source: mainline/uspace/lib/drv/generic/remote_audio_pcm.c@ ad1aedc

/*
 * Copyright (c) 2012 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 libdrv
 * @{
 */
/** @file
 */

#include <async.h>
#include <devman.h>
#include <ddf/log.h>
#include <errno.h>
#include <str.h>
#include <as.h>
#include <sys/mman.h>

#include "audio_pcm_iface.h"
#include "ddf/driver.h"

typedef enum {
        IPC_M_AUDIO_PCM_GET_INFO_STR,
        IPC_M_AUDIO_PCM_QUERY_CAPS,
        IPC_M_AUDIO_PCM_TEST_FORMAT,
        IPC_M_AUDIO_PCM_GET_BUFFER,
        IPC_M_AUDIO_PCM_RELEASE_BUFFER,
        IPC_M_AUDIO_PCM_GET_BUFFER_POS,
        IPC_M_AUDIO_PCM_START_PLAYBACK,
        IPC_M_AUDIO_PCM_STOP_PLAYBACK,
        IPC_M_AUDIO_PCM_START_CAPTURE,
        IPC_M_AUDIO_PCM_STOP_CAPTURE,
} audio_pcm_iface_funcs_t;

/*
 * CLIENT SIDE
 */

/**
 * Open audio session with device identified by location service string.
 *
 * @param name Location service string.
 * @return Pointer to a new audio device session, NULL on failure.
 */
audio_pcm_sess_t *audio_pcm_open(const char *name)
{
        devman_handle_t device_handle = 0;
        const int ret = devman_fun_get_handle(name, &device_handle, 0);
        if (ret != EOK)
                return NULL;
        return devman_device_connect(EXCHANGE_SERIALIZE, device_handle,
            IPC_FLAG_BLOCKING);
}

/**
 * Open audio session with device identified by location service id
 *
 * @param name Location service id.
 * @return Pointer to a new audio device session, NULL on failure.
 */
audio_pcm_sess_t *audio_pcm_open_service(service_id_t id)
{
        return loc_service_connect(EXCHANGE_SERIALIZE, id, IPC_FLAG_BLOCKING);
}

/**
 * Close open audio device session.
 *
 * @param name Open audio device session.
 *
 * @note Calling this function on already closed or invalid session results
 * in undefined behavior.
 */
void audio_pcm_close(audio_pcm_sess_t *sess)
{
        if (sess)
                async_hangup(sess);
}

/**
 * Get a short description string.
 *
 * @param sess Audio device session.
 * @param name Place to store newly allocated string.
 *
 * @return Error code.
 *
 * @note Caller is responsible for freeing newly allocated memory.
 */
int audio_pcm_get_info_str(audio_pcm_sess_t *sess, const char **name)
{
        async_exch_t *exch = async_exchange_begin(sess);
        sysarg_t name_size;
        const int ret = async_req_1_1(exch,
            DEV_IFACE_ID(AUDIO_PCM_BUFFER_IFACE),
            IPC_M_AUDIO_PCM_GET_INFO_STR, &name_size);
        if (ret == EOK && name) {
                char *name_place = calloc(1, name_size);
                if (!name_place) {
                        /* Make the other side fail
                         * as it waits for read request */
                        async_data_read_start(exch, (void*)-1, 0);
                        async_exchange_end(exch);
                        return ENOMEM;
                }
                const int ret =
                    async_data_read_start(exch, name_place, name_size);
                if (ret != EOK) {
                        free(name_place);
                        async_exchange_end(exch);
                        return ret;
                }
                *name = name_place;
        }
        async_exchange_end(exch);
        return ret;
}


/**
 * Query value of specified capability.
 *
 * @param sess Audio device session.
 * @param cap  Audio device capability.
 * @param val  Place to store queried value.
 *
 * @return Error code.
 */
int audio_pcm_query_cap(audio_pcm_sess_t *sess, audio_cap_t cap, unsigned *val)
{
        if (!val)
                return EINVAL;
        async_exch_t *exch = async_exchange_begin(sess);
        sysarg_t value = 0;
        const int ret = async_req_2_1(exch,
            DEV_IFACE_ID(AUDIO_PCM_BUFFER_IFACE), IPC_M_AUDIO_PCM_QUERY_CAPS,
            cap, &value);
        if (ret == EOK)
                *val = value;
        async_exchange_end(exch);
        return ret;
}

