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

[alsa-devel] Support for floating point controls?
by Eliot Blennerhassett 02 Feb '10

02 Feb '10

Greetings, some of our products now have control paramters that are 32 bit floating point values. I wonder if a patch implementing float ctl elements would be accepted? Note that the floats are not manipulated in the kernel at all, the values are just passed through from the hardware to userspace (?) Adding something like: #define SNDRV_CTL_ELEM_TYPE_FLOAT ((__force snd_ctl_elem_type_t) 7) struct snd_ctl_elem_info { ... union { struct { float min; /* R: minimum value */ float max; /* R: maximum value */ float step; /* R: step (0 variable) */ } float; ... } struct snd_ctl_elem_value { union { union { float value[128]; } float; -- Eliot Blennerhassett AudioScience Inc

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[alsa-devel] [PATCH] alsa: Add support for timberdale I2S block
by Richard Röjfors 01 Feb '10

01 Feb '10

This driver adds support for the I2S block of the timberdale FPGA. The timberdale is a FPGA found on intel development boards for In-Vehicle Infotainment. The block has support for up to 8 I2S channels, can be clocked from either the FPGA or the device side. This driver introduces support for this block, by exposing each I2S channel as an ALSA PCM channel. Signed-off-by: Richard Röjfors <richard.rojfors(a)mocean-labs.com> --- diff --git a/include/sound/timbi2s.h b/include/sound/timbi2s.h new file mode 100644 index 0000000..4ab810c --- /dev/null +++ b/include/sound/timbi2s.h @@ -0,0 +1,32 @@ +/* + * timbi2s.h timberdale FPGA I2S platform data + * Copyright (c) 2009 Intel Corporation + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * 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 __INCLUDE_SOUND_TIMBI2S_H +#define __INCLUDE_SOUND_TIMBI2S_H + +struct timbi2s_bus_data { + u8 rx; + u16 sample_rate; +}; + +struct timbi2s_platform_data { + const struct timbi2s_bus_data *busses; + int num_busses; + u32 main_clk; +}; + +#endif diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 84714a6..54ad4e7 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -182,4 +182,17 @@ 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. +config SND_TIMBERDALE_I2S + tristate "The timberdale FPGA I2S driver" + depends on MFD_TIMBERDALE && HAS_IOMEM + default y + help + Say Y here to enable driver for the I2S block found within the + Timberdale FPGA. + There is support for up to 8 I2S channels, in either transmitter + or receiver mode. + + To compile this driver as a module, choose M here: the module + will be called snd-timbi2s. + endif # SND_DRIVERS diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile index d4a07f9..dea2eed 100644 --- a/sound/drivers/Makefile +++ b/sound/drivers/Makefile @@ -10,6 +10,7 @@ snd-portman2x4-objs := portman2x4.o snd-serial-u16550-objs := serial-u16550.o snd-virmidi-objs := virmidi.o snd-ml403-ac97cr-objs := ml403-ac97cr.o pcm-indirect2.o +snd-timbi2s-objs := timbi2s.o # Toplevel Module Dependency obj-$(CONFIG_SND_DUMMY) += snd-dummy.o @@ -19,5 +20,6 @@ obj-$(CONFIG_SND_MTPAV) += snd-mtpav.o obj-$(CONFIG_SND_MTS64) += snd-mts64.o obj-$(CONFIG_SND_PORTMAN2X4) += snd-portman2x4.o obj-$(CONFIG_SND_ML403_AC97CR) += snd-ml403-ac97cr.o +obj-$(CONFIG_SND_TIMBERDALE_I2S) += snd-timbi2s.o obj-$(CONFIG_SND) += opl3/ opl4/ mpu401/ vx/ pcsp/ diff --git a/sound/drivers/timbi2s.c b/sound/drivers/timbi2s.c new file mode 100644 index 0000000..a9c34c2 --- /dev/null +++ b/sound/drivers/timbi2s.c @@ -0,0 +1,755 @@ +/* + * timbi2s.c timberdale FPGA I2S driver + * Copyright (c) 2009 Intel Corporation + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * 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. + */ + +/* Supports: + * Timberdale FPGA I2S + * + */ + +#include <linux/io.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/timbi2s.h> + +#define DRIVER_NAME "timb-i2s" + +#define MAX_BUSSES 8 + +#define TIMBI2S_REG_VER 0x00 +#define TIMBI2S_REG_UIR 0x04 + +#define TIMBI2S_BUS_PRESCALE 0x00 +#define TIMBI2S_BUS_ICLR 0x04 +#define TIMBI2S_BUS_IPR 0x08 +#define TIMBI2S_BUS_ISR 0x0c +#define TIMBI2S_BUS_IER 0x10 + + +#define TIMBI2S_IRQ_TX_FULL 0x01 +#define TIMBI2S_IRQ_TX_ALMOST_FULL 0x02 +#define TIMBI2S_IRQ_TX_ALMOST_EMPTY 0x04 +#define TIMBI2S_IRQ_TX_EMPTY 0x08 + +#define TIMBI2S_IRQ_RX_FULL 0x10 +#define TIMBI2S_IRQ_RX_ALMOST_FULL 0x20 +#define TIMBI2S_IRQ_RX_ALMOST_EMPTY 0x40 +#define TIMBI2S_IRQ_RX_NOT_EMPTY 0x80 + +#define TIMBI2S_BUS_ICOR 0x14 +#define TIMBI2S_ICOR_TX_ENABLE 0x00000001 +#define TIMBI2S_ICOR_RX_ENABLE 0x00000002 +#define TIMBI2S_ICOR_LFIFO_RST 0x00000004 +#define TIMBI2S_ICOR_RFIFO_RST 0x00000008 +#define TIMBI2S_ICOR_FIFO_RST (TIMBI2S_ICOR_LFIFO_RST | TIMBI2S_ICOR_RFIFO_RST) +#define TIMBI2S_ICOR_SOFT_RST 0x00000010 +#define TIMBI2S_ICOR_WORD_SEL_LEFT_SHIFT 8 +#define TIMBI2S_ICOR_WORD_SEL_LEFT_MASK (0xff << 8) +#define TIMBI2S_ICOR_WORD_SEL_RIGHT_SHIFT 16 +#define TIMBI2S_ICOR_WORD_SEL_RIGHT_MASK (0xff << 16) +#define TIMBI2S_ICOR_CLK_MASTER 0x10000000 +#define TIMBI2S_ICOR_RX_ID 0x20000000 +#define TIMBI2S_ICOR_TX_ID 0x40000000 +#define TIMBI2S_ICOR_WORD_SEL 0x80000000 +#define TIMBI2S_BUS_FIFO 0x18 + +#define TIMBI2S_BUS_REG_AREA_SIZE (TIMBI2S_BUS_FIFO - \ + TIMBI2S_BUS_PRESCALE + 4) +#define TIMBI2S_FIRST_BUS_AREA_OFS 0x08 + +struct timbi2s_bus { + u32 flags; + u32 prescale; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_pcm_substream *substream; + unsigned buf_pos; + spinlock_t lock; /* mutual exclusion */ + u16 sample_rate; +}; + +#define BUS_RX 0x200 +#define BUS_MASTER 0x100 +#define BUS_INDEX_MASK 0xff +#define BUS_INDEX(b) ((b)->flags & BUS_INDEX_MASK) +#define BUS_IS_MASTER(b) ((b)->flags & BUS_MASTER) +#define BUS_IS_RX(b) ((b)->flags & BUS_RX) + +#define SET_BUS_INDEX(b, id) ((b)->flags = ((b)->flags & ~BUS_INDEX_MASK) | id) +#define SET_BUS_MASTER(b) ((b)->flags |= BUS_MASTER) +#define SET_BUS_RX(b) ((b)->flags |= BUS_RX) + +#define TIMBI2S_BUS_OFFSET(bus) (TIMBI2S_FIRST_BUS_AREA_OFS + \ + TIMBI2S_BUS_REG_AREA_SIZE * BUS_INDEX(bus)) + +struct timbi2s { + void __iomem *membase; + int irq; + struct tasklet_struct tasklet; + u32 main_clk; + unsigned num_busses; + struct timbi2s_bus busses[0]; +}; + +#define BITS_PER_CHANNEL 16 +#define NUM_CHANNELS 2 + +#define SAMPLE_SIZE ((NUM_CHANNELS * BITS_PER_CHANNEL) / 8) +#define NUM_PERIODS 32 +#define NUM_SAMPLES 256 + +static struct snd_pcm_hardware timbi2s_rx_hw = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, /* only stereo */ + .channels_max = 2, + .buffer_bytes_max = NUM_PERIODS * SAMPLE_SIZE * NUM_SAMPLES, + .period_bytes_min = SAMPLE_SIZE * NUM_SAMPLES, + .period_bytes_max = SAMPLE_SIZE * NUM_SAMPLES, + .periods_min = NUM_PERIODS, + .periods_max = NUM_PERIODS, +}; + +static struct snd_pcm_hardware timbi2s_tx_hw = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, /* only stereo */ + .channels_max = 2, + .buffer_bytes_max = NUM_PERIODS * SAMPLE_SIZE * NUM_SAMPLES, + .period_bytes_min = SAMPLE_SIZE * NUM_SAMPLES, + .period_bytes_max = SAMPLE_SIZE * NUM_SAMPLES, + .periods_min = NUM_PERIODS, + .periods_max = NUM_PERIODS, +}; + +static inline void timbi2s_bus_write(struct timbi2s_bus *bus, u32 val, u32 reg) +{ + struct timbi2s *i2s = snd_pcm_chip(bus->card); + + iowrite32(val, i2s->membase + TIMBI2S_BUS_OFFSET(bus) + reg); +} + +static inline u32 timbi2s_bus_read(struct timbi2s_bus *bus, u32 reg) +{ + struct timbi2s *i2s = snd_pcm_chip(bus->card); + + return ioread32(i2s->membase + TIMBI2S_BUS_OFFSET(bus) + reg); +} + +static u32 timbi2s_calc_prescale(u32 main_clk, u32 sample_rate) +{ + u32 halfbit_rate = sample_rate * BITS_PER_CHANNEL * NUM_CHANNELS * 2; + return main_clk / halfbit_rate; +} + +static int timbi2s_open(struct snd_pcm_substream *substream) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + struct snd_pcm_runtime *runtime = substream->runtime; + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d\n", __func__, substream, + BUS_INDEX(bus)); + + if (BUS_IS_RX(bus)) { + runtime->hw = timbi2s_rx_hw; + if (bus->sample_rate == 8000) { + runtime->hw.rates = SNDRV_PCM_RATE_8000; + runtime->hw.rate_min = 8000; + runtime->hw.rate_max = 8000; + } + } else + runtime->hw = timbi2s_tx_hw; + + bus->substream = substream; + + return 0; +} + +static int timbi2s_close(struct snd_pcm_substream *substream) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d\n", __func__, substream, + BUS_INDEX(bus)); + + bus->substream = NULL; + + return 0; +} + +static int timbi2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + struct timbi2s *i2s = snd_pcm_chip(card); + int err; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d\n", __func__, + substream, BUS_INDEX(bus)); + + bus->prescale = timbi2s_calc_prescale(i2s->main_clk, + params_rate(hw_params)); + + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + dev_dbg(snd_card_get_device_link(card), + "%s: Rate: %d, format: %d\n", __func__, params_rate(hw_params), + params_format(hw_params)); + + return 0; +} + +static int timbi2s_hw_free(struct snd_pcm_substream *substream) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + unsigned long flags; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p\n", __func__, substream); + + spin_lock_irqsave(&bus->lock, flags); + /* disable interrupts */ + timbi2s_bus_write(bus, 0, TIMBI2S_BUS_IER); + spin_unlock_irqrestore(&bus->lock, flags); + + /* disable TX and RX */ + timbi2s_bus_write(bus, TIMBI2S_ICOR_FIFO_RST | TIMBI2S_ICOR_SOFT_RST, + TIMBI2S_BUS_ICOR); + + return snd_pcm_lib_free_pages(substream); +} + +static int timbi2s_prepare(struct snd_pcm_substream *substream) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + struct snd_pcm_runtime *runtime = substream->runtime; + u32 data; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d, buffer: %d, period: %d\n", + __func__, substream, + BUS_INDEX(bus), (int)snd_pcm_lib_buffer_bytes(substream), + (int)snd_pcm_lib_period_bytes(substream)); + + if (runtime->dma_addr & 3 || runtime->buffer_size & 3) { + dev_err(snd_card_get_device_link(card), + "%s: Only word aligned data allowed\n", __func__); + return -EINVAL; + } + + if (runtime->channels != NUM_CHANNELS) { + dev_err(snd_card_get_device_link(card), + "%s: Number of channels unsupported %d\n", __func__, + runtime->channels); + return -EINVAL; + } + + /* reset */ + timbi2s_bus_write(bus, TIMBI2S_ICOR_FIFO_RST | TIMBI2S_ICOR_SOFT_RST, + TIMBI2S_BUS_ICOR); + + /* only masters have prescaling, don't write if not needed */ + if (BUS_IS_MASTER(bus)) + timbi2s_bus_write(bus, bus->prescale, TIMBI2S_BUS_PRESCALE); + + /* write word select */ + data = ((BITS_PER_CHANNEL << TIMBI2S_ICOR_WORD_SEL_LEFT_SHIFT) & + TIMBI2S_ICOR_WORD_SEL_LEFT_MASK) | + ((BITS_PER_CHANNEL << TIMBI2S_ICOR_WORD_SEL_RIGHT_SHIFT) & + TIMBI2S_ICOR_WORD_SEL_RIGHT_MASK); + timbi2s_bus_write(bus, data, TIMBI2S_BUS_ICOR); + + bus->buf_pos = 0; + + return 0; +} + +static int +timbi2s_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + unsigned long flags; + u32 data; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d, cmd: %d\n", __func__, + substream, BUS_INDEX(bus), cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dev_dbg(snd_card_get_device_link(card), + "%s: Got TRIGGER_START command\n", __func__); + + /* start */ + data = timbi2s_bus_read(bus, TIMBI2S_BUS_ICOR); + data |= TIMBI2S_ICOR_TX_ENABLE; + timbi2s_bus_write(bus, data, TIMBI2S_BUS_ICOR); + + /* enable interrupts */ + timbi2s_bus_write(bus, TIMBI2S_IRQ_TX_ALMOST_EMPTY, + TIMBI2S_BUS_IER); + dev_dbg(snd_card_get_device_link(card), + "%s: ISR: %x, ICOR: %x\n", __func__, + timbi2s_bus_read(bus, TIMBI2S_BUS_ISR), + timbi2s_bus_read(bus, TIMBI2S_BUS_ICOR)); + break; + case SNDRV_PCM_TRIGGER_STOP: + dev_dbg(snd_card_get_device_link(card), + "%s: Got TRIGGER_STOP command\n", __func__); + + spin_lock_irqsave(&bus->lock, flags); + /* disable interrupts */ + timbi2s_bus_write(bus, 0, TIMBI2S_BUS_IER); + spin_unlock_irqrestore(&bus->lock, flags); + + /* reset */ + data = timbi2s_bus_read(bus, TIMBI2S_BUS_ICOR); + data &= ~TIMBI2S_ICOR_TX_ENABLE; + + timbi2s_bus_write(bus, data, TIMBI2S_BUS_ICOR); + break; + default: + dev_dbg(snd_card_get_device_link(card), + "%s: Got unsupported command\n", __func__); + + return -EINVAL; + } + + return 0; +} + +static int +timbi2s_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + unsigned long flags; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d, cmd: %d\n", __func__, + substream, BUS_INDEX(bus), cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dev_dbg(snd_card_get_device_link(card), + "%s: Got TRIGGER_START command\n", __func__); + + timbi2s_bus_write(bus, TIMBI2S_ICOR_RX_ENABLE | + TIMBI2S_ICOR_FIFO_RST, TIMBI2S_BUS_ICOR); + + timbi2s_bus_write(bus, TIMBI2S_IRQ_RX_ALMOST_FULL, + TIMBI2S_BUS_IER); + break; + case SNDRV_PCM_TRIGGER_STOP: + dev_dbg(snd_card_get_device_link(card), + "%s: Got TRIGGER_STOP command\n", __func__); + /* disable interrupts */ + spin_lock_irqsave(&bus->lock, flags); + timbi2s_bus_write(bus, 0, TIMBI2S_BUS_IER); + spin_unlock_irqrestore(&bus->lock, flags); + /* Stop RX */ + timbi2s_bus_write(bus, 0, TIMBI2S_BUS_ICOR); + break; + default: + dev_dbg(snd_card_get_device_link(card), + "%s: Got unsupported command\n", __func__); + + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t +timbi2s_pointer(struct snd_pcm_substream *substream) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + snd_pcm_uframes_t ret; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p\n", __func__, substream); + + ret = bytes_to_frames(substream->runtime, bus->buf_pos); + if (ret >= substream->runtime->buffer_size) + ret -= substream->runtime->buffer_size; + + return ret; +} + +static struct snd_pcm_ops timbi2s_playback_ops = { + .open = timbi2s_open, + .close = timbi2s_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = timbi2s_hw_params, + .hw_free = timbi2s_hw_free, + .prepare = timbi2s_prepare, + .trigger = timbi2s_playback_trigger, + .pointer = timbi2s_pointer, +}; + +static struct snd_pcm_ops timbi2s_capture_ops = { + .open = timbi2s_open, + .close = timbi2s_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = timbi2s_hw_params, + .hw_free = timbi2s_hw_free, + .prepare = timbi2s_prepare, + .trigger = timbi2s_capture_trigger, + .pointer = timbi2s_pointer, +}; + +static void timbi2s_irq_process_rx(struct timbi2s_bus *bus) +{ + struct snd_pcm_runtime *runtime = bus->substream->runtime; + u32 buffer_size = snd_pcm_lib_buffer_bytes(bus->substream); + u32 ipr = timbi2s_bus_read(bus, TIMBI2S_BUS_IPR); + int i; + + dev_dbg(snd_card_get_device_link(bus->card), + "%s: Entry, bus: %d, IPR %x\n", __func__, BUS_INDEX(bus), ipr); + + for (i = 0; i < NUM_SAMPLES; i++) { + *(u32 *)(runtime->dma_area + bus->buf_pos) = + timbi2s_bus_read(bus, TIMBI2S_BUS_FIFO); + bus->buf_pos += SAMPLE_SIZE; + bus->buf_pos %= buffer_size; + } + + timbi2s_bus_write(bus, ipr, TIMBI2S_BUS_ICLR); + + /* inform ALSA that a period was received */ + snd_pcm_period_elapsed(bus->substream); +} + +static void timbi2s_irq_process_tx(struct timbi2s_bus *bus) +{ + struct snd_pcm_runtime *runtime = bus->substream->runtime; + u32 buffer_size = snd_pcm_lib_buffer_bytes(bus->substream); + u32 ipr = timbi2s_bus_read(bus, TIMBI2S_BUS_IPR); + int i; + + dev_dbg(snd_card_get_device_link(bus->card), + "%s: Entry, bus: %d, IPR %x\n", __func__, BUS_INDEX(bus), ipr); + + for (i = 0; i < NUM_SAMPLES; i++) { + timbi2s_bus_write(bus, + *(u32 *)(runtime->dma_area + bus->buf_pos), + TIMBI2S_BUS_FIFO); + bus->buf_pos += SAMPLE_SIZE; + bus->buf_pos %= buffer_size; + } + + dev_dbg(snd_card_get_device_link(bus->card), "%s: ISR: %x, ICOR: %x\n", + __func__, timbi2s_bus_read(bus, TIMBI2S_BUS_ISR), + timbi2s_bus_read(bus, TIMBI2S_BUS_ICOR)); + + timbi2s_bus_write(bus, ipr, TIMBI2S_BUS_ICLR); + + /* inform ALSA that a period was received */ + snd_pcm_period_elapsed(bus->substream); +} + +static void timbi2s_tasklet(unsigned long arg) +{ + struct snd_card *card = (struct snd_card *)arg; + struct timbi2s *i2s = snd_pcm_chip(card); + u32 uir = ioread32(i2s->membase + TIMBI2S_REG_UIR); + unsigned i; + + dev_dbg(snd_card_get_device_link(card), "%s: Entry, UIR %x\n", + __func__, uir); + + for (i = 0; i < i2s->num_busses; i++) + if (uir & (1 << i)) { + struct timbi2s_bus *bus = i2s->busses + i; + if (BUS_IS_RX(bus)) + timbi2s_irq_process_rx(bus); + else + timbi2s_irq_process_tx(bus); + } + + enable_irq(i2s->irq); +} + +static irqreturn_t timbi2s_irq(int irq, void *devid) +{ + struct timbi2s *i2s = devid; + + tasklet_schedule(&i2s->tasklet); + disable_irq_nosync(i2s->irq); + + return IRQ_HANDLED; +} + +static int timbi2s_setup_busses(struct snd_card *card, + struct platform_device *pdev) +{ + const struct timbi2s_platform_data *pdata = pdev->dev.platform_data; + unsigned i; + + dev_dbg(&pdev->dev, "%s: Entry, no busses: %d, busses: %p\n", __func__, + pdata->num_busses, pdata->busses); + + for (i = 0; i < pdata->num_busses; i++) { + int capture = pdata->busses[i].rx; + int err; + u32 ctl; + struct timbi2s *i2s = snd_pcm_chip(card); + struct timbi2s_bus *bus = i2s->busses + i; + + dev_dbg(&pdev->dev, "%s: Setting up bus: %d\n", __func__, i); + + SET_BUS_INDEX(bus, i); + bus->sample_rate = pdata->busses[i].sample_rate; + bus->card = card; + /* prescaling only applies to master busses, we use the + * knowledge of that to identify the direction later + * eg, bus->prescale != 0 -> master bus + */ + if (capture) + SET_BUS_RX(bus); + + spin_lock_init(&bus->lock); + + if (bus->sample_rate != 44100 && bus->sample_rate != 8000) { + dev_err(&pdev->dev, + "Unsupported bitrate: %d\n", bus->sample_rate); + return -EINVAL; + } + + dev_dbg(&pdev->dev, "%s: Will check HW direction on bus: %d\n", + __func__, BUS_INDEX(bus)); + + /* check that the HW agrees with the direction */ + ctl = timbi2s_bus_read(bus, TIMBI2S_BUS_ICOR); + if ((capture && !(ctl & TIMBI2S_ICOR_RX_ID)) || + (!capture && !(ctl & TIMBI2S_ICOR_TX_ID))) { + dev_dbg(&pdev->dev, + "HW and platform data disagree on direction\n"); + return -EINVAL; + } + + dev_dbg(&pdev->dev, "%s: Will create PCM channel for bus: %d\n", + __func__, BUS_INDEX(bus)); + err = snd_pcm_new(card, card->shortname, i, !capture, + capture, &bus->pcm); + if (err) { + dev_dbg(&pdev->dev, "%s, Failed to create pcm: %d\n", + __func__, err); + return err; + } + + if (capture) + snd_pcm_set_ops(bus->pcm, SNDRV_PCM_STREAM_CAPTURE, + &timbi2s_capture_ops); + if (!capture) + snd_pcm_set_ops(bus->pcm, SNDRV_PCM_STREAM_PLAYBACK, + &timbi2s_playback_ops); + + dev_dbg(&pdev->dev, "%s: Will preallocate buffers to bus: %d\n", + __func__, BUS_INDEX(bus)); + + err = snd_pcm_lib_preallocate_pages_for_all(bus->pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + NUM_SAMPLES * NUM_PERIODS * SAMPLE_SIZE * 2, + NUM_SAMPLES * NUM_PERIODS * SAMPLE_SIZE * 2); + if (err) { + dev_dbg(&pdev->dev, "%s, Failed to create pcm: %d\n", + __func__, err); + + return err; + } + + bus->pcm->private_data = bus; + bus->pcm->info_flags = 0; + strcpy(bus->pcm->name, card->shortname); + i2s->num_busses++; + } + + return 0; +} + +static int __devinit timbi2s_probe(struct platform_device *pdev) +{ + int err; + int irq; + struct timbi2s *i2s; + struct resource *iomem; + const struct timbi2s_platform_data *pdata = pdev->dev.platform_data; + struct snd_card *card; + u32 ver; + + if (!pdata) { + err = -ENODEV; + goto out; + } + + if (pdata->num_busses > MAX_BUSSES) { + err = -EINVAL; + goto out; + } + + iomem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iomem) { + err = -ENODEV; + goto out; + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + err = -ENODEV; + goto out; + } + + err = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, sizeof(struct timbi2s) + + sizeof(struct timbi2s_bus) * pdata->num_busses, &card); + if (err) + goto out; + + strcpy(card->driver, "Timberdale I2S"); + strcpy(card->shortname, "Timberdale I2S"); + sprintf(card->longname, "Timberdale I2S Driver"); + + snd_card_set_dev(card, &pdev->dev); + + i2s = snd_pcm_chip(card); + + if (!request_mem_region(iomem->start, resource_size(iomem), + DRIVER_NAME)) { + err = -EBUSY; + goto err_region; + } + + i2s->membase = ioremap(iomem->start, resource_size(iomem)); + if (!i2s->membase) { + err = -ENOMEM; + goto err_ioremap; + } + + err = timbi2s_setup_busses(card, pdev); + if (err) + goto err_setup; + + tasklet_init(&i2s->tasklet, timbi2s_tasklet, (unsigned long)card); + i2s->irq = irq; + i2s->main_clk = pdata->main_clk; + + err = request_irq(irq, timbi2s_irq, 0, DRIVER_NAME, i2s); + if (err) + goto err_request_irq; + + err = snd_card_register(card); + if (err) + goto err_register; + + platform_set_drvdata(pdev, card); + + ver = ioread32(i2s->membase + TIMBI2S_REG_VER); + + printk(KERN_INFO + "Driver for Timberdale I2S (ver: %d.%d) successfully probed.\n", + ver >> 16 , ver & 0xffff); + + return 0; + +err_register: + free_irq(irq, card); +err_request_irq: +err_setup: + iounmap(i2s->membase); +err_ioremap: + release_mem_region(iomem->start, resource_size(iomem)); +err_region: + snd_card_free(card); +out: + printk(KERN_ERR DRIVER_NAME": Failed to register: %d\n", err); + + return err; +} + +static int __devexit timbi2s_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct timbi2s *i2s = snd_pcm_chip(card); + struct resource *iomem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + tasklet_kill(&i2s->tasklet); + free_irq(i2s->irq, i2s); + + iounmap(i2s->membase); + release_mem_region(iomem->start, resource_size(iomem)); + snd_card_free(card); + + platform_set_drvdata(pdev, 0); + return 0; +} + +static struct platform_driver timbi2s_platform_driver = { + .driver = { + .name = DRIVER_NAME, + .owner = THIS_MODULE, + }, + .probe = timbi2s_probe, + .remove = __devexit_p(timbi2s_remove), +}; + +/*--------------------------------------------------------------------------*/ + +static int __init timbi2s_init(void) +{ + return platform_driver_register(&timbi2s_platform_driver); +} + +static void __exit timbi2s_exit(void) +{ + platform_driver_unregister(&timbi2s_platform_driver); +} + +module_init(timbi2s_init); +module_exit(timbi2s_exit); + +MODULE_ALIAS("platform:"DRIVER_NAME); +MODULE_DESCRIPTION("Timberdale I2S bus driver"); +MODULE_AUTHOR("Mocean Laboratories <info(a)mocean-labs.com>"); +MODULE_LICENSE("GPL v2");

