From 92429069d0fc9f52d436c9067c5b5c392e3f8876 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Thu, 19 Mar 2009 09:32:01 +0100 Subject: [PATCH 01/13] ASoC: pxa-ssp: Use 16-bit DMA for magician stereo Now magician and similar boards can use network mode with only one active slot to explicitly set 16 bit frame width, even for S16_LE stereo sound. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index b0bf40973d5b..c7c1996a5447 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -627,12 +627,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); + int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; /* select correct DMA params */ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) dma = 1; /* capture DMA offset is 1,3 */ - if (chn == 2) - dma += 2; /* stereo DMA offset is 2, mono is 0 */ + /* Network mode with one active slot (ttsa == 1) can be used + * to force 16-bit frame width on the wire (for S16_LE), even + * with two channels. Use 16-bit DMA transfers for this case. + */ + if (((chn == 2) && (ttsa != 1)) || (width == 32)) + dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */ + cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); @@ -712,7 +718,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ - if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) { + if ((sscr0 & SSCR0_MOD) && !ttsa) { dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); return -EINVAL; } From 7377226c344a7295a7573dce400ce9ddd42f0ca4 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Thu, 19 Mar 2009 09:34:46 +0100 Subject: [PATCH 02/13] ASoC: Add Magician machine support HTC Magician has a Philips UDA1380 codec connected via SSP1 (playback) and I2S (capture). There is a flip-flop between the SSP frame clock output and the codec's word select input pin. To make the codec see proper I2S input, the SSP has to send two frames per sample. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 10 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/magician.c | 560 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 572 insertions(+) create mode 100644 sound/soc/pxa/magician.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 5998ab366e83..ad8a10fe6298 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -116,6 +116,16 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_PXA2XX_SOC_MAGICIAN + tristate "SoC Audio support for HTC Magician" + depends on SND_PXA2XX_SOC && MACH_MAGICIAN + select SND_PXA2XX_SOC_I2S + select SND_PXA_SOC_SSP + select SND_SOC_UDA1380 + help + Say Y if you want to add support for SoC audio on the + HTC Magician. + config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 8ed881c5e5cc..4b90c3ccae45 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -20,6 +20,7 @@ snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o +snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o @@ -31,5 +32,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c new file mode 100644 index 000000000000..f7c4544f7859 --- /dev/null +++ b/sound/soc/pxa/magician.c @@ -0,0 +1,560 @@ +/* + * SoC audio for HTC Magician + * + * Copyright (c) 2006 Philipp Zabel + * + * based on spitz.c, + * Authors: Liam Girdwood + * Richard Purdie + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include "../codecs/uda1380.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-i2s.h" +#include "pxa-ssp.h" + +#define MAGICIAN_MIC 0 +#define MAGICIAN_MIC_EXT 1 + +static int magician_hp_switch; +static int magician_spk_switch = 1; +static int magician_in_sel = MAGICIAN_MIC; + +static void magician_ext_control(struct snd_soc_codec *codec) +{ + if (magician_spk_switch) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + if (magician_hp_switch) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + switch (magician_in_sel) { + case MAGICIAN_MIC: + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); + break; + case MAGICIAN_MIC_EXT: + snd_soc_dapm_disable_pin(codec, "Call Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + break; + } + + snd_soc_dapm_sync(codec); +} + +static int magician_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->card->codec; + + /* check the jack status at stream startup */ + magician_ext_control(codec); + + return 0; +} + +/* + * Magician uses SSP port for playback. + */ +static int magician_playback_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; + unsigned int acps, acds, width, rate; + unsigned int div4 = PXA_SSP_CLK_SCDB_4; + int ret = 0; + + rate = params_rate(params); + width = snd_pcm_format_physical_width(params_format(params)); + + /* + * rate = SSPSCLK / (2 * width(16 or 32)) + * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) + */ + switch (params_rate(params)) { + case 8000: + /* off by a factor of 2: bug in the PXA27x audio clock? */ + acps = 32842000; + switch (width) { + case 16: + /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_16; + break; + case 32: + /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_8; + } + break; + case 11025: + acps = 5622000; + switch (width) { + case 16: + /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_4; + break; + case 32: + /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + } + break; + case 22050: + acps = 5622000; + switch (width) { + case 16: + /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 44100: + acps = 5622000; + switch (width) { + case 16: + /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 48000: + acps = 12235000; + switch (width) { + case 16: + /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 96000: + acps = 12235000; + switch (width) { + case 16: + /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + break; + case 32: + /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + div4 = PXA_SSP_CLK_SCDB_1; + break; + } + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | + 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_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + if (ret < 0) + return ret; + + /* set audio clock as clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set the SSP audio system clock ACDS divider */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + PXA_SSP_AUDIO_DIV_ACDS, acds); + if (ret < 0) + return ret; + + /* set the SSP audio system clock SCDB divider4 */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + PXA_SSP_AUDIO_DIV_SCDB, div4); + if (ret < 0) + return ret; + + /* set SSP audio pll clock */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps); + if (ret < 0) + return ret; + + return 0; +} + +/* + * Magician uses I2S for capture. + */ +static int magician_capture_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 ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_MSB | 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_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the I2S system clock as output */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops magician_capture_ops = { + .startup = magician_startup, + .hw_params = magician_capture_hw_params, +}; + +static struct snd_soc_ops magician_playback_ops = { + .startup = magician_startup, + .hw_params = magician_playback_hw_params, +}; + +static int magician_get_hp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_hp_switch; + return 0; +} + +static int magician_set_hp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (magician_hp_switch == ucontrol->value.integer.value[0]) + return 0; + + magician_hp_switch = ucontrol->value.integer.value[0]; + magician_ext_control(codec); + return 1; +} + +static int magician_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_spk_switch; + return 0; +} + +static int magician_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (magician_spk_switch == ucontrol->value.integer.value[0]) + return 0; + + magician_spk_switch = ucontrol->value.integer.value[0]; + magician_ext_control(codec); + return 1; +} + +static int magician_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_in_sel; + return 0; +} + +static int magician_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (magician_in_sel == ucontrol->value.integer.value[0]) + return 0; + + magician_in_sel = ucontrol->value.integer.value[0]; + + switch (magician_in_sel) { + case MAGICIAN_MIC: + gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1); + break; + case MAGICIAN_MIC_EXT: + gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0); + } + + return 1; +} + +static int magician_spk_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int magician_hp_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int magician_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +/* magician machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power), + SND_SOC_DAPM_SPK("Speaker", magician_spk_power), + SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias), + SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias), +}; + +/* magician machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + + /* Headphone connected to VOUTL, VOUTR */ + {"Headphone Jack", NULL, "VOUTL"}, + {"Headphone Jack", NULL, "VOUTR"}, + + /* Speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* Mics are connected to VINM */ + {"VINM", NULL, "Headset Mic"}, + {"VINM", NULL, "Call Mic"}, +}; + +static const char *input_select[] = {"Call Mic", "Headset Mic"}; +static const struct soc_enum magician_in_sel_enum = + SOC_ENUM_SINGLE_EXT(2, input_select); + +static const struct snd_kcontrol_new uda1380_magician_controls[] = { + SOC_SINGLE_BOOL_EXT("Headphone Switch", + (unsigned long)&magician_hp_switch, + magician_get_hp, magician_set_hp), + SOC_SINGLE_BOOL_EXT("Speaker Switch", + (unsigned long)&magician_spk_switch, + magician_get_spk, magician_set_spk), + SOC_ENUM_EXT("Input Select", magician_in_sel_enum, + magician_get_input, magician_set_input), +}; + +/* + * Logic for a uda1380 as connected on a HTC Magician + */ +static int magician_uda1380_init(struct snd_soc_codec *codec) +{ + int err; + + /* NC codec pins */ + snd_soc_dapm_nc_pin(codec, "VOUTLHP"); + snd_soc_dapm_nc_pin(codec, "VOUTRHP"); + + /* FIXME: is anything connected here? */ + snd_soc_dapm_nc_pin(codec, "VINL"); + snd_soc_dapm_nc_pin(codec, "VINR"); + + /* Add magician specific controls */ + err = snd_soc_add_controls(codec, uda1380_magician_controls, + ARRAY_SIZE(uda1380_magician_controls)); + if (err < 0) + return err; + + /* Add magician specific widgets */ + snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + + /* Set up magician specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + return 0; +} + +/* magician digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link magician_dai[] = { +{ + .name = "uda1380", + .stream_name = "UDA1380 Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK], + .init = magician_uda1380_init, + .ops = &magician_playback_ops, +}, +{ + .name = "uda1380", + .stream_name = "UDA1380 Capture", + .cpu_dai = &pxa_i2s_dai, + .