diff options
40 files changed, 2321 insertions, 491 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7f4dcebda9c..d0eb696d32e 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -300,6 +300,74 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. control correctly. If you have problems regarding this, try another ALSA compliant mixer (alsamixer works). + Module snd-azt1605 + ------------------ + + Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605 + chipset. + + port - port # for BASE (0x220,0x240,0x260,0x280) + wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) + irq - IRQ # for WSS (7,9,10,11) + dma1 - DMA # for WSS playback (0,1,3) + dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) + mpu_irq - IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default) + fm_port - port # for OPL3 (0x388), -1 = disabled (default) + + This module supports multiple cards. It does not support autoprobe: port, + wss_port, irq and dma1 have to be specified. The other values are + optional. + + "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) + or the value stored in the card's EEPROM for cards that have an EEPROM and + their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can + be choosen freely from the options enumerated above. + + If dma2 is specified and different from dma1, the card will operate in + full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to + enable capture since only channels 0 and 1 are available for capture. + + Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 + mpu_port=0x330 mpu_irq=9 fm_port=0x388". + + Whatever IRQ and DMA channels you pick, be sure to reserve them for + legacy ISA in your BIOS. + + Module snd-azt2316 + ------------------ + + Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316 + chipset. + + port - port # for BASE (0x220,0x240,0x260,0x280) + wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) + irq - IRQ # for WSS (7,9,10,11) + dma1 - DMA # for WSS playback (0,1,3) + dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default) + fm_port - port # for OPL3 (0x388), -1 = disabled (default) + + This module supports multiple cards. It does not support autoprobe: port, + wss_port, irq and dma1 have to be specified. The other values are + optional. + + "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) + or the value stored in the card's EEPROM for cards that have an EEPROM and + their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can + be choosen freely from the options enumerated above. + + If dma2 is specified and different from dma1, the card will operate in + full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to + enable capture since only channels 0 and 1 are available for capture. + + Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 + mpu_port=0x330 mpu_irq=9 fm_port=0x388". + + Whatever IRQ and DMA channels you pick, be sure to reserve them for + legacy ISA in your BIOS. + Module snd-aw2 -------------- @@ -1641,20 +1709,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This card is also known as Audio Excel DSP 16 or Zoltrix AV302. - Module snd-sgalaxy - ------------------ - - Module for Aztech Sound Galaxy sound card. - - sbport - Port # for SB16 interface (0x220,0x240) - wssport - Port # for WSS interface (0x530,0xe80,0xf40,0x604) - irq - IRQ # (7,9,10,11) - dma1 - DMA # - - This module supports multiple cards. - - The power-management is supported. - Module snd-sscape ----------------- diff --git a/include/sound/jack.h b/include/sound/jack.h index d90b9fa3270..c140fc7cbd3 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -47,6 +47,9 @@ enum snd_jack_types { SND_JACK_BTN_0 = 0x4000, SND_JACK_BTN_1 = 0x2000, SND_JACK_BTN_2 = 0x1000, + SND_JACK_BTN_3 = 0x0800, + SND_JACK_BTN_4 = 0x0400, + SND_JACK_BTN_5 = 0x0200, }; struct snd_jack { @@ -55,7 +58,7 @@ struct snd_jack { int type; const char *id; char name[100]; - unsigned int key[3]; /* Keep in sync with definitions above */ + unsigned int key[6]; /* Keep in sync with definitions above */ void *private_data; void (*private_free)(struct snd_jack *); }; diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 85f1c6bf856..dfd9b76b185 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -278,6 +278,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ + unsigned long hw_ptr_buffer_jiffies; /* buffer time in jiffies */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ /* -- HW params -- */ diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index f50ebf20df9..86afb13cd24 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -77,7 +77,7 @@ static int snd_mixer_oss_release(struct inode *inode, struct file *file) struct snd_mixer_oss_file *fmixer; if (file->private_data) { - fmixer = (struct snd_mixer_oss_file *) file->private_data; + fmixer = file->private_data; module_put(fmixer->card->module); snd_card_file_remove(fmixer->card, file); kfree(fmixer); @@ -368,7 +368,7 @@ static int snd_mixer_oss_ioctl1(struct snd_mixer_oss_file *fmixer, unsigned int