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

[alsa-devel] [PATCH 8/8] sound: asihpi: fix sign bug
by Kulikov Vasiliy 15 Jul '10

15 Jul '10

bytes_per_sec is unsigned, so if snd_pcm_format_width() return error we would not see it. Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/pci/asihpi/asihpi.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 91218f7..c80b0b8 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -460,6 +460,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); int err; u16 format; + int width; unsigned int bytes_per_sec; print_hwparams(params); @@ -512,9 +513,10 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, dpcm->hpi_buffer_attached); } bytes_per_sec = params_rate(params) * params_channels(params); - bytes_per_sec *= snd_pcm_format_width(params_format(params)); + width = snd_pcm_format_width(params_format(params)); + bytes_per_sec *= width; bytes_per_sec /= 8; - if (bytes_per_sec <= 0) + if (width < 0 || bytes_per_sec == 0) return -EINVAL; dpcm->bytes_per_sec = bytes_per_sec; -- 1.7.0.4

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[alsa-devel] ALSA Help?
by Matthew Hudson 15 Jul '10

15 Jul '10

Hello, I posted similar messages to the alsa-user mail list and to ubuntu forums ( http://ubuntuforums.org/showthread.php?t=1528442 ), but haven't been able to get help. Could any of you help with this? So, I've got a great Acer Aspire 7736. Seems to have a RealTek ALC888 card & Intel HDA. I've upgraded Alsa. I've used the Drivers at the RealTek site. I've checked alsamixer over and over and over. Yet, currently I cannot get sound through the headphone jack. Originally, I had problems w/ headphone/speakers playing simlutaneously. Somewhere in my adventures I fixed that, but then only one app would use sound during a login (UGH). So, I kept poking...as all good linux users do...now I'm afraid I've totally wonked it. I'll put all pertinent alsa-info up here for anyone who will help!! http://www.alsa-project.org/db/?f=58850a92a9ea0edbf28a8d014a875cdbd81b621d I guess I'm bright enough to really mess it up, but dumb enough not to have a clue how to fix it. It has NEVER been able to recognize a headphone plugged in the jack automatically. I hope someone knows... -- Soli Deo Gloria! Matt Hudson "There is a kind of happiness and wonder that makes you serious." C.S. Lewis

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[alsa-devel] [PATCH 1/3] ASoC: multi-component - Fix S3C compile error
by Seungwhan Youn 15 Jul '10

15 Jul '10

This patch fix compile error on using S3C I2S. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index ad92e10..5017e31 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -56,11 +56,11 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, switch (clk_id) { case S3C64XX_CLKSRC_PCLK: - iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX; + iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; break; case S3C64XX_CLKSRC_MUX: - iismod |= S3C64XX_IISMOD_IMS_SYSMUX; + iismod |= S3C2412_IISMOD_IMS_SYSMUX; break; case S3C64XX_CLKSRC_CDCLK: @@ -90,7 +90,7 @@ struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(dai); u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - if (iismod & S3C64XX_IISMOD_IMS_SYSMUX) + if (iismod & S3C2412_IISMOD_IMS_SYSMUX) return i2s->iis_cclk; else return i2s->iis_pclk; -- 1.6.2.5

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Re: [alsa-devel] [PATCH 1/2] Kconfig: SND_AC97_POWER_SAVE description improvement
by Jiri Kosina 15 Jul '10