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE], + .ops = &magician_capture_ops, +} +}; + +/* magician audio machine driver */ +static struct snd_soc_card snd_soc_card_magician = { + .name = "Magician", + .dai_link = magician_dai, + .num_links = ARRAY_SIZE(magician_dai), + .platform = &pxa2xx_soc_platform, +}; + +/* magician audio private data */ +static struct uda1380_setup_data magician_uda1380_setup = { + .i2c_address = 0x18, + .dac_clk = UDA1380_DAC_CLK_WSPLL, +}; + +/* magician audio subsystem */ +static struct snd_soc_device magician_snd_devdata = { + .card = &snd_soc_card_magician, + .codec_dev = &soc_codec_dev_uda1380, + .codec_data = &magician_uda1380_setup, +}; + +static struct platform_device *magician_snd_device; + +static int __init magician_init(void) +{ + int ret; + + if (!machine_is_magician()) + return -ENODEV; + + ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER"); + if (ret) + goto err_request_power; + ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET"); + if (ret) + goto err_request_reset; + ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER"); + if (ret) + goto err_request_spk; + ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER"); + if (ret) + goto err_request_ep; + ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER"); + if (ret) + goto err_request_mic; + ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0"); + if (ret) + goto err_request_in_sel0; + ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1"); + if (ret) + goto err_request_in_sel1; + + gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1); + gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0); + + /* we may need to have the clock running here - pH5 */ + gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1); + udelay(5); + gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0); + + magician_snd_device = platform_device_alloc("soc-audio", -1); + if (!magician_snd_device) { + ret = -ENOMEM; + goto err_pdev; + } + + platform_set_drvdata(magician_snd_device, &magician_snd_devdata); + magician_snd_devdata.dev = &magician_snd_device->dev; + ret = platform_device_add(magician_snd_device); + if (ret) { + platform_device_put(magician_snd_device); + goto err_pdev; + } + + return 0; + +err_pdev: + gpio_free(EGPIO_MAGICIAN_IN_SEL1); +err_request_in_sel1: + gpio_free(EGPIO_MAGICIAN_IN_SEL0); +err_request_in_sel0: + gpio_free(EGPIO_MAGICIAN_MIC_POWER); +err_request_mic: + gpio_free(EGPIO_MAGICIAN_EP_POWER); +err_request_ep: + gpio_free(EGPIO_MAGICIAN_SPK_POWER); +err_request_spk: + gpio_free(EGPIO_MAGICIAN_CODEC_RESET); +err_request_reset: + gpio_free(EGPIO_MAGICIAN_CODEC_POWER); +err_request_power: + return ret; +} + +static void __exit magician_exit(void) +{ + platform_device_unregister(magician_snd_device); + + gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0); + + gpio_free(EGPIO_MAGICIAN_IN_SEL1); + gpio_free(EGPIO_MAGICIAN_IN_SEL0); + gpio_free(EGPIO_MAGICIAN_MIC_POWER); + gpio_free(EGPIO_MAGICIAN_EP_POWER); + gpio_free(EGPIO_MAGICIAN_SPK_POWER); + gpio_free(EGPIO_MAGICIAN_CODEC_RESET); + gpio_free(EGPIO_MAGICIAN_CODEC_POWER); +} + +module_init(magician_init); +module_exit(magician_exit); + +MODULE_AUTHOR("Philipp Zabel"); +MODULE_DESCRIPTION("ALSA SoC Magician"); +MODULE_LICENSE("GPL"); From a4d11fe50c238a7da5225d1399314c3505cbd792 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Wed, 25 Mar 2009 18:20:37 -0500 Subject: [PATCH 03/13] ASoC: remove trigger delay in Freescale MPC8610 sound driver Remove the delay from the trigger function in the Freescale MPC8610 sound driver when capture is started. This delay was used to ensure that the DMA controller was active when ALSA call the .pointer function to request a DMA transfer status. A better approach is for the .pointer function to detect that DMA has not started, and return zero instead. This change eliminates the need for the delay. Also add some related code to check for a DMA programming error, and report XRUN if it occurs. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 17 +++++++++++++++++ sound/soc/fsl/fsl_ssi.c | 20 ++------------------ 2 files changed, 19 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index b3eb8570cd7b..2c4892c853cf 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -697,6 +697,23 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) else position = in_be32(&dma_channel->dar); + /* + * When capture is started, the SSI immediately starts to fill its FIFO. + * This means that the DMA controller is not started until the FIFO is + * full. However, ALSA calls this function before that happens, when + * MR.DAR is still zero. In this case, just return zero to indicate + * that nothing has been received yet. + */ + if (!position) + return 0; + + if ((position < dma_private->dma_buf_phys) || + (position > dma_private->dma_buf_end)) { + dev_err(substream->pcm->card->dev, + "dma pointer is out of range, halting stream\n"); + return SNDRV_PCM_POS_XRUN; + } + frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); /* diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 169bca295b78..72823a2b33d6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -466,28 +466,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); - } else { - long timeout = jiffies + 10; - + else setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); - - /* Wait until the SSI has filled its FIFO. Without this - * delay, ALSA complains about overruns. When the FIFO - * is full, the DMA controller initiates its first - * transfer. Until then, however, the DMA's DAR - * register is zero, which translates to an - * out-of-bounds pointer. This makes ALSA think an - * overrun has occurred. - */ - while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) && - (jiffies < timeout)); - if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0)) - return -EIO; - } break; case SNDRV_PCM_TRIGGER_STOP: From 057de50c0d34b4ef7e15b7a8442a36a396d99c00 Mon Sep 17 00:00:00 2001 From: Luotao Fu Date: Thu, 26 Mar 2009 13:18:03 +0100 Subject: [PATCH 04/13] pxa2xx-ac97: fix displaying GSR after reset timeout the variable gsr_bit is set in isr. It is however set to 0 and interrupts are disabled prior to reset. Hence it doesn't make a lot of sense to show the content of gsr_bit in case of a reset timeout. Signed-off-by: Luotao Fu Signed-off-by: Mark Brown --- sound/arm/pxa2xx-ac97-lib.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 2e6355f4cbb9..71bef45e9d31 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -239,6 +239,8 @@ static inline void pxa_ac97_cold_pxa3xx(void) bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) { + unsigned long gsr; + #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) pxa_ac97_warm_pxa25x(); @@ -255,10 +257,10 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) else #endif BUG(); - - if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { + gsr = GSR | gsr_bits; + if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + __func__, gsr); return false; } @@ -269,6 +271,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) { + unsigned long gsr; + #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) pxa_ac97_cold_pxa25x(); @@ -286,9 +290,10 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) #endif BUG(); - if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { + gsr = GSR | gsr_bits; + if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + __func__, gsr); return false; } From d5a908b27adfd7e67b5ab98f674892badcca19c6 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 26 Mar 2009 11:42:38 -0500 Subject: [PATCH 05/13] ASoC: trim SSI sysfs statistics in Freescale MPC8610 sound drivers Optimize the display of SSI statistics in the Freescale MPC8610 sound driver to display the status count only of the interrupts that were actually enabled. Previously, it would display the counts of all SISR status bits, even those that were not enabled. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 77 ++++++++++++++++++++++++----------------- 1 file changed, 46 insertions(+), 31 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 72823a2b33d6..3711d8454d96 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -60,6 +60,13 @@ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE) #endif +/* SIER bitflag of interrupts to enable */ +#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ + CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ + CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ + CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ + CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) + /** * fsl_ssi_private: per-SSI private data * @@ -140,7 +147,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) were interrupted for. We mask it with the Interrupt Enable register so that we only check for events that we're interested in. */ - sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier); + sisr = in_be32(&ssi->sisr) & SIER_FLAGS; if (sisr & CCSR_SSI_SISR_RFRC) { ssi_private->stats.rfrc++; @@ -324,12 +331,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ /* 4. Enable the interrupts and DMA requests */ - out_be32(&ssi->sier, - CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | - CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | - CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | - CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN); + out_be32(&ssi->sier, SIER_FLAGS); /* * Set the watermark for transmit FIFI 0 and receive FIFO 0. We @@ -590,39 +592,52 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .ops = &fsl_ssi_dai_ops, }; +/* Show the statistics of a flag only if its interrupt is enabled. The + * compiler will optimze this code to a no-op if the interrupt is not + * enabled. + */ +#define