June 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] alsa-info.sh
by Justin P. Mattock 04 Jun '10

04 Jun '10

hello.. Im looking for the codec#0 for the MacBookAir2,1 can anybody supply this?(doing a google search gives nothing but crudd). cheers, Justin P. Mattock

2 2

[alsa-devel] ASoC: Defining "signed integer" mixer controls?
by Stuart Longland 04 Jun '10

04 Jun '10

Hi all... I'm posting this via GMane since my server at home seems to be down, hopefully it gets through. I hope to put a version of the TLV320AIC3204 driver that I've been working on up there -- I did try posting it to the list, but I suspect it's either been left in the moderation queue or silently dropped. In short, I got the sound working; just testing playback for now... Recording should work, once I get the mixer set up. My query though; the mixer on this CODEC provides gain settings, helpfully scaled in dB, as signed integers. For instance; the four line level output drivers: Headphone left/right,and Line Out left/right, can all be adjusted in gain from -6dB through to +29dB. These are set in individual registers as the lower 6 bits; so: -6 dB is represented as 111010 (0x3a) 29 dB is represented as 011101 (0x1d) At the moment, things "half work" with the following code: /* * DAC digital volumes. From -63.5 to +24 dB in 0.5 dB steps */ static DECLARE_TLV_DB_MINMAX(dac_tlv, -6350, 2400); /* ADC PGA gain volumes. From 0 to 59.5 dB in 0.5 dB steps */ static DECLARE_TLV_DB_MINMAX(adc_tlv, 0, 0); /* * Output driver stage volumes; -6dB through to +29dB */ static DECLARE_TLV_DB_MINMAX(output_stage_tlv, -600, 2900); static const struct snd_kcontrol_new aic3204_snd_controls[] = { /* Output */ SOC_DOUBLE_R_TLV("Headphone Output Volume", AIC3204_HPLGAIN, AIC3204_HPRGAIN, 0, 29, 0, output_stage_tlv), SOC_DOUBLE_R_TLV("Line Output Volume", AIC3204_LOLGAIN, AIC3204_LORGAIN, 0, 29, 0, output_stage_tlv), SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC3204_LDACVOL, AIC3204_RDACVOL, 0, 127, 0, dac_tlv), ... However; it treats the bitfield as unsigned, which is wrong in this case. The bitfield is a signed integer in two's complement format. How do I convince ALSA that these integers have a sign bit? Regards, Stuart Longland

4 9

[alsa-devel] [PATCH] alsa-lib: snd_pcm_delay and friends do not account for a write being currently in progress
by John Lindgren 04 Jun '10

04 Jun '10

Hi, In a multi-threaded application it is possible for snd_pcm_delay or an equivalent function to be called by one thread while another is sitting in snd_pcm_writei. In this case, snd_pcm_delay does not take into account that there may not be enough space for all the data passed to snd_pcm_writei to be written to the ring buffer at once, and will return incorrect values. To illustrate: Suppose a sound card has a maximum buffer size of 0.1 second. Then, suppose an application writes 0.5 second of audio. snd_pcm_writei will immediately write the first 0.1 second to the ring buffer, then will incrementally write the remaining 0.4 second as the buffer drains. During this time, snd_pcm_delay will return a constant fill level of 0.1 second. On the application side, the playback time counter will be calculated during this time as 0.5 seconds written to ALSA minus a delay of 0.1 second, giving a constant stream position of 0.4 second. (This problem showed up when a recent change in Audacious caused it to call snd_pcm_writei with larger chunks of audio than formerly -- the result was that the visualization, which is synchronized with the time counter, began to jump, where it should be a smooth 30 FPS animation.) The problem is a little tough to fix, since there must be some thread semantics involved so that the thread calling snd_pcm_delay knows at what instant snd_pcm_delay will begin taking into account the data passed to snd_pcm_writei. My solution is to create a variant of snd_pcm_writei ("snd_pcm_writei_locked"), which is the passed the additional argument of a pointer to a mutex locked in advance by the caller. snd_pcm_writei_locked will then unlock the mutex once the application can safely assume that snd_pcm_delay will take into account the new data. I realize this is a substantial change, so please tell me whether this patch is along the right track or if a different solution is called for. Signed-off-by: John Lindgren <john.lindgren(a)tds.net> --- include/pcm.h | 8 +++++ src/pcm/pcm.c | 76 ++++++++++++++++++++++++++++++++++++++++++++++++--- src/pcm/pcm_hw.c | 39 +++++++++++++++++++++++++- src/pcm/pcm_local.h | 3 ++ 4 files changed, 121 insertions(+), 5 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index f3618c3..2234279 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -33,6 +33,10 @@ extern "C" { #endif +#include <pthread.h> + +#define SND_PCM_HAVE_LOCKED_WRITE + /** * \defgroup PCM PCM Interface * See the \ref pcm page for more details. @@ -448,8 +452,12 @@ snd_pcm_sframes_t snd_pcm_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t frames); snd_pcm_sframes_t snd_pcm_forwardable(snd_pcm_t *pcm); snd_pcm_sframes_t snd_pcm_forward(snd_pcm_t *pcm, snd_pcm_uframes_t frames); snd_pcm_sframes_t snd_pcm_writei(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size); +snd_pcm_sframes_t snd_pcm_writei_locked (snd_pcm_t * pcm, const void * buffer, + snd_pcm_uframes_t size, pthread_mutex_t * mutex); snd_pcm_sframes_t snd_pcm_readi(snd_pcm_t *pcm, void *buffer, snd_pcm_uframes_t size); snd_pcm_sframes_t snd_pcm_writen(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t size); +snd_pcm_sframes_t snd_pcm_writen_locked (snd_pcm_t * pcm, void * * bufs, + snd_pcm_uframes_t size, pthread_mutex_t * mutex); snd_pcm_sframes_t snd_pcm_readn(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t size); int snd_pcm_wait(snd_pcm_t *pcm, int timeout); diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index f910189..db03067 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -903,8 +903,12 @@ int snd_pcm_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t *params) */ int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status) { - assert(pcm && status); - return pcm->fast_ops->status(pcm->fast_op_arg, status); + int result; + + assert(pcm && status); + result = pcm->fast_ops->status (pcm->fast_op_arg, status); + status->delay += pcm->fast_op_arg->write_delay; + return result; } /** @@ -973,12 +977,17 @@ link_warning(snd_pcm_hwsync, "Warning: snd_pcm_hwsync() is deprecated, consider */ int snd_pcm_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) { + int result; + assert(pcm); if (CHECK_SANITY(! pcm->setup)) { SNDMSG("PCM not set up"); return -EIO; } - return pcm->fast_ops->delay(pcm->fast_op_arg, delayp); + + result = pcm->fast_ops->delay (pcm->fast_op_arg, delayp); + * delayp += pcm->fast_op_arg->write_delay; + return result; } /** @@ -1248,6 +1257,34 @@ snd_pcm_sframes_t snd_pcm_writei(snd_pcm_t *pcm, const void *buffer, snd_pcm_ufr } /** + * \brief Variant of snd_pcm_writei for multithreaded applications + * \param mutex Mutex already locked by caller + * + * In a multithreaded application, there is uncertainty inherent in calling + * snd_pcm_delay while another thread is calling and_pcm_writei, since there is + * no way to know whether the data passed to snd_pcm_writei has yet been + * accounted for in the delay calculation. This function provides a solution to + * that problem by unlocking the mutex when it is safe to call snd_pcm_delay. + * The mutex is locked again before the function returns. An application using + * this function should be careful to avoid deadlocks. + * + * Programmers should check that the macro SND_PCM_HAVE_LOCKED_WRITE is defined + * before using this function. + */ + +snd_pcm_sframes_t snd_pcm_writei_locked (snd_pcm_t * pcm, const void * buffer, + snd_pcm_uframes_t size, pthread_mutex_t * mutex) +{ + snd_pcm_sframes_t result; + + assert (pcm && pcm->fast_op_arg && ! pcm->fast_op_arg->write_mutex); + pcm->fast_op_arg->write_mutex = mutex; + result = snd_pcm_writei (pcm, buffer, size); + pcm->fast_op_arg->write_mutex = NULL; + return result; +} + +/** * \brief Write non interleaved frames to a PCM * \param pcm PCM handle * \param bufs frames containing buffers (one for each channel) @@ -1280,6 +1317,25 @@ snd_pcm_sframes_t snd_pcm_writen(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t } /** + * \brief Variant of snd_pcm_writen for multithreaded applications + * \param Mutex already locked by caller + * + * (See snd_pcm_writei_locked for a full description.) + */ + +snd_pcm_sframes_t snd_pcm_writen_locked (snd_pcm_t * pcm, void * * bufs, + snd_pcm_uframes_t size, pthread_mutex_t * mutex) +{ + snd_pcm_sframes_t result; + + assert (pcm && pcm->fast_op_arg && ! pcm->fast_op_arg->write_mutex); + pcm->fast_op_arg->write_mutex = mutex; + result = snd_pcm_writen (pcm, bufs, size); + pcm->fast_op_arg->write_mutex = NULL; + return result; +} + +/** * \brief Read interleaved frames from a PCM * \param pcm PCM handle * \param buffer frames containing buffer @@ -2480,6 +2536,7 @@ int snd_pcm_avail_delay(snd_pcm_t *pcm, sf = pcm->fast_ops->delay(pcm->fast_op_arg, delayp); if (sf < 0) return (int)sf; + * delayp += pcm->fast_op_arg->write_delay; sf = pcm->fast_ops->avail_update(pcm->fast_op_arg); if (sf < 0) return (int)sf; @@ -6652,7 +6709,18 @@ snd_pcm_sframes_t snd_pcm_write_areas(snd_pcm_t *pcm, const snd_pcm_channel_area goto _end; } - err = snd_pcm_wait(pcm, -1); + if (pcm->write_mutex) { + pcm->write_delay = size; + assert (! pthread_mutex_unlock (pcm->write_mutex)); + } + + err = snd_pcm_wait(pcm, -1); + + if (pcm->write_mutex) { + assert (! pthread_mutex_lock (pcm->write_mutex)); + pcm->write_delay = 0; + } + if (err < 0) break; goto _again; diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 9d243d5..6a428d8 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -753,7 +753,8 @@ static int snd_pcm_hw_unlink(snd_pcm_t *pcm) return 0; } -static snd_pcm_sframes_t snd_pcm_hw_writei(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size) +static snd_pcm_sframes_t writei_real (snd_pcm_t * pcm, const void * buffer, + snd_pcm_uframes_t size) { int err; snd_pcm_hw_t *hw = pcm->private_data; @@ -772,6 +773,42 @@ static snd_pcm_sframes_t snd_pcm_hw_writei(snd_pcm_t *pcm, const void *buffer, s return xferi.result; } +static snd_pcm_sframes_t snd_pcm_hw_writei (snd_pcm_t * pcm, const void * + buffer, snd_pcm_uframes_t size) +{ + snd_pcm_sframes_t remain = size; + snd_pcm_sframes_t result; + + while (1) { + if ((result = snd_pcm_hw_avail_update (pcm)) < 0) + return result; + if ((result = writei_real (pcm, buffer, result < remain ? result : + remain)) < 0) + return result; + + buffer = (const char *) buffer + snd_pcm_frames_to_bytes (pcm, result); + remain -= result; + + if (! remain) + return size; + + if (pcm->write_mutex) { + pcm->write_delay = remain; + assert (! pthread_mutex_unlock (pcm->write_mutex)); + } + + result = snd_pcm_wait (pcm, -1); + + if (pcm->write_mutex) { + assert (! pthread_mutex_lock (pcm->write_mutex)); + pcm->write_delay = 0; + } + + if (result < 0) + return result; + } +} + static snd_pcm_sframes_t snd_pcm_hw_writen(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t size) { int err; diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 9aa81e1..53eef78 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -229,6 +229,9 @@ struct _snd_pcm { snd_pcm_t *fast_op_arg; void *private_data; struct list_head async_handlers; + snd_pcm_uframes_t write_delay; /* size of data passed to snd_pcm_write* + * but not yet outputted to ring buffer */ + pthread_mutex_t * write_mutex; /* mutex passed to snd_pcm_write*_locked */ }; /* make local functions really local */

