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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

2 1

[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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Re: [alsa-devel] [LAD] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 14 Jul '10

14 Jul '10

On Tue, Jul 13, 2010 at 1:19 AM, <fons(a)kokkinizita.net> wrote: >On Mon, Jul 12, 2010 at 06:18:29PM -0700, Niels Mayer wrote: >> Speaking of making my brain hurt -- I really would not have done the >> meters as done in envy24control. I really hate the concept of >> emulating the look of real gear down to the level that you have to >> "quantize" to individual LED's... > > That is indeed quite absurd. Yes, the meters in envy24control -- http://git.alsa-project.org/?p=alsa-tools.git;a=blob_plain;f=envy24control/… -- are absurd! What's even more absurd, is to have the benefits of the ice1712's hardware metering squandered by drawing thousands of rectangles per second (each individual "LED" on each meter is a separate x draw-rectangle call). The existing envy24control meters are an abomination of inefficiency! Especially when used with a remote X display. As a second patch (coming soon), I've rewritten the meters in a more sensible fashion, drawing a single rectangle to represent the instantaneous level (2-3 X-primitive draws per meter total and one blit, versus hundreds of draws and a blit per value change per meter in the original code). I'm also being a little more careful about not needing to redraw the entire meter for each change -- using the monotonicity of the audio signal being represented to optimize the way you draw it; not redrawing if nothing has changed, etc. And since the hardware metering already quantizes to 256 levels -- there's no point in re-quantizing those to individual "virtual LEDs" as is done w/ the current envy24control. The result so far are promising: Here's 'top' output (*) using the new envy24control. Note the load is lower, and 'envy24control' and 'X' both use much less CPU. In fact running this meter lowers the system load. whereas running the old envy24control raises it (and makes the fans run faster!). top - 21:30:13 up 11:30, 9 users, load average: 0.02, 0.11, 0.06 2664 root 20 0 544m 134m 34m S 2.6 3.4 26:03.51 X 9404 npm 20 0 192m 9340 6908 S 0.7 0.2 0:01.72 envy24control The new metering gives 2.6% CPU for X, and 0.7% CPU for the new-metered envy24control; Versus 6.3% CPU for X and 2.7% for the old metering in envy24control (the one from my first patch): top - 21:25:57 up 11:25, 9 users, load average: 0.10, 0.09, 0.03 2664 root 20 0 544m 134m 34m S 6.3 3.4 25:50.67 X 9398 npm 20 0 192m 9368 6940 S 2.7 0.2 0:16.81 envy24control Now granted this is not a scientific test, and "top" is not exactly the best way to gauge performance. But it does suggest that the computational complexity caused by the bad metering algorithm directly translates to higher-power consumption, higher loads, and poorer performance. Compared to the existing envy24control metering -- It would probably me more efficient to actually compute RMS levels off the audio stream -- or even hook up jkmeter through jack -- instead of using the hardware metering data through these computationally wasteful level meters. -- Niels http://nielsmayer.com

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[alsa-devel] [PATCH] wm8988: remove a redundant snd_soc_unregister_codec call in wm8988_register
by Axel Lin 14 Jul '10

14 Jul '10

snd_soc_unregister_codec is called twice if snd_soc_register_dai fail. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8988.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 0417dae..19ad590 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -885,7 +885,6 @@ static int wm8988_register(struct wm8988_priv *wm8988, ret = snd_soc_register_dai(&wm8988_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); goto err_codec; } -- 1.7.0.4

1 0

[alsa-devel] ASoC updates for-2.6.36
by Mark Brown 13 Jul '10

13 Jul '10

The following changes since commit 66b47fdb851924956b6e4696fb43a3496ae2c462: ASoC: Implement WM8994 OPCLK support (2010-07-09 08:50:12 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Arnaud Patard (2): ASoC: Fix kirkwood i2s mono playback ASoC: kirkwood-i2s: Handle mute/unmute playback/record Joe Perches (2): ASoC: Remove unnecessary casts of private_data ASoC: Remove unnecessary casts of private_data Kuninori Morimoto (8): ASoC: fsi: fixup clock inversion operation ASoC: fsi: fixup wrong value setting order of TDM ASoC: fsi: modify format area definition on flags ASoC: fsi: remove un-used variable on fsi_dai_startup ASoC: fsi: remove noisy CR_FMT macro ASoC: fsi: Change struct fsi_regs to fsi_core ASoC: fsi: Add pr_err for noticing unsupported access ASoC: fsi: Fixup for master mode Manuel Lauss (1): ASoC: au1x: fix section mismatch in psc-i2s.c Mark Brown (1): Merge branch 'for-2.6.35' into for-2.6.36 include/sound/sh_fsi.h | 44 +++++++- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 6 +- sound/soc/blackfin/bf5xx-tdm.c | 6 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 27 ++++-- sound/soc/kirkwood/kirkwood.h | 3 + sound/soc/sh/fsi.c | 213 ++++++++++++++++++++++++++----------- 8 files changed, 219 insertions(+), 84 deletions(-)