SIER_SHOW(flag, name) \ + do { \ + if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \ + length += sprintf(buf + length, #name "=%u\n", \ + ssi_private->stats.name); \ + } while (0) + + /** * fsl_sysfs_ssi_show: display SSI statistics * - * Display the statistics for the current SSI device. + * Display the statistics for the current SSI device. To avoid confusion, + * we only show those counts that are enabled. */ static ssize_t fsl_sysfs_ssi_show(struct device *dev, struct device_attribute *attr, char *buf) { struct fsl_ssi_private *ssi_private = - container_of(attr, struct fsl_ssi_private, dev_attr); - ssize_t length; + container_of(attr, struct fsl_ssi_private, dev_attr); + ssize_t length = 0; - length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc); - length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc); - length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau); - length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu); - length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt); - length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1); - length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0); - length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1); - length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0); - length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1); - length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0); - length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1); - length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0); - length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs); - length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs); - length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls); - length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls); - length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1); - length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0); - length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1); - length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0); + SIER_SHOW(RFRC_EN, rfrc); + SIER_SHOW(TFRC_EN, tfrc); + SIER_SHOW(CMDAU_EN, cmdau); + SIER_SHOW(CMDDU_EN, cmddu); + SIER_SHOW(RXT_EN, rxt); + SIER_SHOW(RDR1_EN, rdr1); + SIER_SHOW(RDR0_EN, rdr0); + SIER_SHOW(TDE1_EN, tde1); + SIER_SHOW(TDE0_EN, tde0); + SIER_SHOW(ROE1_EN, roe1); + SIER_SHOW(ROE0_EN, roe0); + SIER_SHOW(TUE1_EN, tue1); + SIER_SHOW(TUE0_EN, tue0); + SIER_SHOW(TFS_EN, tfs); + SIER_SHOW(RFS_EN, rfs); + SIER_SHOW(TLS_EN, tls); + SIER_SHOW(RLS_EN, rls); + SIER_SHOW(RFF1_EN, rff1); + SIER_SHOW(RFF0_EN, rff0); + SIER_SHOW(TFE1_EN, tfe1); + SIER_SHOW(TFE0_EN, tfe0); return length; } From 31ad0f31c3a45ba489203eef7e71d3215005afbc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 27 Mar 2009 10:39:07 +0200 Subject: [PATCH 06/13] ASoC: TWL4030: 96KHz playback support TWL4030 supports 96KHz sample playback, but only playback. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 97738e2ece04..b07d8d68a939 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1395,7 +1395,7 @@ struct snd_soc_dai twl4030_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = TWL4030_RATES, + .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000, .formats = TWL4030_FORMATS,}, .capture = { .stream_name = "Capture", From 7220b9f4bd4fad41f6f7299fe74c2c38ec85d793 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 27 Mar 2009 10:39:08 +0200 Subject: [PATCH 07/13] ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 54 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 54 insertions(+) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b07d8d68a939..4199498a8918 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -122,6 +122,9 @@ struct twl4030_priv { unsigned int bypass_state; unsigned int codec_powered; unsigned int codec_muted; + + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; }; /* @@ -1217,6 +1220,50 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, return 0; } +static int twl4030_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct twl4030_priv *twl4030 = codec->private_data; + + /* If we already have a playback or capture going then constrain + * this substream to match it. + */ + if (twl4030->master_substream) { + struct snd_pcm_runtime *master_runtime; + master_runtime = twl4030->master_substream->runtime; + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + twl4030->slave_substream = substream; + } else + twl4030->master_substream = substream; + + return 0; +} + +static void twl4030_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct twl4030_priv *twl4030 = codec->private_data; + + if (twl4030->master_substream == substream) + twl4030->master_substream = twl4030->slave_substream; + + twl4030->slave_substream = NULL; +} + static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1224,8 +1271,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; u8 mode, old_mode, format, old_format; + if (substream == twl4030->slave_substream) + /* Ignoring hw_params for slave substream */ + return 0; + /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; @@ -1384,6 +1436,8 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) static