3 6

[alsa-devel] hw_ptr_interrupt removal broke interrupt pointer updates
by Clemens Ladisch 01 Feb '10

01 Feb '10

Commit "cleanup & merge hw_ptr update functions" says: > The main change is hw_ptr_interrupt variable removal to simplify code > logic. This variable can be computed directly from hw_ptr. The hw_ptr_interrupt variable was needed to differentiate between the position at the last normal pointer update and the position of the last signaled period boundary. if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ delta = old_hw_ptr - (old_hw_ptr % runtime->period_size) + runtime->period_size; if (delta > new_hw_ptr) { hw_base += runtime->buffer_size; It is possible for the status/delay ioctls to be called when the sound card's pointer register alreay shows a position at the beginning of the new period, but immediately before the interrupt is actually executed. (This happens regularly on a SMP machine with mplayer.) When that happens, the code thinks that the position must be at least one period ahead of the current position and drops an entire buffer of data. Best regards, Clemens

4 8

[alsa-devel] [PATCH] ASoC: improve MCLKDIV calculation in wm8978, when OPCLK is not used
by Guennadi Liakhovetski 01 Feb '10

01 Feb '10

In case, if OPCLK is not used, and PLL is used for driving the codec, the choice of PLL output frequency could result in a needlessly imprecise system clock frequency. Use an iterative process to select a precise configuration. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 6d807b1..61b0925 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -58,6 +58,7 @@ struct wm8978_priv { unsigned int f_mclk; unsigned int f_256fs; unsigned int f_opclk; + int mclk_idx; enum wm8978_sysclk_src sysclk; u16 reg_cache[WM8978_CACHEREGNUM]; }; @@ -402,6 +403,35 @@ static void pll_factors(struct wm8978_pll_div *pll_div, unsigned int target, pll_div->k = k; } + +/* MCLK dividers */ +static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; +static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; + +/* + * find index >= idx, such that, for a given f_out, + * 3 * f_mclk / 4 <= f_PLLOUT < 13 * f_mclk / 4 + * f_out can be f_256fs or f_opclk, currently only used for f_256fs. Can be + * generalised for f_opclk with suitable coefficient arrays, but currently + * the OPCLK divisor is calculated directly, not iteratively. + */ +static int wm8978_enum_mclk(unsigned int f_out, unsigned int f_mclk, + unsigned int *f_pllout) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + unsigned int f_pllout_x4 = 4 * f_out * mclk_numerator[i] / + mclk_denominator[i]; + if (3 * f_mclk <= f_pllout_x4 && f_pllout_x4 < 13 * f_mclk) { + *f_pllout = f_pllout_x4 / 4; + return i; + } + } + + return -EINVAL; +} + /* * Calculate internal frequencies and dividers, according to Figure 40 * "PLL and Clock Select Circuit" in WM8978 datasheet Rev. 2.6 @@ -412,12 +442,16 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) struct wm8978_pll_div pll_div; unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk, f_256fs = wm8978->f_256fs; - unsigned int f2, opclk_div; + unsigned int f2; if (!f_mclk) return -EINVAL; if (f_opclk) { + unsigned int opclk_div; + /* Cannot set up MCLK divider now, do later */ + wm8978->mclk_idx = -1; + /* * The user needs OPCLK. Choose OPCLKDIV to put * 6 <= R = f2 / f1 < 13, 1 <= OPCLKDIV <= 4. @@ -444,7 +478,7 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) wm8978->f_pllout = f_opclk * opclk_div; } else if (f_256fs) { /* - * Not using OPCLK, choose R: + * Not using OPCLK, but PLL is used for the codec, choose R: * 6 <= R = f2 / f1 < 13, to put 1 <= MCLKDIV <= 12. * f_256fs = f_mclk * prescale * R / 4 / MCLKDIV, where * prescale = 1, or prescale = 2. Prescale is calculated inside @@ -453,18 +487,11 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) * f_mclk * 3 / 48 <= f_256fs < f_mclk * 13 / 4. This means MCLK * must be 3.781MHz <= f_MCLK <= 32.768MHz */ - if (48 * f_256fs < 3 * f_mclk || 4 * f_256fs >= 13 * f_mclk) - return -EINVAL; + int idx = wm8978_enum_mclk(f_256fs, f_mclk, &wm8978->f_pllout); + if (idx < 0) + return idx; - /* - * MCLKDIV will be selected in .hw_params(), just choose a - * suitable f_PLLOUT - */ - if (4 * f_256fs < 3 * f_mclk) - /* Will have to use MCLKDIV */ - wm8978->f_pllout = wm8978->f_mclk * 3 / 4; - else - wm8978->f_pllout = f_256fs; + wm8978->mclk_idx = idx; /* GPIO1 into default mode as input - before configuring PLL */ snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); @@ -515,6 +542,20 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, wm8978->f_opclk = div; if (wm8978->f_mclk) + /* + * We know the MCLK frequency, the user has requested + * OPCLK, configure the PLL based on that and start it + * and OPCLK immediately. We will configure PLL to match + * user-requested OPCLK frquency as good as possible. + * In fact, it is likely, that matching the sampling + * rate, when it becomes known, is more important, and + * we will not be reconfiguring PLL then, because we + * must not interrupt OPCLK. But it should be fine, + * because typically the user will request OPCLK to run + * at 256fs or 512fs, and for these cases we will also + * find an exact MCLK divider configuration - it will + * be equal to or double the OPCLK divisor. + */ ret = wm8978_configure_pll(codec); break; case WM8978_BCLKDIV: @@ -640,10 +681,6 @@ static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -/* MCLK dividers */ -static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; -static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; - /* * Set PCM DAI bit size and sample rate. */ @@ -709,9 +746,11 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, wm8978->f_256fs = params_rate(params) * 256; if (wm8978->sysclk == WM8978_MCLK) { + wm8978->mclk_idx = -1; f_sel = wm8978->f_mclk; } else { if (!wm8978->f_pllout) { + /* We only enter here, if OPCLK is not used */ int ret = wm8978_configure_pll(codec); if (ret < 0) return ret; @@ -719,32 +758,34 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, f_sel = wm8978->f_pllout; } - /* - * In some cases it is possible to reconfigure PLL to a higher frequency - * by raising OPCLKDIV, but normally OPCLK is configured to 256 * fs or - * 512 * fs, so, we should be fine. - */ - if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) - return -EINVAL; + if (wm8978->mclk_idx < 0) { + /* Either MCLK is used directly, or OPCLK is used */ + if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) + return -EINVAL; - for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { - diff = abs(wm8978->f_256fs * 3 - - f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + diff = abs(wm8978->f_256fs * 3 - + f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); - if (diff < diff_best) { - diff_best = diff; - best = i; - } + if (diff < diff_best) { + diff_best = diff; + best = i; + } - if (!diff) - break; + if (!diff) + break; + } + } else { + /* OPCLK not used, codec driven by PLL */ + best = wm8978->mclk_idx; + diff = 0; } if (diff) - dev_warn(codec->dev, "Imprecise clock: %u%s\n", - f_sel * mclk_denominator[best] / mclk_numerator[best], - wm8978->sysclk == WM8978_MCLK ? - ", consider using PLL" : ""); + dev_warn(codec->dev, "Imprecise sampling rate: %uHz%s\n", + f_sel * mclk_denominator[best] / mclk_numerator[best] / 256, + wm8978->sysclk == WM8978_MCLK ? + ", consider using PLL" : ""); dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, params_format(params), params_rate(params), best);