/**
 * Query current position in device buffer.
 *
 * @param sess Audio device session.
 * @param pos Place to store the result.
 *
 * @return Error code.
 *
 * Works for both playback and capture.
 */
int audio_pcm_get_buffer_pos(audio_pcm_sess_t *sess, size_t *pos)
{
        if (!pos)
                return EINVAL;
        async_exch_t *exch = async_exchange_begin(sess);
        sysarg_t value = 0;;
        const int ret = async_req_1_1(exch,
            DEV_IFACE_ID(AUDIO_PCM_BUFFER_IFACE),
            IPC_M_AUDIO_PCM_GET_BUFFER_POS, &value);
        if (ret == EOK)
                *pos = value;
        async_exchange_end(exch);
        return ret;
}

/**
 * Test format parameters for device support.
 *
 * @param sess Audio device session.
 * @param channels Number of channels
 * @param rate Sampling rate.
 * @format Sample format.
 *
 * @return Error code.
 *
 * Works for both playback and capture. This function modifies provided
 * parameters to the nearest values supported by the device.
 */
int audio_pcm_test_format(audio_pcm_sess_t *sess, unsigned *channels,
    unsigned *rate, pcm_sample_format_t *format)
{
        async_exch_t *exch = async_exchange_begin(sess);
        sysarg_t channels_arg = channels ? *channels : 0;
        sysarg_t rate_arg = rate ? *rate : 0;
        sysarg_t format_arg = format ? *format : 0;
        const int ret = async_req_4_3(exch,
            DEV_IFACE_ID(AUDIO_PCM_BUFFER_IFACE),
            IPC_M_AUDIO_PCM_TEST_FORMAT, channels_arg, rate_arg, format_arg,
            &channels_arg, &rate_arg, &format_arg);
        async_exchange_end(exch);

        /* All OK or something has changed. Verify that it was not one of the
         * params we care about */
        if ((ret == EOK || ret == ELIMIT)
            && (!channels || *channels == channels_arg)
            && (!rate || *rate == rate_arg)
            && (!format || *format == format_arg))
                return EOK;
        if (channels)
                *channels = channels_arg;
        if (rate)
                *rate = rate_arg;
        if (format)
                *format = format_arg;
        return ret;
}

/**
 * Get device accessible playback/capture buffer.
 *
 * @param sess Audio device session.
 * @param buffer Place to store pointer to the buffer.
 * @param size Place to store buffer size (bytes).
 * @param event_rec Event callback function.
 * @param arg Event callback custom parameter.
 *
 * @return Error code.
 */
int audio_pcm_get_buffer(audio_pcm_sess_t *sess, void **buffer, size_t *size,
    async_client_conn_t event_rec, void* arg)
{
        if (!buffer || !size)
                return EINVAL;

        async_exch_t *exch = async_exchange_begin(sess);

        sysarg_t buffer_size = *size;
        const int ret = async_req_2_1(exch,
            DEV_IFACE_ID(AUDIO_PCM_BUFFER_IFACE), IPC_M_AUDIO_PCM_GET_BUFFER,
            (sysarg_t)buffer_size, &buffer_size);
        if (ret == EOK) {
                void *dst = NULL;
                int ret = async_share_in_start_0_0(exch, buffer_size, &dst);
                if (ret != EOK) {
                        async_exchange_end(exch);
                        return ret;
                }
                ret = async_connect_to_me(exch, 0, 0, 0, event_rec, arg);
                if (ret != EOK) {
                        async_exchange_end(exch);
                        return ret;
                }

                *buffer = dst;
                *size = buffer_size;
        }
        async_exchange_end(exch);
        return ret;
}

