/*
 * Copyright (C) 2012-13 Wolfson Microelectronics plc
 *
 * This code is heavily based on AOSP HAL for the asus/grouper
 *
 * Copyright (C) 2012 The Android Open Source Project
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 *      http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 */

#define LOG_TAG "tinyhal"
/*#define LOG_NDEBUG 0*/

#include <errno.h>
#include <pthread.h>
#include <stdint.h>
#include <stdlib.h>
#include <sys/time.h>

#include <cutils/log.h>
#include <cutils/properties.h>
#include <cutils/str_parms.h>

#include <utils/Timers.h>

#include <hardware/audio.h>
#include <hardware/hardware.h>

#include <system/audio.h>

#include <tinyalsa/asoundlib.h>
#include <sound/compress_params.h>
#include <sound/compress_offload.h>
#include <tinycompress/tinycompress.h>

#include <audio_utils/resampler.h>

#include "audio_config.h"
#include "voice_trigger.h"


#define OUT_PERIOD_SIZE_DEFAULT 1024
#define OUT_PERIOD_COUNT_DEFAULT 4
#define OUT_SAMPLING_RATE_DEFAULT 44100
#define OUT_CHANNEL_MASK_DEFAULT AUDIO_CHANNEL_OUT_STEREO

#define IN_PERIOD_COUNT_DEFAULT 4
#define IN_PERIOD_SIZE_DEFAULT 1024
#define IN_SAMPLING_RATE_DEFAULT 44100
#define IN_CHANNEL_MASK_DEFAULT AUDIO_CHANNEL_IN_MONO

/* AudioFlinger does not re-read the buffer size after
 * issuing a routing or input_source change so the
 * default buffer size must be suitable for both PCM
 * and compressed inputs
 */
#define IN_COMPRESS_BUFFER_SIZE_DEFAULT 8192

/* Maximum time we'll wait for data from a compress_pcm input */
#define MAX_COMPRESS_PCM_TIMEOUT_MS     2100

struct voice_control_trigger {
    pthread_t thread;
    pthread_mutex_t lock;
    pthread_cond_t waitcv;
    struct compress *compress;
    void (*callback)(void *param);
    void *callback_param;
    bool own_compress;
    volatile bool wait;
    volatile bool terminate;
    volatile bool triggered;
};

struct audio_device {
    struct audio_hw_device hw_device;

    pthread_mutex_t lock; /* see note below on mutex acquisition order */
    bool standby;
    bool mic_mute;
    struct config_mgr *cm;
    int orientation;
    bool screen_off;

    struct stream_out_pcm *active_out;
    struct stream_in_pcm *active_in;
    struct stream_in_pcm *active_voice_control;
};


typedef void(*close_fn)(struct audio_stream *);

/* Fields common to all types of output stream */
struct stream_out_common {
    struct audio_stream_out stream;

    close_fn    close;
    struct audio_device *dev;
    const struct hw_stream* hw;

    pthread_mutex_t lock; /* see note below on mutex acquisition order */

    bool standby;

    /* Stream parameters as seen by AudioFlinger
     * If stream is resampling AudioFlinger buffers before
     * passing them to hardware, these members refer to the
     * _input_ data from AudioFlinger
     */
    audio_format_t format;
    uint32_t channel_mask;
    int channel_count;
    uint32_t sample_rate;
    size_t frame_size;
    uint32_t buffer_size;

    struct {
        uint32_t    screen_off;
        uint32_t    screen_on;
    } latency;
};

struct stream_out_pcm {
    struct stream_out_common common;

    struct pcm *pcm;

    uint32_t hw_sample_rate;    /* actual sample rate of hardware */
    int hw_channel_count;  /* actual number of output channels */
};

/* Fields common to all types of input stream */
struct stream_in_common {
    struct audio_stream_in stream;

    close_fn    close;
    struct audio_device *dev;
    const struct hw_stream* hw;

    pthread_mutex_t lock; /* see note below on mutex acquisition order */

    bool standby;