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 13 Jul '10

13 Jul '10

The following changes since commit 1c5474a65bf15a4cb162dfff86d6d0b5a08a740c: Linux 2.6.35-rc5 (2010-07-12 14:55:33 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Kuninori Morimoto (2): ASoC: fsi: fixup clock inversion operation ASoC: fsi: fixup wrong value setting order of TDM Small, targetted fixes for a specific platform. sound/soc/sh/fsi.c | 27 +++++++++++---------------- 1 files changed, 11 insertions(+), 16 deletions(-)

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[alsa-devel] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 13 Jul '10

13 Jul '10

I've been long annoyed by alsa-tools' envy24control (*) lack of peak-level indication in it's metering, and the non-implementation of its "Reset Peaks" button: https://bugzilla.redhat.com/show_bug.cgi?id=602903 I've fixed that annoyance. Now, a small white line appears above the highest level on each meter, and can be reset via "Reset Peaks" button. Original source: http://git.alsa-project.org/?p=alsa-tools.git;a=blob_plain;f=envy24control/… Patch: http://nielsmayer.com/npm/602903.patch To run this version of envy24control grab the most recent stable release ( ftp://ftp.alsa-project.org/pub/tools/alsa-tools-1.0.23.tar.bz2 ) or git pull from trunk of the "alsa-tools" project ( http://alsa-project.org/main/index.php/GIT_Server ). After unpacking and assuming you've got the patch in ~/602903.patch do: >> cd alsa-tools-1.0.23 >> cat ~/602903.patch | patch -p1 it should give message "patching file envy24control/levelmeters.c" ... follow the directions to compile alsa-tools.... Can whoever's in charge of these things commit this change "upstream" and get it to propagate to the distros? Alternately, give me commit access and I'll do it myself, if people agree this patch is worth having. Next steps: (1) Think about building-in meter ballistics from Fons Adriaensen's jkmeter ( http://www.kokkinizita.net/linuxaudio/downloads/jkmeter-0.4.0.tar.bz2 ). Which helps implent Bob Katz' "K System" ( http://www.digido.com/level-practices-part-2-includes-the-k-system.html ). Although it probably most makes sense to implement such metering on the main "digital mixer" meter of envy24control, if it's not too inefficient, I'd want it available on all meters, despite the warning from Fons ( http://old.nabble.com/First-release-of-jkmeter-td18798950.html ). > This is the type of meter you want for live recording, > mixing and mastering. It probably makes no sense to > use it on all tracks of a DAW, where keeping digital > level within limits is the main purpose of metering. (2) Fix https://bugzilla.redhat.com/show_bug.cgi?id=602900 which causes 'envy24control' to crash when given "-D" argument. and behave incorrectly when given a correct non-numeric name for an audio device: e.g. >> envy24control --card M66 >> invalid card type (driver is USB-Audio) >> envy24control --card M66 --device M66 >> Segmentation fault (core dumped) (3) Apply more reasonable defaults and automatically size to correct dimensions. I use, for example: >> envy24control --card 2 --outputs 4 --input 4 --pcm_output 8 --view_spdif_playback --midichannel 1 --midienhanced --window_width 1275 --tall_eq_mixer_heights 1 & (4) Add midi output, in addition to midi input, which would monitor the hardware peak metering from the ice1712 ("amixer -c M66 cget iface=PCM,name='Multi Track Peak',numid=45") and output a MIDI note-on on the corresponding MIDI channel whenever a specified threshold value was exceeded. This could be used to implement "automatic gain control" in association with the midi-controllable ADC gain provided by envy24control. Niels http://nielsmayer.com PS: (*) http://alsa.opensrc.org/index.php/Envy24control Cards supported by this fix: * M Audio Delta 1010 * M Audio Delta 1010LT * M Audio Delta DiO 2496 * M Audio Delta 66 * M Audio Delta 44 * M Audio Delta 410 * M Audio Audiophile 2496 * TerraTec EWS 88MT * TerraTec EWS 88D * TerraTec EWX 24/96 * TerraTec DMX 6Fire * TerraTec Phase 88 * Hoontech SoundTrack DSP 24 * Hoontech SoundTrack DSP 24 Value * Hoontech SoundTrack DSP 24 Media 7.1 * Event Electronics, EZ8 * Digigram VX442 * Lionstracs, Mediastaton * Terrasoniq TS 88

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[alsa-devel] [PATCH] ASoC: au1x: fix section mismatch in psc-i2s.c
by Manuel Lauss 13 Jul '10

13 Jul '10

Annotate platform probe callback with __devinit instead of plain __init. Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- sound/soc/au1x/psc-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 495be6e..24454c9 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -300,7 +300,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = { }; EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev) +static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) { struct resource *r; unsigned long sel; -- 1.7.1.1 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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