2 1

[alsa-devel] Some installation questions
by Klaus Schulz 01 Feb '10

01 Feb '10

Hi there. I am still maintaining the (inoffical) Ubuntu Alsa Upgrade Script over at http://ubuntuforums.org/showthread.php?t=1046137 Perhaps somebody can give me a hint on some installation problems I am running into: 1. Quite some people reporting that /proc/asound/version is still showing the "original" revision after running the upgrade-script instead of 1.0.22.1. What could be the issue? What is setting /proc/asound/version? 2. I also introduced a kind of restore function. What it does is basically reinstalling all the required Ubuntu packages from the repo to overwrite the (by my script) upgraded files again. a. My first problem was that I was only able to get the original modules activated when running the module assistant. ( module-assistant -f -t -q a-i alsa-source ) Just a re-installation of kernel packages and alsa-packages didn't work. b. When redoing the Alsa Upgrade after the restore (by my script) , I still see in /proc/asound the old Alsa version. Looks again a bit like I am having slight problems with the module handling FYI: Below I run after the "make install" of the driver package The way the .ko files are handled is that they are copied to /lib/modules/`uname -r`/kernel/sound/ ##################### cd ${ALSASRCDIR}/${DRIVER}/ find ./ -name ''*.ko'' > /tmp/alsa_modules header "Copy modules to target directories..." if [ "`uname -a| grep "2.6.24"`" != "" ] ; then tar -cv -T /tmp/alsa_modules -f /lib/modules/`uname -r`/ubuntu/sound/alsa-driver/${DRIVER}.tar cd /lib/modules/`uname -r`/ubuntu/sound/alsa-driver else tar -cv -T /tmp/alsa_modules -f /lib/modules/`uname -r`/kernel/sound/${DRIVER}.tar cd /lib/modules/`uname -r`/kernel/sound/ fi #Extract new modules, overwriting old ones tar -xvf ${DRIVER}.tar rm *.tar depmod -a ############################## I obviously lack some kind of understanding how all this works. Perhaps somebody can show me directions how to tackle the subject. THX \Klaus

2 1

[alsa-devel] No sound with DH55TC motherboard
by Fabienne Ducroquet 31 Jan '10

31 Jan '10

Hi, I recently bought an Intel DH55TC motherboard, and I can't get the sound card to work. The sound card is: 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) with a Realtek ALC888S codec chip. Output from dmesg: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: irq 28 for MSI/MSI-X HDA Intel 0000:00:1b.0: setting latency timer to 64 hda_codec: ALC888: BIOS auto-probing. autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) speaker_outs=1 (0x15/0x0/0x0/0x0/0x0) hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) mono: mono_out=0x0 dig-out=0x1e/0x0 inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x0, aux=0x0 realtek: No valid SSID, checking pincfg 0x4016c629 for NID 0x1d realtek: Enabling init ASM_ID=0xc629 CODEC_ID=10ec0888 The snd_hda_codec_intelhdmi and snd_hda_codec_realtek modules are loaded, but there is no /proc/asound/modules file and any program trying to use sound says there is no sound card. I tried all of the models listed under ALC888 in HD-Audio-Models.txt, but none of them worked. I tried with a 2.6.32 kernel from Debian, as well as with a 2.6.33-rc6 kernel and with the sound-2.6 tree from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git pulled on top of a 2.6.33-rc6 kernel, and the results were the same in all three cases. If I compile snd_hda_intel in the kernel, I get the lines: ALSA device list: #0: HDA Intel at 0xfe420000 irq 28 after the lines from HDA quoted above in dmesg, while when it's compiled as a module I get: ALSA device list: No soundcards found before those lines, but apart from that the behavior is the same. Attached is the output from alsa-info.sh. Regards, Fabienne

1 0

[alsa-devel] Warning/error from ALSA’s pcm_min.c example. Possible problem?
by Louise Hoffman 31 Jan '10

31 Jan '10

Dear ALSA developers, When I compile ALSA's pcm_min.c[0] example with gcc -Wall -lasound pcm_min.c -o pcm_min Everything is fine, but running it, I get the white noise as expected, but I also get this warning/error: Short write (expected 16384, wrote 7616) Which comes from the last if-statement. One bug is in this line frames = snd_pcm_writei(handle, buffer, sizeof(buffer)); where "sizeof(buffer)" should be "number of frames in the buffer". But there are still some strange things going on. I still get those errors, and if I remove the for-loop, nothing is played. Here is my code. [1] Can someone see what's wrong? Hugs, Louise [0] http://www.google.com/codesearch/p?hl=en&sa=N&cd=1&ct=rc#4FSOSMZ6Pxc/distfi… [1] http://pastebin.com/m2f28b578

3 4

[alsa-devel] [Q] playing / recording mono on a stereo-only hardware
by Guennadi Liakhovetski 30 Jan '10

30 Jan '10

Hi I know this question belongs rather to alsa-users, and I did ask it there, but got no solution so far. Maybe someone here can help. The recently posted SuperH SIU driver only supports stereo playback and capture (channels_min = channels_max = 2). Is there a way to tell alsa-lib to duplicate the mono-channel during playback and to mix recorded channels (or use one of them), if mono recording is requested? Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

2 2

[alsa-devel] [PATCH 5/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 30 Jan '10

30 Jan '10

5/5 Patchin' patcher: This patch updates alsa-driver echoaudio.patch . Short description: This patch updates alsa-driver echoaudio.patch . Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- alsa-driver-1.0.22.1/pci/echoaudio/echoaudio.patch__orig 2009-12-28 16:15:54.000000000 +0100 +++ alsa-driver-1.0.22.1/pci/echoaudio/echoaudio.patch 2010-01-29 23:35:16.000000000 +0100 @@ -1,6 +1,6 @@ ---- ../../alsa-kernel/pci/echoaudio/echoaudio.c 2006-08-05 14:41:12.000000000 +0200 -+++ echoaudio.c 2006-10-07 20:30:53.000000000 +0200 -@@ -1926,6 +1926,7 @@ +--- ../../alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:17:31.000000000 +0100 ++++ echoaudio.c 2010-01-29 23:28:45.000000000 +0100 +@@ -1943,6 +1943,7 @@ static __devinit int snd_echo_create(str struct echoaudio *chip; int err; size_t sz; @@ -8,7 +8,7 @@ static struct snd_device_ops ops = { .dev_free = snd_echo_dev_free, }; -@@ -1988,7 +1989,12 @@ +@@ -2014,7 +2015,12 @@ static __devinit int snd_echo_create(str chip->comm_page_phys = chip->commpage_dma_buf.addr; chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area; @@ -19,6 +19,28 @@ + pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsystem_device); +#endif + err = init_hw(chip, chip->pci->device, subsystem_device); - if (err) { - DE_INIT(("init_hw err=%d\n", err)); - snd_echo_free(chip); + if (err >= 0) + err = set_mixer_defaults(chip); + if (err < 0) { +@@ -2245,6 +2251,7 @@ static int snd_echo_resume(struct pci_de + struct comm_page *commpage, *commpage_bak; + u32 pipe_alloc_mask; + int err; ++ u16 subsystem_device; + + DE_INIT(("resume start\n")); + pci_restore_state(pci); +@@ -2252,7 +2259,12 @@ static int snd_echo_resume(struct pci_de + commpage = chip->comm_page; + memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + +- err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); ++#if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 3, 0) ++ subsystem_device = pci->subsystem_device; ++#else ++ pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsystem_device); ++#endif ++ err = init_hw(chip, chip->pci->device, subsystem_device); + if (err < 0) { + kfree(commpage_bak); + DE_INIT(("resume init_hw err=%d\n", err)); -- Giuliano.

1 0

[alsa-devel] [PATCH 4/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 30 Jan '10

