source: mainline/uspace/srv/audio/hound/audio_device.c@ 7373117

/*
 * 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 audio
 * @brief HelenOS sound server.
 * @{
 */
/** @file
 */

#include <assert.h>
#include <async.h>
#include <errno.h>
#include <inttypes.h>
#include <loc.h>
#include <str.h>
#include <str_error.h>


#include "audio_device.h"
#include "log.h"

/* hardwired to provide ~21ms per fragment */
#define BUFFER_PARTS   16

static int device_sink_connection_callback(audio_sink_t *sink, bool new);
static int device_source_connection_callback(audio_source_t *source, bool new);
static void device_event_callback(ipc_callid_t iid, ipc_call_t *icall, void *arg);
static int device_check_format(audio_sink_t* sink);
static int get_buffer(audio_device_t *dev);
static int release_buffer(audio_device_t *dev);
static void advance_buffer(audio_device_t *dev, size_t size);
static inline bool is_running(audio_device_t *dev)
{
        assert(dev);
        /* we release buffer on stop so this should be enough */
        return (bool)dev->buffer.base;
}

/**
 * Initialize audio device structure.
 * @param dev The structure to initialize.
 * @param id Location service id of the device driver.
 * @param name Name of the device.
 * @return Error code.
 */
int audio_device_init(audio_device_t *dev, service_id_t id, const char *name)
{
        assert(dev);
        link_initialize(&dev->link);
        dev->id = id;
        dev->name = str_dup(name);
        dev->sess = audio_pcm_open_service(id);
        if (!dev->sess) {
                log_debug("Failed to connect to device \"%s\"", name);
                return ENOMEM;
        }

        audio_sink_init(&dev->sink, name, dev, device_sink_connection_callback,
            device_check_format, NULL, &AUDIO_FORMAT_ANY);
        audio_source_init(&dev->source, name, dev,
            device_source_connection_callback, NULL, &AUDIO_FORMAT_ANY);

        /* Init buffer members */
        dev->buffer.base = NULL;
        dev->buffer.position = NULL;
        dev->buffer.size = 0;
        dev->buffer.fragment_size = 0;

        log_verbose("Initialized device (%p) '%s' with id %" PRIun ".",
            dev, dev->name, dev->id);

        return EOK;
}

/**
 * Restore resource cplaimed during initialization.
 * @param dev The device to release.
 *
 * NOT IMPLEMENTED
 */
void audio_device_fini(audio_device_t *dev)
{
        //TODO implement;
}

/**
 * Get device provided audio source.
 * @param dev Th device.
 * @return pointer to aa audio source structure, NULL if the device is not
 *         capable of capturing audio.
 */
audio_source_t * audio_device_get_source(audio_device_t *dev)
{
        assert(dev);
        if (audio_pcm_query_cap(dev->sess, AUDIO_CAP_CAPTURE))
                return &dev->source;
        return NULL;
}

/**
 * Get device provided audio sink.
 * @param dev Th device.
 * @return pointer to aa audio source structure, NULL if the device is not
 *         capable of audio playback.
 */
audio_sink_t * audio_device_get_sink(audio_device_t *dev)
{
        assert(dev);
        if (audio_pcm_query_cap(dev->sess, AUDIO_CAP_PLAYBACK))
                return &dev->sink;
        return NULL;
}

/**
 * Handle connection addition and removal.
 * @param sink audio sink that is connected or disconnected.
 * @param new True of a connection was added, false otherwise.
 * @return Error code.
 *
 * Starts playback on first connection. Stops playback when there are no
 * connections.
 */
static int device_sink_connection_callback(audio_sink_t* sink, bool new)
{
        assert(sink);
        audio_device_t *dev = sink->private_data;
        if (new && list_count(&sink->connections) == 1) {
                log_verbose("First connection on device sink '%s'", sink->name);

                int ret = get_buffer(dev);
                if (ret != EOK) {
                        log_error("Failed to get device buffer: %s",
                            str_error(ret));
                        return ret;
                }
                audio_pcm_register_event_callback(dev->sess,
                    device_event_callback, dev);\

                /* Fill the buffer first. Fill the first two fragments,
                 * so that we stay one fragment ahead */
                pcm_format_silence(dev->buffer.base, dev->buffer.size,
                    &dev->sink.format);
                //TODO add underrun detection.
                const size_t size = dev->buffer.fragment_size * 2;
                /* We never cross the end of the buffer here */
                audio_sink_mix_inputs(&dev->sink, dev->buffer.position, size);
                advance_buffer(dev, size);

                const unsigned frames = dev->buffer.fragment_size /
                    pcm_format_frame_size(&dev->sink.format);
                log_verbose("Fragment frame count %u", frames);
                ret = audio_pcm_start_playback_fragment(dev->sess, frames,
                    dev->sink.format.channels, dev->sink.format.sampling_rate,
                    dev->sink.format.sample_format);
                if (ret != EOK) {
                        log_error("Failed to start playback: %s",
                            str_error(ret));
                        release_buffer(dev);
                        return ret;
                }
        }
        if (list_count(&sink->connections) == 0) {
                assert(!new);
                log_verbose("Removed last connection on device sink '%s'",
                    sink->name);
                int ret = audio_pcm_stop_playback_immediate(dev->sess);
                if (ret != EOK) {
                        log_error("Failed to stop playback: %s",
                            str_error(ret));
                        return ret;
                }
                dev->sink.format = AUDIO_FORMAT_ANY;
                ret = release_buffer(dev);
                if (ret != EOK) {
                        log_error("Failed to release buffer: %s",
                            str_error(ret));
                        return ret;
                }
        }
        return EOK;
}