/**
 * Release device accessible playback/capture buffer.
 *
 * @param sess Audio device session.
 *
 * @return Error code.
 */
int audio_pcm_release_buffer(audio_pcm_sess_t *sess)
{
        async_exch_t *exch = async_exchange_begin(sess);
        const int ret = async_req_1_0(exch,
            DEV_IFACE_ID(AUDIO_PCM_BUFFER_IFACE),
            IPC_M_AUDIO_PCM_RELEASE_BUFFER);
        async_exchange_end(exch);
        return ret;
}

/**
 * Start playback on buffer from position 0.
 *
 * @param sess Audio device session.
 * @param frames Size of fragment (in frames).
 * @param channels Number of channels.
 * @param sample_rate Sampling rate (for one channel).
 * @param format Sample format.
 *
 * @return Error code.
 *
 * Event will be generated after every fragment. Set fragment size to
 * 0 to turn off event generation.
 */
int audio_pcm_start_playback(audio_pcm_sess_t *sess, unsigned frames,
    unsigned channels, unsigned sample_rate, pcm_sample_format_t format)
{
        if (channels > UINT16_MAX)
                return EINVAL;
        assert((format & UINT16_MAX) == format);
        const sysarg_t packed = (channels << 16) | (format & UINT16_MAX);
        async_exch_t *exch = async_exchange_begin(sess);
        const int ret = async_req_4_0(exch,
            DEV_IFACE_ID(AUDIO_PCM_BUFFER_IFACE),
            IPC_M_AUDIO_PCM_START_PLAYBACK,
            frames, sample_rate, packed);
        async_exchange_end(exch);
        return ret;
}

/**
 * Stop current playback.
 *
 * @param sess Audio device session.
 *
 * @return Error code.
 */
int audio_pcm_stop_playback(audio_pcm_sess_t *sess)
{
        async_exch_t *exch = async_exchange_begin(sess);
        const int ret = async_req_1_0(exch,
            DEV_IFACE_ID(AUDIO_PCM_BUFFER_IFACE),
            IPC_M_AUDIO_PCM_STOP_PLAYBACK);
        async_exchange_end(exch);
        return ret;
}

/**
 * Start capture on buffer from position 0.
 *
 * @param sess Audio device session.
 * @param frames Size of fragment (in frames).
 * @param channels Number of channels.
 * @param sample_rate Sampling rate (for one channel).
 * @param format Sample format.
 *
 * @return Error code.
 *
 * Event will be generated after every fragment. Set fragment size to
 * 0 to turn off event generation.
 */
int audio_pcm_start_capture(audio_pcm_sess_t *sess, unsigned frames,
    unsigned channels, unsigned sample_rate, pcm_sample_format_t format)
{
        if (channels > UINT16_MAX)
                return EINVAL;
        assert((format & UINT16_MAX) == format);
        const sysarg_t packed = (channels << 16) | (format & UINT16_MAX);
        async_exch_t *exch = async_exchange_begin(sess);
        const int ret = async_req_4_0(exch,
            DEV_IFACE_ID(AUDIO_PCM_BUFFER_IFACE), IPC_M_AUDIO_PCM_START_CAPTURE,
            frames, sample_rate, packed);
        async_exchange_end(exch);
        return ret;
}

/**
 * Stop current playback.
 *
 * @param sess Audio device session.
 *
 * @return Error code.
 */
int audio_pcm_stop_capture(audio_pcm_sess_t *sess)
{
        async_exch_t *exch = async_exchange_begin(sess);
        const int ret = async_req_1_0(exch,
            DEV_IFACE_ID(AUDIO_PCM_BUFFER_IFACE), IPC_M_AUDIO_PCM_STOP_CAPTURE);
        async_exchange_end(exch);
        return ret;
}