struct snd_soc_dai_ops twl4030_dai_ops = { + .startup = twl4030_startup, + .shutdown = twl4030_shutdown, .hw_params = twl4030_hw_params, .set_sysclk = twl4030_set_dai_sysclk, .set_fmt = twl4030_set_dai_fmt, From 6984992bf0520a07b931124d33f46b46437f6e1c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 27 Mar 2009 15:32:01 +0200 Subject: [PATCH 08/13] ASoC: OMAP: Set minimum buffer size constraint for McBSP2 in OMAP3 McBSP2 in OMAP3 has 1 ksample (1k x 32 bit) internal FIFO. During initial playback startup, this FIFO is keeping the DMA request active until the FIFO is full. So now if ALSA buffer size is smaller, DMA is looping around it while filling up the HW FIFO, generating burst of interrupts as well and SW doesn't have any change to fill enough data. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d6882be33452..9c09b94f0cf8 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -146,6 +146,17 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; + if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) { + /* + * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer. + * Set constraint for minimum buffer size to the same than FIFO + * size in order to avoid underruns in playback startup because + * HW is keeping the DMA request active until FIFO is filled. + */ + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX); + } + if (!cpu_dai->active) err = omap_mcbsp_request(mcbsp_data->bus_id); From a7808331f1ea6c7f89a14d1d94eafc62615b997b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 27 Mar 2009 17:14:52 +0000 Subject: [PATCH 09/13] ASoC: Add some documentation for the ASoC jack API A brief overview of how the components of the API fit together. Signed-off-by: Mark Brown --- Documentation/sound/alsa/soc/jack.txt | 71 +++++++++++++++++++++++++++ 1 file changed, 71 insertions(+) create mode 100644 Documentation/sound/alsa/soc/jack.txt diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/alsa/soc/jack.txt new file mode 100644 index 000000000000..fcf82a417293 --- /dev/null +++ b/Documentation/sound/alsa/soc/jack.txt @@ -0,0 +1,71 @@ +ASoC jack detection +=================== + +ALSA has a standard API for representing physical jacks to user space, +the kernel side of which can be seen in include/sound/jack.h. ASoC +provides a version of this API adding two additional features: + + - It allows more than one jack detection method to work together on one + user visible jack. In embedded systems it is common for multiple + to be present on a single jack but handled by separate bits of + hardware. + + - Integration with DAPM, allowing DAPM endpoints to be updated + automatically based on the detected jack status (eg, turning off the + headphone outputs if no headphones are present). + +This is done by splitting the jacks up into three things working +together: the jack itself represented by a struct snd_soc_jack, sets of +snd_soc_jack_pins representing DAPM endpoints to update and blocks of +code providing jack reporting mechanisms. + +For example, a system may have a stereo headset jack with two reporting +mechanisms, one for the headphone and one for the microphone. Some +systems won't be able to use their speaker output while a headphone is +connected and so will want to make sure to update both speaker and +headphone when the headphone jack status changes. + +The jack - struct snd_soc_jack +============================== + +This represents a physical jack on the system and is what is visible to +user space. The jack itself is completely passive, it is set up by the +machine driver and updated by jack detection methods. + +Jacks are created by the machine driver calling snd_soc_jack_new(). + +snd_soc_jack_pin +================ + +These represent a DAPM pin to update depending on some of the status +bits supported by the jack. Each snd_soc_jack has zero or more of these +which are updated automatically. They are created by the machine driver +and associated with the jack using snd_soc_jack_add_pins(). The status +of the endpoint may configured to be the opposite of the jack status if +required (eg, enabling a built in microphone if a microphone is not +connected via a jack). + +Jack detection methods +====================== + +Actual jack detection is done by code which is able to monitor some +input to the system and update a jack by calling snd_soc_jack_report(), +specifying a subset of bits to update. The jack detection code should +be set up by the machine driver, taking configuration for the jack to +update and the set of things to report when the jack is connected. + +Often this is done based on the status of a GPIO - a handler for this is +provided by the snd_soc_jack_add_gpio() function. Other methods are +also available, for example integrated into CODECs. One example of +CODEC integrated jack detection can be see in the WM8350 driver. + +Each jack may have multiple reporting mechanisms, though it will need at +least one to be useful. + +Machine drivers +=============== + +These are all hooked together by the machine driver depending on the +system hardware. The machine driver will set up the snd_soc_jack and +the list