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[alsa-devel] [RFC][PATCH 20/26] alsa: ASoC: Add JZ4740 codec driver
by Lars-Peter Clausen 04 Jun '10

04 Jun '10

This patch adds support for the JZ4740 internal codec. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/jz4740-codec.c | 502 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/jz4740-codec.h | 20 ++ 4 files changed, 528 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/jz4740-codec.c create mode 100644 sound/soc/codecs/jz4740-codec.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 31ac553..b8008df 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -23,6 +23,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4671 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS4270 if I2C + select SND_SOC_JZ4740 if SOC_JZ4740 select SND_SOC_MAX9877 if I2C select SND_SOC_DA7210 if I2C select SND_SOC_PCM3008 @@ -269,6 +270,9 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate +config SND_SOC_JZ4740_CODEC + tristate + # Amp config SND_SOC_MAX9877 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 91429ea..4c7ee31 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,6 +56,7 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o +snd-soc-jz4740-codec-objs := jz4740-codec.o # Amp snd-soc-max9877-objs := max9877.o @@ -121,6 +122,7 @@ obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o +obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o diff --git a/sound/soc/codecs/jz4740-codec.c b/sound/soc/codecs/jz4740-codec.c new file mode 100644 index 0000000..6e4b741 --- /dev/null +++ b/sound/soc/codecs/jz4740-codec.c @@ -0,0 +1,502 @@ +/* + * Copyright (C) 2009-2010, Lars-Peter Clausen <lars(a)metafoo.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> +#include <sound/soc.h> + +#define JZ4740_REG_CODEC_1 0x0 +#define JZ4740_REG_CODEC_2 0x1 + +#define JZ4740_CODEC_1_LINE_ENABLE BIT(29) +#define JZ4740_CODEC_1_MIC_ENABLE BIT(28) +#define JZ4740_CODEC_1_SW1_ENABLE BIT(27) +#define JZ4740_CODEC_1_ADC_ENABLE BIT(26) +#define JZ4740_CODEC_1_SW2_ENABLE BIT(25) +#define JZ4740_CODEC_1_DAC_ENABLE BIT(24) +#define JZ4740_CODEC_1_VREF_DISABLE BIT(20) +#define JZ4740_CODEC_1_VREF_AMP_DISABLE BIT(19) +#define JZ4740_CODEC_1_VREF_PULL_DOWN BIT(18) +#define JZ4740_CODEC_1_VREF_LOW_CURRENT BIT(17) +#define JZ4740_CODEC_1_VREF_HIGH_CURRENT BIT(16) +#define JZ4740_CODEC_1_HEADPHONE_DISABLE BIT(14) +#define JZ4740_CODEC_1_HEADPHONE_AMP_CHANGE_ANY BIT(13) +#define JZ4740_CODEC_1_HEADPHONE_CHANGE BIT(12) +#define JZ4740_CODEC_1_HEADPHONE_PULL_DOWN_M BIT(11) +#define JZ4740_CODEC_1_HEADPHONE_PULL_DOWN_R BIT(10) +#define JZ4740_CODEC_1_HEADPHONE_POWER_DOWN_M BIT(9) +#define JZ4740_CODEC_1_HEADPHONE_POWER_DOWN BIT(8) +#define JZ4740_CODEC_1_SUSPEND BIT(1) +#define JZ4740_CODEC_1_RESET BIT(0) + +#define JZ4740_CODEC_1_LINE_ENABLE_OFFSET 29 +#define JZ4740_CODEC_1_MIC_ENABLE_OFFSET 28 +#define JZ4740_CODEC_1_SW1_ENABLE_OFFSET 27 +#define JZ4740_CODEC_1_ADC_ENABLE_OFFSET 26 +#define JZ4740_CODEC_1_SW2_ENABLE_OFFSET 25 +#define JZ4740_CODEC_1_DAC_ENABLE_OFFSET 24 +#define JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET 14 +#define JZ4740_CODEC_1_HEADPHONE_POWER_DOWN_OFFSET 8 + +#define JZ4740_CODEC_2_INPUT_VOLUME_MASK 0x1f0000 +#define JZ4740_CODEC_2_SAMPLE_RATE_MASK 0x000f00 +#define JZ4740_CODEC_2_MIC_BOOST_GAIN_MASK 0x000030 +#define JZ4740_CODEC_2_HEADPHONE_VOLUME_MASK 0x000003 + +#define JZ4740_CODEC_2_INPUT_VOLUME_OFFSET 16 +#define JZ4740_CODEC_2_SAMPLE_RATE_OFFSET 8 +#define JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET 4 +#define JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET 0 + +struct jz4740_codec { + void __iomem *base; + struct resource *mem; + + uint32_t reg_cache[2]; + struct snd_soc_codec codec; +}; + +static inline struct jz4740_codec *codec_to_jz4740(struct snd_soc_codec *codec) +{ + return container_of(codec, struct jz4740_codec, codec); +} + +static unsigned int jz4740_codec_read(struct snd_soc_codec *codec, unsigned int reg) +{ + struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec); + return readl(jz4740_codec->base + (reg << 2)); +} + +static int jz4740_codec_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec); + jz4740_codec->reg_cache[reg] = val; + + writel(val, jz4740_codec->base + (reg << 2)); + return 0; +} + +static const struct snd_kcontrol_new jz4740_codec_controls[] = { + SOC_SINGLE("Master Playback Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET, 3, 0), + SOC_SINGLE("Capture Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_INPUT_VOLUME_OFFSET, 31, 0), + SOC_SINGLE("Master Playback Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET, 1, 1), + SOC_SINGLE("Mic Capture Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET, 3, 0), +}; + +static const struct snd_kcontrol_new jz4740_codec_output_controls[] = { + SOC_DAPM_SINGLE("Bypass Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW1_ENABLE_OFFSET, 1, 0), + SOC_DAPM_SINGLE("DAC Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW2_ENABLE_OFFSET, 1, 0), +}; + +static const struct snd_kcontrol_new jz4740_codec_input_controls[] = { + SOC_DAPM_SINGLE("Line Capture Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_LINE_ENABLE_OFFSET, 1, 0), + SOC_DAPM_SINGLE("Mic Capture Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_MIC_ENABLE_OFFSET, 1, 0), +}; + +static const struct snd_soc_dapm_widget jz4740_codec_dapm_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_ADC_ENABLE_OFFSET, 0), + SND_SOC_DAPM_DAC("DAC", "Playback", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_DAC_ENABLE_OFFSET, 0), + + SND_SOC_DAPM_MIXER("Output Mixer", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_HEADPHONE_POWER_DOWN_OFFSET, 1, + jz4740_codec_output_controls, + ARRAY_SIZE(jz4740_codec_output_controls)), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + jz4740_codec_input_controls, + ARRAY_SIZE(jz4740_codec_input_controls)), + SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_INPUT("RIN"), +}; + +static const struct snd_soc_dapm_route jz4740_codec_dapm_routes[] = { + + {"Line Input", NULL, "LIN"}, + {"Line Input", NULL, "RIN"}, + + {"Input Mixer", "Line Capture Switch", "Line Input"}, + {"Input Mixer", "Mic Capture Switch", "MIC"}, + + {"ADC", NULL, "Input Mixer"}, + + {"Output Mixer", "Bypass Switch", "Input Mixer"}, + {"Output Mixer", "DAC Switch", "DAC"}, + + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, +}; + +static int jz4740_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + uint32_t val; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S18_3LE: + break; + default: + return -EINVAL; + break; + } + + switch (params_rate(params)) { + case 8000: + val = 0; + break; + case 11025: + val = 1; + break; + case 12000: + val = 2; + break; + case 16000: + val = 3; + break; + case 22050: + val = 4; + break; + case 24000: + val = 5; + break; + case 32000: + val = 6; + break; + case 44100: + val = 7; + break; + case 48000: + val = 8; + break; + default: + return -EINVAL; + } + + val <<= JZ4740_CODEC_2_SAMPLE_RATE_OFFSET; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_SAMPLE_RATE_MASK, val); + + return 0; +} + +static int jz4740_codec_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops jz4740_codec_dai_ops = { + .hw_params = jz4740_codec_hw_params, + .set_fmt = jz4740_codec_set_fmt, +}; + +struct snd_soc_dai jz4740_codec_dai = { + .name = "jz-codec", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + }, + .ops = &jz4740_codec_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(jz4740_codec_dai); + +static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + + if (codec->bias_level == SND_SOC_BIAS_OFF && level != SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_RESET, JZ4740_CODEC_1_RESET); + udelay(2); + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SUSPEND | JZ4740_CODEC_1_RESET, 0); + } + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_VREF_DISABLE | + JZ4740_CODEC_1_VREF_AMP_DISABLE | + JZ4740_CODEC_1_HEADPHONE_POWER_DOWN_M | + JZ4740_CODEC_1_VREF_LOW_CURRENT | + JZ4740_CODEC_1_VREF_HIGH_CURRENT, + 0); + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_VREF_LOW_CURRENT | + JZ4740_CODEC_1_VREF_HIGH_CURRENT, + JZ4740_CODEC_1_VREF_LOW_CURRENT | + JZ4740_CODEC_1_VREF_HIGH_CURRENT); + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_VREF_DISABLE | JZ4740_CODEC_1_VREF_AMP_DISABLE, + JZ4740_CODEC_1_VREF_DISABLE | JZ4740_CODEC_1_VREF_AMP_DISABLE); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SUSPEND, JZ4740_CODEC_1_SUSPEND); + break; + } + codec->bias_level = level; + + return 0; +} + + +static struct snd_soc_codec *jz4740_codec_codec; + +static int jz4740_codec_dev_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = jz4740_codec_codec; + + BUG_ON(!codec); + + socdev->card->codec = codec; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret) { + dev_err(&pdev->dev, "Failed to create pcms: %d\n", ret); + return ret; + } + + snd_soc_add_controls(codec, jz4740_codec_controls, + ARRAY_SIZE(jz4740_codec_controls)); + + snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets, + ARRAY_SIZE(jz4740_codec_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes, + ARRAY_SIZE(jz4740_codec_dapm_routes)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int jz4740_codec_dev_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_jz4740_codec = { + .probe = jz4740_codec_dev_probe, + .remove = jz4740_codec_dev_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_jz4740_codec); + +static int __devinit jz4740_codec_probe(struct platform_device *pdev) +{ + int ret; + struct jz4740_codec *jz4740_codec; + struct snd_soc_codec *codec; + + jz4740_codec = kzalloc(sizeof(*jz4740_codec), GFP_KERNEL); + + if (!jz4740_codec) + return -ENOMEM; + + jz4740_codec->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + if (!jz4740_codec->mem) { + dev_err(&pdev->dev, "Failed to get mmio memory resource\n"); + ret = -ENOENT; + goto err_free_codec; + } + + jz4740_codec->mem = request_mem_region(jz4740_codec->mem->start, + resource_size(jz4740_codec->mem), pdev->name); + + if (!jz4740_codec->mem) { + dev_err(&pdev->dev, "Failed to request mmio memory region\n"); + ret = -EBUSY; + goto err_free_codec; + } + + jz4740_codec->base = ioremap(jz4740_codec->mem->start, resource_size(jz4740_codec->mem)); + + if (!jz4740_codec->base) { + dev_err(&pdev->dev, "Failed to ioremap mmio memory\n"); + ret = -EBUSY; + goto err_release_mem_region; + } + + jz4740_codec_dai.dev = &pdev->dev; + + codec = &jz4740_codec->codec; + + codec->dev = &pdev->dev; + codec->name = "jz-codec"; + codec->owner = THIS_MODULE; + + codec->read = jz4740_codec_read; + codec->write = jz4740_codec_write; + codec->set_bias_level = jz4740_codec_set_bias_level; + codec->bias_level = SND_SOC_BIAS_OFF; + + codec->dai = &jz4740_codec_dai; + codec->num_dai = 1; + + codec->reg_cache = jz4740_codec->reg_cache; + codec->reg_cache_size = 2; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + jz4740_codec_codec = codec; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE); + + + platform_set_drvdata(pdev, jz4740_codec); + ret = snd_soc_register_codec(codec); + + if (ret) { + dev_err(&pdev->dev, "Failed to register codec\n"); + goto err_iounmap; + } + + ret = snd_soc_register_dai(&jz4740_codec_dai); + if (ret) { + dev_err(&pdev->dev, "Failed to register codec dai\n"); + goto err_unregister_codec; + } + + jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +err_unregister_codec: + snd_soc_unregister_codec(codec); +err_iounmap: + iounmap(jz4740_codec->base); +err_release_mem_region: + release_mem_region(jz4740_codec->mem->start, resource_size(jz4740_codec->mem)); +err_free_codec: + kfree(jz4740_codec); + + return ret; +} + +static int __devexit jz4740_codec_remove(struct platform_device *pdev) +{ + struct jz4740_codec *jz4740_codec = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&jz4740_codec_dai); + snd_soc_unregister_codec(&jz4740_codec->codec); + + iounmap(jz4740_codec->base); + release_mem_region(jz4740_codec->mem->start, resource_size(jz4740_codec->mem)); + + platform_set_drvdata(pdev, NULL); + kfree(jz4740_codec); + + return 0; +} + +static struct platform_driver jz4740_codec_driver = { + .probe = jz4740_codec_probe, + .remove = __devexit_p(jz4740_codec_remove), + .driver = { + .name = "jz4740-codec", + .owner = THIS_MODULE, + }, +}; + +static int __init jz4740_codec_init(void) +{ + return platform_driver_register(&jz4740_codec_driver); +} +module_init(jz4740_codec_init); + +static void __exit jz4740_codec_exit(void) +{ + platform_driver_unregister(&jz4740_codec_driver); +} +module_exit(jz4740_codec_exit); + +MODULE_DESCRIPTION("JZ4740 SoC internal codec driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:jz-codec"); diff --git a/sound/soc/codecs/jz4740-codec.h b/sound/soc/codecs/jz4740-codec.h new file mode 100644 index 0000000..b5a0691 --- /dev/null +++ b/sound/soc/codecs/jz4740-codec.h @@ -0,0 +1,20 @@ +/* + * Copyright (C) 2009, Lars-Peter Clausen <lars(a)metafoo.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#ifndef __SND_SOC_CODECS_JZ4740_CODEC_H__ +#define __SND_SOC_CODECS_JZ4740_CODEC_H__ + +extern struct snd_soc_dai jz4740_codec_dai; +extern struct snd_soc_codec_device soc_codec_dev_jz4740_codec; + +#endif -- 1.5.6.5

