diff options
author | David Dillow <dave@thedillows.org> | 2007-12-14 14:40:23 +0100 |
---|---|---|
committer | Jaroslav Kysela <perex@perex.cz> | 2008-01-31 17:29:36 +0100 |
commit | 175859bf1602c7ee38d720daa14a287072cc2b72 (patch) | |
tree | a05379ad2adcd91d8e8dc5d907ab27ccdd87b695 /sound/pci/sis7019.c | |
parent | 3b378e1f7e5b372a88189398f3946fd55da5923a (diff) |
[ALSA] sis7019: support the SiS 7019 Audio Accelerator
Basic audio support for the SiS 7019 Audio Accelerator as found in the
SiS 55x SoC. There is currently no synth support at the moment, but
audio playback and capture with two periods per buffer has seen
extensive use. Arbitrary period and buffer sizes (with multiple periods
per buffer) have seen light testing, but are believed to be production
ready.
Signed-off-by: David Dillow <dave@thedillows.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Diffstat (limited to 'sound/pci/sis7019.c')
-rw-r--r-- | sound/pci/sis7019.c | 1461 |
1 files changed, 1461 insertions, 0 deletions
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c new file mode 100644 index 000000000000..2f178598186d --- /dev/null +++ b/sound/pci/sis7019.c @@ -0,0 +1,1461 @@ +/* + * Driver for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave@thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed 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 <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/time.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include "sis7019.h" + +MODULE_AUTHOR("David Dillow <dave@thedillows.org>"); +MODULE_DESCRIPTION("SiS7019"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int enable = 1; + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); + +static struct pci_device_id snd_sis7019_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_sis7019_ids); + +/* There are three timing modes for the voices. + * + * For both playback and capture, when the buffer is one or two periods long, + * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt + * to let us know when the periods have ended. + * + * When performing playback with more than two periods per buffer, we set + * the "Stop Sample Offset" and tell the hardware to interrupt us when we + * reach it. We then update the offset and continue on until we are + * interrupted for the next period. + * + * Capture channels do not have a SSO, so we allocate a playback channel to + * use as a timer for the capture periods. We use the SSO on the playback + * channel to clock out virtual periods, and adjust the virtual period length + * to maintain synchronization. This algorithm came from the Trident driver. + * + * FIXME: It'd be nice to make use of some of the synth features in the + * hardware, but a woeful lack of documentation is a significant roadblock. + */ +struct voice { + u16 flags; +#define VOICE_IN_USE 1 +#define VOICE_CAPTURE 2 +#define VOICE_SSO_TIMING 4 +#define VOICE_SYNC_TIMING 8 + u16 sync_cso; + u16 period_size; + u16 buffer_size; + u16 sync_period_size; + u16 sync_buffer_size; + u32 sso; + u32 vperiod; + struct snd_pcm_substream *substream; + struct voice *timing; + void __iomem *ctrl_base; + void __iomem *wave_base; + void __iomem *sync_base; + int num; +}; + +/* We need four pages to store our wave parameters during a suspend. If + * we're not doing power management, we still need to allocate a page + * for the silence buffer. + */ +#ifdef CONFIG_PM +#define SIS_SUSPEND_PAGES 4 +#else +#define SIS_SUSPEND_PAGES 1 +#endif + +struct sis7019 { + unsigned long ioport; + void __iomem *ioaddr; + int irq; + int codecs_present; + + struct pci_dev *pci; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_ac97 *ac97[3]; + + /* Protect against more than one thread hitting the AC97 + * registers (in a more polite manner than pounding the hardware + * semaphore) + */ + struct mutex ac97_mutex; + + /* voice_lock protects allocation/freeing of the voice descriptions + */ + spinlock_t voice_lock; + + struct voice voices[64]; + struct voice capture_voice; + + /* Allocate pages to store the internal wave state during + * suspends. When we're operating, this can be used as a silence + * buffer for a timing channel. + */ + void *suspend_state[SIS_SUSPEND_PAGES]; + + int silence_users; + dma_addr_t silence_dma_addr; +}; + +#define SIS_PRIMARY_CODEC_PRESENT 0x0001 +#define SIS_SECONDARY_CODEC_PRESENT 0x0002 +#define SIS_TERTIARY_CODEC_PRESENT 0x0004 + +/* The HW offset parameters (Loop End, Stop Sample, End Sample) have a + * documented range of 8-0xfff8 samples. Given that they are 0-based, + * that places our period/buffer range at 9-0xfff9 samples. That makes the + * max buffer size 0xfff9 samples * 2 channels * 2 bytes per sample, and + * max samples / min samples gives us the max periods in a buffer. + * + * We'll add a constraint upon open that limits the period and buffer sample + * size to values that are legal for the hardware. + */ +static struct