/*
 * SERVER SIDE
 */
static void remote_audio_pcm_get_info_str(ddf_fun_t *, void *, ipc_callid_t, ipc_call_t *);
static void remote_audio_pcm_query_caps(ddf_fun_t *, void *, ipc_callid_t, ipc_call_t *);
static void remote_audio_pcm_get_buffer_pos(ddf_fun_t *, void *, ipc_callid_t, ipc_call_t *);
static void remote_audio_pcm_test_format(ddf_fun_t *, void *, ipc_callid_t, ipc_call_t *);
static void remote_audio_pcm_get_buffer(ddf_fun_t *, void *, ipc_callid_t, ipc_call_t *);
static void remote_audio_pcm_release_buffer(ddf_fun_t *, void *, ipc_callid_t, ipc_call_t *);
static void remote_audio_pcm_start_playback(ddf_fun_t *, void *, ipc_callid_t, ipc_call_t *);
static void remote_audio_pcm_stop_playback(ddf_fun_t *, void *, ipc_callid_t, ipc_call_t *);
static void remote_audio_pcm_start_capture(ddf_fun_t *, void *, ipc_callid_t, ipc_call_t *);
static void remote_audio_pcm_stop_capture(ddf_fun_t *, void *, ipc_callid_t, ipc_call_t *);

/** Remote audio pcm buffer interface operations. */
static remote_iface_func_ptr_t remote_audio_pcm_iface_ops[] = {
        [IPC_M_AUDIO_PCM_GET_INFO_STR] = remote_audio_pcm_get_info_str,
        [IPC_M_AUDIO_PCM_QUERY_CAPS] = remote_audio_pcm_query_caps,
        [IPC_M_AUDIO_PCM_GET_BUFFER_POS] = remote_audio_pcm_get_buffer_pos,
        [IPC_M_AUDIO_PCM_TEST_FORMAT] = remote_audio_pcm_test_format,
        [IPC_M_AUDIO_PCM_GET_BUFFER] = remote_audio_pcm_get_buffer,
        [IPC_M_AUDIO_PCM_RELEASE_BUFFER] = remote_audio_pcm_release_buffer,
        [IPC_M_AUDIO_PCM_START_PLAYBACK] = remote_audio_pcm_start_playback,
        [IPC_M_AUDIO_PCM_STOP_PLAYBACK] = remote_audio_pcm_stop_playback,
        [IPC_M_AUDIO_PCM_START_CAPTURE] = remote_audio_pcm_start_capture,
        [IPC_M_AUDIO_PCM_STOP_CAPTURE] = remote_audio_pcm_stop_capture,
};

/** Remote audio mixer interface structure. */
remote_iface_t remote_audio_pcm_iface = {
        .method_count = sizeof(remote_audio_pcm_iface_ops) /
            sizeof(remote_audio_pcm_iface_ops[0]),
        .methods = remote_audio_pcm_iface_ops
};

void remote_audio_pcm_get_info_str(ddf_fun_t *fun, void *iface,
    ipc_callid_t callid, ipc_call_t *call)
{
        const audio_pcm_iface_t *pcm_iface = iface;

        if (!pcm_iface->get_info_str) {
                async_answer_0(callid, ENOTSUP);
                return;
        }
        const char *name = NULL;
        const int ret = pcm_iface->get_info_str(fun, &name);
        const size_t name_size = name ? str_size(name) + 1 : 0;
        async_answer_1(callid, ret, name_size);
        /* Send the string. */
        if (ret == EOK && name_size > 0) {
                size_t size;
                ipc_callid_t name_id;
                if (!async_data_read_receive(&name_id, &size)) {
                        async_answer_0(name_id, EPARTY);
                        return;
                }
                if (size != name_size) {
                        async_answer_0(name_id, ELIMIT);
                        return;
                }
                async_data_read_finalize(name_id, name, name_size);
        }
}

void remote_audio_pcm_query_caps(ddf_fun_t *fun, void *iface, ipc_callid_t callid, ipc_call_t *call)
{
        const audio_pcm_iface_t *pcm_iface = iface;
        const audio_cap_t cap = DEV_IPC_GET_ARG1(*call);
        if (pcm_iface->query_cap) {
                const unsigned value = pcm_iface->query_cap(fun, cap);
                async_answer_1(callid, EOK, value);
        } else {
                async_answer_0(callid, ENOTSUP);
        }
}
void remote_audio_pcm_get_buffer_pos(ddf_fun_t *fun, void *iface, ipc_callid_t callid, ipc_call_t *call)
{
        const audio_pcm_iface_t *pcm_iface = iface;
        size_t pos = 0;
        const int ret = pcm_iface->get_buffer_pos ?
            pcm_iface->get_buffer_pos(fun, &pos) : ENOTSUP;
        async_answer_1(callid, ret, pos);
}