4 8

[alsa-devel] [RFC][PATCH 21/26] alsa: ASoC: Add JZ4740 ASoC support
by Lars-Peter Clausen 03 Jun '10

03 Jun '10

This patch adds ASoC support for JZ4740 SoCs I2S module. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: alsa-devel(a)alsa-project.org --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/jz4740/Kconfig | 13 + sound/soc/jz4740/Makefile | 9 + sound/soc/jz4740/jz4740-i2s.c | 568 +++++++++++++++++++++++++++++++++++++++++ sound/soc/jz4740/jz4740-i2s.h | 18 ++ sound/soc/jz4740/jz4740-pcm.c | 350 +++++++++++++++++++++++++ sound/soc/jz4740/jz4740-pcm.h | 22 ++ 8 files changed, 982 insertions(+), 0 deletions(-) create mode 100644 sound/soc/jz4740/Kconfig create mode 100644 sound/soc/jz4740/Makefile create mode 100644 sound/soc/jz4740/jz4740-i2s.c create mode 100644 sound/soc/jz4740/jz4740-i2s.h create mode 100644 sound/soc/jz4740/jz4740-pcm.c create mode 100644 sound/soc/jz4740/jz4740-pcm.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index b1749bc..5a7a724 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -36,6 +36,7 @@ source "sound/soc/s3c24xx/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/txx9/Kconfig" +source "sound/soc/jz4740/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1470141..fdbe74d 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,3 +14,4 @@ obj-$(CONFIG_SND_SOC) += s3c24xx/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += txx9/ +obj-$(CONFIG_SND_SOC) += jz4740/ diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig new file mode 100644 index 0000000..39df949 --- /dev/null +++ b/sound/soc/jz4740/Kconfig @@ -0,0 +1,13 @@ +config SND_JZ4740_SOC + tristate "SoC Audio for Ingenic JZ4740 SoC" + depends on SOC_JZ4740 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the Jz4740 AC97, I2S or SSP interface. You will also need + to select the audio interfaces to support below. + +config SND_JZ4740_SOC_I2S + depends on SND_JZ4740_SOC + tristate "SoC Audio (I2S protocol) for Ingenic jz4740 chip" + help + Say Y if you want to use I2S protocol and I2S codec on Ingenic Jz4740 QI_LB60 board. diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile new file mode 100644 index 0000000..1be8d19 --- /dev/null +++ b/sound/soc/jz4740/Makefile @@ -0,0 +1,9 @@ +# +# Jz4740 Platform Support +# +snd-soc-jz4740-objs := jz4740-pcm.o +snd-soc-jz4740-i2s-objs := jz4740-i2s.o + +obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o +obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o + diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c new file mode 100644 index 0000000..2b139fd --- /dev/null +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -0,0 +1,568 @@ +/* + * Copyright (C) 2010, Lars-Peter Clausen <lars(a)metafoo.de> + * + * 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. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/init.h> +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <linux/clk.h> +#include <linux/delay.h> + +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "jz4740-i2s.h" +#include "jz4740-pcm.h" + +#define JZ_REG_AIC_CONF 0x00 +#define JZ_REG_AIC_CTRL 0x04 +#define JZ_REG_AIC_I2S_FMT 0x10 +#define JZ_REG_AIC_FIFO_STATUS 0x14 +#define JZ_REG_AIC_I2S_STATUS 0x1c +#define JZ_REG_AIC_CLK_DIV 0x30 +#define JZ_REG_AIC_FIFO 0x34 + +#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_MASK (0xf << 12) +#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_MASK (0xf << 8) +#define JZ_AIC_CONF_OVERFLOW_PLAY_LAST BIT(6) +#define JZ_AIC_CONF_INTERNAL_CODEC BIT(5) +#define JZ_AIC_CONF_I2S BIT(4) +#define JZ_AIC_CONF_RESET BIT(3) +#define JZ_AIC_CONF_BIT_CLK_MASTER BIT(2) +#define JZ_AIC_CONF_SYNC_CLK_MASTER BIT(1) +#define JZ_AIC_CONF_ENABLE BIT(0) + +#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 12 +#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 8 + +#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK (0x7 << 19) +#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK (0x7 << 16) +#define JZ_AIC_CTRL_ENABLE_RX_DMA BIT(15) +#define JZ_AIC_CTRL_ENABLE_TX_DMA BIT(14) +#define JZ_AIC_CTRL_MONO_TO_STEREO BIT(11) +#define JZ_AIC_CTRL_SWITCH_ENDIANNESS BIT(10) +#define JZ_AIC_CTRL_SIGNED_TO_UNSIGNED BIT(9) +#define JZ_AIC_CTRL_FLUSH BIT(8) +#define JZ_AIC_CTRL_ENABLE_ROR_INT BIT(6) +#define JZ_AIC_CTRL_ENABLE_TUR_INT BIT(5) +#define JZ_AIC_CTRL_ENABLE_RFS_INT BIT(4) +#define JZ_AIC_CTRL_ENABLE_TFS_INT BIT(3) +#define JZ_AIC_CTRL_ENABLE_LOOPBACK BIT(2) +#define JZ_AIC_CTRL_ENABLE_PLAYBACK BIT(1) +#define JZ_AIC_CTRL_ENABLE_CAPTURE BIT(0) + +#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_OFFSET 19 +#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET 16 + +#define JZ_AIC_I2S_FMT_DISABLE_BIT_CLK BIT(12) +#define JZ_AIC_I2S_FMT_ENABLE_SYS_CLK BIT(4) +#define JZ_AIC_I2S_FMT_MSB BIT(0) + +#define JZ_AIC_I2S_STATUS_BUSY BIT(2) + +#define JZ_AIC_CLK_DIV_MASK 0xf + +struct jz4740_i2s { + struct resource *mem; + void __iomem *base; + dma_addr_t phys_base; + + struct clk *clk_aic; + struct clk *clk_i2s; + + struct jz4740_pcm_config pcm_config_playback; + struct jz4740_pcm_config pcm_config_capture; +}; + +static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s, + unsigned int reg) +{ + return readl(i2s->base + reg); +} + +static inline void jz4740_i2s_write(const struct jz4740_i2s *i2s, + unsigned int reg, uint32_t value) +{ + writel(value, i2s->base + reg); +} + +static inline struct jz4740_i2s *jz4740_dai_to_i2s(struct snd_soc_dai *dai) +{ + return dai->private_data; +} + +static int jz4740_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf, ctrl; + + if (dai->active) + return 0; + + + ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + ctrl |= JZ_AIC_CTRL_FLUSH; + jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + + clk_enable(i2s->clk_i2s); + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf |= JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + return 0; +} + +static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + if (!dai->active) + return; + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf &= ~JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + clk_disable(i2s->clk_i2s); +} + + +static int jz4740_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + uint32_t ctrl; + uint32_t mask; + + if (playback) + mask = JZ_AIC_CTRL_ENABLE_PLAYBACK | JZ_AIC_CTRL_ENABLE_TX_DMA; + else + mask = JZ_AIC_CTRL_ENABLE_CAPTURE | JZ_AIC_CTRL_ENABLE_RX_DMA; + + ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ctrl |= mask; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ctrl &= ~mask; + break; + default: + return -EINVAL; + } + + jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + + return 0; +} + + +static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + + uint32_t format = 0; + uint32_t conf; + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + + conf &= ~(JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + conf |= JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER; + format |= JZ_AIC_I2S_FMT_ENABLE_SYS_CLK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + conf |= JZ_AIC_CONF_SYNC_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: + conf |= JZ_AIC_CONF_BIT_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_MSB: + format |= JZ_AIC_I2S_FMT_MSB; + break; + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + jz4740_i2s_write(i2s, JZ_REG_AIC_I2S_FMT, format); + + return 0; +} + +static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + enum jz4740_dma_width dma_width; + struct jz4740_pcm_config *pcm_config; + unsigned int sample_size; + uint32_t ctrl; + + ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + sample_size = 0; + dma_width = JZ4740_DMA_WIDTH_8BIT; + break; + case SNDRV_PCM_FORMAT_S16: + sample_size = 1; + dma_width = JZ4740_DMA_WIDTH_16BIT; + break; + default: + return -EINVAL; + } + + if (playback) { + ctrl &= ~JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK; + ctrl |= sample_size << JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_OFFSET; + } else { + ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK; + ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; + } + + switch (params_channels(params)) { + case 2: + break; + case 1: + if (playback) { + ctrl |= JZ_AIC_CTRL_MONO_TO_STEREO; + break; + } + default: /* Falltrough */ + return -EINVAL; + } + + jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + + if (playback) { + pcm_config = &i2s->pcm_config_playback; + pcm_config->dma_config.dst_width = dma_width; + } else { + pcm_config = &i2s->pcm_config_capture; + pcm_config->dma_config.src_width = dma_width; + } + + + snd_soc_dai_set_dma_data(dai, substream, pcm_config); + + return 0; +} + +static int jz4740_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + + switch (div_id) { + case JZ4740_I2S_BIT_CLK: + if (div & 1 || div > 16) + return -EINVAL; + jz4740_i2s_write(i2s, JZ_REG_AIC_CLK_DIV, div - 1); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int jz4740_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + int ret = 0; + struct clk *parent; + + switch (clk_id) { + case JZ4740_I2S_CLKSRC_EXT: + parent = clk_get(NULL, "ext"); + clk_set_parent(i2s->clk_i2s, parent); + break; + case JZ4740_I2S_CLKSRC_PLL: + parent = clk_get(NULL, "pll half"); + clk_set_parent(i2s->clk_i2s, parent); + ret = clk_set_rate(i2s->clk_i2s, freq); + break; + default: + return -EINVAL; + } + clk_put(parent); + + return ret; +} + +static int jz4740_i2s_suspend(struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + if (dai->active) { + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf &= ~JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + clk_disable(i2s->clk_i2s); + } + + clk_disable(i2s->clk_aic); + + return 0; +} + +static int jz4740_i2s_resume(struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + clk_enable(i2s->clk_aic); + + if (dai->active) { + clk_enable(i2s->clk_i2s); + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf |= JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + } + + return 0; +} + +static int jz4740_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + conf = (7 << JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | + (8 << JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | + JZ_AIC_CONF_OVERFLOW_PLAY_LAST | + JZ_AIC_CONF_I2S | + JZ_AIC_CONF_INTERNAL_CODEC; + + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, JZ_AIC_CONF_RESET); + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + return 0; +} + + +static struct snd_soc_dai_ops jz4740_i2s_dai_ops = { + .startup = jz4740_i2s_startup, + .shutdown = jz4740_i2s_shutdown, + .trigger = jz4740_i2s_trigger, + .hw_params = jz4740_i2s_hw_params, + .set_fmt = jz4740_i2s_set_fmt, + .set_clkdiv = jz4740_i2s_set_clkdiv, + .set_sysclk = jz4740_i2s_set_sysclk, +}; + +#define JZ4740_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE) + +struct snd_soc_dai jz4740_i2s_dai = { + .name = "jz4740-i2s", + .probe = jz4740_i2s_probe, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = JZ4740_I2S_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = JZ4740_I2S_FMTS, + }, + .symmetric_rates = 1, + .ops = &jz4740_i2s_dai_ops, + .suspend = jz4740_i2s_suspend, + .resume = jz4740_i2s_resume, +}; +EXPORT_SYMBOL_GPL(jz4740_i2s_dai); + +static void __devinit jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) +{ + struct jz4740_dma_config *dma_config; + + /* Playback */ + dma_config = &i2s->pcm_config_playback.dma_config; + dma_config->src_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; + dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT; + dma_config->flags = JZ4740_DMA_SRC_AUTOINC; + dma_config->mode = JZ4740_DMA_MODE_SINGLE; + i2s->pcm_config_playback.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO; + + /* Capture */ + dma_config = &i2s->pcm_config_capture.dma_config; + dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; + dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE; + dma_config->flags = JZ4740_DMA_DST_AUTOINC; + dma_config->mode = JZ4740_DMA_MODE_SINGLE; + i2s->pcm_config_capture.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO; +} + +static int __devinit jz4740_i2s_dev_probe(struct platform_device *pdev) +{ + struct jz4740_i2s *i2s; + int ret; + + i2s = kzalloc(sizeof(*i2s), GFP_KERNEL); + + if (!i2s) + return -ENOMEM; + + i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + if (!i2s->mem) { + ret = -ENOENT; + goto err_free; + } + + i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem), + pdev->name); + + if (!i2s->mem) { + ret = -EBUSY; + goto err_free; + } + + i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem)); + + if (!i2s->base) { + ret = -EBUSY; + goto err_release_mem_region; + } + + i2s->phys_base = i2s->mem->start; + + i2s->clk_aic = clk_get(&pdev->dev, "aic"); + if (IS_ERR(i2s->clk_aic)) { + ret = PTR_ERR(i2s->clk_aic); + goto err_iounmap; + } + + i2s->clk_i2s = clk_get(&pdev->dev, "i2s"); + if (IS_ERR(i2s->clk_i2s)) { + ret = PTR_ERR(i2s->clk_i2s); + goto err_iounmap; + } + + clk_enable(i2s->clk_aic); + + jz4740_i2c_init_pcm_config(i2s); + + jz4740_i2s_dai.private_data = i2s; + ret = snd_soc_register_dai(&jz4740_i2s_dai); + + platform_set_drvdata(pdev, i2s); + + return 0; + +err_iounmap: + iounmap(i2s->base); +err_release_mem_region: + release_mem_region(i2s->mem->start, resource_size(i2s->mem)); +err_free: + kfree(i2s); + + return ret; +} + +static int __devexit jz4740_i2s_dev_remove(struct platform_device *pdev) +{ + struct jz4740_i2s *i2s = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&jz4740_i2s_dai); + + clk_disable(i2s->clk_aic); + clk_put(i2s->clk_i2s); + clk_put(i2s->clk_aic); + + iounmap(i2s->base); + release_mem_region(i2s->mem->start, resource_size(i2s->mem)); + + platform_set_drvdata(pdev, NULL); + kfree(i2s); + + return 0; +} + +static struct platform_driver jz4740_i2s_driver = { + .probe = jz4740_i2s_dev_probe, + .remove = __devexit_p(jz4740_i2s_dev_remove), + .driver = { + .name = "jz4740-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init jz4740_i2s_init(void) +{ + return platform_driver_register(&jz4740_i2s_driver); +} +module_init(jz4740_i2s_init); + +static void __exit jz4740_i2s_exit(void) +{ + platform_driver_unregister(&jz4740_i2s_driver); +} +module_exit(jz4740_i2s_exit); + +MODULE_AUTHOR("Lars-Peter Clausen, <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("Ingenic JZ4740 SoC I2S driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:jz4740-i2s"); diff --git a/sound/soc/jz4740/jz4740-i2s.h b/sound/soc/jz4740/jz4740-i2s.h new file mode 100644 index 0000000..da22ed8 --- /dev/null +++ b/sound/soc/jz4740/jz4740-i2s.h @@ -0,0 +1,18 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _JZ4740_I2S_H +#define _JZ4740_I2S_H + +/* I2S clock source */ +#define JZ4740_I2S_CLKSRC_EXT 0 +#define JZ4740_I2S_CLKSRC_PLL 1 + +#define JZ4740_I2S_BIT_CLK 0 + +extern struct snd_soc_dai jz4740_i2s_dai; + +#endif diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c new file mode 100644 index 0000000..fd1c203 --- /dev/null +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -0,0 +1,350 @@ +/* + * Copyright (C) 2010, Lars-Peter Clausen <lars(a)metafoo.de> + * + * 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. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/slab.h> + +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mach-jz4740/dma.h> +#include "jz4740-pcm.h" + +struct jz4740_runtime_data { + unsigned int dma_period; + dma_addr_t dma_start; + dma_addr_t dma_pos; + dma_addr_t dma_end; + + struct jz4740_dma_chan *dma; + + dma_addr_t fifo_addr; +}; + +/* identify hardware playback capabilities */ +static const struct snd_pcm_hardware jz4740_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + + .rates = SNDRV_PCM_RATE_8000_48000, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 16, + .period_bytes_max = 2 * PAGE_SIZE, + .periods_min = 2, + .periods_max = 128, + .buffer_bytes_max = 128 * 2 * PAGE_SIZE, + .fifo_size = 32, +}; + +static void jz4740_pcm_start_transfer(struct jz4740_runtime_data *prtd, int stream) +{ + unsigned int count; + + if (prtd->dma_pos + prtd->dma_period > prtd->dma_end) + count = prtd->dma_end - prtd->dma_pos; + else + count = prtd->dma_period; + + jz4740_dma_disable(prtd->dma); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + jz4740_dma_set_src_addr(prtd->dma, prtd->dma_pos); + jz4740_dma_set_dst_addr(prtd->dma, prtd->fifo_addr); + } else { + jz4740_dma_set_src_addr(prtd->dma, prtd->fifo_addr); + jz4740_dma_set_dst_addr(prtd->dma, prtd->dma_pos); + } + + jz4740_dma_set_transfer_count(prtd->dma, count); + + jz4740_dma_enable(prtd->dma); + + prtd->dma_pos += prtd->dma_period; + if (prtd->dma_pos >= prtd->dma_end) + prtd->dma_pos = prtd->dma_start; +} + +static void jz4740_pcm_dma_transfer_done(struct jz4740_dma_chan *dma, int err, + void *dev_id) +{ + struct snd_pcm_substream *substream = dev_id; + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + + snd_pcm_period_elapsed(substream); + + jz4740_pcm_start_transfer(prtd, substream->stream); +} + +static int jz4740_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct jz4740_pcm_config *config; + + config = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + if (!prtd->dma) { + const char *dma_channel_name; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_channel_name = "PCM Playback"; + else + dma_channel_name = "PCM Capture"; + + prtd->dma = jz4740_dma_request(substream, dma_channel_name); + } + + if (!prtd->dma) + return -EBUSY; + + jz4740_dma_configure(prtd->dma, &config->dma_config); + prtd->fifo_addr = config->fifo_addr; + + jz4740_dma_set_complete_cb(prtd->dma, jz4740_pcm_dma_transfer_done); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + runtime->dma_bytes; + + return 0; +} + +static int jz4740_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct jz4740_runtime_data *prtd = substream->runtime->private_data; + + snd_pcm_set_runtime_buffer(substream, NULL); + if (prtd->dma) { + jz4740_dma_free(prtd->dma); + prtd->dma = NULL; + } + + return 0; +} + +static int jz4740_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct jz4740_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + if (!prtd->dma) + return 0; + + prtd->dma_pos = prtd->dma_start; + + return ret; +} + +static int jz4740_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + jz4740_pcm_start_transfer(prtd, substream->stream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + jz4740_dma_disable(prtd->dma); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t jz4740_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + unsigned long count, pos; + snd_pcm_uframes_t offset; + struct jz4740_dma_chan *dma = prtd->dma; + + count = jz4740_dma_get_residue(dma); + if (prtd->dma_pos == prtd->dma_start) + pos = prtd->dma_end - prtd->dma_start - count; + else + pos = prtd->dma_pos - prtd->dma_start - count; + + offset = bytes_to_frames(runtime, pos); + if (offset >= runtime->buffer_size) + offset = 0; + + return offset; +} + +static int jz4740_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd; + + snd_soc_set_runtime_hwparams(substream, &jz4740_pcm_hardware); + prtd = kzalloc(sizeof(struct jz4740_runtime_data), GFP_KERNEL); + + if (prtd == NULL) + return -ENOMEM; + + runtime->private_data = prtd; + return 0; +} + +static int jz4740_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + + return 0; +} + +static int jz4740_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static struct snd_pcm_ops jz4740_pcm_ops = { + .open = jz4740_pcm_open, + .close = jz4740_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = jz4740_pcm_hw_params, + .hw_free = jz4740_pcm_hw_free, + .prepare = jz4740_pcm_prepare, + .trigger = jz4740_pcm_trigger, + .pointer = jz4740_pcm_pointer, + .mmap = jz4740_pcm_mmap, +}; + +static int jz4740_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = jz4740_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + buf->area = dma_alloc_noncoherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + + return 0; +} + +static void jz4740_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_noncoherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); + +int jz4740_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &jz4740_pcm_dmamask; + + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->playback.channels_min) { + ret = jz4740_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto err; + } + + if (dai->capture.channels_min) { + ret = jz4740_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto err; + } + +err: + return ret; +} + +struct snd_soc_platform jz4740_soc_platform = { + .name = "jz4740-pcm", + .pcm_ops = &jz4740_pcm_ops, + .pcm_new = jz4740_pcm_new, + .pcm_free = jz4740_pcm_free, +}; +EXPORT_SYMBOL_GPL(jz4740_soc_platform); + +static int __init jz4740_soc_platform_init(void) +{ + return snd_soc_register_platform(&jz4740_soc_platform); +} +module_init(jz4740_soc_platform_init); + +static void __exit jz4740_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&jz4740_soc_platform); +} +module_exit(jz4740_soc_platform_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("Ingenic SoC JZ4740 PCM driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/jz4740/jz4740-pcm.h b/sound/soc/jz4740/jz4740-pcm.h new file mode 100644 index 0000000..e3f221e --- /dev/null +++ b/sound/soc/jz4740/jz4740-pcm.h @@ -0,0 +1,22 @@ +/* + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _JZ4740_PCM_H +#define _JZ4740_PCM_H + +#include <linux/dma-mapping.h> +#include <asm/mach-jz4740/dma.h> + +/* platform data */ +extern struct snd_soc_platform jz4740_soc_platform; + +struct jz4740_pcm_config { + struct jz4740_dma_config dma_config; + phys_addr_t fifo_addr; +}; + +#endif -- 1.5.6.5

