Alsa-devel December 2011

alsa-devel@alsa-project.org

88 participants
272 discussions

[alsa-devel] patch for jack plugin
by Fons Adriaensen 09 Jan '12

09 Jan '12

The following patch will make snd_pcm_poll_descriptors_revents() return the correct value when used on a jack plugin device. The current version returns POLLIN regardless of capture or playback. --- pcm_jack_orig.c 2011-12-29 16:42:07.000000000 +0100 +++ pcm_jack.c 2011-12-29 16:51:18.000000000 +0100 @@ -77,17 +77,18 @@ return 0; } -static int snd_pcm_jack_poll_revents(snd_pcm_ioplug_t *io ATTRIBUTE_UNUSED, +static int snd_pcm_jack_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { static char buf[1]; assert(pfds && nfds == 1 && revents); - read(pfds[0].fd, buf, 1); - - *revents = pfds[0].revents; + *revents = pfds[0].revents & ~(POLLIN | POLLOUT); + if (pfds[0].revents & POLLIN) { + *revents |= (io->stream == SND_PCM_STREAM_PLAYBACK) ? POLLOUT : POLLIN; + } return 0; } Ciao, -- FA Vor uns liegt ein weites Tal, die Sonne scheint - ein Glitzerstrahl.

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[alsa-devel] imac27 12,2 (2011) model for patch_cirrus.c
by Jérémy Lal 09 Jan '12

09 Jan '12

Hi, it'd be glad to have patch_cirrus.c recognize model=imac27_122, (i'm adding a model because it's harder to auto-detect it), please find attached patch, tested on linux 3.2-rc7. I don't know anything about pin configs, just that it works with the ones in the patch. Note that i tried with model=auto (no sound) and model=imac27 (front speakers not muted when headphones are plugged in, and surround speakers not properly detected). Regards, Jérémy.

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[alsa-devel] TWL4030 low-latency
by Edgar Berdahl 09 Jan '12

09 Jan '12

Dear Alsa Devel List, I would like to send some feedback on the TWL4030 driver when used on the Beagle Board. We have tested it under Ubuntu and Angstrom, and we find it does not work well at low latencies (e.g. input-to-output in 5ms, 10ms, or 20ms). Perhaps it was never tested extensively at these latencies. I thought I would send you this information in case you would look into it, or at least pass it on to the relevant contact. I can also run further tests if someone informs me what I should try. 1) To get the drivers to run at particularly low latencies (e.g. 20ms), it is necessary to operate the drivers in NON-realtime mode. It would interest me to know whether this means that the realtime functionality of the Beagle Board xM hardware or OS is hampered, or whether the TWL4030 driver is not operating up to specification. (If instead the TWL4030 driver is operating in realtime mode, then Jack stops running when an unrelated low-latency USB-serial device is opened.) 2) Some audio software (e.g. Pure Data, ChucK, etc.) will not connect to the TWL4030 ALSA drivers directly, instead they will only start when jack intermediates between the audio software and the TWL4030 ALSA drivers. 3) In some situations, the TWL4030 drivers appear not to start using the requested settings. For example, when we start jack requesting two periods per buffer, it is apparently getting initialized at eight periods per buffer, see the final lines below: Calling $ jackd -r -dalsa -dhw:0 -p128 -n2 -i2 -o2 -s -S -r on the Beagle under Ubuntu 10.10 and Angstrom demo image results in the following jackd 0.118.0 Copyright 2001-2009 Paul Davis, Stephane Letz, Jack O'Quinn, Torben Hohn and ot. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with System V SHM support. loading driver .. apparent rate = 44100 creating alsa driver ... hw:0|hw:0|128|2|44100|2|2|nomon|swmeter|soft-mode|16bit control device hw:0 configuring for 44100Hz, period = 128 frames (2.9 ms), buffer = 2 periods ALSA: final selected sample format for capture: 16bit little-endian ALSA: use **8** periods for capture ALSA: final selected sample format for playback: 16bit little-endian ALSA: use **8** periods for playback Thanks very much for listening! Best, Edgar Berdahl Lecturer Center for Computer Research in Music and Acoustics (CCRMA) Stanford University http://ccrma.stanford.edu/courses/250a

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[alsa-devel] [PATCH v2 4/4] ASoC: OMAP: ams-delta: drop .set_bias_level callback
by Janusz Krzysztofik 09 Jan '12