15 Jul '10

On Wed, 14 Jul 2010, Michael Witten wrote: > The description has been expanded to explain the time-out > value provided by the power_save module parameter. Michael, thanks a lot for the patch. I'd rather let it go through Takashi still anyway (added to CC). Thanks. > > Signed-off-by: Michael Witten <mfwitten(a)gmail.com> > --- > sound/drivers/Kconfig | 24 +++++++++++++++++++++--- > 1 files changed, 21 insertions(+), 3 deletions(-) > > diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig > index 84714a6..3264600 100644 > --- a/sound/drivers/Kconfig > +++ b/sound/drivers/Kconfig > @@ -170,9 +170,25 @@ config SND_AC97_POWER_SAVE > AC97 codecs. In this mode, the power-mode is dynamically > controlled at each open/close. > > - The mode is activated by passing power_save=1 option to > - snd-ac97-codec driver. You can toggle it dynamically over > - sysfs, too. > + The mode is activated by passing 'power_save=X' to the > + snd-ac97-codec driver module, where 'X' is the time-out > + value, a nonnegative integer that specifies how many > + seconds of idle time the driver must count before it may > + put the AC97 into power-save mode; a value of 0 (zero) > + disables the use of this power-save mode. > + > + After the snd-ac97-codec driver module has been loaded, > + the 'power_save' parameter can be set via sysfs as follows: > + > + echo 10 > /sys/module/snd_ac97_codec/parameters/power_save > + > + In this case, the time-out is set to 10 seconds; setting > + the time-out to 1 second (the minimum activation value) > + isn't recommended because many applications try to reopen > + the device frequently. A value of 10 seconds would be a > + good choice for normal operations. > + > + See Documentation/sound/alsa/powersave.txt for more details. > > config SND_AC97_POWER_SAVE_DEFAULT > int "Default time-out for AC97 power-save mode" > @@ -182,4 +198,6 @@ config SND_AC97_POWER_SAVE_DEFAULT > The default time-out value in seconds for AC97 automatic > power-save mode. 0 means to disable the power-save mode. > > + See SND_AC97_POWER_SAVE for more details. > + > endif # SND_DRIVERS > -- > 1.7.2.rc1.9.g5b527 > -- Jiri Kosina SUSE Labs, Novell Inc.

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[alsa-devel] [RFC][PATCH 0/3] ASoC: FSI - codecs settings
by Kuninori Morimoto 15 Jul '10

15 Jul '10

Dear Mark, Liam These patches are FSI - codecs settings Kuninori Morimoto (3): ASoC: ak4642: make sure snd_soc_codec value for ak4642 ASoC: da7210: make sure snd_soc_codec value for da7210 ASoC: Add fsi-codec support But I'm not sure this type of fix is correct or not. So, please check and teach me. Now, FSI driver has 2 connections like below. fsi - ak4642 fsi - da7210 In future, I will send "fsi - HDMI" pair. On AP4 board, I would like to use "fsi - ak4642" and "fsi - HDMI" in same time. To use it, I guess I should modify snd_soc_card :: dai_link correct ? I expected this is correct, I tried to use "fsi - ak4642" and "fsi - da7210" in same time for test. But when I used "soc_codec_dev_ak4642" for snd_soc_device :: codec_dev, da7210 didn't work. because i2c address of ak4642 is used for da7210. Same things happen on "soc_codec_dev_da7210" - ak4642. My patches solve above issue for me. But I'm not sure this is correct way. Best regards -- Kuninori Morimoto

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[alsa-devel] controlling access to the sound card
by Jason Butler 14 Jul '10

14 Jul '10

I have three applications that want to use the sound card, two audio stream players, and a voip phone. Is it possible to use alsa so that if a call comes in on the phone alsa will disconnect the audio players, give exclusive access to the voip phone, and then when the phone is done reconnect the audio players to the sound card. Can this be done using alsa and if so how? Could it be done using just the alsa command line tools and config files and if so how?

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[alsa-devel] [PATCH] wm8727: add a missing return in wm8727_platform_probe
by Axel Lin 14 Jul '10

14 Jul '10

otherwise the error path will always be executed. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8727.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 1072621..9d1df26 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -127,6 +127,8 @@ static __devinit int wm8727_platform_probe(struct platform_device *pdev) goto err_codec; } + return 0; + err_codec: snd_soc_unregister_codec(codec); err: -- 1.7.0.4

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Re: [alsa-devel] [PATCH 4/4] ASoC: DaVinci: pcm, fix underrun by using sram
by troy.kisky＠boundarydevices.com 14 Jul '10

