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[alsa-devel] Vendor support in Alsa Drivers
by Emmanuel Kasper 15 Jun '08

15 Jun '08

Hello With a couple of other french folks, we are building a kind of webshop for Linux/FreeBSD compatible hardware. We will reference only devices which work with a free driver, so no nvidia binary thingie or atheros HAL. Based on the information found on the alsa wiki we plan to reference the following sound devices: M-Audio Audiophile 24/96 ( snd-ice1712 driver ) Terratec AUREON 5.1 M-Audio Fast Track usb Hercules Muse Pocket LT Ion TTUSB usb turnable ( these 4 working with snd-usb-driver ) We would be also interested to indicate on each product description, if the vendor of the product, or the chipset maker for that matters had been helpful with developers trying to write a free driver. So for the devices mentionned above, was any of the vendor helpful by providing programming documentation or sample hardware ? If yes we will mention it on the product description with an extra tag like "programmer friendly" Regards -- Emmanuel Kasper System Administrator www.libera.cc Find, compare, buy Linux & FreeBSD compatible hardware

6 7

[alsa-devel] input source selection not working (snd-hda-intel)
by Bruno Schneider 14 Jun '08

14 Jun '08

Hi all, I'm using a computer with a ASUS P5LD2-X motherboard. The audio controller, is a Realtek ALC662. I'm using an updated Debian Lenny (testing) system. No matter what I set as the input source, sound is always recorded from the back microphone jack. Before posting here, I tried to get help from my Debian list and from the alsa-user list [1]. I'm starting to think this is a bug on the snd-hda-intel module. Please help. [1] http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg22667.html -- Bruno Schneider http://www.dcc.ufla.br/~bruno/

1 0

[alsa-devel] [PATCH][ASoC v2] Update Freescale MPC8610HPCD fabric driver to support multiple codecs
by Timur Tabi 14 Jun '08