30 Jan '10

4/5 Add suspend support #2: This patch adds suspend and resume callbacks. It also rearranges parts of the initialization code so it can be used in both the first initialization (when the module is loaded we also have to load default settings) and the resume callback (where we have to restore the previous settings). P.S. I'll fix "rettings" later. Short description: This patch adds rearranges parts of the initialization code and adds suspend and resume callbacks. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:09:09.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:11:43.000000000 +0100 @@ -753,6 +753,8 @@ static int pcm_trigger(struct snd_pcm_su spin_lock(&chip->lock); switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + DE_ACT(("pcm_trigger resume\n")); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: DE_ACT(("pcm_trigger start\n")); @@ -776,6 +778,8 @@ static int pcm_trigger(struct snd_pcm_su err = start_transport(chip, channelmask, chip->pipe_cyclic_mask); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + DE_ACT(("pcm_trigger suspend\n")); case SNDRV_PCM_TRIGGER_STOP: DE_ACT(("pcm_trigger stop\n")); for (i = 0; i < DSP_MAXPIPES; i++) { @@ -1951,18 +1955,27 @@ static __devinit int snd_echo_create(str return err; pci_set_master(pci); - /* allocate a chip-specific data */ - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (!chip) { - pci_disable_device(pci); - return -ENOMEM; + /* Allocate chip if needed */ + if (!*rchip) { + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + pci_disable_device(pci); + return -ENOMEM; + } + DE_INIT(("chip=%p\n", chip)); + spin_lock_init(&chip->lock); + chip->card = card; + chip->pci = pci; + chip->irq = -1; + atomic_set(&chip->opencount, 0); + mutex_init(&chip->mode_mutex); + chip->can_set_rate = 1; + } else { + /* If this was called from the resume function, chip is + * already allocated and it contains current card settings. + */ + chip = *rchip; } - DE_INIT(("chip=%p\n", chip)); - - spin_lock_init(&chip->lock); - chip->card = card; - chip->pci = pci; - chip->irq = -1; /* PCI resource allocation */ chip->dsp_registers_phys = pci_resource_start(pci, 0); @@ -2002,7 +2015,9 @@ static __devinit int snd_echo_create(str chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area; err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); - if (err) { + if (err >= 0) + err = set_mixer_defaults(chip); + if (err < 0) { DE_INIT(("init_hw err=%d\n", err)); snd_echo_free(chip); return err; @@ -2013,9 +2028,6 @@ static __devinit int snd_echo_create(str snd_echo_free(chip); return err; } - atomic_set(&chip->opencount, 0); - mutex_init(&chip->mode_mutex); - chip->can_set_rate = 1; *rchip = chip; /* Init done ! */ return 0; @@ -2048,6 +2060,7 @@ static int __devinit snd_echo_probe(stru snd_card_set_dev(card, &pci->dev); + chip = NULL; /* Tells snd_echo_create to allocate chip */ if ((err = snd_echo_create(card, pci, &chip)) < 0) { snd_card_free(card); return err; @@ -2187,6 +2200,112 @@ ctl_error: +#if defined(CONFIG_PM) + +static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + + DE_INIT(("suspend start\n")); + snd_pcm_suspend_all(chip->analog_pcm); + snd_pcm_suspend_all(chip->digital_pcm); + +#ifdef ECHOCARD_HAS_MIDI + /* This call can sleep */ + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 0); +#endif + spin_lock_irq(&chip->lock); + if (wait_handshake(chip)) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + clear_handshake(chip); + if (send_vector(chip, DSP_VC_GO_COMATOSE) < 0) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + spin_unlock_irq(&chip->lock); + + chip->dsp_code = NULL; + free_irq(chip->irq, chip); + chip->irq = -1; + pci_save_state(pci); + pci_disable_device(pci); + + DE_INIT(("suspend done\n")); + return 0; +} + + + +static int snd_echo_resume(struct pci_dev *pci) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + struct comm_page *commpage, *commpage_bak; + u32 pipe_alloc_mask; + int err; + + DE_INIT(("resume start\n")); + pci_restore_state(pci); + commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + commpage = chip->comm_page; + memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + + err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); + if (err < 0) { + kfree(commpage_bak); + DE_INIT(("resume init_hw err=%d\n", err)); + snd_echo_free(chip); + return err; + } + DE_INIT(("resume init OK\n")); + + /* Temporarily set chip->pipe_alloc_mask=0 otherwise + * restore_dsp_settings() fails. + */ + pipe_alloc_mask = chip->pipe_alloc_mask; + chip->pipe_alloc_mask = 0; + err = restore_dsp_rettings(chip); + chip->pipe_alloc_mask = pipe_alloc_mask; + if (err < 0) { + kfree(commpage_bak); + return err; + } + DE_INIT(("resume restore OK\n")); + + memcpy(&commpage->audio_format, &commpage_bak->audio_format, + sizeof(commpage->audio_format)); + memcpy(&commpage->sglist_addr, &commpage_bak->sglist_addr, + sizeof(commpage->sglist_addr)); + memcpy(&commpage->midi_output, &commpage_bak->midi_output, + sizeof(commpage->midi_output)); + kfree(commpage_bak); + + if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, + ECHOCARD_NAME, chip)) { + snd_echo_free(chip); + snd_printk(KERN_ERR "cannot grab irq\n"); + return -EBUSY; + } + chip->irq = pci->irq; + DE_INIT(("resume irq=%d\n", chip->irq)); + +#ifdef ECHOCARD_HAS_MIDI + if (chip->midi_input_enabled) + enable_midi_input(chip, TRUE); + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 1); +#endif + + DE_INIT(("resume done\n")); + return 0; +} + +#endif /* CONFIG_PM */ + + + static void __devexit snd_echo_remove(struct pci_dev *pci) { struct echoaudio *chip; @@ -2209,6 +2328,10 @@ static struct pci_driver driver = { .id_table = snd_echo_ids, .probe = snd_echo_probe, .remove = __devexit_p(snd_echo_remove), +#ifdef CONFIG_PM + .suspend = snd_echo_suspend, + .resume = snd_echo_resume, +#endif /* CONFIG_PM */ }; Only in alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/: echoaudio.c.orig diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -497,9 +497,6 @@ static int load_firmware(struct echoaudi if ((box_type = load_asic(chip)) < 0) return box_type; /* error */ - if ((err = restore_dsp_rettings(chip)) < 0) - return err; - return box_type; } @@ -659,51 +656,106 @@ static void get_audio_meters(struct echo static int restore_dsp_rettings(struct echoaudio *chip) { - int err; + int i, o, err; DE_INIT(("restore_dsp_settings\n")); if ((err = check_asic_status(chip)) < 0) return err; - /* @ Gina20/Darla20 only. Should be harmless for other cards. */ + /* Gina20/Darla20 only. Should be harmless for other cards. */ chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF; chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF; chip->comm_page->handshake = 0xffffffff; - if ((err = set_sample_rate(chip, chip->sample_rate)) < 0) + /* Restore output busses */ + for (i = 0; i < num_busses_out(chip); i++) { + err = set_output_gain(chip, i, chip->output_gain[i]); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_VMIXER + for (i = 0; i < num_pipes_out(chip); i++) + for (o = 0; o < num_busses_out(chip); o++) { + err = set_vmixer_gain(chip, o, i, + chip->vmixer_gain[o][i]); + if (err < 0) + return err; + } + if (update_vmixer_level(chip) < 0) + return -EIO; +#endif /* ECHOCARD_HAS_VMIXER */ + +#ifdef ECHOCARD_HAS_MONITOR + for (o = 0; o < num_busses_out(chip); o++) + for (i = 0; i < num_busses_in(chip); i++) { + err = set_monitor_gain(chip, o, i, + chip->monitor_gain[o][i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_MONITOR */ + +#ifdef ECHOCARD_HAS_INPUT_GAIN + for (i = 0; i < num_busses_in(chip); i++) { + err = set_input_gain(chip, i, chip->input_gain[i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_INPUT_GAIN */ + + err = update_output_line_level(chip); + if (err < 0) return err; - if (chip->meters_enabled) - if (send_vector(chip, DSP_VC_METERS_ON) < 0) - return -EIO; + err = update_input_line_level(chip); + if (err < 0) + return err; -#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK - if (set_input_clock(chip, chip->input_clock) < 0) + err = set_sample_rate(chip, chip->sample_rate); + if (err < 0) + return err; + + if (chip->meters_enabled) { + err = send_vector(chip, DSP_VC_METERS_ON); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH + if (set_digital_mode(chip, chip->digital_mode) < 0) return -EIO; #endif -#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH - if (set_output_clock(chip, chip->output_clock) < 0) +#ifdef ECHOCARD_HAS_DIGITAL_IO + if (set_professional_spdif(chip, chip->professional_spdif) < 0) return -EIO; #endif - if (update_output_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_PHANTOM_POWER + if (set_phantom_power(chip, chip->phantom_power) < 0) return -EIO; +#endif - if (update_input_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK + /* set_input_clock() also restores automute setting */ + if (set_input_clock(chip, chip->input_clock) < 0) return -EIO; +#endif -#ifdef ECHOCARD_HAS_VMIXER - if (update_vmixer_level(chip) < 0) +#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH + if (set_output_clock(chip, chip->output_clock) < 0) return -EIO; #endif if (wait_handshake(chip) < 0) return -EIO; clear_handshake(chip); + if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) + return -EIO; DE_INIT(("restore_dsp_rettings done\n")); - return send_vector(chip, DSP_VC_UPDATE_FLAGS); + return 0; } @@ -920,9 +972,6 @@ static int init_dsp_comm_page(struct ech chip->card_name = ECHOCARD_NAME; chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->digital_mode = DIGITAL_MODE_NONE; - chip->input_clock = ECHO_CLOCK_INTERNAL; - chip->output_clock = ECHO_CLOCK_WORD; chip->asic_loaded = FALSE; memset(chip->comm_page, 0, sizeof(struct comm_page)); @@ -933,7 +982,6 @@ static int init_dsp_comm_page(struct ech chip->comm_page->midi_out_free_count = cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); chip->comm_page->sample_rate = cpu_to_le32(44100); - chip->sample_rate = 44100; /* Set line levels so we don't blast any inputs on startup */ memset(chip->comm_page->monitors, ECHOGAIN_MUTED, MONITOR_ARRAY_SIZE); @@ -944,50 +992,21 @@ static int init_dsp_comm_page(struct ech -/* This function initializes the several volume controls for busses and pipes. -This MUST be called after the DSP is up and running ! */ +/* This function initializes the chip structure with default values, ie. all + * muted and internal clock source. Then it copies the settings to the DSP. + * This MUST be called after the DSP is up and running ! + */ static int init_line_levels(struct echoaudio *chip) { - int st, i, o; - DE_INIT(("init_line_levels\n")); - - /* Mute output busses */ - for (i = 0; i < num_busses_out(chip); i++) - if ((st = set_output_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; - -#ifdef ECHOCARD_HAS_VMIXER - /* Mute the Vmixer */ - for (i = 0; i < num_pipes_out(chip); i++) - for (o = 0; o < num_busses_out(chip); o++) - if ((st = set_vmixer_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_vmixer_level(chip))) - return st; -#endif /* ECHOCARD_HAS_VMIXER */ - -#ifdef ECHOCARD_HAS_MONITOR - /* Mute the monitor mixer */ - for (o = 0; o < num_busses_out(chip); o++) - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_monitor_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_MONITOR */ - -#ifdef ECHOCARD_HAS_INPUT_GAIN - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_input_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_input_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_INPUT_GAIN */ - - return 0; + memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); + memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); + memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); + memset(chip->vmixer_gain, ECHOGAIN_MUTED, sizeof(chip->vmixer_gain)); + chip->input_clock = ECHO_CLOCK_INTERNAL; + chip->output_clock = ECHO_CLOCK_WORD; + chip->sample_rate = 44100; + return restore_dsp_rettings(chip); } diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:08:36.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:11:43.000000000 +0100 @@ -472,6 +472,8 @@ static void free_firmware(const struct f #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); +static void snd_echo_midi_output_trigger( + struct snd_rawmidi_substream *substream, int up); static int midi_service_irq(struct echoaudio *chip); static int __devinit snd_echo_midi_create(struct snd_card *card, struct echoaudio *chip); -- Giuliano.

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[alsa-devel] [PATCH 3/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 30 Jan '10

30 Jan '10

3/5 Add suspend support #1: This patch moves the code that initializes the card's controls with default valued from the init_hw() function into a separated set_mixer_defaults() function (one for each of the 16 supported cards). This change is necessary because during resume we must resurrect the hardware without losing the previous settings. set_mixer_defaults() must be called only once when the module is loaded. Short description: Move the controls init code outside the init_hw() function because is must not be called during resume. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla20_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -57,15 +57,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + /* The Darla20 has no external clock sources */ static u32 detect_input_clocks(const struct echoaudio *chip) { diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla24_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -56,15 +56,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echo3g_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echo3g_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -97,20 +97,6 @@ static int init_hw(struct echoaudio *chi chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->non_audio_spdif = FALSE; - chip->bad_board = FALSE; - - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_phantom_power(chip, 0); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); DE_INIT(("init_hw done\n")); return err; @@ -118,6 +104,18 @@ static int init_hw(struct echoaudio *chi +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->non_audio_spdif = FALSE; + chip->bad_board = FALSE; + chip->phantom_power = FALSE; + return init_line_levels(chip); +} + + + static int set_phantom_power(struct echoaudio *chip, char on) { u32 control_reg = le32_to_cpu(chip->comm_page->control_register); diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina20_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -62,17 +62,20 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina24_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -57,9 +57,6 @@ static int init_hw(struct echoaudio *chi ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 | ECHO_CLOCK_BIT_ADAT; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { @@ -81,19 +78,22 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodj_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodj_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodjx_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_express_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_express_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_express_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_express_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -61,6 +61,7 @@ static int set_sample_rate(struct echoau control_reg |= clock; if (control_reg != old_control_reg) { + DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock)); chip->comm_page->control_register = cpu_to_le32(control_reg); chip->sample_rate = rate; clear_handshake(chip); diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoio_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoiox_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla20_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -64,17 +64,20 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla24_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -61,9 +61,6 @@ static int init_hw(struct echoaudio *chi ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; if ((err = load_firmware(chip)) < 0) return err; @@ -72,17 +69,22 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip)) < 0) return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mia_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mia_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -66,15 +66,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip))) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mona_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mona_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -67,28 +67,26 @@ static int init_hw(struct echoaudio *chi else chip->dsp_code_to_load = FW_MONA_301_DSP; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - if ((err = load_firmware(chip)) < 0) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; -- Giuliano.