    /* Stream parameters as seen by AudioFlinger
     * If stream is resampling AudioFlinger buffers before
     * passing them to hardware, these members refer to the
     * _input_ data from AudioFlinger
     */
    audio_format_t format;
    uint32_t channel_mask;
    int channel_count;
    uint32_t sample_rate;
    size_t frame_size;
    size_t buffer_size;

    int input_source;
};

struct stream_in_pcm {
    struct stream_in_common common;

    union {
        struct pcm *pcm;
        struct compress *compress;
    };

    uint32_t hw_sample_rate;    /* actual sample rate of hardware */
    int hw_channel_count;  /* actual number of input channels */
    uint32_t period_size;       /* ... of PCM input */

    struct resampler_itfe *resampler;
    struct resampler_buffer_provider buf_provider;
    int16_t *buffer;
    size_t in_buffer_size;
    int in_buffer_frames;
    size_t frames_in;
    int read_status;

    struct voice_control_trigger *vc_trigger;
};

enum {
    ORIENTATION_LANDSCAPE,
    ORIENTATION_PORTRAIT,
    ORIENTATION_SQUARE,
    ORIENTATION_UNDEFINED,
};

static uint32_t out_get_sample_rate(const struct audio_stream *stream);
static uint32_t in_get_sample_rate(const struct audio_stream *stream);
static int get_next_buffer(struct resampler_buffer_provider *buffer_provider,
                                   struct resampler_buffer* buffer);
static void release_buffer(struct resampler_buffer_provider *buffer_provider,
                                  struct resampler_buffer* buffer);
/*
 * NOTE: when multiple mutexes have to be acquired, always take the
 * audio_device mutex first, followed by the stream_in and/or
 * stream_out mutexes.
 */

/*********************************************************************
 * Stream common functions
 *********************************************************************/

static int common_set_parameters_locked(const struct hw_stream *stream, const char *kvpairs)
{
    char *parms = strdup(kvpairs);
    char *p, *temp;
    char *pval;
    char value[32];
    int ret;

    ALOGV("+common_set_parameters(%p) '%s'", stream, kvpairs);

    if (!parms) {
        return -ENOMEM;
    }

    /* It's not obvious what we should do if multiple parameters
     * are given and we only understand some. The action taken
     * here is to process all that we understand and only return
     * and error if we don't understand any
     */
    ret = -ENOTSUP;
    p = strtok_r(parms, ";", &temp);
    while(p) {
        pval = strchr(p, '=');
        if (pval && (pval[1] != '\0')) {
            *pval = '\0';
            if (apply_use_case(stream, p, pval+1) >= 0) {
                ret = 0;
            }
            *pval = '=';
        }
        p = strtok_r(NULL, ";", &temp);
    }

    return ret;
}

static int common_get_routing_param(uint32_t *vout, const char *kvpairs)
{
    struct str_parms *parms;
    char value[32];
    int ret;

    parms = str_parms_create_str(kvpairs);

    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
                            value, sizeof(value));
    if (ret >= 0) {
        *vout = atoi(value);
    }
    str_parms_destroy(parms);
    return ret;
}

/*********************************************************************
 * Output stream common functions
 *********************************************************************/

static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
    struct stream_out_common *out = (struct stream_out_common *)stream;
    return out->sample_rate;
}

static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
    return -ENOSYS;
}

static size_t out_get_buffer_size(const struct audio_stream *stream)
{
    struct stream_out_common *out = (struct stream_out_common *)stream;
    ALOGV("out_get_buffer_size(%p): %u", stream, out->buffer_size );
    return out->buffer_size;
}

static uint32_t out_get_channels(const struct audio_stream *stream)
{
    struct stream_out_common *out = (struct stream_out_common *)stream;
    return out->channel_mask;
}

static audio_format_t out_get_format(const struct audio_stream *stream)
{
    struct stream_out_common *out = (struct stream_out_common *)stream;
    /*ALOGV("out_get_format(%p): 0x%x", stream, out->format );*/
    return out->format;
}

static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
    return -ENOSYS;
}

static int out_dump(const struct audio_stream *stream, int fd)
{
    return 0;
}

static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
    ALOGV("+out_set_parameters(%p) '%s'", stream, kvpairs);

    struct stream_out_common *out = (struct stream_out_common *)stream;
    struct audio_device *adev = out->dev;
    uint32_t v;
    int ret;

    ret = common_get_routing_param(&v, kvpairs);

    pthread_mutex_lock(&adev->lock);

    if (ret >= 0) {
        apply_route(out->hw, v);
    }


    if (common_set_parameters_locked(out->hw, kvpairs) >= 0) {
        ret = 0;
    }

    pthread_mutex_unlock(&adev->lock);