09 Jan '12

This functionality has just been implemented in the cx20442 codec driver, no need to keep it here duplicated. Once done, remove the no longer used AMS_DELTA_LATCH2_MODEM_NRESET symbol from the board header file and a call to the regulator_toggle() helper function from the old API wrapper found in the board file. While being at it, simplify the way the modem .pm callback handles the regulator, and drop that no longer used helper function and its related consumer setup elements completely. Depends on patch 3/4 "ARM: OMAP1: ams-delta: update the modem to use regulator API" Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Changes against initial version: * don't move consumer setup elements, now named to indicated their modem related purpose, down the file, * don't track the regulator enavble/disable state, compare new target power state with the old one instead; thanks to Mark Brown for suggesting this simplification, * actually drop all references to AMS_DELTA_LATCH2_MODEM_NRESET. arch/arm/mach-omap1/board-ams-delta.c | 42 +++----------------- arch/arm/plat-omap/include/plat/board-ams-delta.h | 1 - sound/soc/omap/ams-delta.c | 32 ---------------- 3 files changed, 7 insertions(+), 68 deletions(-) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index 3389237..704fee6 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -18,7 +18,6 @@ #include <linux/input.h> #include <linux/interrupt.h> #include <linux/leds.h> -#include <linux/mutex.h> #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/regulator/fixed.h> @@ -291,35 +290,8 @@ static struct platform_device modem_nreset_device = { struct modem_private_data { struct regulator *regulator; - struct { - struct mutex lock; - bool enabled; - } consumer; }; -static int regulator_toggle(struct modem_private_data *priv, bool enable) -{ - int err = 0; - - mutex_lock(&priv->consumer.lock); - if (IS_ERR(priv->regulator)) { - err = PTR_ERR(priv->regulator); - } else if (enable) { - if (!priv->consumer.enabled) { - err = regulator_enable(priv->regulator); - priv->consumer.enabled = true; - } - } else { - if (priv->consumer.enabled) { - err = regulator_disable(priv->regulator); - priv->consumer.enabled = false; - } - } - mutex_unlock(&priv->consumer.lock); - - return err; -} - static struct modem_private_data modem_priv; void ams_delta_latch_write(int base, int ngpio, u16 mask, u16 value) @@ -330,8 +302,6 @@ void ams_delta_latch_write(int base, int ngpio, u16 mask, u16 value) for (; bit < ngpio; bit++, bitpos = bitpos << 1) { if (!(mask & bitpos)) continue; - else if (base + bit == AMS_DELTA_GPIO_PIN_MODEM_NRESET) - regulator_toggle(&modem_priv, (value & bitpos) != 0); else gpio_set_value(base + bit, (value & bitpos) != 0); } @@ -530,10 +500,16 @@ static void modem_pm(struct uart_port *port, unsigned int state, unsigned old) { struct modem_private_data *priv = port->private_data; + if (IS_ERR(priv->regulator)) + return; + if (state == old) return; - regulator_toggle(priv, state == 0); + if (state == 0) + regulator_enable(priv->regulator); + else if (old == 0) + regulator_disable(priv->regulator); } static struct plat_serial8250_port ams_delta_modem_ports[] = { @@ -593,7 +569,6 @@ static int __init late_init(void) gpio_direction_input(AMS_DELTA_GPIO_PIN_MODEM_IRQ); /* Initialize the modem_nreset regulator consumer before use */ - mutex_init(&modem_priv.consumer.lock); modem_priv.regulator = ERR_PTR(-ENODEV); ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, @@ -606,9 +581,6 @@ static int __init late_init(void) /* * Once the modem device is registered, the modem_nreset * regulator can be requested on behalf of that device. - * In addition to the modem .pm callback, that regulator - * is still used via the ams_delta_latch_write() wrapper - * by the ASoC driver until updated. */ modem_priv.regulator = regulator_get(&ams_delta_modem_device.dev, "RESET#"); diff --git a/arch/arm/plat-omap/include/plat/board-ams-delta.h b/arch/arm/plat-omap/include/plat/board-ams-delta.h index 027e79e..ad6f865 100644 --- a/arch/arm/plat-omap/include/plat/board-ams-delta.h +++ b/arch/arm/plat-omap/include/plat/board-ams-delta.h @@ -30,7 +30,6 @@ #define AMD_DELTA_LATCH2_SCARD_RSTIN 0x0400 #define AMD_DELTA_LATCH2_SCARD_CMDVCC 0x0800 -#define AMS_DELTA_LATCH2_MODEM_NRESET 0x1000 #define AMS_DELTA_LATCH2_MODEM_CODEC 0x2000 #define AMS_DELTA_GPIO_PIN_KEYBRD_DATA 0 diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 0462c7e..d213d04 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -426,29 +426,6 @@ static struct snd_soc_ops ams_delta_ops = { }; -/* Board specific codec bias level control */ -static int ams_delta_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - switch (level) { - case SND_SOC_BIAS_ON: - case SND_SOC_BIAS_PREPARE: - case SND_SOC_BIAS_STANDBY: - if (card->dapm.bias_level == SND_SOC_BIAS_OFF) - ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, - AMS_DELTA_LATCH2_MODEM_NRESET); - break; - case SND_SOC_BIAS_OFF: - if (card->dapm.bias_level != SND_SOC_BIAS_OFF) - ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, - 0); - } - card->dapm.bias_level = level; - - return 0; -} - /* Digital mute implemented using modem/CPU multiplexer. * Shares hardware with codec config pulse generation */ static bool ams_delta_muted = 1; @@ -512,9 +489,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) ams_delta_ops.shutdown = ams_delta_shutdown; } - /* Set codec bias level */ - ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); - /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ ret = snd_soc_jack_new(rtd->codec, "hook_switch", @@ -597,7 +571,6 @@ static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", .dai_link = &ams_delta_dai_link, .num_links = 1, - .set_bias_level = ams_delta_set_bias_level, }; /* Module init/exit */ @@ -646,11 +619,6 @@ static void __exit ams_delta_module_exit(void) ARRAY_SIZE(ams_delta_hook_switch_gpios), ams_delta_hook_switch_gpios); - /* Keep modem power on */ - ams_delta_set_bias_level(&ams_delta_audio_card, - &ams_delta_audio_card.rtd[0].codec->dapm, - SND_SOC_BIAS_STANDBY); - platform_device_unregister(cx20442_platform_device); platform_device_unregister(ams_delta_audio_platform_device); } -- 1.7.3.4