1 0

[alsa-devel] [PATCH 2/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 30 Jan '10

30 Jan '10

2/4 Add firmware cache #2: This patch changes get_firmware(), free_firmware() and adds free_firmware_cache(). The first two functions implement a very simple cache and the latter is used to actually release all the stored firmwares when the module is unloaded. When CONFIG_PM is not enabled those functions act as before, that is free_firmware() releases the firmware immediately and free_firmware_cache() does nothing. Short description: This patch implements a simple cache for the firmware files when CONFIG_PM is defined. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:09:09.000000000 +0100 @@ -43,12 +43,24 @@ static int get_firmware(const struct fir { int err; char name[30]; - const struct firmware *frm = &card_fw[fw_index]; - DE_ACT(("firmware requested: %s\n", frm->data)); - snprintf(name, sizeof(name), "ea/%s", frm->data); - if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) +#ifdef CONFIG_PM + if (chip->fw_cache[fw_index]) { + DE_ACT(("firmware requested: %s is cached\n", card_fw[fw_index].data)); + *fw_entry = chip->fw_cache[fw_index]; + return 0; + } +#endif + + DE_ACT(("firmware requested: %s\n", card_fw[fw_index].data)); + snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data); + err = request_firmware(fw_entry, name, pci_device(chip)); + if (err < 0) snd_printk(KERN_ERR "get_firmware(): Firmware not available (%d)\n", err); +#ifdef CONFIG_PM + else + chip->fw_cache[fw_index] = *fw_entry; +#endif return err; } @@ -56,8 +68,29 @@ static int get_firmware(const struct fir static void free_firmware(const struct firmware *fw_entry) { +#ifdef CONFIG_PM + DE_ACT(("firmware not released (kept in cache)\n")); +#else release_firmware(fw_entry); DE_ACT(("firmware released\n")); +#endif +} + + + +static void free_firmware_cache(struct echoaudio *chip) +{ +#ifdef CONFIG_PM + int i; + + for (i = 0; i < 8 ; i++) + if (chip->fw_cache[i]) { + release_firmware(chip->fw_cache[i]); + DE_ACT(("release_firmware(%d)\n", i)); + } + + DE_ACT(("firmware_cache released\n")); +#endif } @@ -1880,6 +1913,7 @@ static int snd_echo_free(struct echoaudi pci_disable_device(chip->pci); /* release chip data */ + free_firmware_cache(chip); kfree(chip); DE_INIT(("Chip freed.\n")); return 0; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:08:36.000000000 +0100 @@ -449,6 +449,9 @@ struct echoaudio { volatile u32 __iomem *dsp_registers; /* DSP's register base */ u32 active_mask; /* Chs. active mask or * punks out */ +#ifdef CONFIG_PM + const struct firmware *fw_cache[8]; /* Cached firmwares */ +#endif #ifdef ECHOCARD_HAS_MIDI u16 mtc_state; /* State for MIDI input parsing state machine */ -- Giuliano.

1 0

[alsa-devel] [PATCH 1/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 30 Jan '10

30 Jan '10

1/5 Add firmware cache #1: When CONFIG_PM is enabled the firmware cannot be released because the driver will need it again to resume the card. With this patch the firmware is passed as an index of the struct firmware card_fw[] in place of a pointer. That same index is then used to locate the firmware in the firmware cache. Short description: Changes the way the firmware is passed through functions. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla20_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA20_DSP]; + chip->dsp_code_to_load = FW_DARLA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla24_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA24_DSP]; + chip->dsp_code_to_load = FW_DARLA24_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echo3g_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echo3g_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -61,7 +61,7 @@ static int init_hw(struct echoaudio *chi chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_ECHO3G_DSP]; + chip->dsp_code_to_load = FW_ECHO3G_DSP; /* Load the DSP code and the ASIC on the PCI card and get what type of external box is attached */ diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_3g.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_3g.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_3g.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_3g.c 2010-01-02 21:27:05.000000000 +0100 @@ -227,12 +227,11 @@ static int load_asic(struct echoaudio *c /* Give the DSP a few milliseconds to settle down */ mdelay(2); - err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, - &card_fw[FW_3G_ASIC]); + err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, FW_3G_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_3G_ASIC]; + chip->asic_code = FW_3G_ASIC; /* Now give the new ASIC some time to set up */ msleep(1000); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-02 21:27:05.000000000 +0100 @@ -36,11 +36,15 @@ MODULE_PARM_DESC(enable, "Enable " ECHOC static unsigned int channels_list[10] = {1, 2, 4, 6, 8, 10, 12, 14, 16, 999999}; static const DECLARE_TLV_DB_SCALE(db_scale_output_gain, -12800, 100, 1); + + static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip) + struct echoaudio *chip, const short fw_index) { int err; char name[30]; + const struct firmware *frm = &card_fw[fw_index]; + DE_ACT(("firmware requested: %s\n", frm->data)); snprintf(name, sizeof(name), "ea/%s", frm->data); if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) @@ -48,6 +52,8 @@ static int get_firmware(const struct fir return err; } + + static void free_firmware(const struct firmware *fw_entry) { release_firmware(fw_entry); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -175,15 +175,15 @@ static inline int check_asic_status(stru #ifdef ECHOCARD_HAS_ASIC /* Load ASIC code - done after the DSP is loaded */ -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic) +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) { const struct firmware *fw; int err; u32 i, size; u8 *code; - if ((err = get_firmware(&fw, asic, chip)) < 0) { + err = get_firmware(&fw, chip, asic); + if (err < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return err; } @@ -245,7 +245,8 @@ static int install_resident_loader(struc return 0; } - if ((i = get_firmware(&fw, &card_fw[FW_361_LOADER], chip)) < 0) { + i = get_firmware(&fw, chip, FW_361_LOADER); + if (i < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return i; } @@ -485,7 +486,8 @@ static int load_firmware(struct echoaudi chip->dsp_code = NULL; } - if ((err = get_firmware(&fw, chip->dsp_code_to_load, chip)) < 0) + err = get_firmware(&fw, chip, chip->dsp_code_to_load); + if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); free_firmware(fw); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.h 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-02 21:27:05.000000000 +0100 @@ -442,8 +442,8 @@ struct echoaudio { u16 device_id, subdevice_id; u16 *dsp_code; /* Current DSP code loaded, * NULL if nothing loaded */ - const struct firmware *dsp_code_to_load;/* DSP code to load */ - const struct firmware *asic_code; /* Current ASIC code */ + short dsp_code_to_load; /* DSP code to load */ + short asic_code; /* Current ASIC code */ u32 comm_page_phys; /* Physical address of the * memory seen by DSP */ volatile u32 __iomem *dsp_registers; /* DSP's register base */ @@ -464,7 +464,7 @@ static int load_firmware(struct echoaudi static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip); + struct echoaudio *chip, const short fw_index); static void free_firmware(const struct firmware *fw_entry); #ifdef ECHOCARD_HAS_MIDI diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina20_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -49,7 +49,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_GINA20_DSP]; + chip->dsp_code_to_load = FW_GINA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina24_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -33,8 +33,7 @@ static int write_control_reg(struct echo static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,13 +63,13 @@ static int init_hw(struct echoaudio *chi /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { - chip->dsp_code_to_load = &card_fw[FW_GINA24_361_DSP]; + chip->dsp_code_to_load = FW_GINA24_361_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; } else { - chip->dsp_code_to_load = &card_fw[FW_GINA24_301_DSP]; + chip->dsp_code_to_load = FW_GINA24_301_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | @@ -125,7 +124,7 @@ static int load_asic(struct echoaudio *c { u32 control_reg; int err; - const struct firmware *fw; + short asic; if (chip->asic_loaded) return 1; @@ -135,14 +134,15 @@ static int load_asic(struct echoaudio *c /* Pick the correct ASIC for '301 or '361 Gina24 */ if (chip->device_id == DEVICE_ID_56361) - fw = &card_fw[FW_GINA24_361_ASIC]; + asic = FW_GINA24_361_ASIC; else - fw = &card_fw[FW_GINA24_301_ASIC]; + asic = FW_GINA24_301_ASIC; - if ((err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, fw)) < 0) + err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, asic); + if (err < 0) return err; - chip->asic_code = fw; + chip->asic_code = asic; /* Now give the new ASIC a little time to set up */ mdelay(10); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJ_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigo_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigo_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IOX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla20_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -31,8 +31,7 @@ static int read_dsp(struct echoaudio *chip, u32 *data); static int set_professional_spdif(struct echoaudio *chip, char prof); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); static int update_flags(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chi chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA20_DSP]; + chip->dsp_code_to_load = FW_LAYLA20_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER; @@ -144,7 +143,7 @@ static int load_asic(struct echoaudio *c return 0; err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA_ASIC, - &card_fw[FW_LAYLA20_ASIC]); + FW_LAYLA20_ASIC); if (err < 0) return err; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla24_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -32,8 +32,7 @@ static int write_control_reg(struct echo static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chi chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA24_DSP]; + chip->dsp_code_to_load = FW_LAYLA24_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT; @@ -123,18 +122,18 @@ static int load_asic(struct echoaudio *c /* Load the ASIC for the PCI card */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC, - &card_fw[FW_LAYLA24_1_ASIC]); + FW_LAYLA24_1_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_LAYLA24_2S_ASIC]; + chip->asic_code = FW_LAYLA24_2S_ASIC; /* Now give the new ASIC a little time to set up */ mdelay(10); /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC, - &card_fw[FW_LAYLA24_2S_ASIC]); + FW_LAYLA24_2S_ASIC); if (err < 0) return FALSE; @@ -299,7 +298,7 @@ static int set_input_clock(struct echoau /* Depending on what digital mode you want, Layla24 needs different ASICs loaded. This function checks the ASIC needed for the new mode and sees if it matches the one already loaded. */ -static int switch_asic(struct echoaudio *chip, const struct firmware *asic) +static int switch_asic(struct echoaudio *chip, short asic) { s8 *monitors; @@ -335,7 +334,7 @@ static int dsp_set_digital_mode(struct e { u32 control_reg; int err, incompatible_clock; - const struct firmware *asic; + short asic; /* Set clock to "internal" if it's not compatible with the new mode */ incompatible_clock = FALSE; @@ -344,12 +343,12 @@ static int dsp_set_digital_mode(struct e case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2S_ASIC]; + asic = FW_LAYLA24_2S_ASIC; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2A_ASIC]; + asic = FW_LAYLA24_2A_ASIC; break; default: DE_ACT(("Digital mode not supported: %d\n", mode)); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mia_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mia_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -53,7 +53,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_MIA_DSP]; + chip->dsp_code_to_load = FW_MIA_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mona_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mona_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -33,8 +33,7 @@ static int write_control_reg(struct echo static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,9 +63,9 @@ static int init_hw(struct echoaudio *chi /* Mona comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) - chip->dsp_code_to_load = &card_fw[FW_MONA_361_DSP]; + chip->dsp_code_to_load = FW_MONA_361_DSP; else - chip->dsp_code_to_load = &card_fw[FW_MONA_301_DSP]; + chip->dsp_code_to_load = FW_MONA_301_DSP; chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; chip->professional_spdif = FALSE; @@ -120,7 +119,7 @@ static int load_asic(struct echoaudio *c { u32 control_reg; int err; - const struct firmware *asic; + short asic; if (chip->asic_loaded) return 0; @@ -128,9 +127,9 @@ static int load_asic(struct echoaudio *c mdelay(10); if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC, asic); if (err < 0) @@ -141,7 +140,7 @@ static int load_asic(struct echoaudio *c /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_EXTERNAL_ASIC, - &card_fw[FW_MONA_2_ASIC]); + FW_MONA_2_ASIC); if (err < 0) return err; @@ -165,22 +164,22 @@ loaded. This function checks the ASIC n if it matches the one already loaded. */ static int switch_asic(struct echoaudio *chip, char double_speed) { - const struct firmware *asic; int err; + short asic; /* Check the clock detect bits to see if this is a single-speed clock or a double-speed clock; load a new ASIC if necessary. */ if (chip->device_id == DEVICE_ID_56361) { if (double_speed) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; } else { if (double_speed) - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } if (asic != chip->asic_code) { @@ -200,7 +199,7 @@ static int switch_asic(struct echoaudio static int set_sample_rate(struct echoaudio *chip, u32 rate) { u32 control_reg, clock; - const struct firmware *asic; + short asic; char force_write; /* Only set the clock for internal mode. */ @@ -218,14 +217,14 @@ static int set_sample_rate(struct echoau if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EINVAL; if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; } else { if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } force_write = 0; @@ -410,8 +409,8 @@ static int dsp_set_digital_mode(struct e case DIGITAL_MODE_ADAT: /* If the current ASIC is the 96KHz ASIC, switch the ASIC and set to 48 KHz */ - if (chip->asic_code == &card_fw[FW_MONA_361_1_ASIC96] || - chip->asic_code == &card_fw[FW_MONA_301_1_ASIC96]) { + if (chip->asic_code == FW_MONA_361_1_ASIC96 || + chip->asic_code == FW_MONA_301_1_ASIC96) { set_sample_rate(chip, 48000); } control_reg |= GML_ADAT_MODE; -- Giuliano.