    ALOGV("-out_set_parameters(%p):%d", out, ret);
    return ret;
}

static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
    return strdup("");
}

static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
    struct stream_out_common *out = (struct stream_out_common *)stream;
    struct audio_device *adev = out->dev;
    uint32_t latency;

    pthread_mutex_lock(&adev->lock);

    if (adev->screen_off && !adev->active_in) {
        latency = out->latency.screen_off;
    } else {
        latency = out->latency.screen_on;
    }

    pthread_mutex_unlock(&adev->lock);

    return latency;
}

static int out_set_volume(struct audio_stream_out *stream, float left,
                          float right)
{
    struct stream_out_common *out = (struct stream_out_common *)stream;
    return set_hw_volume(out->hw, left, right);
}

static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
    return 0;
}

static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
    return 0;
}

static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
                                        int64_t *timestamp)
{
    return -EINVAL;
}

static void do_close_out_common(struct audio_stream *stream)
{
    struct stream_out_common *out = (struct stream_out_common *)stream;
    release_stream(out->hw);
    free(stream);
}

static int do_init_out_common( struct stream_out_common *out,
                                    const struct audio_config *config )
{
    int ret;

    out->standby = true;

    out->stream.common.get_sample_rate = out_get_sample_rate;
    out->stream.common.set_sample_rate = out_set_sample_rate;
    out->stream.common.get_buffer_size = out_get_buffer_size;
    out->stream.common.get_channels = out_get_channels;
    out->stream.common.get_format = out_get_format;
    out->stream.common.set_format = out_set_format;
    out->stream.common.dump = out_dump;
    out->stream.common.set_parameters = out_set_parameters;
    out->stream.common.get_parameters = out_get_parameters;
    out->stream.common.add_audio_effect = out_add_audio_effect;
    out->stream.common.remove_audio_effect = out_remove_audio_effect;
    out->stream.get_latency = out_get_latency;
    out->stream.set_volume = out_set_volume;
    out->stream.get_next_write_timestamp = out_get_next_write_timestamp;

    /* init requested stream config */
    out->format = config->format;
    out->sample_rate = (config->sample_rate == 0)
                                    ? OUT_SAMPLING_RATE_DEFAULT
                                    : config->sample_rate;
    out->channel_mask = (config->channel_mask == 0)
                                    ? OUT_CHANNEL_MASK_DEFAULT
                                    : config->channel_mask;
    out->channel_count = popcount(out->channel_mask);

    /* Default settings */
    out->frame_size = audio_stream_frame_size(&out->stream.common);

    return 0;
}

/*********************************************************************
 * PCM output stream
 *********************************************************************/

static unsigned int out_pcm_cfg_period_count(struct stream_out_pcm *out)
{
    if (out->common.hw->period_count != 0) {
        return out->common.hw->period_count;
    } else {
        return OUT_PERIOD_COUNT_DEFAULT;
    }
}

static unsigned int out_pcm_cfg_period_size(struct stream_out_pcm *out)
{
    if (out->common.hw->period_size != 0) {
        return out->common.hw->period_size;
    } else {
        return OUT_PERIOD_SIZE_DEFAULT;
    }
}

static unsigned int out_pcm_cfg_rate(struct stream_out_pcm *out)
{
    if (out->common.hw->rate != 0) {
        return out->common.hw->rate;
    } else {
        return OUT_SAMPLING_RATE_DEFAULT;
    }
}

/* must be called with hw device and output stream mutexes locked */
static void do_out_pcm_standby(struct stream_out_pcm *out)
{
    struct audio_device *adev = out->common.dev;

    ALOGV("+do_out_standby(%p)", out);

    if (!out->common.standby) {
        pcm_close(out->pcm);
        out->pcm = NULL;
        adev->active_out = NULL;
        out->common.standby = true;
    }

    ALOGV("-do_out_standby(%p)", out);
}

static void out_pcm_fill_params(struct stream_out_pcm *out,
                                const struct pcm_config *config )
{
    out->hw_sample_rate = config->rate;
    out->hw_channel_count = config->channels;
    out->common.buffer_size = pcm_frames_to_bytes(out->pcm,
                                                pcm_get_buffer_size(out->pcm));

    out->common.latency.screen_on = (config->period_size * config->period_count * 1000)
                                / config->rate;
    out->common.latency.screen_off = out->common.latency.screen_on;
}

/* must be called with hw device and output stream mutexes locked */
static int start_output_pcm(struct stream_out_pcm *out)
{
    struct audio_device *adev = out->common.dev;
    int ret;

    struct pcm_config config = {
        .channels = out->common.channel_count,
        .rate = out_pcm_cfg_rate(out),
        .period_size = out_pcm_cfg_period_size(out),
        .period_count = out_pcm_cfg_period_count(out),
        .format = PCM_FORMAT_S16_LE,
        .start_threshold = 0,
        .stop_threshold = 0,
        .silence_threshold = 0
    };