14 Jul '10

BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; } On Tue 13/07/10 6:00 AM , "Nori, Sekhar" nsekhar(a)ti.com sent: Hi Troy, On Tue, Nov 17, 2009 at 05:22:34, Troy Kisky wrote: > Fix underruns by using dma to copy 1st to sram > in a ping/pong buffer style and then copying from > the sram to the ASP. This also has the advantage > of tolerating very long interrupt latency on dma > completion. > > Signed-off-by: Troy Kisky [...] > +static int request_ping_pong(struct snd_pcm_substream *substream, > + struct davinci_runtime_data *prtd, > + struct snd_dma_buffer *iram_dma) > +{ > + dma_addr_t asp_src_ping; > + dma_addr_t asp_dst_ping; > + int link; > + struct davinci_pcm_dma_params *dma_data = prtd->params; > + > + /* Request ram master channel */ > + link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, > + davinci_pcm_dma_irq, substream, > + EVENTQ_1); What is the reason for choosing EVENTQ_1 for this channel? EVENTQ_0 is already being used for ASP channel. I imagine it will be much easier to tune the queue usage in the system if all of audio data was using the same queue. I am working on a patch which lets platform specify the event queues for audio DMA. I am not sure if I really need to make a provision for two different queues to be specified - that's why I ask. Thanks, Sekhar The reason is so that the IRAM data can be fetched and used while EVENTQ_1 fetches the next buffer of data from sdram into IRAM. Troy

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[alsa-devel] [PATCH] ASoC: Handle read failures in codec_reg
by Mark Brown 14 Jul '10

14 Jul '10

When a device is powered down volatile registers can't be read so attempts to display codec_reg will show error values, and obviously it is also possible for there to be hardware errors too. Check for errors from reads and display them more clearly when formatting codec_reg. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 22 +++++++++++++++++----- 1 files changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96f9a8e..844ae82 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -84,7 +84,7 @@ static int run_delayed_work(struct delayed_work *dwork) /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { - int i, step = 1, count = 0; + int ret, i, step = 1, count = 0; if (!codec->reg_cache_size) return 0; @@ -101,12 +101,24 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (count >= PAGE_SIZE - 1) break; - if (codec->display_register) + if (codec->display_register) { count += codec->display_register(codec, buf + count, PAGE_SIZE - count, i); - else - count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", codec->read(codec, i)); + } else { + /* If the read fails it's almost certainly due to + * the register being volatile and the device being + * powered off. + */ + ret = codec->read(codec, i); + if (ret >= 0) + count += snprintf(buf + count, + PAGE_SIZE - count, + "%4x", ret); + else + count += snprintf(buf + count, + PAGE_SIZE - count, + "<no data: %d>", ret); + } if (count >= PAGE_SIZE - 1) break; -- 1.7.1

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[alsa-devel] [PATCH] Asoc Davinci Voicecodec: Added support based on copy_from_user instead of DMA
by Raffaele Recalcati 14 Jul '10