void remote_audio_pcm_test_format(ddf_fun_t *fun, void *iface, ipc_callid_t callid, ipc_call_t *call)
{
        const audio_pcm_iface_t *pcm_iface = iface;
        unsigned channels = DEV_IPC_GET_ARG1(*call);
        unsigned rate = DEV_IPC_GET_ARG2(*call);
        pcm_sample_format_t format = DEV_IPC_GET_ARG3(*call);
        const int ret = pcm_iface->test_format ?
            pcm_iface->test_format(fun, &channels, &rate, &format) : ENOTSUP;
        async_answer_3(callid, ret, channels, rate, format);
}

void remote_audio_pcm_get_buffer(ddf_fun_t *fun, void *iface,
    ipc_callid_t callid, ipc_call_t *call)
{
        const audio_pcm_iface_t *pcm_iface = iface;

        if (!pcm_iface->get_buffer ||
            !pcm_iface->release_buffer ||
            !pcm_iface->set_event_session) {
                async_answer_0(callid, ENOTSUP);
                return;
        }
        void *buffer = NULL;
        size_t size = DEV_IPC_GET_ARG1(*call);
        int ret = pcm_iface->get_buffer(fun, &buffer, &size);
        async_answer_1(callid, ret, size);
        if (ret != EOK || size == 0)
                return;

        /* Share the buffer. */
        size_t share_size = 0;
        ipc_callid_t share_id = 0;

        ddf_msg(LVL_DEBUG2, "Receiving share request.");
        if (!async_share_in_receive(&share_id, &share_size)) {
                ddf_msg(LVL_DEBUG, "Failed to share pcm buffer.");
                pcm_iface->release_buffer(fun);
                async_answer_0(share_id, EPARTY);
                return;
        }

        ddf_msg(LVL_DEBUG2, "Checking requested share size.");
        if (share_size != size) {
                ddf_msg(LVL_DEBUG, "Incorrect pcm buffer size requested.");
                pcm_iface->release_buffer(fun);
                async_answer_0(share_id, ELIMIT);
                return;
        }

        ddf_msg(LVL_DEBUG2, "Calling share finalize.");
        ret = async_share_in_finalize(share_id, buffer, AS_AREA_WRITE
        | AS_AREA_READ);
        if (ret != EOK) {
                ddf_msg(LVL_DEBUG, "Failed to share buffer.");
                pcm_iface->release_buffer(fun);
                return;
        }

        ddf_msg(LVL_DEBUG2, "Buffer shared with size %zu, creating callback.",
            share_size);
        {
                ipc_call_t call;
                ipc_callid_t callid = async_get_call(&call);
                async_sess_t *sess =
                    async_callback_receive_start(EXCHANGE_ATOMIC, &call);
                if (sess == NULL) {
                        ddf_msg(LVL_DEBUG, "Failed to create event callback");
                        pcm_iface->release_buffer(fun);
                        async_answer_0(callid, EAGAIN);
                        return;
                }
                ret = pcm_iface->set_event_session(fun, sess);
                if (ret != EOK) {
                        ddf_msg(LVL_DEBUG, "Failed to set event callback.");
                        pcm_iface->release_buffer(fun);
                        async_answer_0(callid, ret);
                        return;
                }
                ddf_msg(LVL_DEBUG2, "Buffer and event session setup OK.");
                async_answer_0(callid, EOK);
        }
}

void remote_audio_pcm_release_buffer(ddf_fun_t *fun, void *iface,
    ipc_callid_t callid, ipc_call_t *call)
{
        const audio_pcm_iface_t *pcm_iface = iface;

        const int ret = pcm_iface->release_buffer ?
            pcm_iface->release_buffer(fun) : ENOTSUP;
        async_answer_0(callid, ret);
}