/**
 * Handle connection addition and removal.
 * @param source audio source that is connected or disconnected.
 * @param new True of a connection was added, false otherwise.
 * @return Error code.
 *
 * Starts capture on first connection. Stops capture when there are no
 * connections.
 */
static int device_source_connection_callback(audio_source_t *source, bool new)
{
        assert(source);
        audio_device_t *dev = source->private_data;
        if (new && list_count(&source->connections) == 1) {
                int ret = get_buffer(dev);
                if (ret != EOK) {
                        log_error("Failed to get device buffer: %s",
                            str_error(ret));
                        return ret;
                }

                //TODO set and test format

                const unsigned frames = dev->buffer.fragment_size /
                    pcm_format_frame_size(&dev->sink.format);
                ret = audio_pcm_start_capture_fragment(dev->sess, frames,
                    dev->source.format.channels,
                    dev->source.format.sampling_rate,
                    dev->source.format.sample_format);
                if (ret != EOK) {
                        log_error("Failed to start recording: %s",
                            str_error(ret));
                        release_buffer(dev);
                        return ret;
                }
        }
        if (list_count(&source->connections) == 0) { /* Disconnected */
                assert(!new);
                int ret = audio_pcm_stop_capture_immediate(dev->sess);
                if (ret != EOK) {
                        log_error("Failed to start recording: %s",
                            str_error(ret));
                        return ret;
                }
                source->format = AUDIO_FORMAT_ANY;
                ret = release_buffer(dev);
                if (ret != EOK) {
                        log_error("Failed to release buffer: %s",
                            str_error(ret));
                        return ret;
                }
                audio_pcm_unregister_event_callback(dev->sess);
        }

        return EOK;
}

/**
 * Audio device event handler.
 * @param iid initial call id.
 * @param icall initial call structure.
 * @param arg (unused)
 */
static void device_event_callback(ipc_callid_t iid, ipc_call_t *icall, void *arg)
{
        /* Answer initial request */
        async_answer_0(iid, EOK);
        audio_device_t *dev = arg;
        assert(dev);
        while (1) {
                ipc_call_t call;
                ipc_callid_t callid = async_get_call(&call);
                async_answer_0(callid, EOK);
                switch(IPC_GET_IMETHOD(call)) {
                case PCM_EVENT_FRAMES_PLAYED: {
                        struct timeval time1;
                        getuptime(&time1);
                        //TODO add underrun detection.
                        /* We never cross the end of the buffer here */
                        audio_sink_mix_inputs(&dev->sink, dev->buffer.position,
                            dev->buffer.fragment_size);
                        advance_buffer(dev, dev->buffer.fragment_size);
                        struct timeval time2;
                        getuptime(&time2);
                        log_verbose("Time to mix sources: %li\n",
                            tv_sub(&time2, &time1));
                        break;
                }
                case PCM_EVENT_PLAYBACK_TERMINATED:
                        log_verbose("Playback terminated!");
                        return;
                case PCM_EVENT_FRAMES_CAPTURED: {
                        const int ret = audio_source_push_data(&dev->source,
                            dev->buffer.position, dev->buffer.fragment_size);
                        advance_buffer(dev, dev->buffer.fragment_size);
                        if (ret != EOK)
                                log_warning("Failed to push recorded data");
                        break;
                }
                case PCM_EVENT_CAPTURE_TERMINATED:
                        log_verbose("Recording terminated!");
                        return;
                }

        }
}

/**
 * Test format against hardware limits.
 * @param sink audio playback device.
 * @return Error code.
 */
static int device_check_format(audio_sink_t* sink)
{
        assert(sink);
        audio_device_t *dev = sink->private_data;
        assert(dev);
        /* Check whether we are running */
        if (is_running(dev))
                return EBUSY;
        log_verbose("Checking format on sink %s", sink->name);
        return audio_pcm_test_format(dev->sess, &sink->format.channels,
            &sink->format.sampling_rate, &sink->format.sample_format);
}

/**
 * Get access to device buffer.
 * @param dev Audio device.
 * @return Error code.
 */
static int get_buffer(audio_device_t *dev)
{
        assert(dev);
        if (!dev->sess) {
                log_debug("No connection to device");
                return EIO;
        }
        if (dev->buffer.base) {
                log_debug("We already have a buffer");
                return EBUSY;
        }

        /* Ask for largest buffer possible */
        size_t preferred_size = 0;

        const int ret = audio_pcm_get_buffer(dev->sess, &dev->buffer.base,
            &preferred_size);
        if (ret == EOK) {
                dev->buffer.size = preferred_size;
                dev->buffer.fragment_size = dev->buffer.size / BUFFER_PARTS;
                dev->buffer.position = dev->buffer.base;
        }
        return ret;

}

/**
 * Surrender access to device buffer.
 * @param dev Audio device.
 * @return Error code.
 */
static int release_buffer(audio_device_t *dev)
{
        assert(dev);
        assert(dev->buffer.base);

        const int ret = audio_pcm_release_buffer(dev->sess);
        if (ret == EOK) {
                dev->buffer.base = NULL;
                dev->buffer.size = 0;
                dev->buffer.position = NULL;
        } else {
                log_warning("Failed to release buffer: %s", str_error(ret));
        }
        return ret;
}

/**
 * Move buffer position pointer.
 * @param dev Audio device.
 * @param size number of bytes to move forward
 */
static void advance_buffer(audio_device_t *dev, size_t size)
{
        assert(dev);
        assert(dev->buffer.position >= dev->buffer.base);
        assert(dev->buffer.position < (dev->buffer.base + dev->buffer.size));
        dev->buffer.position += size;
        if (dev->buffer.position == (dev->buffer.base + dev->buffer.size))
                dev->buffer.position = dev->buffer.base;
}
/**
 * @}
 */