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[alsa-devel] [RFC][PATCH 26/26] alsa: ASoC: JZ4740: Add qi_lb60 board driver
by Lars-Peter Clausen 03 Jun '10

03 Jun '10

This patch adds ASoC support for the qi_lb60 board. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: alsa-devel(a)alsa-project.org --- sound/soc/jz4740/Kconfig | 8 ++ sound/soc/jz4740/Makefile | 4 + sound/soc/jz4740/qi_lb60.c | 175 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 187 insertions(+), 0 deletions(-) create mode 100644 sound/soc/jz4740/qi_lb60.c diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig index 39df949..aaa116f 100644 --- a/sound/soc/jz4740/Kconfig +++ b/sound/soc/jz4740/Kconfig @@ -11,3 +11,11 @@ config SND_JZ4740_SOC_I2S tristate "SoC Audio (I2S protocol) for Ingenic jz4740 chip" help Say Y if you want to use I2S protocol and I2S codec on Ingenic Jz4740 QI_LB60 board. + +config SND_JZ4740_SOC_QI_LB60 + tristate "SoC Audio support for Qi Hardware Ben Nanonote" + depends on SND_JZ4740_SOC && JZ4740_QI_LB60 + select SND_JZ4740_SOC_I2S + select SND_SOC_JZ4740_CODEC + help + Say Y if you want to add support for SoC audio of internal codec on Ingenic Jz4740 QI_LB60 board. diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile index 1be8d19..be873c1 100644 --- a/sound/soc/jz4740/Makefile +++ b/sound/soc/jz4740/Makefile @@ -7,3 +7,7 @@ snd-soc-jz4740-i2s-objs := jz4740-i2s.o obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o +# Jz4740 Machine Support +snd-soc-qi-lb60-objs := qi_lb60.o + +obj-$(CONFIG_SND_JZ4740_SOC_QI_LB60) += snd-soc-qi-lb60.o diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c new file mode 100644 index 0000000..c3f1f91 --- /dev/null +++ b/sound/soc/jz4740/qi_lb60.c @@ -0,0 +1,175 @@ +/* + * Copyright (C) 2009, Lars-Peter Clausen <lars(a)metafoo.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> + +#include "../codecs/jz4740-codec.h" +#include "jz4740-pcm.h" +#include "jz4740-i2s.h" + + +#define QI_LB60_SND_GPIO JZ_GPIO_PORTB(29) +#define QI_LB60_AMP_GPIO JZ_GPIO_PORTD(4) + +static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *ctrl, int event) +{ + int on = 0; + if (event & SND_SOC_DAPM_POST_PMU) + on = 1; + else if (event & SND_SOC_DAPM_PRE_PMD) + on = 0; + + gpio_set_value(QI_LB60_SND_GPIO, on); + gpio_set_value(QI_LB60_AMP_GPIO, on); + + return 0; +} + +static const struct snd_soc_dapm_widget qi_lb60_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", qi_lb60_spk_event), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route qi_lb60_routes[] = { + {"Mic", NULL, "MIC"}, + {"Speaker", NULL, "LOUT"}, + {"Speaker", NULL, "ROUT"}, +}; + +#define QI_LB60_DAIFMT (SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + +static int qi_lb60_codec_init(struct snd_soc_codec *codec) +{ + int ret; + struct snd_soc_dai *cpu_dai = codec->socdev->card->dai_link->cpu_dai; + struct snd_soc_dai *codec_dai = codec->socdev->card->dai_link->codec_dai; + + snd_soc_dapm_nc_pin(codec, "LIN"); + snd_soc_dapm_nc_pin(codec, "RIN"); + + ret = snd_soc_dai_set_fmt(codec_dai, QI_LB60_DAIFMT); + if (ret < 0) { + dev_err(codec->dev, "Failed to set codec dai format: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cpu dai format: %d\n", ret); + return ret; + } + + snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets)); + + snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link qi_lb60_dai = { + .name = "jz4740-codec", + .stream_name = "jz4740-codec", + .cpu_dai = &jz4740_i2s_dai, + .codec_dai = &jz4740_codec_dai, + .init = qi_lb60_codec_init, +}; + +static struct snd_soc_card qi_lb60 = { + .name = "QI LB60", + .dai_link = &qi_lb60_dai, + .num_links = 1, + .platform = &jz4740_soc_platform, +}; + +static struct snd_soc_device qi_lb60_snd_devdata = { + .card = &qi_lb60, + .codec_dev = &soc_codec_dev_jz4740_codec, +}; + +static struct platform_device *qi_lb60_snd_device; + +static int __init qi_lb60_init(void) +{ + int ret; + + qi_lb60_snd_device = platform_device_alloc("soc-audio", -1); + + if (!qi_lb60_snd_device) + return -ENOMEM; + + + ret = gpio_request(QI_LB60_SND_GPIO, "SND"); + if (ret) { + pr_err("qi_lb60 snd: Failed to request SND GPIO(%d): %d\n", + QI_LB60_SND_GPIO, ret); + goto err_device_put; + } + + ret = gpio_request(QI_LB60_AMP_GPIO, "AMP"); + if (ret) { + pr_err("qi_lb60 snd: Failed to request AMP GPIO(%d): %d\n", + QI_LB60_AMP_GPIO, ret); + goto err_gpio_free_snd; + } + + gpio_direction_output(JZ_GPIO_PORTB(29), 0); + gpio_direction_output(JZ_GPIO_PORTD(4), 0); + + platform_set_drvdata(qi_lb60_snd_device, &qi_lb60_snd_devdata); + qi_lb60_snd_devdata.dev = &qi_lb60_snd_device->dev; + ret = platform_device_add(qi_lb60_snd_device); + if (ret) { + pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret); + goto err_unset_pdata; + } + + return 0; + +err_unset_pdata: + platform_set_drvdata(qi_lb60_snd_device, NULL); +/*err_gpio_free_amp:*/ + gpio_free(QI_LB60_AMP_GPIO); +err_gpio_free_snd: + gpio_free(QI_LB60_SND_GPIO); +err_device_put: + platform_device_put(qi_lb60_snd_device); + + return ret; +} +module_init(qi_lb60_init); + +static void __exit qi_lb60_exit(void) +{ + gpio_free(QI_LB60_AMP_GPIO); + gpio_free(QI_LB60_SND_GPIO); + platform_device_unregister(qi_lb60_snd_device); +} +module_exit(qi_lb60_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support"); +MODULE_LICENSE("GPL v2"); -- 1.5.6.5

