source: mainline/uspace/drv/audio/sb16/dsp.c@ d2765ab3

/*
 * Copyright (c) 2011 Jan Vesely
 * All rights reserved.
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 *
 * - Redistributions of source code must retain the above copyright
 *   notice, this list of conditions and the following disclaimer.
 * - Redistributions in binary form must reproduce the above copyright
 *   notice, this list of conditions and the following disclaimer in the
 *   documentation and/or other materials provided with the distribution.
 * - The name of the author may not be used to endorse or promote products
 *   derived from this software without specific prior written permission.
 *
 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
 * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
 * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
 * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
 * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
 * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
 * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
 * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
 * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 */
/** @addtogroup drvaudiosb16
 * @{
 */
/** @file
 * @brief DSP helper functions implementation
 */

#include <as.h>
#include <bool.h>
#include <ddi.h>
#include <devman.h>
#include <device/hw_res.h>
#include <libarch/ddi.h>
#include <libarch/barrier.h>
#include <str_error.h>
#include <audio_pcm_iface.h>

#include "ddf_log.h"
#include "dsp_commands.h"
#include "dsp.h"

/* Maximum allowed transfer size for ISA DMA transfers is 64kB */
#define MAX_BUFFER_SIZE (4 * 1024) // use 4kB for now

#ifndef DSP_RETRY_COUNT
#define DSP_RETRY_COUNT 100
#endif

#define DSP_RESET_RESPONSE 0xaa
#define DSP_RATE_LIMIT 45000

#define AUTO_DMA_MODE

static inline int sb_dsp_read(sb_dsp_t *dsp, uint8_t *data)
{
        assert(data);
        assert(dsp);
        uint8_t status;
        size_t attempts = DSP_RETRY_COUNT;
        do {
                status = pio_read_8(&dsp->regs->dsp_read_status);
        } while (--attempts && ((status & DSP_READ_READY) == 0));

        if ((status & DSP_READ_READY) == 0)
                return EIO;

        *data = pio_read_8(&dsp->regs->dsp_data_read);
        return EOK;
}

static inline int sb_dsp_write(sb_dsp_t *dsp, uint8_t data)
{
        assert(dsp);
        uint8_t status;
        size_t attempts = DSP_RETRY_COUNT;
        do {
                status = pio_read_8(&dsp->regs->dsp_write);
        } while (--attempts && ((status & DSP_WRITE_BUSY) != 0));

        if ((status & DSP_WRITE_BUSY))
                return EIO;

        pio_write_8(&dsp->regs->dsp_write, data);
        return EOK;
}

static inline void sb_dsp_reset(sb_dsp_t *dsp)
{
        assert(dsp);
        /* Reset DSP, see Chapter 2 of Sound Blaster HW programming guide */
        pio_write_8(&dsp->regs->dsp_reset, 1);
        udelay(3); /* Keep reset for 3 us */
        pio_write_8(&dsp->regs->dsp_reset, 0);
}

static inline int sb_setup_dma(sb_dsp_t *dsp, uintptr_t pa, size_t size)
{
        async_sess_t *sess = devman_parent_device_connect(EXCHANGE_ATOMIC,
            dsp->sb_dev->handle, IPC_FLAG_BLOCKING);

        const int ret = hw_res_dma_channel_setup(sess,
            dsp->dma16_channel, pa, size,
            DMA_MODE_READ | DMA_MODE_AUTO | DMA_MODE_ON_DEMAND);

        async_hangup(sess);
        return ret;
}

static inline int sb_setup_buffer(sb_dsp_t *dsp, size_t size)
{
        assert(dsp);
        if (size > MAX_BUFFER_SIZE || size == 0 || (size % 2) == 1)
                size = MAX_BUFFER_SIZE;
        void *buffer = NULL, *pa = NULL;
        int ret = dmamem_map_anonymous(size, AS_AREA_WRITE | AS_AREA_READ,
            0, &pa, &buffer);
        if (ret != EOK) {
                ddf_log_error("Failed to allocate DMA buffer.");
                return ENOMEM;
        }

        ddf_log_verbose("Setup dma buffer at %p(%p).", buffer, pa, size);
        assert((uintptr_t)pa < (1 << 25));