14 Jul '10

From: Davide Bonfanti <davide.bonfanti(a)bticino.it> Since DM365 has the same DMA event for McBSP and Voicecodec this two peripherals cannot be used at the same time. This driver implements Voicecodec without the use of a DMA but with a copy_from_user. There's a buffer of BUF_SIZE (tested with 2KB) bytes in the driver that is filled with davinci_pcm_copy. When pcm is running, a TIMER interrupt is activated each 1.36ms in order to fill HW FIFO (dm_vc_irq). BUG: It happens sometimes that the peripheral stops working so there's a trap. This driver was tested with timer1 and timer4. The measures made using a GPIO and scope give a time for copy_from_user about 8/10us (triggered by alsa layer) while TIMER interrupt takes 1.5us. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and tested on bmx board. Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> --- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-pcm-copyfromuser.c | 333 ++++++++++++++++++++++++++ sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/soc-core.c | 31 ++- 4 files changed, 354 insertions(+), 12 deletions(-) create mode 100644 sound/soc/davinci/davinci-pcm-copyfromuser.c diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a93679d..7d6a9a1 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,5 +1,6 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o +snd-soc-davinci-objs += davinci-pcm-copyfromuser.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o snd-soc-davinci-vcif-objs:= davinci-vcif.o diff --git a/sound/soc/davinci/davinci-pcm-copyfromuser.c b/sound/soc/davinci/davinci-pcm-copyfromuser.c new file mode 100644 index 0000000..df37873 --- /dev/null +++ b/sound/soc/davinci/davinci-pcm-copyfromuser.c @@ -0,0 +1,333 @@ +/* + * + * Copyright (C) 2010 Bticino S.p.a + * Author: Davide Bonfanti <davide.bonfanti(a)bticino.it> + * + * Contributors: + * Raffaele Recalcati <raffaele.recalcati(a)bticino.it> + * + * 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 version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any + * kind, whether express or implied; without even the implied warranty + * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/kernel.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/gpio.h> + +#include "davinci-pcm.h" + +/* Timer register offsets */ +#define PID12 0x00 +#define TIM12 0x10 +#define TIM34 0x14 +#define PRD12 0x18 +#define PRD34 0x1c +#define TCR 0x20 +#define TGCR 0x24 + +/* Timer register bitfields */ +#define TCR_ENAMODE_DISABLE 0x0 +#define TCR_ENAMODE_ONESHOT 0x1 +#define TCR_ENAMODE_PERIODIC 0x2 +#define TCR_ENAMODE_MASK 0x3 + +#define TGCR_TIMMODE_SHIFT 2 +#define TGCR_TIMMODE_64BIT_GP 0x0 +#define TGCR_TIMMODE_32BIT_UNCHAINED 0x1 +#define TGCR_TIMMODE_64BIT_WDOG 0x2 +#define TGCR_TIMMODE_32BIT_CHAINED 0x3 + +#define TGCR_TIM12RS_SHIFT 0 +#define TGCR_TIM34RS_SHIFT 1 +#define TGCR_RESET 0x0 +#define TGCR_UNRESET 0x1 +#define TGCR_RESET_MASK 0x3 + +#define TIMER1_BASE 0x01C21800 +#define TIMER4_BASE 0x01C23800 + +/* choose the timer to use */ +#define TIMER_BASE TIMER4_BASE +/* driver buffer dimension */ +#define BUF_SIZE 2048 + +#if (TIMER_BASE == TIMER1_BASE) +#define TINT IRQ_TINT1_TINT12 +#define TIMER "timer1" +#elif (TIMER_BASE == TIMER4_BASE) +#define TINT IRQ_PWMINT2 +#define TIMER "timer4" +#endif + +int pointer_sub; +u16 local_buffer[BUF_SIZE/2]; + +static struct snd_pcm_hardware pcm_hardware_playback = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_8000), + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = BUF_SIZE, + .period_bytes_min = 512, + .period_bytes_max = 512, + .periods_min = BUF_SIZE / 512, + .periods_max = BUF_SIZE / 512, + .fifo_size = 0, +}; + +/* + * How this driver works... + * + * Since DM365 have the same DMA event for I2S and Voicecodec this two + * peripheralscannot be used at the same time. + * This driver implements voicecodec without the use of a DMA but with + * a copy_from_user. + * There's a buffer of BUF_SIZE bytes in the driver