14 Jun '08

Update the Freescale MPC8610 HPCD fabric driver to create a new platform device for each SSI in the system, instead of registering multiple PCMs under one card. This is necessary for alsamixer to control the individual codecs. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 236 +++++++++++++++++++++--------------------- 1 files changed, 117 insertions(+), 119 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 8813291..2cf7d95 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -40,13 +40,10 @@ struct mpc8610_hpcd_data { struct snd_soc_card soc_card; struct ccsr_guts __iomem *guts; - unsigned int num_configs; - struct snd_soc_pcm_config configs[MAX_SSI]; - struct mpc8610_hpcd_names { - char pcm_name[32]; - char ssi_name[32]; - char codec_name[32]; - } names[MAX_SSI]; + struct snd_soc_pcm_config pcm_config; + char pcm_name[32]; + char ssi_name[32]; + char codec_name[32]; }; /** @@ -91,29 +88,25 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) { struct snd_soc_codec *codec; struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; - unsigned int i; int ret = 0; - for (i = 0; i < soc_card_data->num_configs; i++) { - codec = snd_soc_card_get_codec(soc_card, - soc_card_data->names[i].codec_name, - soc_card_data->configs[i].codec_num); + codec = snd_soc_card_get_codec(soc_card, soc_card_data->codec_name, + soc_card_data->pcm_config.codec_num); - if (!codec) { - dev_err(soc_card->dev, "could not find codec\n"); - return -ENODEV; - } + if (!codec) { + dev_err(soc_card->dev, "could not find codec\n"); + return -ENODEV; + } - /* The codec driver should have called - * snd_soc_card_config_codec() by now. - */ - ret = snd_soc_card_init_codec(codec, soc_card); - if (ret < 0) { - dev_err(soc_card->dev, - "could not initialize codec %s-%u\n", - codec->name, codec->num); - continue; - } + /* The codec driver should have called + * snd_soc_card_config_codec() by now. + */ + ret = snd_soc_card_init_codec(codec, soc_card); + if (ret < 0) { + dev_err(soc_card->dev, + "could not initialize codec %s-%u\n", + codec->name, codec->num); + return -ENODEV; } return 0; @@ -316,10 +309,16 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) struct snd_soc_card *soc_card = NULL; struct mpc8610_hpcd_data *soc_card_data; struct device_node *guts_np = NULL; - struct device_node *ssi_np; - unsigned int ssi = 0; + struct device_node *ssi_np = + (struct device_node *)platform_get_resource(pdev, 0, 0)->start; int ret = -ENODEV; + struct device_node *codec_np; + const char *compat; + const char *p; + unsigned int bus; + unsigned int address; + soc_card_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); if (!soc_card_data) { dev_err(&pdev->dev, "could not allocate card structure\n"); @@ -353,100 +352,68 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) soc_card->private_data = soc_card_data; soc_card->dev = &pdev->dev; - /* Scan the device tree for SSI nodes. Each one we find that has a - * codec is registered. Remember, we only support MAX_SSI of these. - */ - for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") { - struct snd_soc_pcm_config *pcm_config; - struct mpc8610_hpcd_names *name; - struct device_node *codec_np; - const char *compat; - const char *p; - unsigned int bus; - unsigned int address; - int i; - - name = &soc_card_data->names[ssi]; - pcm_config = &soc_card_data->configs[ssi]; - - pcm_config->name = name->pcm_name; - pcm_config->codec = name->codec_name; - pcm_config->codec_dai = name->codec_name; - pcm_config->platform = "fsl-elo"; - pcm_config->cpu_dai = name->ssi_name; - pcm_config->ops = &mpc8610_hpcd_ops; - pcm_config->playback = 1; - pcm_config->capture = 1; - - sprintf(name->pcm_name, "MPC8610HPCD.%u", ssi); - - /* Get the SSI name */ - i = of_get_integer(ssi_np, "cell-index"); - if (i < 0) { - dev_err(soc_card->dev, "no cell-index for %s\n", - ssi_np->full_name); - continue; - } - sprintf(name->ssi_name, "ssi%u", i); - - /* Get the codec name and ID */ - codec_np = find_codec(ssi_np); - if (!codec_np) { - dev_err(soc_card->dev, "missing codec node for %s\n", - ssi_np->full_name); - continue; - } - - /* We assume that the first (or only) string in the compatible - * field is the one that counts. - */ - compat = of_get_property(codec_np, "compatible", NULL); - if (!compat) { - dev_err(soc_card->dev, - "missing compatible property for %s\n", - ssi_np->full_name); - continue; - } - - /* We only care about the part after the comma */ - p = strchr(compat, ','); - strcpy(name->codec_name, p ? p + 1 : compat); - - /* Now determine the I2C bus and address of the codec */ - bus = of_get_integer(of_get_parent(codec_np), "cell-index"); - if (bus < 0) { - dev_err(soc_card->dev, - "cannot determine I2C bus number for %s\n", - codec_np->full_name); - continue; - } + soc_card_data->pcm_config.name = soc_card_data->pcm_name; + soc_card_data->pcm_config.codec = soc_card_data->codec_name; + soc_card_data->pcm_config.codec_dai = soc_card_data->codec_name; + soc_card_data->pcm_config.platform = "fsl-elo"; + soc_card_data->pcm_config.cpu_dai = soc_card_data->ssi_name; + soc_card_data->pcm_config.ops = &mpc8610_hpcd_ops; + soc_card_data->pcm_config.playback = 1; + soc_card_data->pcm_config.capture = 1; + + sprintf(soc_card_data->pcm_name, "MPC8610HPCD.%u", pdev->id); + + /* Get the SSI name */ + sprintf(soc_card_data->ssi_name, "ssi%u", pdev->id); + + /* Get the codec name and ID */ + codec_np = find_codec(ssi_np); + if (!codec_np) { + dev_err(soc_card->dev, "missing codec node for %s\n", + ssi_np->full_name); + goto error; + } - address = of_get_integer(codec_np, "reg"); - if (address < 0) { - dev_err(soc_card->dev, - "cannot determine I2C address for %s\n", - codec_np->full_name); - continue; - } + /* We assume that the first (or only) string in the compatible + * field is the one that counts. + */ + compat = of_get_property(codec_np, "compatible", NULL); + if (!compat) { + dev_err(soc_card->dev, + "missing compatible property for %s\n", + ssi_np->full_name); + goto error; + } - pcm_config->codec_num = bus << 16 | address; + /* We only care about the part after the comma */ + p = strchr(compat, ','); + strcpy(soc_card_data->codec_name, p ? p + 1 : compat); - dev_dbg(soc_card->dev, "registering cpu %s with codec %s-%x\n", - pcm_config->cpu_dai, pcm_config->codec, - pcm_config->codec_num); + /* Now determine the I2C bus and address of the codec */ + bus = of_get_integer(of_get_parent(codec_np), "cell-index"); + if (bus < 0) { + dev_err(soc_card->dev, + "cannot determine I2C bus number for %s\n", + codec_np->full_name); + goto error; + } - if (++ssi == MAX_SSI) - break; + address = of_get_integer(codec_np, "reg"); + if (address < 0) { + dev_err(soc_card->dev, + "cannot determine I2C address for %s\n", + codec_np->full_name); + goto error; } - ret = snd_soc_card_create_pcms(soc_card, soc_card_data->configs, ssi); + soc_card_data->pcm_config.codec_num = bus << 16 | address; + + ret = snd_soc_card_create_pcms(soc_card, &soc_card_data->pcm_config, 1); if (ret) { dev_err(soc_card->dev, "could not create PCMs\n"); return ret; } - soc_card_data->num_configs = ssi; - platform_set_drvdata(pdev, soc_card); ret = snd_soc_card_register(soc_card); @@ -493,7 +460,8 @@ static struct platform_driver mpc8610_hpcd_driver = { .remove = __devexit_p(mpc8610_hpcd_remove), }; -static struct platform_device *pdev; +/* Platform device pointers for each SSI */ +static struct platform_device *pdev[MAX_SSI]; /** * mpc8610_hpcd_init: fabric driver initialization. @@ -502,6 +470,9 @@ static struct platform_device *pdev; */ static int __init mpc8610_hpcd_init(void) { + struct resource res; + struct device_node *ssi_np; + unsigned int i = 0; int ret; pr_info("Freescale MPC8610 HPCD ASoC fabric driver\n"); @@ -512,12 +483,35 @@ static int __init mpc8610_hpcd_init(void) return ret; } - pdev = platform_device_register_simple(mpc8610_hpcd_driver.driver.name, - 0, NULL, 0); - if (!pdev) { - pr_err("mpc8610-hpcd: could not register device\n"); - platform_driver_unregister(&mpc8610_hpcd_driver); - return ret; + memset(pdev, 0, sizeof(pdev)); + + memset(&res, 0, sizeof(res)); + res.name = "ssi"; + + for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") { + int id; + + id = of_get_integer(ssi_np, "cell-index"); + if (id < 0) { + pr_err("mpc8610-hpcd: no cell-index for %s\n", + ssi_np->full_name); + continue; + } + if (!find_codec(ssi_np)) + /* No codec node? Skip it */ + continue; + + res.start = (resource_size_t) ssi_np; + + pdev[i] = platform_device_register_simple( + mpc8610_hpcd_driver.driver.name, id, &res, 1); + if (!pdev[i]) { + pr_err("mpc8610-hpcd: could not register %s\n", + ssi_np->full_name); + continue; + } + + i++; } return 0; @@ -530,7 +524,11 @@ static int __init mpc8610_hpcd_init(void) */ static void __exit mpc8610_hpcd_exit(void) { - platform_device_unregister(pdev); + unsigned int i; + + for (i = 0; i < MAX_SSI; i++) + platform_device_unregister(pdev[i]); + platform_driver_unregister(&mpc8610_hpcd_driver); } -- 1.5.5