        /* Setup 16 bit channel */
        ret = sb_setup_dma(dsp, (uintptr_t)pa, size);
        if (ret == EOK) {
                dsp->buffer.data = buffer;
                dsp->buffer.size = size;
        } else {
                ddf_log_error("Failed to setup DMA16 channel: %s.",
                    str_error(ret));
                dmamem_unmap_anonymous(buffer);
        }
        return ret;
}

static inline void sb_clear_buffer(sb_dsp_t *dsp)
{
        assert(dsp);
        dmamem_unmap_anonymous(dsp->buffer.data);
        dsp->buffer.data = NULL;
        dsp->buffer.size = 0;
}

static inline size_t sample_count(pcm_sample_format_t format, size_t byte_count)
{
        return byte_count / pcm_sample_format_size(format);
}

int sb_dsp_init(sb_dsp_t *dsp, sb16_regs_t *regs, ddf_dev_t *dev,
    int dma8, int dma16)
{
        assert(dsp);
        dsp->regs = regs;
        dsp->dma8_channel = dma8;
        dsp->dma16_channel = dma16;
        dsp->event_session = NULL;
        dsp->event_exchange = NULL;
        dsp->sb_dev = dev;
        dsp->status = DSP_STOPPED;
        dsp->ignore_interrupts = false;
        sb_dsp_reset(dsp);
        /* "DSP takes about 100 microseconds to initialize itself" */
        udelay(100);
        uint8_t response;
        const int ret = sb_dsp_read(dsp, &response);
        if (ret != EOK) {
                ddf_log_error("Failed to read DSP reset response value.");
                return ret;
        }
        if (response != DSP_RESET_RESPONSE) {
                ddf_log_error("Invalid DSP reset response: %x.", response);
                return EIO;
        }

        /* Get DSP version number */
        sb_dsp_write(dsp, DSP_VERSION);
        sb_dsp_read(dsp, &dsp->version.major);
        sb_dsp_read(dsp, &dsp->version.minor);

        return ret;
}

void sb_dsp_interrupt(sb_dsp_t *dsp)
{
        assert(dsp);

#ifndef AUTO_DMA_MODE
        if (dsp->status == DSP_PLAYBACK) {
                sb_dsp_write(dsp, SINGLE_DMA_16B_DA);
                sb_dsp_write(dsp, dsp->active.mode);
                sb_dsp_write(dsp, (dsp->active.samples - 1) & 0xff);
                sb_dsp_write(dsp, (dsp->active.samples - 1) >> 8);
        }

        if (dsp->status == DSP_RECORDING) {
                sb_dsp_write(dsp, SINGLE_DMA_16B_AD);
                sb_dsp_write(dsp, dsp->active.mode);
                sb_dsp_write(dsp, (dsp->active.samples - 1) & 0xff);
                sb_dsp_write(dsp, (dsp->active.samples - 1) >> 8);
        }
#endif

        if (dsp->ignore_interrupts)
                return;

        dsp->active.frame_count +=
            dsp->active.samples / ((dsp->active.mode & DSP_MODE_STEREO) ? 2 : 1);

        if (dsp->event_exchange) {
                switch (dsp->status) {
                case DSP_PLAYBACK:
                        async_msg_1(dsp->event_exchange,
                            PCM_EVENT_FRAMES_PLAYED, dsp->active.frame_count);
                        break;
                case DSP_RECORDING:
                        async_msg_1(dsp->event_exchange,
                            PCM_EVENT_FRAMES_RECORDED, dsp->active.frame_count);
                        break;
                default:
                case DSP_STOPPED:
                        ddf_log_warning("Interrupt while DSP stopped and "
                            "event exchange present. Terminating exchange");
                        async_exchange_end(dsp->event_exchange);
                        dsp->event_exchange = NULL;
                }
        } else {
                ddf_log_warning("Interrupt with no event consumer.");
        }
}