snd_pcm_hardware sis_playback_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static struct snd_pcm_hardware sis_capture_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static void sis_update_sso(struct voice *voice, u16 period) +{ + void __iomem *base = voice->ctrl_base; + + voice->sso += period; + if (voice->sso >= voice->buffer_size) + voice->sso -= voice->buffer_size; + + /* Enforce the documented hardware minimum offset */ + if (voice->sso < 8) + voice->sso = 8; + + /* The SSO is in the upper 16 bits of the register. */ + writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2); +} + +static void sis_update_voice(struct voice *voice) +{ + if (voice->flags & VOICE_SSO_TIMING) { + sis_update_sso(voice, voice->period_size); + } else if (voice->flags & VOICE_SYNC_TIMING) { + int sync; + + /* If we've not hit the end of the virtual period, update + * our records and keep going. + */ + if (voice->vperiod > voice->period_size) { + voice->vperiod -= voice->period_size; + if (voice->vperiod < voice->period_size) + sis_update_sso(voice, voice->vperiod); + else + sis_update_sso(voice, voice->period_size); + return; + } + + /* Calculate our relative offset between the target and + * the actual CSO value. Since we're operating in a loop, + * if the value is more than half way around, we can + * consider ourselves wrapped. + */ + sync = voice->sync_cso; + sync -= readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO); + if (sync > (voice->sync_buffer_size / 2)) + sync -= voice->sync_buffer_size; + + /* If sync is positive, then we interrupted too early, and + * we'll need to come back in a few samples and try again. + * There's a minimum wait, as it takes some time for the DMA + * engine to startup, etc... + */ + if (sync > 0) { + if (sync < 16) + sync = 16; + sis_update_sso(voice, sync); + return; + } + + /* Ok, we interrupted right on time, or (hopefully) just + * a bit late. We'll adjst our next waiting period based + * on how close we got. + * + * We need to stay just behind the actual channel to ensure + * it really is past a period when we get our interrupt -- + * otherwise we'll fall into the early code above and have + * a minimum wait time, which makes us quite late here, + * eating into the user's time to refresh the buffer, esp. + * if using small periods. + * + * If we're less than 9 samples behind, we're on target. + */ + if (sync > -9) + voice->vperiod = voice->sync_period_size + 1; + else + voice->vperiod = voice->sync_period_size - 4; + + if (voice->vperiod < voice->buffer_size) { + sis_update_sso(voice, voice->vperiod); + voice->vperiod = 0; + } else + sis_update_sso(voice, voice->period_size); + + sync = voice->sync_cso + voice->sync_period_size; + if (sync >= voice->sync_buffer_size) + sync -= voice->sync_buffer_size; + voice->sync_cso = sync; + } + + snd_pcm_period_elapsed(voice->substream); +} + +static void sis_voice_irq(u32 status, struct voice *voice) +{ + int bit; + + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + sis_update_voice(voice); + voice++; + } +} + +static irqreturn_t sis_interrupt(int irq, void *dev) +{ + struct sis7019 *sis = dev; + unsigned long io = sis->ioport; + struct voice *voice; + u32 intr, status; + + /* We only use the DMA interrupts, and we don't enable any other + * source of interrupts. But, it is possible to see an interupt + * status that didn't actually interrupt us, so eliminate anything + * we're not expecting to avoid falsely claiming an IRQ, and an + * ensuing endless loop. + */ + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + if (!intr) + return IRQ_NONE; + + do { + status = inl(io + SIS_PISR_A); + if (status) { + sis_voice_irq(status, sis->voices); + outl(status, io + SIS_PISR_A); + } + + status = inl(io + SIS_PISR_B); + if (status) { + sis_voice_irq(status, &sis->voices[32]); + outl(status, io + SIS_PISR_B); + } + + status = inl(io + SIS_RISR); + if (status) { + voice = &sis->capture_voice; + if (!voice->timing) + snd_pcm_period_elapsed(voice->substream); + + outl(status, io + SIS_RISR); + } + + outl(intr, io + SIS_GISR); + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + } while (intr); + + return IRQ_HANDLED; +} + +static u32 sis_rate_to_delta(unsigned int rate) +{ + u32 delta; + + /* This was copied from the trident driver, but it seems its gotten + * around a bit... nevertheless, it works well. + * + * We special case 44100 and 8000 since rounding with the equation + * does not give us an accurate enough value. For 11025 and 22050 + * the equation gives us the best answer. All other frequencies will + * also use the equation. JDW + */ + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff; + return delta; +} + +static void __sis_map_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + if (!sis->silence_users) + sis->silence_dma_addr = pci_map_single(sis->pci, + sis->suspend_state[0], + 4096, PCI_DMA_TODEVICE); + sis->silence_users++; +} + +static