that is filled with + * davinci_pcm_copy. + * When pcm is running, a TIMER interrupt is activated each 1.36ms in + * order to fill HW FIFO (dm_vc_irq). + * It happens that the peripheral stop working so there's a trap... + * Driver was proved with timer1 and timer4. if timer2 is used the device + * resets during power-up. Timer3 have a structure a bit different and, + * in fact, doesn't work. + * Measures using a GPIO and scope give a time for copy_from_user about + * 8/10us while interrupt takes 1.5us + */ + +static snd_pcm_uframes_t +davinci_pcm_pointer(struct snd_pcm_substream *substream) +{ + return pointer_sub; +} + +static int davinci_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd; + struct snd_pcm_hardware *ppcm; + int ret = 0; + struct davinci_pcm_dma_params *pa; + struct davinci_pcm_dma_params *params; + struct clk *clk; + + pointer_sub = 0; + +/* + * ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + * &pcm_hardware_playback : &pcm_hardware_capture; + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ppcm = &pcm_hardware_playback; + else + return -ENODEV; + + snd_soc_set_runtime_hwparams(substream, ppcm); + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + __raw_writel(0x0, IO_ADDRESS(0x01D0C008)); + __raw_writel(0x7400, IO_ADDRESS(0x01D0C004)); + clk = clk_get(NULL, TIMER); + if (!IS_ERR(clk)) + clk_enable(clk); + return 0; +} + +static int davinci_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct clk *clk; + + clk = clk_get(NULL, TIMER); + if (!IS_ERR(clk)) + clk_disable(clk); + return 0; +} + +static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int davinci_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + if (copy_from_user(local_buffer + hwoff, buf, + frames_to_bytes(runtime, frames))) { + printk(KERN_ERR "ERROR COPY_FROM_USER\n"); + return -EFAULT; + } + return 0; +} + +static struct snd_pcm_ops davinci_pcm_ops = { + .open = davinci_pcm_open, + .close = davinci_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = davinci_pcm_hw_params, + .hw_free = davinci_pcm_hw_free, + .pointer = davinci_pcm_pointer, + .copy = davinci_pcm_copy, +}; + +static u64 davinci_pcm_dmamask = 0xffffffff; + +static irqreturn_t dm_vc_irq(int irq, void *sbst) +{ + struct snd_pcm_substream *substream = + (struct snd_pcm_substream *) sbst; + int fifo, diff, per_size, buf_size; + static int last_ptr; + + if (substream->runtime && substream->runtime->status && + snd_pcm_running(substream)) { + fifo = __raw_readl(IO_ADDRESS(0x01D0C028)); + fifo = (fifo & 0x1f00) >> 8; + if (fifo == 15) { + /* peripheral blocked! restart */ + __raw_writel(0, IO_ADDRESS(0x01D0C004)); + __raw_writel(0x5400, IO_ADDRESS(0x01D0C004)); + } + buf_size = substream->runtime->buffer_size; + per_size = substream->runtime->period_size; + for (; fifo < 0x10; fifo++) { + __raw_writew(local_buffer[pointer_sub++], + IO_ADDRESS(0x01D0C024)); + pointer_sub %= buf_size; + do { + diff = __raw_readl(IO_ADDRESS(0x01D0C00C)); + } while (!(diff & 0x8)); + } + if (last_ptr >= pointer_sub) + diff = buf_size + pointer_sub - last_ptr; + else + diff = pointer_sub - last_ptr; + if (diff >= per_size) { + snd_pcm_period_elapsed(substream); + last_ptr += per_size; + if (last_ptr >= buf_size) + last_ptr -= buf_size; + } + } else + last_ptr = 0; + return IRQ_HANDLED; +} + +static int davinci_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int tcr, tgcr; + struct snd_dma_buffer *buf; + struct snd_pcm_substream *substream; + struct clk *clk; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &davinci_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + buf = &substream->dma_buffer; + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->bytes = pcm_hardware_playback.buffer_bytes_max; + } + clk = clk_get(NULL, TIMER); + if (IS_ERR(clk)) + printk(KERN_ERR "ERROR getting timer\n"); + + /* Disabled, Internal clock source */ + __raw_writel(0, IO_ADDRESS(TIMER_BASE + TCR)); + + /* reset both timers, no pre-scaler for timer34 */ + tgcr = 