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Re: [alsa-devel] [PATCH] AsoC/s3c: patch for the unnecessary variable 'state' removal
by Mark Brown 03 Jun '10

03 Jun '10

On Thu, Jun 03, 2010 at 12:54:59AM +0800, Wan ZongShun wrote: > The variable 'state' of structure 's3c_ac97_info' seems no use here, > so this patch is to remove the unnecessary variable. > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> Applied, thanks.

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Re: [alsa-devel] [PATCH] AsoC/atmel: patch for the unnecessary variable removal
by Mark Brown 03 Jun '10

03 Jun '10

On Thu, Jun 03, 2010 at 12:28:46AM +0800, Wan ZongShun wrote: > The variable 'periods' of structure 'atmel_runtime_data' > seems no use in whole atmel alsa driver,so I make a patch > to remove the unnecessary variable. > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> Applied, thanks. ASoC, BTW - not AsoC.

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[alsa-devel] [PATCH v2 1/4] imx-ssi.c: add new choices to platform configuration
by Eric Bénard 03 Jun '10

03 Jun '10

* introduce 3 new flags to allow a more detailed configuration of the SSI link : IMX_SSI_NET : enable Network Mode IMX_SSI_SYN : enable Synchronous Mode IMX_SSI_USE_I2S_SLAVE : enable I2S Slave Mode * new platform can use these settings without breaking actual platforms. Signed-off-by: Eric Bénard <eric(a)eukrea.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- arch/arm/plat-mxc/include/mach/ssi.h | 3 +++ sound/soc/imx/imx-ssi.c | 11 +++++++++-- 2 files changed, 12 insertions(+), 2 deletions(-) diff --git a/arch/arm/plat-mxc/include/mach/ssi.h b/arch/arm/plat-mxc/include/mach/ssi.h index c34ded5..63f3c28 100644 --- a/arch/arm/plat-mxc/include/mach/ssi.h +++ b/arch/arm/plat-mxc/include/mach/ssi.h @@ -10,6 +10,9 @@ struct imx_ssi_platform_data { unsigned int flags; #define IMX_SSI_DMA (1 << 0) #define IMX_SSI_USE_AC97 (1 << 1) +#define IMX_SSI_NET (1 << 2) +#define IMX_SSI_SYN (1 << 3) +#define IMX_SSI_USE_I2S_SLAVE (1 << 4) void (*ac97_reset) (struct snd_ac97 *ac97); void (*ac97_warm_reset)(struct snd_ac97 *ac97); }; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 80b4fee..50f5162 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -83,8 +83,6 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, /* * SSI DAI format configuration. * Should only be called when port is inactive (i.e. SSIEN = 0). - * Note: We don't use the I2S modes but instead manually configure the - * SSI for I2S because the I2S mode is only a register preset. */ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -99,6 +97,10 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* data on rising edge of bclk, frame low 1clk before data */ strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; scr |= SSI_SCR_NET; + if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) { + scr &= ~SSI_I2S_MODE_MASK; + scr |= SSI_SCR_I2S_MODE_SLAVE; + } break; case SND_SOC_DAIFMT_LEFT_J: /* data on rising edge of bclk, frame high with data */ @@ -143,6 +145,11 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) strcr |= SSI_STCR_TFEN0; + if (ssi->flags & IMX_SSI_NET) + scr |= SSI_SCR_NET; + if (ssi->flags & IMX_SSI_SYN) + scr |= SSI_SCR_SYN; + writel(strcr, ssi->base + SSI_STCR); writel(strcr, ssi->base + SSI_SRCR); writel(scr, ssi->base + SSI_SCR); -- 1.6.3.3

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[alsa-devel] [PATCH v5 0/5] OMAP/ASoC: McBSP: FIFO handling related fixes
by Peter Ujfalusi 03 Jun '10

03 Jun '10

Hello, Tony: Could you take a look at the arch/arm patches in this series? Thanks! Changes since v3/4: - Fixed commit subjects and messages for OMAP3 related patches - Added Acked-by from Mark, and Jarkko Intro message from the original series: This series aims to correct how the McBSP FIFO is viewed, and handled. Introduction of the problem: OMAP McBSP FIFO is word structured: McBSP2 has 1024 + 256 = 1280 word long buffer, McBSP1,3,4,5 has 128 word long buffer This means, that the size of the FIFO depends on the McBSP word size configuration. For example on McBSP3: 16bit samples: size is 128 * 2 = 256 bytes 32bit samples: size is 128 * 4 = 512 bytes It is simpler to place constraint for buffer and period based on channels. McBSP3 as example again (16 or 32 bit samples): 1 channel (mono): size is 128 frames (128 words) 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) Since now the McBSP codec supports not only 16bit samples (32biut has been recently added), the FIFO size handling is no longer correct, since it has been hard wired for 16bit word length. The series changes how the users of McBSP are configuring the FIFO: It used to be 0 based (0 meant 1 word threshold). After this series users can configure the threshold in 1 base mode (1 means 1 word threshold). The platform code now provides the full size of the FIFO in words, instead of the already limited value used in the past. In ASoC omap-mcbsp code hw_rule based constraint refinement is going to be used instead of the hardwired static constraint, which was correct only in case of 16bit word length. The hw_rule is refining the minimum buffer size based on the channel number going to be used by the coming stream. In case of threshold mode additional hw_rule refines the maximum allowed period size. The series are generated agains Takashi's sound-2.6: topic/asoc branch. CCing also Eduardo, and Eero since they have worked on the original FIFO/threshold implementation. All commetns and testers are welcome! Peter --- Peter Ujfalusi (5): OMAP: McBSP: Function to query the FIFO size OMAP3: McBSP: Change the way how the FIFO is handled OMAP3: McBSP: Use the port's buffer_size when calculating tx delay ASoC: omap-mcbsp: Save, and use wlen for threshold configuration ASoC: omap-mcbsp: Place correct constraints for streams arch/arm/mach-omap2/mcbsp.c | 10 ++-- arch/arm/plat-omap/include/plat/mcbsp.h | 2 + arch/arm/plat-omap/mcbsp.c | 51 ++++++++++----- sound/soc/omap/omap-mcbsp.c | 112 +++++++++++++++++++++++------- 4 files changed, 128 insertions(+), 47 deletions(-)

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