unsigned sb_dsp_query_cap(sb_dsp_t *dsp, audio_cap_t cap)
{
        switch(cap) {
        case AUDIO_CAP_RECORD:
        case AUDIO_CAP_PLAYBACK:
        case AUDIO_CAP_INTERRUPT:
                return 1;
        case AUDIO_CAP_MAX_BUFFER:
                return MAX_BUFFER_SIZE;
        case AUDIO_CAP_INTERRUPT_MIN_FRAMES:
                return 1;
        case AUDIO_CAP_INTERRUPT_MAX_FRAMES:
                return 16535;
        case AUDIO_CAP_BUFFER_POS:
        default:
                return 0;
        }
}

int sb_dsp_test_format(sb_dsp_t *dsp, unsigned *channels, unsigned *rate,
  pcm_sample_format_t *format)
{
        int ret = EOK;
        if (*channels == 0 || *channels > 2) {
                *channels = 2;
                ret = ELIMIT;
        }
        if (*rate > DSP_RATE_LIMIT) {
                *rate = DSP_RATE_LIMIT;
                ret = ELIMIT;
        }
        //TODO 8bit DMA supports 8bit formats
        if (*format != PCM_SAMPLE_SINT16_LE && *format != PCM_SAMPLE_UINT16_LE) {
                *format = pcm_sample_format_is_signed(*format) ?
                    PCM_SAMPLE_SINT16_LE : PCM_SAMPLE_UINT16_LE;
                ret = ELIMIT;
        }
        return ret;
}

int sb_dsp_get_buffer(sb_dsp_t *dsp, void **buffer, size_t *size)
{
        assert(dsp);
        assert(size);

        /* buffer is already setup by for someone, refuse to work until
         * it's released */
        if (dsp->buffer.data)
                return EBUSY;

        const int ret = sb_setup_buffer(dsp, *size);
        if (ret == EOK) {
                ddf_log_debug("Providing buffer: %p, %zu B.",
                    dsp->buffer.data, dsp->buffer.size);

                if (buffer)
                        *buffer = dsp->buffer.data;
                if (size)
                        *size = dsp->buffer.size;
        }
        return ret;
}

int sb_dsp_set_event_session(sb_dsp_t *dsp, async_sess_t *session)
{
        assert(dsp);
        assert(session);
        if (dsp->event_session)
                return EBUSY;
        dsp->event_session = session;
        ddf_log_debug("Set event session.");
        return EOK;
}

int sb_dsp_release_buffer(sb_dsp_t *dsp)
{
        assert(dsp);
        sb_clear_buffer(dsp);
        async_exchange_end(dsp->event_exchange);
        dsp->event_exchange = NULL;
        if (dsp->event_session)
                async_hangup(dsp->event_session);
        dsp->event_session = NULL;
        ddf_log_debug("DSP buffer released.");
        return EOK;
}

int sb_dsp_start_playback(sb_dsp_t *dsp, unsigned frames,
    unsigned channels, unsigned sampling_rate, pcm_sample_format_t format)
{
        assert(dsp);

        if (!dsp->event_session)
                return EINVAL;

        /* Check supported parameters */
        ddf_log_debug("Requested playback: %u frames, %uHz, %s, %u channel(s).",
            frames, sampling_rate, pcm_sample_format_str(format), channels);
        if (sb_dsp_test_format(dsp, &channels, &sampling_rate, &format) != EOK)
                return ENOTSUP;

        /* Client requested regular interrupts */
        if (frames) {
                dsp->event_exchange = async_exchange_begin(dsp->event_session);
                if (!dsp->event_exchange)
                        return ENOMEM;
                dsp->ignore_interrupts = false;
        }

        const bool sign = (format == PCM_SAMPLE_SINT16_LE);

        sb_dsp_write(dsp, SET_SAMPLING_RATE_OUTPUT);
        sb_dsp_write(dsp, sampling_rate >> 8);
        sb_dsp_write(dsp, sampling_rate & 0xff);

        ddf_log_verbose("Sample rate: %hhx:%hhx.",
            sampling_rate >> 8, sampling_rate & 0xff);