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[alsa-devel] A driver for Zoom R16 (USB)?
by Jani Frilander 30 Jan '10

30 Jan '10

Hi. I have a Zoom R16 multitrack recorder / USB audio interface / control surface unit. Unfortunately it's audio interface/control surface part currently don't work with linux. When I plug it in kernel (2.6.31-9-rt) won't recognize it as a audio interface. Is there a way to make it work? It probably needs to be hacked, because Zoom won't provide any technical information, according to their email I received. I'm a total newbie in driver development, but I'll be willing to work with this. Regards, Jani Frilander

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[alsa-devel] overrun in second capture device after stopping first capture device
by Tobias Schneider 29 Jan '10

29 Jan '10

Dear ALSA Dev-Team, while testing my driver, I figured out that under some circumstances there occur overruns when recording two streams: first I startet a long capture at device 1: -bash-3.2# -bash-3.2# arecord -D hw:0,1 -d 180 -r 32000 -c 1 -t wav -f S16_LE arecord_049.wav Recording WAVE 'arecord_049.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono overrun!!! (at least 121.536 ms long) overrun!!! (at least 115.361 ms long) overrun!!! (at least 190.883 ms long) overrun!!! (at least 121.772 ms long) overrun!!! (at least 119.487 ms long) In another terminal, I started a second capture on the second device. Everytime this capture is done, I got an overrun in the first capture (see above) -bash-3.2# -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# It seems, that the duration of the overrun relates to the capture time of the second capture, as it increases with bigger capture times. But if capture times are bigger than 25s, it seems there is no overrun...just some logs to clarify: First Terminal: -bash-3.2# -bash-3.2# arecord -D hw:0,1 -d 180 -r 32000 -c 1 -t wav -f S16_LE arecord_049.wav Recording WAVE 'arecord_049.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono overrun!!! (at least 99.074 ms long) Second Terminal: -bash-3.2# -bash-3.2# arecord -D hw:0,0 -d 30 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 30 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 30 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# First Terminal: -bash-3.2# arecord -D hw:0,1 -d 180 -r 32000 -c 1 -t wav -f S16_LE arecord_049.wav Recording WAVE 'arecord_049.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono overrun!!! (at least 193.656 ms long) Second Terminal: -bash-3.2# arecord -D hw:0,0 -d 25 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 20 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# So I am wondering why such an overrun occurs, can someone please give me some hints what my be the reason? Thanks in Advance! Tobias

1 2

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 29 Jan '10

29 Jan '10

The following changes since commit 64a028a6de08545a2c94f302bc7694bf48aee5b5: Linus Torvalds (1): Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs-2.6 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Pull request against Linus' current master branch since fix/asoc doesn't seem to have been updated - should I be looking at a different branch? Anuj Aggarwal (2): ASoC: AIC23: Fixing writes to non-existing registers in resume function ASoC: AM3517: ASoC driver not getting compiled sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/omap/Makefile | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-)

2 1

[alsa-devel] [PATCH 0/2] McBSP: OMAP3: Add sidetone feature
by Ilkka Koskinen 29 Jan '10

29 Jan '10

The first patch implements McBSP sidetone feature on OMAP3 while the second one provides ALSA interface for using it. The patch set is based on the patch Eduardo Valentin sent to alsa-devel mailing list in October 2009. Channel gain control and enabling the sidetone have been moved to ALSA interface as suggested in comments. Eero Nurkkala (1): McBSP: OMAP3: Add sidetone feature Ilkka Koskinen (1): ASoC: OMAP-McBSP: ASoC interface for McBSP sidetone arch/arm/mach-omap2/mcbsp.c | 2 + arch/arm/plat-omap/include/plat/mcbsp.h | 63 +++++ arch/arm/plat-omap/mcbsp.c | 402 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 187 ++++++++++++++ sound/soc/omap/omap-mcbsp.h | 2 + 5 files changed, 655 insertions(+), 1 deletions(-)

5 7

[alsa-devel] [PATCH v2] ASoC: clean up wm8974 and wm8978 clock divider handling
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their .set_clkdiv() methods, which is wrong, because these are simple boolean switches and not clock dividers. Move these bits to sound controls. Also remove manual configuration of the MCLK divider in wm8978, since it is configured automatically. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- v1 -> v2: 1. Added " Switch" to wm8974 and wm8978 DAC/ADC oversampling controls sound/soc/codecs/wm8974.c | 12 ++++-------- sound/soc/codecs/wm8974.h | 12 +----------- sound/soc/codecs/wm8978.c | 19 ++++--------------- sound/soc/codecs/wm8978.h | 3 --- sound/soc/sh/migor.c | 4 ---- 5 files changed, 9 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 8812751..ee637af 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -170,6 +170,10 @@ SOC_ENUM("Aux Mode", wm8974_auxmode), SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1), + +/* DAC / ADC oversampling */ +SOC_SINGLE("DAC 128x Oversampling Switch", WM8974_DAC, 8, 1, 0), +SOC_SINGLE("ADC 128x Oversampling Switch", WM8974_ADC, 8, 1, 0), }; /* Speaker Output Mixer */ @@ -381,14 +385,6 @@ static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai, reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f; snd_soc_write(codec, WM8974_CLOCK, reg | div); break; - case WM8974_ADCCLK: - reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7; - snd_soc_write(codec, WM8974_ADC, reg | div); - break; - case WM8974_DACCLK: - reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7; - snd_soc_write(codec, WM8974_DAC, reg | div); - break; case WM8974_BCLKDIV: reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3; snd_soc_write(codec, WM8974_CLOCK, reg | div); diff --git a/sound/soc/codecs/wm8974.h b/sound/soc/codecs/wm8974.h index 98de956..896a7f0 100644 --- a/sound/soc/codecs/wm8974.h +++ b/sound/soc/codecs/wm8974.h @@ -57,17 +57,7 @@ /* Clock divider Id's */ #define WM8974_OPCLKDIV 0 #define WM8974_MCLKDIV 1 -#define WM8974_ADCCLK 2 -#define WM8974_DACCLK 3 -#define WM8974_BCLKDIV 4 - -/* DAC clock dividers */ -#define WM8974_DACCLK_F2 (1 << 3) -#define WM8974_DACCLK_F4 (0 << 3) - -/* ADC clock dividers */ -#define WM8974_ADCCLK_F2 (1 << 3) -#define WM8974_ADCCLK_F4 (0 << 3) +#define WM8974_BCLKDIV 2 /* PLL Out dividers */ #define WM8974_OPCLKDIV_1 (0 << 4) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 8dcebaa..ec2624b 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -210,6 +210,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { /* Speaker */ SOC_DOUBLE_R("Speaker Switch", WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), + + /* DAC / ADC oversampling */ + SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ @@ -513,21 +517,6 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, if (wm8978->f_mclk) ret = wm8978_configure_pll(codec); break; - case WM8978_MCLKDIV: - if (div & ~0xe0) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); - break; - case WM8978_ADCCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); - break; - case WM8978_DACCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); - break; case WM8978_BCLKDIV: if (div & ~0x1c) return -EINVAL; diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h index b58f0bf..56ec832 100644 --- a/sound/soc/codecs/wm8978.h +++ b/sound/soc/codecs/wm8978.h @@ -72,9 +72,6 @@ /* Clock divider Id's */ enum wm8978_clk_id { WM8978_OPCLKRATE, - WM8978_MCLKDIV, - WM8978_ADCCLK, - WM8978_DACCLK, WM8978_BCLKDIV, }; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 3ccd9b3..b823a5c 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -59,10 +59,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); if (ret < 0) return ret; -- 1.6.2.4

2 1

[alsa-devel] AT91 AC97 Support broken in menuconfig.
by javier Martin 29 Jan '10

29 Jan '10

Hello, I was trying to get AC97 support for an AT91SAM9G45 in kernel 2.6.32 and I have noticed that it requires that AVR32 is selected, however the following commit states that driver for AVR32 was adapted for AT91 too: commit d9fbd9a2cd5ac1b286a7d7cdb1a180ce1edaee2f Merge: f25f60b... baea7b9... Author: Linus Torvalds <torvalds(a)linux-foundation.org> Date: Thu Sep 24 17:06:51 2009 -0700 Merge branch 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm * 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm: (103 commits) .... ARM: 5710/1: at91: add AC97 support to at91sam9rl and at91sam9rlek board ARM: 5709/1: at91: add AC97 support to at91sam9g45 series and at91sam9m10g45ek board ARM: 5621/1: at91/dmaengine: integration of at_hdmac driver in at91sam9g45 series ARM: 5620/1: at91/dmaengine: integration of at_hdmac driver in at91sam9rl ... I just wanted to know if this is just an error in Kconfig tree or the driver under "sound/atmel/" does not support at91sam9g54. Thanks. -- Javier Martin Vista Silicon S.L. CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

1 0

[alsa-devel] [PATCH] ASoC: remove bogus SLEEP mode from wm8978 driver
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

Tests showed, that bit 6 of the WM8978_POWER_MANAGEMENT_2 register of wm8978 affects codec clocks. Being useless for suspend / resume, it cannot be used in bias-level control either. Remove this bit handling. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- sound/soc/codecs/wm8978.c | 7 +------ 1 files changed, 1 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index b86ef47..8dcebaa 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -873,9 +873,7 @@ static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); /* Also switch PLL off */ snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0); - /* Put to sleep */ - snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x40); - + return 0; } @@ -887,9 +885,6 @@ static int wm8978_resume(struct platform_device *pdev) int i; u16 *cache = codec->reg_cache; - /* Wake up the codec */ - snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); - /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { if (i == WM8978_RESET) -- 1.6.2.4