static long snd_mixer_oss_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { - return snd_mixer_oss_ioctl1((struct snd_mixer_oss_file *) file->private_data, cmd, arg); + return snd_mixer_oss_ioctl1(file->private_data, cmd, arg); } int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned long arg) @@ -582,7 +582,7 @@ static int snd_mixer_oss_get_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *left, int *right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; *left = *right = 100; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { @@ -691,7 +691,7 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int left, int right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); @@ -740,7 +740,7 @@ static int snd_mixer_oss_get_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -753,7 +753,7 @@ static int snd_mixer_oss_get_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -766,7 +766,7 @@ static int snd_mixer_oss_put_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], active, active, 0); return 0; @@ -776,7 +776,7 @@ static int snd_mixer_oss_put_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], active, active, 1); return 0; @@ -813,7 +813,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -861,7 +861,7 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -925,7 +925,7 @@ static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *sl static void snd_mixer_oss_slot_free(struct snd_mixer_oss_slot *chn) { - struct slot *p = (struct slot *)chn->private_data; + struct slot *p = chn->private_data; if (p) { if (p->allocated && p->assigned) { kfree(p->assigned->name); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 204af48c5cc..88525a95829 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -364,8 +364,7 @@ static void snd_pcm_stream_proc_info_read(struct snd_info_entry *entry, static void snd_pcm_substream_proc_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - snd_pcm_proc_info_read((struct snd_pcm_substream *)entry->private_data, - buffer); + snd_pcm_proc_info_read(entry->private_data, buffer); } static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e23e0e7ab26..a1707cca9c6 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -334,11 +334,15 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* delta = "expected next hw_ptr" for in_interrupt != 0 */ delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { - hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) - hw_base = 0; - new_hw_ptr = hw_base + pos; - goto __delta; + /* check for double acknowledged interrupts */ + hdelta = jiffies - runtime->hw_ptr_jiffies; + if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + goto __delta; + } } } /* new_hw_ptr might be lower than old_hw_ptr in case when */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 134fc6c2e08..e2e73895db1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -864,6 +864,8 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->hw_ptr_jiffies = jiffies; + runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / + runtime->rate; runtime->status->state = state; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 480c38623da..c8961165277 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -74,6 +74,25 @@ config SND_DUMMY To compile this driver as a module, choose M here: the module will be called snd-dummy. +config SND_ALOOP + tristate "Generic loopback driver (PCM)" + select SND_PCM + help + Say 'Y' or 'M' to include support for the PCM loopback device. + This module returns played samples back to the user space using + the standard ALSA PCM device. The devices are routed 0->1 and + 1->0, where first number is the playback PCM device and second + number is the capture device. Module creates two PCM devices and + configured number of substreams (see the pcm_substreams module + parameter). + + The looback device allow time sychronization with an external + timing source using the time shift universal control (+-20% + of system time). + + To compile this driver as a module, choose M here: the module + will be called snd-aloop. + config SND_VIRMIDI tristate "Virtual MIDI soundcard" depends on SND_SEQUENCER diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile index d4a07f9ff2c..1a8440c8b13 100644 --- a/sound/drivers/Makefile +++ b/sound/drivers/Makefile @@ -4,6 +4,7 @@ # snd-dummy-objs := dummy.o +snd-aloop-objs := aloop.o snd-mtpav-objs := mtpav.o snd-mts64-objs := mts64.o snd-portman2x4-objs := portman2x4.o @@ -13,6 +14,7 @@ snd-ml403-ac97cr-objs := ml403-ac97cr.o pcm-indirect2.o # Toplevel Module Dependency obj-$(CONFIG_SND_DUMMY) += snd-dummy.o +obj-$(CONFIG_SND_ALOOP) += snd-aloop.o obj-$(CONFIG_SND_VIRMIDI) += snd-virmidi.o obj-$(CONFIG_SND_SERIAL_U16550) += snd-serial-u16550.o obj-$(CONFIG_SND_MTPAV) += snd-mtpav.o diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c new file mode 100644 index 00000000000..3123a15d23f --- /dev/null +++ b/sound/drivers/aloop.c @@ -0,0 +1,1055 @@ +/* + * Loopback soundcard + * + * Original code: + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> + * + * More accurate positioning and full-duplex support: + * Copyright (c) Ahmet İnan <ainan at mathematik.uni-freiburg.de> + * + * Major (almost complete) rewrite: + * Copyright (c) by Takashi Iwai <tiwai@suse.de> + * + * A next major update in 2010 (separate timers for playback and capture): + * Copyright (c) Jaroslav Kysela <perex@perex.cz> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/init.h> +#include <linux/jiffies.h> +#include <linux/slab.h> +#include <linux/time.h> +#include <linux/wait.