#ifdef AUTO_DMA_MODE
        sb_dsp_write(dsp, AUTO_DMA_16B_DA_FIFO);
#else
        sb_dsp_write(dsp, SINGLE_DMA_16B_DA);
#endif

        dsp->active.mode = 0
            | (sign ? DSP_MODE_SIGNED : 0)
            | (channels == 2 ? DSP_MODE_STEREO : 0);
        dsp->active.samples = frames * channels;
        dsp->active.frame_count = 0;

        sb_dsp_write(dsp, dsp->active.mode);
        sb_dsp_write(dsp, (dsp->active.samples - 1) & 0xff);
        sb_dsp_write(dsp, (dsp->active.samples - 1) >> 8);

        ddf_log_verbose("Playback started, interrupt every %u samples "
            "(~1/%u sec)", dsp->active.samples,
            sampling_rate / (dsp->active.samples * channels));

        dsp->status = DSP_PLAYBACK;

        return EOK;
}

int sb_dsp_stop_playback(sb_dsp_t *dsp)
{
        assert(dsp);
        sb_dsp_write(dsp, DMA_16B_EXIT);
        ddf_log_debug("Stopping playback on buffer.");
        async_msg_0(dsp->event_exchange, PCM_EVENT_PLAYBACK_TERMINATED);
        async_exchange_end(dsp->event_exchange);
        dsp->event_exchange = NULL;
        return EOK;
}

int sb_dsp_start_record(sb_dsp_t *dsp, unsigned frames,
    unsigned channels, unsigned sampling_rate, pcm_sample_format_t format)
{
        assert(dsp);

        if (!dsp->event_session)
                return EINVAL;

        /* Check supported parameters */
        ddf_log_debug("Requested record: %u frames, %uHz, %s, %u channel(s).",
            frames, sampling_rate, pcm_sample_format_str(format), channels);
        if (sb_dsp_test_format(dsp, &channels, &sampling_rate, &format) != EOK)
                return ENOTSUP;

        /* client requested regular interrupts */
        if (frames) {
                dsp->event_exchange = async_exchange_begin(dsp->event_session);
                if (!dsp->event_exchange)
                        return ENOMEM;
                dsp->ignore_interrupts = false;
        }

        const bool sign = (format == PCM_SAMPLE_SINT16_LE);

        sb_dsp_write(dsp, SET_SAMPLING_RATE_OUTPUT);
        sb_dsp_write(dsp, sampling_rate >> 8);
        sb_dsp_write(dsp, sampling_rate & 0xff);

        ddf_log_verbose("Sampling rate: %hhx:%hhx.",
            sampling_rate >> 8, sampling_rate & 0xff);

#ifdef AUTO_DMA_MODE
        sb_dsp_write(dsp, AUTO_DMA_16B_AD_FIFO);
#else
        sb_dsp_write(dsp, SINGLE_DMA_16B_AD);
#endif

        dsp->active.mode = 0 |
            (sign ? DSP_MODE_SIGNED : 0) |
            (channels == 2 ? DSP_MODE_STEREO : 0);
        dsp->active.samples = frames * channels;
        dsp->active.frame_count = 0;

        sb_dsp_write(dsp, dsp->active.mode);
        sb_dsp_write(dsp, (dsp->active.samples - 1) & 0xff);
        sb_dsp_write(dsp, (dsp->active.samples - 1) >> 8);

        ddf_log_verbose("Recording started started, interrupt every %u samples "
            "(~1/%u sec)", dsp->active.samples,
            sampling_rate / (dsp->active.samples * channels));
        dsp->status = DSP_RECORDING;

        return EOK;
}

int sb_dsp_stop_record(sb_dsp_t *dsp)
{
        assert(dsp);
        sb_dsp_write(dsp, DMA_16B_EXIT);
        ddf_log_debug("Stopped recording");
        async_msg_0(dsp->event_exchange, PCM_EVENT_RECORDING_TERMINATED);
        async_exchange_end(dsp->event_exchange);
        dsp->event_exchange = NULL;
        return EOK;
}
/**
 * @}
 */