3 2

[alsa-devel] [PATCH] ASoC: clean up wm8974 and wm8978 clock divider handling
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their .set_clkdiv() methods, which is wrong, because these are simple boolean switches and not clock dividers. Move these bits to sound controls. Also remove manual configuration of the MCLK divider in wm8978, since it is configured automatically. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- sound/soc/codecs/wm8974.c | 12 ++++-------- sound/soc/codecs/wm8974.h | 12 +----------- sound/soc/codecs/wm8978.c | 19 ++++--------------- sound/soc/codecs/wm8978.h | 3 --- sound/soc/sh/migor.c | 4 ---- 5 files changed, 9 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 8812751..051d58d 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -170,6 +170,10 @@ SOC_ENUM("Aux Mode", wm8974_auxmode), SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1), + +/* DAC / ADC oversampling */ +SOC_SINGLE("DAC 128x Oversampling", WM8974_DAC, 8, 1, 0), +SOC_SINGLE("ADC 128x Oversampling", WM8974_ADC, 8, 1, 0), }; /* Speaker Output Mixer */ @@ -381,14 +385,6 @@ static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai, reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f; snd_soc_write(codec, WM8974_CLOCK, reg | div); break; - case WM8974_ADCCLK: - reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7; - snd_soc_write(codec, WM8974_ADC, reg | div); - break; - case WM8974_DACCLK: - reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7; - snd_soc_write(codec, WM8974_DAC, reg | div); - break; case WM8974_BCLKDIV: reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3; snd_soc_write(codec, WM8974_CLOCK, reg | div); diff --git a/sound/soc/codecs/wm8974.h b/sound/soc/codecs/wm8974.h index 98de956..896a7f0 100644 --- a/sound/soc/codecs/wm8974.h +++ b/sound/soc/codecs/wm8974.h @@ -57,17 +57,7 @@ /* Clock divider Id's */ #define WM8974_OPCLKDIV 0 #define WM8974_MCLKDIV 1 -#define WM8974_ADCCLK 2 -#define WM8974_DACCLK 3 -#define WM8974_BCLKDIV 4 - -/* DAC clock dividers */ -#define WM8974_DACCLK_F2 (1 << 3) -#define WM8974_DACCLK_F4 (0 << 3) - -/* ADC clock dividers */ -#define WM8974_ADCCLK_F2 (1 << 3) -#define WM8974_ADCCLK_F4 (0 << 3) +#define WM8974_BCLKDIV 2 /* PLL Out dividers */ #define WM8974_OPCLKDIV_1 (0 << 4) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 8dcebaa..6d807b1 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -210,6 +210,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { /* Speaker */ SOC_DOUBLE_R("Speaker Switch", WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), + + /* DAC / ADC oversampling */ + SOC_SINGLE("DAC 128x Oversampling", WM8978_DAC_CONTROL, 8, 1, 0), + SOC_SINGLE("ADC 128x Oversampling", WM8978_ADC_CONTROL, 8, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ @@ -513,21 +517,6 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, if (wm8978->f_mclk) ret = wm8978_configure_pll(codec); break; - case WM8978_MCLKDIV: - if (div & ~0xe0) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); - break; - case WM8978_ADCCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); - break; - case WM8978_DACCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); - break; case WM8978_BCLKDIV: if (div & ~0x1c) return -EINVAL; diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h index b58f0bf..56ec832 100644 --- a/sound/soc/codecs/wm8978.h +++ b/sound/soc/codecs/wm8978.h @@ -72,9 +72,6 @@ /* Clock divider Id's */ enum wm8978_clk_id { WM8978_OPCLKRATE, - WM8978_MCLKDIV, - WM8978_ADCCLK, - WM8978_DACCLK, WM8978_BCLKDIV, }; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 3ccd9b3..b823a5c 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -59,10 +59,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); if (ret < 0) return ret; -- 1.6.2.4

2 1

[alsa-devel] [PATCH 2b/4 v2] ASoC: add support for the sh7722 Migo-R board
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

Add support for audio on sh7722-based Migo-R boards, using SIU and wm8978 codec, recording via external microphone and playback via headphones are implemented. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- v2 -> v3: 1. adjusted to the new wm8978 clocking scheme 2. added .hw_free() to switch off PLL diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 8072a6d..a86696b 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -55,4 +61,12 @@ config SND_FSI_DA7210 This option enables generic sound support for the FSI - DA7210 unit +config SND_SIU_MIGOR + tristate "SIU sound support on Migo-R" + depends on SH_MIGOR + select SND_SOC_SH4_SIU + select SND_SOC_WM8978 + help + This option enables sound support for the SH7722 Migo-R board + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 1d0ec0a..8a5a192 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -16,7 +16,9 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o snd-soc-fsi-da7210-objs := fsi-da7210.o +snd-soc-migor-objs := migor.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o +obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c new file mode 100644 index 0000000..3ccd9b3 --- /dev/null +++ b/sound/soc/sh/migor.c @@ -0,0 +1,222 @@ +/* + * ALSA SoC driver for Migo-R + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski(a)gmx.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. + */ + +#include <linux/device.h> +#include <linux/firmware.h> +#include <linux/module.h> + +#include <asm/clock.h> + +#include <cpu/sh7722.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8978.h" +#include "siu.h" + +/* Default 8000Hz sampling frequency */ +static unsigned long codec_freq = 8000 * 512; + +static unsigned int use_count; + +/* External clock, sourced from the codec at the SIUMCKB pin */ +static unsigned long siumckb_recalc(struct clk *clk) +{ + return codec_freq; +} + +static struct clk_ops siumckb_clk_ops = { + .recalc = siumckb_recalc, +}; + +static struct clk siumckb_clk = { + .name = "siumckb_clk", + .id = -1, + .ops = &siumckb_clk_ops, + .rate = 0, /* initialised at run-time */ +}; + +static int migor_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + unsigned int rate = params_rate(params); + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 13000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(rtd->dai->cpu_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + codec_freq = rate * 512; + /* + * This propagates the parent frequency change to children and + * recalculates the frequency table + */ + clk_set_rate(&siumckb_clk, codec_freq); + dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq); + + ret = snd_soc_dai_set_sysclk(rtd->dai->cpu_dai, SIU_CLKB_EXT, + codec_freq / 2, SND_SOC_CLOCK_IN); + + if (!ret) + use_count++; + + return ret; +} + +static int migor_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + if (use_count) { + use_count--; + + if (!use_count) + snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 0, + SND_SOC_CLOCK_IN); + } else { + dev_dbg(codec_dai->dev, "Unbalanced hw_free!\n"); + } + + return 0; +} + +static struct snd_soc_ops migor_dai_ops = { + .hw_params = migor_hw_params, + .hw_free = migor_hw_free, +}; + +static const struct snd_soc_dapm_widget migor_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Onboard Microphone", NULL), + SND_SOC_DAPM_MIC("External Microphone", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Headphone output connected to LHP/RHP, enable OUT4 for VMID */ + { "Headphone", NULL, "OUT4 VMID" }, + { "OUT4 VMID", NULL, "LHP" }, + { "OUT4 VMID", NULL, "RHP" }, + + /* On-board microphone */ + { "RMICN", NULL, "Mic Bias" }, + { "RMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Onboard Microphone" }, + + /* External microphone */ + { "LMICN", NULL, "Mic Bias" }, + { "LMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "External Microphone" }, +}; + +static int migor_dai_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, migor_dapm_widgets, + ARRAY_SIZE(migor_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* migor digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link migor_dai = { + .name = "wm8978", + .stream_name = "WM8978", + .cpu_dai = &siu_i2s_dai, + .codec_dai = &wm8978_dai, + .ops = &migor_dai_ops, + .init = migor_dai_init, +}; + +/* migor audio machine driver */ +static struct snd_soc_card snd_soc_migor = { + .name = "Migo-R", + .platform = &siu_platform, + .dai_link = &migor_dai, + .num_links = 1, +}; + +/* migor audio subsystem */ +static struct snd_soc_device migor_snd_devdata = { + .card = &snd_soc_migor, + .codec_dev = &soc_codec_dev_wm8978, +}; + +static struct platform_device *migor_snd_device; + +static int __init migor_init(void) +{ + int ret; + + ret = clk_register(&siumckb_clk); + if (ret < 0) + return ret; + + /* Port number used on this machine: port B */ + migor_snd_device = platform_device_alloc("soc-audio", 1); + if (!migor_snd_device) { + ret = -ENOMEM; + goto epdevalloc; + } + + platform_set_drvdata(migor_snd_device, &migor_snd_devdata); + + migor_snd_devdata.dev = &migor_snd_device->dev; + + ret = platform_device_add(migor_snd_device); + if (ret) + goto epdevadd; + + return 0; + +epdevadd: + platform_device_put(migor_snd_device); +epdevalloc: + clk_unregister(&siumckb_clk); + return ret; +} + +static void __exit migor_exit(void) +{ + clk_unregister(&siumckb_clk); + platform_device_unregister(migor_snd_device); +} + +module_init(migor_init); +module_exit(migor_exit); + +MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski(a)gmx.de>"); +MODULE_DESCRIPTION("ALSA SoC Migor"); +MODULE_LICENSE("GPL v2");

2 1

[alsa-devel] [PATCH 1/1] ASoC: AM3517: ASoC driver not getting compiled
by Anuj Aggarwal 29 Jan '10

29 Jan '10

Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the Makefile. Whereas the config option defined in Kconfig is SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517 was not getting compiled. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Makefile | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c..19283e5 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o +obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- 1.6.2.4

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[alsa-devel] [PATCH] ASoC: AC97: SMDK-WM9713: Convert notes from cset to sset
by jassisinghbrar＠gmail.com 29 Jan '10

29 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> It's more robust when references are provided in control names rather than numid. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/smdk_wm9713.c | 23 +++++++---------------- 1 files changed, 7 insertions(+), 16 deletions(-) diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 6fa2c9d..24fd39f 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -29,24 +29,15 @@ static struct snd_soc_card smdk; /* Playback (HeadPhone):- - Headphone Playback Switch - On - $ amixer cset numid=4 1 - - Right Headphone Out Mux - Headphone - $ amixer cset numid=92 2 - Left Headphone Out Mux - Headphone - $ amixer cset numid=93 2 - - Right HP Mixer PCM Playback Switch - On - $ amixer cset numid=75 1 - Left HP Mixer PCM Playback Switch - On - $ amixer cset numid=81 1 + $ amixer sset 'Headphone' unmute + $ amixer sset 'Right Headphone Out Mux' 'Headphone' + $ amixer sset 'Left Headphone Out Mux' 'Headphone' + $ amixer sset 'Right HP Mixer PCM' unmute + $ amixer sset 'Left HP Mixer PCM' unmute Capture (LineIn):- - Right Capture Source - Line - $ amixer cset numid=86 2 - Left Capture Source - Line - $ amixer cset numid=87 2 + $ amixer sset 'Right Capture Source' 'Line' + $ amixer sset 'Left Capture Source' 'Line' */ static struct snd_soc_dai_link smdk_dai = { -- 1.6.2.5

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[alsa-devel] [PATCH 1/1] ASoC: AIC23: Fixing writes to non-existing registers in resume function
by Anuj Aggarwal 29 Jan '10

29 Jan '10

Commit e9ff5eb2 (Fixing infinite loop in resume path) uses wrong AIC23 register in resume function because of which register writes happen on some non-existing registers. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/codecs/tlv320aic23.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a9dc5fb..da589d8 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev) u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < TLV320AIC23_RESET; reg++) { + for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } -- 1.6.2.4

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