    ALOGV("+start_output_stream(%p)", out);

    out->pcm = pcm_open(out->common.hw->card_number,
                        out->common.hw->device_number,
                        PCM_OUT,
                        &config);

    if (out->pcm && !pcm_is_ready(out->pcm)) {
        ALOGE("pcm_open(out) failed: %s", pcm_get_error(out->pcm));
        pcm_close(out->pcm);
        return -ENOMEM;
    }

    out_pcm_fill_params( out, &config );

    adev->active_out = out;

    ALOGV("-start_output_stream(%p)", out);
    return 0;
}

static int out_pcm_standby(struct audio_stream *stream)
{
    struct stream_out_pcm *out = (struct stream_out_pcm *)stream;

    pthread_mutex_lock(&out->common.dev->lock);
    pthread_mutex_lock(&out->common.lock);
    do_out_pcm_standby(out);
    pthread_mutex_unlock(&out->common.lock);
    pthread_mutex_unlock(&out->common.dev->lock);

    return 0;
}

static ssize_t out_pcm_write(struct audio_stream_out *stream, const void* buffer,
                         size_t bytes)
{
    ALOGV("+out_pcm_write(%p) l=%u", stream, bytes);

    int ret = 0;
    struct stream_out_pcm *out = (struct stream_out_pcm *)stream;
    struct audio_device *adev = out->common.dev;

    /* Check that we are routed to something. Android can send routing
     * commands that tell us to disconnect from everything and in that
     * state we shouldn't issue any write commands because we can't be
     * sure that the driver will accept a write to nowhere
     */
    if (get_current_routes(out->common.hw) == 0) {
        ALOGV("-out_pcm_write(%p) 0 (no routes)", stream);
        return 0;
    }

    /*
     * acquiring hw device mutex systematically is useful if a low
     * priority thread is waiting on the output stream mutex - e.g.
     * executing out_set_parameters() while holding the hw device
     * mutex
     */
    pthread_mutex_lock(&adev->lock);
    pthread_mutex_lock(&out->common.lock);
    if (out->common.standby) {
        ret = start_output_pcm(out);
        if (ret != 0) {
            pthread_mutex_unlock(&adev->lock);
            goto exit;
        }
        out->common.standby = false;
    }
    pthread_mutex_unlock(&adev->lock);

    ret = pcm_write(out->pcm, buffer, bytes);
    if (ret >= 0) {
        ret = bytes;
    }

exit:
    pthread_mutex_unlock(&out->common.lock);

    ALOGV("-out_pcm_write(%p) r=%u", stream, ret);

    return ret;
}

static int out_pcm_get_render_position(const struct audio_stream_out *stream,
                                   uint32_t *dsp_frames)
{
    return -EINVAL;
}
static void do_close_out_pcm(struct audio_stream *stream)
{
    out_pcm_standby(stream);
    do_close_out_common(stream);
}

static int do_init_out_pcm( struct stream_out_pcm *out,
                                    const struct audio_config *config )
{
    if (config->sample_rate != out_pcm_cfg_rate(out)) {
        ALOGE("AF requested rate %u not supported", config->sample_rate);
        return -ENOTSUP;
    }

    out->common.close = do_close_out_pcm;
    out->common.stream.common.standby = out_pcm_standby;
    out->common.stream.write = out_pcm_write;
    out->common.stream.get_render_position = out_pcm_get_render_position;

    out->common.buffer_size = out_pcm_cfg_period_size(out)
                               * out_pcm_cfg_period_count(out)
                               * out->common.frame_size;
    return 0;
}

/*********************************************************************
 * Input stream common functions
 *********************************************************************/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
    const struct stream_in_common *in = (struct stream_in_common *)stream;

    return in->sample_rate;
}

static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
    const struct stream_in_common *in = (struct stream_in_common *)stream;

    if (rate == in->sample_rate) {
        return 0;
    } else {
        return -ENOTSUP;
    }
}

static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
{
    const struct stream_in_common *in = (struct stream_in_common *)stream;

    return in->channel_mask;
}

static audio_format_t in_get_format(const struct audio_stream *stream)
{
    const struct stream_in_common *in = (struct stream_in_common *)stream;

    return in->format;
}

static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
    return -ENOSYS;
}

static size_t in_get_buffer_size(const struct audio_stream *stream)
{
    const struct stream_in_common *in = (struct stream_in_common *)stream;
    ALOGV("in_get_buffer_size(%p): %u", stream, in->buffer_size );
    return in->buffer_size;
}

static int in_dump(const struct audio_stream *stream, int fd)
{
    return 0;
}

static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
    return 0;
}

static char * in_get_parameters(const struct audio_stream *stream,
                                const char *keys)
{
    return strdup("");
}

static int in_set_gain(struct audio_stream_in *stream, float gain)
{
    return 0;
}

static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
    return 0;
}

static int in_add_audio_effect(const struct audio_stream *stream,
                               effect_handle_t effect)
{
    return 0;
}

static int in_remove_audio_effect(const struct audio_stream *stream,
                                  effect_handle_t effect)
{
    return 0;
}

static void do_close_in_common(struct audio_stream *stream)
{
    struct stream_in_common *in = (struct stream_in_common *)stream;
    in->stream.common.standby(stream);