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[alsa-devel] [PATCH] hda_intel: Add Oaktrail identifiers
by Alan Cox 08 Jan '12

08 Jan '12

From: Li Peng <pengi.li(a)intel.com> Oaktrail has 0x8086, 0x080a - AZX_DRIVER_SCH Taken from the Meego patches for Oaktrail Signed-off-by: Li Peng <peng.li(a)intel.com> Signed-off-by: Alan Cox <alan(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c2f79e6..b339eb5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2972,6 +2972,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */ + { PCI_DEVICE(0x8086, 0x080a), + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |

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[alsa-devel] [PATCH 1/3] ASoC: Use dai_fmt in corgi machine driver
by Axel Lin 02 Jan '12

02 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/corgi.c | 14 ++------------ 1 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 5e5004a..5ff6dac 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -142,18 +142,6 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk, SND_SOC_CLOCK_IN); @@ -311,6 +299,8 @@ static struct snd_soc_dai_link corgi_dai = { .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .init = corgi_wm8731_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &corgi_ops, }; -- 1.7.5.4

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[alsa-devel] [PATCH v2 2/4] ASoC: cx20442: add bias control over a platform provided regulator
by Janusz Krzysztofik 02 Jan '12

02 Jan '12

Now that a regulator device for controlling the codec chip reset state over a platform agnostic regulator API is available on the only board using this driver so far, extend the driver with a bias control function which will request virtual power to the codec chip from that virtual regulator, and will supersede the present implementation existing at the sound card level. Thanks to the regulator sharing mechanism, both the old (the sound card) and the new (the codec) implementations should coexist smoothly until the sound card file is updated. For this to work as expected, update the sound card .set_bias_level callback to not touch codec->dapm.bias_level. While extending the cx20442 structure, drop unused control_type member. Created against linxu-3.2-rc6, tested on top of patch 1/4 "ARM: OMAP1: ams-delta: set up a regulator over the modem reset GPIO pin". Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Changes against initial version: * don't track the regulator enable/disable state, compare new bias level against the old one instead; thanks to Mark Brown for suggesting this change, * update the Amstrad Delta sound card file to not touch the codec bias level for the above change to work as expected, * display a warning if getting the regulator fails. sound/soc/codecs/cx20442.c | 48 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/omap/ams-delta.c | 8 ++---- 2 files changed, 49 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index bc7067d..941c527 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -16,6 +16,7 @@ #include <linux/tty.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/initval.h> @@ -25,8 +26,8 @@ struct cx20442_priv { - enum snd_soc_control_type control_type; void *control_data; + struct regulator *por; }; #define CX20442_PM 0x0 @@ -324,6 +325,38 @@ static struct snd_soc_dai_driver cx20442_dai = { }, }; +static int cx20442_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); + int err = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level != SND_SOC_BIAS_STANDBY) + break; + if (IS_ERR(cx20442->por)) + err = PTR_ERR(cx20442->por); + else + err = regulator_enable(cx20442->por); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level != SND_SOC_BIAS_PREPARE) + break; + if (IS_ERR(cx20442->por)) + err = PTR_ERR(cx20442->por); + else + err = regulator_disable(cx20442->por); + break; + default: + break; + } + if (!err) + codec->dapm.bias_level = level; + + return err; +} + static int cx20442_codec_probe(struct snd_soc_codec *codec) { struct cx20442_priv *cx20442; @@ -331,9 +364,13 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec) cx20442 = kzalloc(sizeof(struct cx20442_priv), GFP_KERNEL); if (cx20442 == NULL) return -ENOMEM; - snd_soc_codec_set_drvdata(codec, cx20442); + cx20442->por = regulator_get(codec->dev, "POR"); + if (IS_ERR(cx20442->por)) + dev_warn(codec->dev, "failed to get the regulator"); cx20442->control_data = NULL; + + snd_soc_codec_set_drvdata(codec, cx20442); codec->hw_write = NULL; codec->card->pop_time = 0; @@ -350,6 +387,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) tty_hangup(tty); } + if (!IS_ERR(cx20442->por)) { + /* should be already in STANDBY, hence disabled */ + regulator_put(cx20442->por); + } + + snd_soc_codec_set_drvdata(codec, NULL); kfree(cx20442); return 0; } @@ -359,6 +402,7 @@ static const u8 cx20442_reg; static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, + .set_bias_level = cx20442_set_bias_level, .reg_cache_default = &cx20442_reg, .reg_cache_size = 1, .reg_word_size = sizeof(u8), diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index be81bc7..0462c7e 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -431,22 +431,20 @@ static int ams_delta_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_codec *codec = card->rtd->codec; - switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + if (card->dapm.bias_level == SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, AMS_DELTA_LATCH2_MODEM_NRESET); break; case SND_SOC_BIAS_OFF: - if (codec->dapm.bias_level != SND_SOC_BIAS_OFF) + if (card->dapm.bias_level != SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, 0); } - codec->dapm.bias_level = level; + card->dapm.bias_level = level; return 0; } -- 1.7.3.4

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[alsa-devel] [PATCH] ASoC: Fix return value of wm8903_gpio_direction_in() and wm8903_gpio_direction_out()
by Axel Lin 02 Jan '12

02 Jan '12

We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8903.c | 18 ++++++++++++++---- 1 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d88b727..c91fb2f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1777,13 +1777,18 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; + int ret; mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK; val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | WM8903_GP1_DIR; - return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); + if (ret < 0) + return ret; + + return 0; } static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) @@ -1803,13 +1808,18 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; + int ret; mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK | WM8903_GP1_LVL_MASK; val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | (value << WM8903_GP2_LVL_SHIFT); - return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); + if (ret < 0) + return ret; + + return 0; } static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) -- 1.7.5.4

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[alsa-devel] [PATCH 1/4] ASoC: Convert saarb to table based DAPM init
by Axel Lin 02 Jan '12

02 Jan '12

Also remove a unused ret variable to silence the build warning. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/saarb.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index b2be225..c34146b 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -51,7 +51,7 @@ static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { }; /* saarb machine audio map */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route saarb_audio_map[] = { {"Headset Stereophone", NULL, "HS1"}, {"Headset Stereophone", NULL, "HS2"}, @@ -121,17 +121,17 @@ static struct snd_soc_card snd_soc_card_saarb = { .owner = THIS_MODULE, .dai_link = saarb_dai, .num_links = ARRAY_SIZE(saarb_dai), + + .dapm_widgets = saarb_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets), + .dapm_routes = saarb_audio_map, + .num_dapm_routes = ARRAY_SIZE(saarb_audio_map), }; static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - snd_soc_dapm_new_controls(dapm, saarb_dapm_widgets, - ARRAY_SIZE(saarb_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); -- 1.7.5.4

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[alsa-devel] [PATCH 1/4] ASoC: Use dai_fmt in saarb machine driver
by Axel Lin 02 Jan '12

02 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/saarb.c | 11 ++--------- 1 files changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index 2e21712..b2be225 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -92,15 +92,6 @@ static int saarb_i2s_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); return ret; @@ -119,6 +110,8 @@ static struct snd_soc_dai_link saarb_dai[] = { .platform_name = "pxa-pcm-audio", .codec_name = "88pm860x-codec", .init = saarb_pm860x_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &saarb_i2s_ops, }, }; -- 1.7.5.4

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Alsa-devel December 2011