void remote_audio_pcm_start_playback(ddf_fun_t *fun, void *iface,
    ipc_callid_t callid, ipc_call_t *call)
{
        const audio_pcm_iface_t *pcm_iface = iface;

        const unsigned frames = DEV_IPC_GET_ARG1(*call);
        const unsigned rate = DEV_IPC_GET_ARG2(*call);
        const unsigned channels = (DEV_IPC_GET_ARG3(*call) >> 16) & UINT8_MAX;
        const pcm_sample_format_t format = DEV_IPC_GET_ARG3(*call) & UINT16_MAX;

        const int ret = pcm_iface->start_playback
            ? pcm_iface->start_playback(fun, frames, channels, rate, format)
            : ENOTSUP;
        async_answer_0(callid, ret);
}

void remote_audio_pcm_stop_playback(ddf_fun_t *fun, void *iface,
    ipc_callid_t callid, ipc_call_t *call)
{
        const audio_pcm_iface_t *pcm_iface = iface;

        const int ret = pcm_iface->stop_playback ?
            pcm_iface->stop_playback(fun) : ENOTSUP;
        async_answer_0(callid, ret);
}

void remote_audio_pcm_start_capture(ddf_fun_t *fun, void *iface,
    ipc_callid_t callid, ipc_call_t *call)
{
        const audio_pcm_iface_t *pcm_iface = iface;

        const unsigned frames = DEV_IPC_GET_ARG1(*call);
        const unsigned rate = DEV_IPC_GET_ARG2(*call);
        const unsigned channels = (DEV_IPC_GET_ARG3(*call) >> 16) & UINT16_MAX;
        const pcm_sample_format_t format = DEV_IPC_GET_ARG3(*call) & UINT16_MAX;

        const int ret = pcm_iface->start_capture
            ? pcm_iface->start_capture(fun, frames, channels, rate, format)
            : ENOTSUP;
        async_answer_0(callid, ret);
}

void remote_audio_pcm_stop_capture(ddf_fun_t *fun, void *iface,
    ipc_callid_t callid, ipc_call_t *call)
{
        const audio_pcm_iface_t *pcm_iface = iface;

        const int ret = pcm_iface->stop_capture ?
            pcm_iface->stop_capture(fun) : ENOTSUP;
        async_answer_0(callid, ret);
}

#if 0

void remote_audio_mixer_get_info(
    ddf_fun_t *fun, void *iface, ipc_callid_t callid, ipc_call_t *call)
{
        audio_mixer_iface_t *mixer_iface = iface;

        if (!mixer_iface->get_info) {
                async_answer_0(callid, ENOTSUP);
                return;
        }
        const char *name = NULL;
        unsigned items = 0;
        const int ret = mixer_iface->get_info(fun, &name, &items);
        const size_t name_size = name ? str_size(name) + 1 : 0;
        async_answer_2(callid, ret, name_size, items);
        /* Send the name. */
        if (ret == EOK && name_size > 0) {
                size_t size;
                ipc_callid_t name_id;
                if (!async_data_read_receive(&name_id, &size)) {
                        async_answer_0(name_id, EPARTY);
                        return;
                }
                if (size != name_size) {
                        async_answer_0(name_id, ELIMIT);
                        return;
                }
                async_data_read_finalize(name_id, name, name_size);
        }
}

void remote_audio_mixer_get_item_info(
    ddf_fun_t *fun, void *iface, ipc_callid_t callid, ipc_call_t *call)
{
        audio_mixer_iface_t *mixer_iface = iface;

        if (!mixer_iface->get_item_info) {
                async_answer_0(callid, ENOTSUP);
                return;
        }

        const unsigned item = DEV_IPC_GET_ARG1(*call);
        const char *name = NULL;
        unsigned channels = 0;
        const int ret = mixer_iface->get_item_info(fun, item, &name, &channels);
        const size_t name_size = name ? str_size(name) + 1 : 0;
        async_answer_2(callid, ret, name_size, channels);
        /* Send the name. */
        if (ret == EOK && name_size > 0) {
                size_t size;
                ipc_callid_t name_id;
                if (!async_data_read_receive(&name_id, &size)) {
                        async_answer_0(name_id, EPARTY);
                        return;
                }
                if (size != name_size) {
                        async_answer_0(name_id, ELIMIT);
                        return;
                }
                async_data_read_finalize(name_id, name, name_size);
        }
}