0; + __raw_writel(tgcr, IO_ADDRESS(TIMER_BASE + TGCR)); + + /* Set both timers to unchained 32-bit */ + tgcr = TGCR_TIMMODE_32BIT_UNCHAINED << TGCR_TIMMODE_SHIFT; + __raw_writel(tgcr, IO_ADDRESS(TIMER_BASE + TGCR)); + + /* Unreset timers */ + tgcr |= (TGCR_UNRESET << TGCR_TIM12RS_SHIFT); + __raw_writel(tgcr, IO_ADDRESS(TIMER_BASE + TGCR)); + + /* Init both counters to zero */ + __raw_writel(0, IO_ADDRESS(TIMER_BASE + TIM12)); + tcr = __raw_readl(IO_ADDRESS(TIMER_BASE + TCR)); + + /* disable timer */ + tcr &= ~(TCR_ENAMODE_MASK << 6); + __raw_writel(tcr, IO_ADDRESS(TIMER_BASE + TCR)); + + /* reset counter to zero, set new period */ + __raw_writel(0, IO_ADDRESS(TIMER_BASE + TIM12)); + __raw_writel(0x8000, IO_ADDRESS(TIMER_BASE + PRD12)); + + /* Set enable mode */ + tcr |= TCR_ENAMODE_PERIODIC << 6; + __raw_writel(tcr, IO_ADDRESS(TIMER_BASE + TCR)); + +#if (TIMER_BASE == TIMER4_BASE) + __raw_writel(__raw_readl(IO_ADDRESS(0x01c40018)) | + 0x1000, IO_ADDRESS(0x01c40018)); +#endif + if (request_irq(TINT, dm_vc_irq, IRQF_TIMER, "VOICE-TOUT", substream)) + printk(KERN_ERR "%s ERROR requesting Interrupt\n", __func__); + return 0; +} + +struct snd_soc_platform davinci_soc_platform_copy = { + .name = "davinci-audio-copy", + .pcm_ops = &davinci_pcm_ops, + .pcm_new = davinci_pcm_new, +}; EXPORT_SYMBOL_GPL(davinci_soc_platform_copy); + +static int __init davinci_soc_copy_platform_init(void) +{ + return snd_soc_register_platform(&davinci_soc_platform_copy); +} +module_init(davinci_soc_copy_platform_init); + +static void __exit davinci_soc_copy_platform_exit(void) +{ + snd_soc_unregister_platform(&davinci_soc_platform_copy); +} +module_exit(davinci_soc_copy_platform_exit); + +MODULE_AUTHOR("bticino s.p.a."); +MODULE_DESCRIPTION("TI DAVINCI PCM copy_from_user module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 0764944..cb7c2aa 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -29,5 +29,6 @@ struct davinci_pcm_dma_params { extern struct snd_soc_platform davinci_soc_platform; +extern struct snd_soc_platform davinci_soc_platform_copy; #endif diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad7f952..696e8b3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -801,7 +801,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } return 0; } - +#if 0 /* ASoC PCM operations */ static struct snd_pcm_ops soc_pcm_ops = { .open = soc_pcm_open, @@ -811,7 +811,7 @@ static struct snd_pcm_ops soc_pcm_ops = { .prepare = soc_pcm_prepare, .trigger = soc_pcm_trigger, }; - +#endif #ifdef CONFIG_PM /* powers down audio subsystem for suspend */ static int soc_suspend(struct device *dev) @@ -1306,6 +1306,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + struct snd_pcm_ops *soc_pcm_ops; rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); if (rtd == NULL) @@ -1335,19 +1336,25 @@ static int soc_new_pcm(struct snd_soc_device *socdev, dai_link->pcm = pcm; pcm->private_data = rtd; - soc_pcm_ops.mmap = platform->pcm_ops->mmap; - soc_pcm_ops.pointer = platform->pcm_ops->pointer; - soc_pcm_ops.ioctl = platform->pcm_ops->ioctl; - soc_pcm_ops.copy = platform->pcm_ops->copy; - soc_pcm_ops.silence = platform->pcm_ops->silence; - soc_pcm_ops.ack = platform->pcm_ops->ack; - soc_pcm_ops.page = platform->pcm_ops->page; + soc_pcm_ops = kmalloc(sizeof(struct snd_pcm_ops), GFP_KERNEL); + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_codec_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->mmap = platform->pcm_ops->mmap; + soc_pcm_ops->pointer = platform->pcm_ops->pointer; + soc_pcm_ops->ioctl = platform->pcm_ops->ioctl; + soc_pcm_ops->copy = platform->pcm_ops->copy; + soc_pcm_ops->silence = platform->pcm_ops->silence; + soc_pcm_ops->ack = platform->pcm_ops->ack; + soc_pcm_ops->page = platform->pcm_ops->page; if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); - + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); ret = platform->pcm_new(codec->card, codec_dai, pcm); if (ret < 0) { -- 1.7.0.4

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