4 26

[alsa-devel] Specs available for Sweex External 5.1 with digital in/out USB 2.0 / C-Media CM6207
by Erik Slagter 14 Jun '08

14 Jun '08

Hi there, Today I bought this sound device, I was hoping it had the C-Media CM650? USB chip which is already supported by alsa (according to the list). Instead it has a CM6207 inside. The good news is that it supports USB audio "class" and it's more or less usable, some mixer outputs are available (although they're all called "speaker") and with some trickery the various outputs can be used. The other good news is that Sweex supplies the "official" datasheet from C-media which I think is quite interesting and may very well be useful to resolve the last few issues. These are the actual issues (so far, I didn't yet test it exhaustively): - using 2 channel output the sound is garbled; it looks like the "wrong" 2 channel alt setting is used (output of /proc/asound/card?/stream0 on request), 4/6 channel output is OK though. - some high squeeking noise although C-Media claims the IC to be high-end (or maybe their definition of high-end differs from mine ;-)) - all outputs are labelled "speaker 1" - only two outputs have a mute knob, the others cannot be silenced in any way, only turned down - one of the two outputs with the mute knob doesn't appear to do anything (or maybe it's the sp/dif output?) - the other (with mute) controls the front output - the others (without mute) don't seem to do anything at all Is this a typical case of (a) bug(s) in the usb audio class driver or should I think of "vendor extensions" here? Would any of the developers have some interest in the datasheet of the chip and or any data of this thing? Of course I'd like it very much if the device would be fully supported but I can only ask nicely ;-)

4 9

[alsa-devel] [PATCH] hda: Add support for 92HD73xxx codecs
by Matthew Ranostay 13 Jun '08

13 Jun '08

Added support for new family of IDT codecs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4f44a0..115cee6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -349,7 +349,7 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { static hda_nid_t stac92hd71bxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x14, 0x18, 0x19, 0x1e, + 0x14, 0x18, 0x19, 0x1e, 0x0f }; static hda_nid_t stac927x_pin_nids[14] = { @@ -641,15 +641,21 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect ports 0d and 0f to audio mixer */ - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* connect port 0d to audio mixer */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x02}, /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + /* unmute right and left channels for nodes 0x0a, 0x0d */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {} +}; + +static struct hda_verb stac92hd71bxx_enable_portf[] = { + /* connect port 0f to audio mixer */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* unmute right and left channels for nodes 0x0f */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; @@ -1313,20 +1319,20 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { static unsigned int ref92hd71bxx_pin_configs[10] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, + 0x0181302e, 0x01019020, 0x90a000f0, 0x90a000f0, + 0x01452050, 0x01114010, }; -static unsigned int dell_m4_1_pin_configs[13] = { +static unsigned int dell_m4_1_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, + 0x23a1902e, 0x40f000f0, 0x90a000f0, 0x40f000f0, + 0x4f0000f0, 0x23014250, }; -static unsigned int dell_m4_2_pin_configs[13] = { +static unsigned int dell_m4_2_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, + 0x23a1902e, 0x40f000f0, 0x40f000f0, 0x40f000f0, + 0x044413b0, 0x23014250, }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -2633,7 +2639,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) + if (def_conf && (def_conf == AC_JACK_PORT_NONE)) continue; nid = spec->dmic_nids[i]; @@ -3103,13 +3109,14 @@ static int stac92xx_init(struct hda_codec *codec) 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = get_defcfg_connect(def_conf); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) continue; - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) + if (def_conf && def_conf != AC_JACK_PORT_FIXED) continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26); @@ -3614,12 +3621,39 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, stac92hd71bxx_cfg_tbl); again: + switch (codec->vendor_id) { + case 0x111d76b6: /* 4 Port without Analog Mixer */ + case 0x111d76b7: + case 0x111d76b4: /* 6 Port without Analog Mixer */ + case 0x111d76b5: + spec->mixer = stac92hd71bxx_mixer; + spec->init = stac92hd71bxx_core_init; + break; + case 0x111d7603: /* 6 Port with Analog Mixer */ + case 0x111d7608: + /* no output amps */ + spec->num_pwrs = 0; + /* fallthru */ + default: + spec->mixer = stac92hd71bxx_analog_mixer; + spec->init = stac92hd71bxx_analog_core_init; + } + + if (codec->vendor_id != 0x111d7608) + snd_hda_sequence_write(codec, stac92hd71bxx_enable_portf); + else { + /* skip the VSW on port f */ + spec->num_pins--; + stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); + } + if (spec->board_config < 0) { snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD71BXX, using BIOS defaults\n"); @@ -3634,19 +3668,6 @@ again: stac92xx_set_config_regs(codec); } - switch (codec->vendor_id) { - case 0x111d76b6: /* 4 Port without Analog Mixer */ - case 0x111d76b7: - case 0x111d76b4: /* 6 Port without Analog Mixer */ - case 0x111d76b5: - spec->mixer = stac92hd71bxx_mixer; - spec->init = stac92hd71bxx_core_init; - break; - default: - spec->mixer = stac92hd71bxx_analog_mixer; - spec->init = stac92hd71bxx_analog_core_init; - } - spec->aloopback_mask = 0x20; spec->aloopback_shift = 0; @@ -3666,7 +3687,6 @@ again: spec->num_dmics = STAC92HD71BXX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->multiout.num_dacs = 1; @@ -4306,10 +4326,11 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, + { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7608, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },

2 3

[alsa-devel] [PATCH 1/2] ASoC: Check for exact register match in wm97xx_reset()
by Mark Brown 13 Jun '08

13 Jun '08

To provide added robustness in case an AC97 controller reads back all zeros in error cases check for an exact match when testing to see if resets have brought the codec back. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 28ac66f..4739011 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -589,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba3301..0db9637 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1076,12 +1076,12 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; } -- 1.5.5.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: Check for exact register match in wm97xx_reset()
by Mark Brown 13 Jun '08

13 Jun '08

To provide added robustness in case an AC97 controller reads back all zeros in error cases check for an exact match when testing to see if resets have brought the codec back. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 28ac66f..09e2985 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -589,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == 0x6174) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != 0x6174) goto err; return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba3301..131bb5b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1076,12 +1076,12 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == 0x6174) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != 0x6174) return -EIO; return 0; } -- 1.5.5.4

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[alsa-devel] ALSA 1.0.17rc1 release
by Jaroslav Kysela 13 Jun '08

13 Jun '08

Hi all, ALSA 1.0.17rc1 packages (except alsa-plugins & alsa-oss - no changes) are available for download on http://www.alsa-project.org . Only difference between testing release is a change in alsa-driver for better support of RedHat kernels. http://www.alsa-project.org/main/index.php/Changes_v1.0.16_v1.0.17rc1 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] Names for ALSA controls
by Juergen Beisert 13 Jun '08

13 Jun '08

Hi, when I register these mixer controls: Line Playback Switch Line Playback Route Mono Playback Switch Mono Playback Route Phone Playback Switch Phone Playback Route Headphone Playback Switch Headphone Playback Route PCM Playback Switch PCM Playback Volume Line Bypass Capture Switch Line Bypass Capture Volume ADC Capture Route Running "alsamixer" fails with: ALSA lib simple_none.c:1218:(simple_add1) helem (MIXER,'Headphone Playback Switch',0,0,0) appears twice or more It seems there are restrictions what kind of control name I can use for each control, but what restrictions? Are there better names for my output amplifiers? Each of it I can enable/disable (-> Switch) and select a source (-> Route) independently. Regards, Juergen -- Dipl.-Ing. Juergen Beisert | http://www.pengutronix.de Pengutronix - Linux Solutions for Science and Industry Handelsregister: Amtsgericht Hildesheim, HRA 2686 Vertretung Sued/Muenchen, Germany Phone: +49-8766-939 228 | Fax: +49-5121-206917-9

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[alsa-devel] [PATCH][ASoC v2] Add checks for codec DAI number during match-up
by Timur Tabi 13 Jun '08

13 Jun '08

Update the Codec DAI matching scheme to take the DAI number into account, and not just the name. This is necessary for systems with multiple codecs that use the Codec ID number to differentiate them. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a88e3a0..cc543e6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -801,6 +801,7 @@ codec_dai: /* yes, then check the codec_dai list */ list_for_each_entry(codec_dai, &codec_dai_list, list) { if (!strcmp(config->codec_dai, codec_dai->name) && + (codec_dai->id == config->codec_num) && try_module_get(codec_dai->dev->driver->owner)) { dbg("ASoC %s %s: Match for %s %s\n", soc_card->name, config->name, @@ -1767,6 +1768,7 @@ struct snd_soc_dai *snd_soc_register_codec_dai( dai->playback = template->playback; dai->capture = template->capture; dai->name = template->name; + dai->id = template->id; dai->ac97_control = template->ac97_control; mutex_lock(&client_mutex); list_add(&dai->list, &codec_dai_list); -- 1.5.5

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