void __sis_unmap_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + sis->silence_users--; + if (!sis->silence_users) + pci_unmap_single(sis->pci, sis->silence_dma_addr, 4096, + PCI_DMA_TODEVICE); +} + +static void sis_free_voice(struct sis7019 *sis, struct voice *voice) +{ + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->timing) { + __sis_unmap_silence(sis); + voice->timing->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | + VOICE_SYNC_TIMING); + voice->timing = NULL; + } + voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING); + spin_unlock_irqrestore(&sis->voice_lock, flags); +} + +static struct voice *__sis_alloc_playback_voice(struct sis7019 *sis) +{ + /* Must hold the voice_lock on entry */ + struct voice *voice; + int i; + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + if (voice->flags & VOICE_IN_USE) + continue; + voice->flags |= VOICE_IN_USE; + goto found_one; + } + voice = NULL; + +found_one: + return voice; +} + +static struct voice *sis_alloc_playback_voice(struct sis7019 *sis) +{ + struct voice *voice; + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + voice = __sis_alloc_playback_voice(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + + return voice; +} + +static int sis_alloc_timing_voice(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + unsigned int period_size, buffer_size; + unsigned long flags; + int needed; + + /* If there are one or two periods per buffer, we don't need a + * timing voice, as we can use the capture channel's interrupts + * to clock out the periods. + */ + period_size = params_period_size(hw_params); + buffer_size = params_buffer_size(hw_params); + needed = (period_size != buffer_size && + period_size != (buffer_size / 2)); + + if (needed && !voice->timing) { + spin_lock_irqsave(&sis->voice_lock, flags); + voice->timing = __sis_alloc_playback_voice(sis); + if (voice->timing) + __sis_map_silence(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + if (!voice->timing) + return -ENOMEM; + voice->timing->substream = substream; + } else if (!needed && voice->timing) { + sis_free_voice(sis, voice); + voice->timing = NULL; + } + + return 0; +} + +static int sis_playback_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice; + + voice = sis_alloc_playback_voice(sis); + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_playback_hw_info; + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_substream_close(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + + sis_free_voice(sis, voice); + return 0; +} + +static int sis_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int sis_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *ctrl_base = voice->ctrl_base; + void __iomem *wave_base = voice->wave_base; + u32 format, dma_addr, control, sso_eso, delta, reg; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= SIS_PLAY_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_PLAY_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_PLAY_DMA_FORMAT_MONO; + + /* The baseline setup is for a single period per buffer, and + * we add bells and whistles as needed from there. + */ + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo; + + if (runtime->period_size == (runtime->buffer_size / 2)) { + control |= SIS_PLAY_DMA_INTR_AT_MLP; + } else if (runtime->period_size != runtime->buffer_size) { + voice->flags |= VOICE_SSO_TIMING; + voice->sso = runtime->period_size - 1; + voice->period_size = runtime->period_size; + voice->buffer_size = runtime->buffer_size; + + control &= ~SIS_PLAY_DMA_INTR_AT_LEO; + control |= SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso |= (runtime->period_size - 1) << 16; + } + + delta = sis_rate_to_delta(runtime->rate); + + /* Ok, we're ready to go, set up the channel. + */ + writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE); + writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + + /* Force PCI writes to post. */ + readl(ctrl_base); + + return 0; +} + +static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + unsigned long io = sis->ioport; + struct snd_pcm_substream *s; + struct voice *voice; + void *chip; + int starting; + u32 record = 0; + u32 play[2] = { 0, 0 }; + + /* No locks needed, as the PCM core will hold the locks on the + * substreams, and the HW will only start/stop the indicated voices + * without changing the state of the others. + */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + starting = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + starting = 0; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + /* Make sure it is for us... */ + chip = snd_pcm_substream_chip(s); + if (chip != sis) + continue; + + voice = s->runtime->private_data; + if (voice->flags & VOICE_CAPTURE) { + record |= 1 << voice->num; + voice = voice->timing; + } + + /* voice could be NULL if this a recording stream, and it + * doesn't have an external timing channel. + */ + if (voice) + play[voice->num / 32] |= 1 << (voice->num & 0x1f); + + snd_pcm_trigger_done(s, substream); + } + + if (starting) { + if (record) + outl(record, io + SIS_RECORD_START_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_START_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_START_B_REG); + } else { + if (record) + outl(record, io + SIS_RECORD_STOP_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_STOP_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_STOP_B_REG); + } + return 0; +} + +static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + u32 cso; + + cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + cso &= 0xffff; + return cso; +} + +static int sis_capture_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = &sis->capture_voice; + unsigned long flags; + + /* FIXME: The driver only supports recording from one channel + * at the moment, but it could support more. + */ + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->flags & VOICE_IN_USE) + voice = NULL; + else + voice->flags |= VOICE_IN_USE; + spin_unlock_irqrestore(&sis->voice_lock, flags); + + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_capture_hw_info; + runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + int rc; + + rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE, + params_rate(hw_params)); + if (rc) + goto out; + + rc = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (rc < 0) + goto out; + + rc = sis_alloc_timing_voice(substream, hw_params); + +out: + return rc; +} + +static void sis_prepare_timing_voice(struct voice *voice, + struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *timing = voice->timing; + void __iomem *play_base = timing->ctrl_base; + void __iomem *wave_base = timing->wave_base; + u16 buffer_size, period_size; + u32 format, control, sso_eso, delta; + u32 vperiod, sso, reg; + + /* Set our initial buffer and period as large as we can given a + * single page of silence. + */ + buffer_size = 4096 / runtime->channels; + buffer_size /= snd_pcm_format_size(runtime->format, 1); + period_size = buffer_size; + + /* Initially, we want to interrupt just a bit behind the end of + * the period we're clocking out. 10 samples seems to give a good + * delay. + * + * We want to spread our interrupts throughout the virtual period, + * so that we don't end up with two interrupts back to back at the + * end -- this helps minimize the effects of any jitter. Adjust our + * clocking period size so that the last period is at least a fourth + * of a full period. + * + * This is all moot if we don't need to use virtual periods. + */ + vperiod = runtime->period_size + 10; + if (vperiod > period_size) { + u16 tail = vperiod % period_size; + u16 quarter_period = period_size / 4; + + if (tail && tail < quarter_period) { + u16 loops = vperiod / period_size; + + tail = quarter_period - tail; + tail += loops - 1; + tail /= loops; + period_size -= tail; + } + + sso = period_size - 1; + } else { + /* The initial period will fit inside the buffer, so we + * don't need to use virtual periods -- disable them. + */ + period_size = runtime->period_size; + sso = vperiod - 1; + vperiod = 0; + } + + /* The interrupt handler implements the timing syncronization, so + * setup its state. + */ + timing->flags |= VOICE_SYNC_TIMING; + timing->sync_base = voice->ctrl_base; + timing->sync_cso = runtime->period_size - 1; + timing->sync_period_size = runtime->period_size; + timing->sync_buffer_size = runtime->buffer_size; + timing->period_size = period_size; + timing->buffer_size = buffer_size; + timing->sso = sso; + timing->vperiod = vperiod; + + /* Using unsigned samples with the all-zero silence buffer + * forces the output to the lower rail, killing playback. + * So ignore unsigned vs signed -- it doesn't change the timing. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + control = timing->buffer_size - 1; + control |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso = timing->buffer_size - 1; + sso_eso |= timing->sso << 16; + + delta = sis_rate_to_delta(runtime->rate); + + /* We've done the math, now configure the channel. + */ + writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(sis->silence_dma_addr, play_base + SIS_PLAY_DMA_BASE); + writel(control, play_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); +} + +static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *rec_base = voice->ctrl_base; + u32 format, dma_addr, control; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_CAPTURE_DMA_LOOP; + + /* If we've got more than two periods per buffer, then we have + * use a timing voice to clock out the periods. Otherwise, we can + * use the capture channel's interrupts. + */ + if (voice->timing) { + sis_prepare_timing_voice(voice, substream); + } else { + control |= SIS_CAPTURE_DMA_INTR_AT_LEO; + if (runtime->period_size != runtime->buffer_size) + control |= SIS_CAPTURE_DMA_INTR_AT_MLP; + } + + writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO); + writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE); + writel(control, rec_base + SIS_CAPTURE_DMA_CONTROL); + + /* Force the writes to post. */ + readl(rec_base); + + return 0; +} + +static struct snd_pcm_ops sis_playback_ops = { + .open = sis_playback_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_playback_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_playback_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static struct snd_pcm_ops sis_capture_ops = { + .open = sis_capture_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_capture_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_capture_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static int __devinit sis_pcm_create(struct sis7019 *sis) +{ + struct snd_pcm *pcm; + int rc; + + /* We have 64 voices, and the driver currently records from + * only one channel, though that could change in the future. + */ + rc = snd_pcm_new(sis->card, "SiS7019", 0, 64, 1, &pcm); + if (rc) + return rc; + + pcm->private_data = sis; + strcpy(pcm->name, "SiS7019"); + sis->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops); + + /* Try to preallocate some memory, but it's not the end of the + * world if this fails. + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + + return 0; +} + +static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd) +{ + unsigned long io = sis->ioport; + unsigned short val = 0xffff; + u16 status; + u16 rdy; + int count; + const static u16 codec_ready[3] = { + SIS_AC97_STATUS_CODEC_READY, + SIS_AC97_STATUS_CODEC2_READY, + SIS_AC97_STATUS_CODEC3_READY, + }; + + rdy = codec_ready[codec]; + + + /* Get the AC97 semaphore -- software first, so we don't spin + * pounding out IO reads on the hardware semaphore... + */ + mutex_lock(&sis->ac97_mutex); + + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + goto timeout; + + /* ... and wait for any outstanding commands to complete ... + */ + count = 0xffff; + do { + status = inw(io + SIS_AC97_STATUS); + if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY)) + break; + + udelay(1); + } while (--count); + + if (!count) + goto timeout_sema; + + /* ... before sending our command and waiting for it to finish ... + */ + outl(cmd, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* ... and reading the results (if any). + */ + val = inl(io + SIS_AC97_CMD) >> 16; + +timeout_sema: + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); +timeout: + mutex_unlock(&sis->ac97_mutex); + + if (!count) { + printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + codec, cmd); + } + + return val; +} + +static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_WRITE, + SIS_AC97_CMD_CODEC2_WRITE, + SIS_AC97_CMD_CODEC3_WRITE, + }; + sis_ac97_rw(ac97->private_data, ac97->num, + (val << 16) | (reg << 8) | cmd[ac97->num]); +} + +static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_READ, + SIS_AC97_CMD_CODEC2_READ, + SIS_AC97_CMD_CODEC3_READ, + }; + return sis_ac97_rw(ac97->private_data, ac97->num, + (reg << 8) | cmd[ac97->num]); +} + +static int __devinit sis_mixer_create(struct sis7019 *sis) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = sis_ac97_write, + .read = sis_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = sis; + + rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus); + if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]); + ac97.num = 1; + if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]); + ac97.num = 2; + if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} + +static void sis_free_suspend(struct sis7019 *sis) +{ + int i; + + for (i = 0; i < SIS_SUSPEND_PAGES; i++) + kfree(sis->suspend_state[i]); +} + +static int sis_chip_free(struct sis7019 *sis) +{ + /* Reset the chip, and disable all interrputs. + */ + outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); + udelay(10); + outl(0, sis->ioport + SIS_GCR); + outl(0, sis->ioport + SIS_GIER); + + /* Now, free everything we allocated. + */ + if (sis->irq >= 0) + free_irq(sis->irq, sis); + + if (sis->ioaddr) + iounmap(sis->ioaddr); + + pci_release_regions(sis->pci); + pci_disable_device(sis->pci); + + sis_free_suspend(sis); + return 0; +} + +static int sis_dev_free(struct snd_device *dev) +{ + struct sis7019 *sis = dev->device_data; + return sis_chip_free(sis); +} + +static int sis_chip_init(struct sis7019 *sis) +{ + unsigned long io = sis->ioport; + void __iomem *ioaddr = sis->ioaddr; + u16 status; + int count; + int i; + + /* Reset the audio controller + */ + outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); + udelay(10); + outl(0, io + SIS_GCR); + + /* Get the AC-link semaphore, and reset the codecs + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + return -EIO; + + outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* Now that we've finished the reset, find out what's attached. + */ + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status &am |