of pins to update then set up one or more jack detection +mechanisms to update that jack based on their current status. From 64ab9baa00fa99070da993f00173c35a8e99abfa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Mar 2009 11:27:03 +0100 Subject: [PATCH 10/13] ASoC: Don't defer resume work for AC97 codecs AC97 devices may have other drivers hanging off them directly so need to have resumed when the resume function returns meaning that we can't defer the resume - complete it immediately for them. Non-AC97 devices should not have other drivers hanging directly off the ASoC devices. We only really need the deferral for non-AC97 devices - it's there since some I2C buses are very slow and non-AC97 codecs often have large numbers of registers to restore and require delays to bring the codec up cleanly leading to a substantial impact on overall resume time. Reported-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6e710f705a74..6c62d4a54cdf 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -767,11 +767,21 @@ static int soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; - dev_dbg(socdev->dev, "scheduling resume work\n"); - - if (!schedule_work(&card->deferred_resume_work)) - dev_err(socdev->dev, "resume work item may be lost\n"); + /* AC97 devices might have other drivers hanging off them so + * need to resume immediately. Other drivers don't have that + * problem and may take a substantial amount of time to resume + * due to I/O costs and anti-pop so handle them out of line. + */ + if (cpu_dai->ac97_control) { + dev_dbg(socdev->dev, "Resuming AC97 immediately\n"); + soc_resume_deferred(&card->deferred_resume_work); + } else { + dev_dbg(socdev->dev, "Scheduling resume work\n"); + if (!schedule_work(&card->deferred_resume_work)) + dev_err(socdev->dev, "resume work item may be lost\n"); + } return 0; } From 4ac5c61f0fc9b01946911a52d827f67947ab01a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Apr 2009 19:35:01 +0100 Subject: [PATCH 11/13] ASoC: Set parent for AC97 devices we register Ensure that any AC97 devices that bind to the CODEC are below the ASoC device in the device tree so the suspend and resume code can figure out what order to handle them in. Reported-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6c62d4a54cdf..99712f652d0d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -98,7 +98,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) int err; codec->ac97->dev.bus = &ac97_bus_type; - codec->ac97->dev.parent = NULL; + codec->ac97->dev.parent = codec->card->dev; codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", From 0a11b16853b642a26eb248ac4db422e6dfa04ae5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Apr 2009 15:49:41 +0100 Subject: [PATCH 12/13] ASoC: Implement suspend and resume operations for WM9705 Without this the WM9705 driver fails badly when resuming. Tested-by: Russell King Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 3265817c5c26..6e23a81dba78 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -317,6 +317,41 @@ static int wm9705_reset(struct snd_soc_codec *codec) return -EIO; } +#ifdef CONFIG_PM +static int wm9705_soc_suspend(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff); + + return 0; +} + +static int wm9705_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i, ret; + u16 *cache = codec->reg_cache; + + ret = wm9705_reset(codec); + if (ret < 0) { + printk(KERN_ERR "could not reset AC97 codec\n"); + return ret; + } + + for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { + soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + } + + return 0; +} +#else +#define wm9705_soc_suspend NULL +#define wm9705_soc_resume NULL +#endif + static int wm9705_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -407,6 +442,8 @@ static int wm9705_soc_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_wm9705 = { .probe = wm9705_soc_probe, .remove = wm9705_soc_remove, + .suspend = wm9705_soc_suspend, + .resume = wm9705_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705); From 103f211d0be2bed75b5739de62a10415ef0bbc25 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Apr 2009 14:39:05 +0300 Subject: [PATCH 13/13] ASoC: TWL4030: Add actual support for 96KHz playback support Adds the needed code to be able to use 96KHz playback. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 3 +++ sound/soc/codecs/twl4030.h | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4199498a8918..bfda7a88e825 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1311,6 +1311,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, case 48000: mode |= TWL4030_APLL_RATE_48000; break; + case 96000: + mode |= TWL4030_APLL_RATE_96000; + break; default: printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", params_rate(params)); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 33dbb144dad1..cb63765db1df 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -109,6 +109,7 @@ #define TWL4030_APLL_RATE_32000 0x80 #define TWL4030_APLL_RATE_44100 0x90 #define TWL4030_APLL_RATE_48000 0xA0 +#define TWL4030_APLL_RATE_96000 0xE0 #define TWL4030_SEL_16K 0x04 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01