_mix_path)) dac = spec->out_mix_path.path[0]; if (dac) { spec->private_dac_nids[i] = dac; spec->multiout.num_dacs++; } } if (!spec->out_path[0].depth && spec->out_mix_path.depth) { spec->out_path[0] = spec->out_mix_path; spec->out_mix_path.depth = 0; } return 0; } This one (and a few other changes) allows the subwoofer of asus g75 work correctly. Sorry for not submitting a patch, I'm quite new to alsa; please tell me if/how I should do it. Ciao Max

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[alsa-devel] [PATCH v2 1/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 31 Oct '12

31 Oct '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 9 --------- 2 files changed, 25 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aa037b2..76a9c4a 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -652,7 +651,6 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; @@ -682,13 +680,6 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) if (saif->master_id < 0) return saif->master_id; } - } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

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[alsa-devel] Query about alsa ucm (use case manager) config files
by Ashish Chavan 31 Oct '12

31 Oct '12

Hi, I am just curious to know if there is any centralized repository for alsa use case manager configuration files. I tried searching and could only find alsa-ucm-conf.git repo at alsa project git server, which is actually an empty repo! How people share/reuse ucm configuration files for a specific codec? In near future, I will be developing few ucm configuration files for some codecs and wish to share them with rest of the world. I think even though ucm configs are board/machine specific, it makes sense to share them. Is there any other, more visible place to put these config files than board specific distributions?

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