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); +MODULE_DESCRIPTION("A loopback soundcard"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{ALSA,Loopback soundcard}}"); + +#define MAX_PCM_SUBSTREAMS 8 + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; +static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; +static int pcm_notify[SNDRV_CARDS]; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for loopback soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for loopback soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable this loopback soundcard."); +module_param_array(pcm_substreams, int, NULL, 0444); +MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-8) for loopback driver."); +module_param_array(pcm_notify, int, NULL, 0444); +MODULE_PARM_DESC(pcm_notify, "Break capture when PCM format/rate/channels changes."); + +#define NO_PITCH 100000 + +struct loopback_pcm; + +struct loopback_cable { + spinlock_t lock; + struct loopback_pcm *streams[2]; + struct snd_pcm_hardware hw; + /* flags */ + unsigned int valid; + unsigned int running; +}; + +struct loopback_setup { + unsigned int notify: 1; + unsigned int rate_shift; + unsigned int format; + unsigned int rate; + unsigned int channels; + struct snd_ctl_elem_id active_id; + struct snd_ctl_elem_id format_id; + struct snd_ctl_elem_id rate_id; + struct snd_ctl_elem_id channels_id; +}; + +struct loopback { + struct snd_card *card; + struct mutex cable_lock; + struct loopback_cable *cables[MAX_PCM_SUBSTREAMS][2]; + struct snd_pcm *pcm[2]; + struct loopback_setup setup[MAX_PCM_SUBSTREAMS][2]; +}; + +struct loopback_pcm { + struct loopback *loopback; + struct snd_pcm_substream *substream; + struct loopback_cable *cable; + unsigned int pcm_buffer_size; + unsigned int buf_pos; /* position in buffer */ + unsigned int silent_size; + /* PCM parameters */ + unsigned int pcm_period_size; + unsigned int pcm_bps; /* bytes per second */ + unsigned int pcm_salign; /* bytes per sample * channels */ + unsigned int pcm_rate_shift; /* rate shift value */ + /* flags */ + unsigned int period_update_pending :1; + /* timer stuff */ + unsigned int irq_pos; /* fractional IRQ position */ + unsigned int period_size_frac; + unsigned long last_jiffies; + struct timer_list timer; +}; + +static struct platform_device *devices[SNDRV_CARDS]; + +static inline unsigned int byte_pos(struct loopback_pcm *dpcm, unsigned int x) +{ + if (dpcm->pcm_rate_shift == NO_PITCH) { + x /= HZ; + } else { + x = div_u64(NO_PITCH * (unsigned long long)x, + HZ * (unsigned long long)dpcm->pcm_rate_shift); + } + return x - (x % dpcm->pcm_salign); +} + +static inline unsigned int frac_pos(struct loopback_pcm *dpcm, unsigned int x) +{ + if (dpcm->pcm_rate_shift == NO_PITCH) { /* no pitch */ + return x * HZ; + } else { + x = div_u64(dpcm->pcm_rate_shift * (unsigned long long)x * HZ, + NO_PITCH); + } + return x; +} + +static inline struct loopback_setup *get_setup(struct loopback_pcm *dpcm) +{ + int device = dpcm->substream->pstr->pcm->device; + + if (dpcm->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + device ^= 1; + return &dpcm->loopback->setup[dpcm->substream->number][device]; +} + +static inline unsigned int get_notify(struct loopback_pcm *dpcm) +{ + return get_setup(dpcm)->notify; +} + +static inline unsigned int get_rate_shift(struct loopback_pcm *dpcm) +{ + return get_setup(dpcm)->rate_shift; +} + +static void loopback_timer_start(struct loopback_pcm *dpcm) +{ + unsigned long tick; + unsigned int rate_shift = get_rate_shift(dpcm); + + if (rate_shift != dpcm->pcm_rate_shift) { + dpcm->pcm_rate_shift = rate_shift; + dpcm->period_size_frac = frac_pos(dpcm, dpcm->pcm_period_size); + } + tick = dpcm->period_size_frac - dpcm->irq_pos; + tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps; + dpcm->timer.expires = jiffies + tick; + add_timer(&dpcm->timer); +} + +static inline void loopback_timer_stop(struct loopback_pcm *dpcm) +{ + del_timer(&dpcm->timer); +} + +#define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK) +#define CABLE_VALID_CAPTURE (1 << SNDRV_PCM_STREAM_CAPTURE) +#define CABLE_VALID_BOTH (CABLE_VALID_PLAYBACK|CABLE_VALID_CAPTURE) + +static int loopback_check_format(struct loopback_cable *cable, int stream) +{ + struct snd_pcm_runtime *runtime; + struct loopback_setup *setup; + struct snd_card *card; + int check; + + if (cable->valid != CABLE_VALID_BOTH) { + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + goto __notify; + return 0; + } + runtime = cable->streams[SNDRV_PCM_STREAM_PLAYBACK]-> + substream->runtime; + check = cable->hw.formats != (1ULL << runtime->format) || + cable->hw.rate_min != runtime->rate || + cable->hw.rate_max != runtime->rate || + cable->hw.channels_min != runtime->channels || + cable->hw.channels_max != runtime->channels; + if (!check) + return 0; + if (stream == SNDRV_PCM_STREAM_CAPTURE) { + return -EIO; + } else { + snd_pcm_stop(cable->streams[SNDRV_PCM_STREAM_CAPTURE]-> + substream, SNDRV_PCM_STATE_DRAINING); + __notify: + runtime = cable->streams[SNDRV_PCM_STREAM_PLAYBACK]-> + substream->runtime; + setup = get_setup(cable->streams[SNDRV_PCM_STREAM_PLAYBACK]); + card = cable->streams[SNDRV_PCM_STREAM_PLAYBACK]->loopback->card; + if (setup->format != runtime->format) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &setup->format_id); + setup->format = runtime->format; + } + if (setup->rate != runtime->rate) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &setup->rate_id); + setup->rate = runtime->rate; + } + if (setup->channels != runtime->channels) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &setup->channels_id); + setup->channels = runtime->channels; + } + } + return 0; +} + +static void loopback_active_notify(struct loopback_pcm *dpcm) +{ + snd_ctl_notify(dpcm->loopback->card, + SNDRV_CTL_EVENT_MASK_VALUE, + &get_setup(dpcm)->active_id); +} + +static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loopback_pcm *dpcm = runtime->private_data; + struct loopback_cable *cable = dpcm->cable; + int err; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + err = loopback_check_format(cable, substream->stream); + if (err < 0) + return err; + dpcm->last_jiffies = jiffies; + dpcm->pcm_rate_shift = 0; + loopback_timer_start(dpcm); + cable->running |= (1 << substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + loopback_active_notify(dpcm); + break; + case SNDRV_PCM_TRIGGER_STOP: + cable->running &= ~(1 << substream->stream); + loopback_timer_stop(dpcm); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + loopback_active_notify(dpcm); + break; + default: + return -EINVAL; + } + return 0; +} + +static int loopback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loopback_pcm *dpcm = runtime->private_data; + struct loopback_cable *cable = dpcm->cable; + unsigned int bps, salign; + + salign = (snd_pcm_format_width(runtime->format) * + runtime->channels) / 8; + bps = salign * runtime->rate; + if (bps <= 0 || salign <= 0) + return -EINVAL; + + dpcm->buf_pos = 0; + dpcm->pcm_buffer_size = frames_to_bytes(runtime, runtime->buffer_size); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + /* clear capture buffer */ + dpcm->silent_size = dpcm->pcm_buffer_size; + snd_pcm_format_set_silence(runtime->format, runtime->dma_area, + runtime->buffer_size * runtime->channels); + } + + dpcm->irq_pos = 0; + dpcm->period_update_pending = 0; + dpcm->pcm_bps = bps; + dpcm->pcm_salign = salign; + dpcm->pcm_period_size = frames_to_bytes(runtime, runtime->period_size); + + mutex_lock(&dpcm->loopback->cable_lock); + if (!(cable->valid & ~(1 << substream->stream))) { + cable->hw.formats = (1ULL << runtime->format); + cable->hw.rate_min = runtime->rate; + cable->hw.rate_max = runtime->rate; + cable->hw.channels_min = runtime->channels; + cable->hw.channels_max = runtime->channels; + } + cable->valid |= 1 << substream->stream; + mutex_unlock(&dpcm->loopback->cable_lock); + + return 0; +} + +static void clear_capture_buf(struct loopback_pcm *dpcm, unsigned int bytes) +{ + struct snd_pcm_runtime *runtime = dpcm->substream->runtime; + char *dst = runtime->dma_area; + unsigned int dst_off = dpcm->buf_pos; + + if (dpcm->silent_size >= dpcm->pcm_buffer_size) + return; + if (dpcm->silent_size + bytes > dpcm->pcm_buffer_size) + bytes = dpcm->pcm_buffer_size - dpcm->silent_size; + + for (;;) { + unsigned int size = bytes; + if (dst_off + size > dpcm->pcm_buffer_size) + size = dpcm->pcm_buffer_size - dst_off; + snd_pcm_format_set_silence(runtime->format, dst + dst_off, + bytes_to_frames(runtime, size) * + runtime->channels); + dpcm->silent_size += size; + bytes -= size; + if (!bytes) + break; + dst_off = 0; + } +} + +static void copy_play_buf(struct loopback_pcm *play, + struct loopback_pcm *capt, + unsigned int bytes) +{ + struct snd_pcm_runtime *runtime = play->substream->runtime; + char *src = play->substream->runtime->dma_area; + char *dst = capt->substream->runtime->dma_area; + unsigned int src_off = play->buf_pos; + unsigned int dst_off = capt->buf_pos; + unsigned int clear_bytes = 0; + + /* check if playback is draining, trim the capture copy size + * when our pointer is at the end of playback ring buffer */ + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING && + snd_pcm_playback_hw_avail(runtime) < runtime->buffer_size) { + snd_pcm_uframes_t appl_ptr, appl_ptr1, diff; + appl_ptr = appl_ptr1 = runtime->control->appl_ptr; + appl_ptr1 -= appl_ptr1 % runtime->buff |