void remote_audio_mixer_get_channel_info(
    ddf_fun_t *fun, void *iface, ipc_callid_t callid, ipc_call_t *call)
{
        audio_mixer_iface_t *mixer_iface = iface;

        if (!mixer_iface->get_channel_info) {
                async_answer_0(callid, ENOTSUP);
                return;
        }

        const unsigned item = DEV_IPC_GET_ARG1(*call);
        const unsigned channel = DEV_IPC_GET_ARG2(*call);
        const char *name = NULL;
        unsigned levels = 0;
        const int ret =
            mixer_iface->get_channel_info(fun, item, channel, &name, &levels);
        const size_t name_size = name ? str_size(name) + 1 : 0;
        async_answer_2(callid, ret, name_size, levels);
        /* Send the name. */
        if (ret == EOK && name_size > 0) {
                size_t size;
                ipc_callid_t name_id;
                if (!async_data_read_receive(&name_id, &size)) {
                        async_answer_0(name_id, EPARTY);
                        return;
                }
                if (size != name_size) {
                        async_answer_0(name_id, ELIMIT);
                        return;
                }
                async_data_read_finalize(name_id, name, name_size);
        }
}

void remote_audio_mixer_channel_mute_set(
    ddf_fun_t *fun, void *iface, ipc_callid_t callid, ipc_call_t *call)
{
        audio_mixer_iface_t *mixer_iface = iface;

        if (!mixer_iface->channel_mute_set) {
                async_answer_0(callid, ENOTSUP);
                return;
        }
        const unsigned item = DEV_IPC_GET_ARG1(*call);
        const unsigned channel = DEV_IPC_GET_ARG2(*call);
        const bool mute = DEV_IPC_GET_ARG3(*call);
        const int ret = mixer_iface->channel_mute_set(fun, item, channel, mute);
        async_answer_0(callid, ret);
}

void remote_audio_mixer_channel_mute_get(
    ddf_fun_t *fun, void *iface, ipc_callid_t callid, ipc_call_t *call)
{
        audio_mixer_iface_t *mixer_iface = iface;

        if (!mixer_iface->channel_mute_get) {
                async_answer_0(callid, ENOTSUP);
                return;
        }
        const unsigned item = DEV_IPC_GET_ARG1(*call);
        const unsigned channel = DEV_IPC_GET_ARG2(*call);
        bool mute = false;
        const int ret =
            mixer_iface->channel_mute_get(fun, item, channel, &mute);
        async_answer_1(callid, ret, mute);
}

void remote_audio_mixer_channel_volume_set(
    ddf_fun_t *fun, void *iface, ipc_callid_t callid, ipc_call_t *call)
{
        audio_mixer_iface_t *mixer_iface = iface;

        if (!mixer_iface->channel_volume_set) {
                async_answer_0(callid, ENOTSUP);
                return;
        }
        const unsigned item = DEV_IPC_GET_ARG1(*call);
        const unsigned channel = DEV_IPC_GET_ARG2(*call);
        const unsigned level = DEV_IPC_GET_ARG3(*call);
        const int ret =
            mixer_iface->channel_volume_set(fun, item, channel, level);
        async_answer_0(callid, ret);
}

void remote_audio_mixer_channel_volume_get(
    ddf_fun_t *fun, void *iface, ipc_callid_t callid, ipc_call_t *call)
{
        audio_mixer_iface_t *mixer_iface = iface;

        if (!mixer_iface->channel_volume_get) {
                async_answer_0(callid, ENOTSUP);
                return;
        }
        const unsigned item = DEV_IPC_GET_ARG1(*call);
        const unsigned channel = DEV_IPC_GET_ARG2(*call);
        unsigned current = 0, max = 0;
        const int ret =
            mixer_iface->channel_volume_get(fun, item, channel, &current, &max);
        async_answer_2(callid, ret, current, max);
}
#endif

/**
 * @}
 */