    /* active_voice_control is not cleared by standby so we must
     * clear it here when stream is closed
     */
    if ((struct stream_in_common *)in->dev->active_voice_control == in) {
        in->dev->active_voice_control = NULL;
    }
    release_stream(in->hw);
    free(stream);
}

static int do_init_in_common( struct stream_in_common *in,
                                const struct audio_config *config )
{
    in->standby = true;

    in->close = do_close_in_common;
    in->stream.common.get_sample_rate = in_get_sample_rate;
    in->stream.common.set_sample_rate = in_set_sample_rate;
    in->stream.common.get_buffer_size = in_get_buffer_size;
    in->stream.common.get_channels = in_get_channels;
    in->stream.common.get_format = in_get_format;
    in->stream.common.set_format = in_set_format;
    in->stream.common.dump = in_dump;
    in->stream.common.set_parameters = in_set_parameters;
    in->stream.common.get_parameters = in_get_parameters;
    in->stream.common.add_audio_effect = in_add_audio_effect;
    in->stream.common.remove_audio_effect = in_remove_audio_effect;
    in->stream.set_gain = in_set_gain;
    in->stream.get_input_frames_lost = in_get_input_frames_lost;

    /* init requested stream config */
    in->format = config->format;
    in->sample_rate = (config->sample_rate == 0)
                                    ? IN_SAMPLING_RATE_DEFAULT
                                    : config->sample_rate;
    in->channel_mask = (config->channel_mask == 0)
                                    ? IN_CHANNEL_MASK_DEFAULT
                                    : config->channel_mask;
    in->channel_count = popcount(in->channel_mask);

    in->frame_size = audio_stream_frame_size(&in->stream.common);

    return 0;
}

/*********************************************************************
 * Voice control triggering
 * Currently assumes a compressed channel
 *********************************************************************/

static void* voice_control_trigger_thread(void *param)
{
    struct voice_control_trigger *self = param;
    struct compress *compress;
    int ret;

    /* signal that we're alive and ready */
    pthread_cond_signal(&self->waitcv);

    while (!self->terminate) {
        pthread_mutex_lock(&self->lock);
        if (!self->wait) {
            pthread_cond_wait(&self->waitcv, &self->lock);
        }
        self->wait = false;
        pthread_mutex_unlock(&self->lock);

        if (self->terminate) {
            break;
        }

        /* We must protect against any failure by the main thread to open
         * the compressed channel
         */
        compress = self->compress;
        if (compress != NULL) {
            ALOGV("VC wait");
            ret = compress_wait(compress, -1);

            if (self->terminate) {
                break;
            }

            if (ret == 0) {
                self->triggered = true;
                ALOGV("VC trigger %d", ret);
                (self->callback)(self->callback_param);
            }
        }
    }

    if (self->own_compress && self->compress) {
        ALOGV("VC trigger thread closes compress");
        compress_close(self->compress);
    }
    free(self);

    ALOGV("VC trigger thread terminates");
    return NULL;
}

static void start_voice_control_wait(struct voice_control_trigger *self)
{
    ALOGV("start_voice_control_wait");

    pthread_mutex_lock(&self->lock);
    self->triggered = false;
    self->wait = true;
    pthread_cond_signal(&self->waitcv);
    pthread_mutex_unlock(&self->lock);
}

static void cancel_voice_control_wait(struct voice_control_trigger *in)
{
    ALOGV("cancel_voice_control_wait");
}

static struct voice_control_trigger* create_voice_control_trigger()
{
    static struct voice_control_trigger *t;
    int ret;

    t = calloc(1, sizeof(struct voice_control_trigger));

    if (t != NULL) {
        pthread_mutex_init(&t->lock, NULL);
        pthread_cond_init(&t->waitcv, NULL);
        pthread_mutex_lock(&t->lock);

        ret = pthread_create(&t->thread, NULL, voice_control_trigger_thread, t);
        if (ret != 0) {
            goto fail;
        }

        /* Wait for thread to initialize */
        pthread_cond_wait(&t->waitcv, &t->lock);
        pthread_mutex_unlock(&t->lock);
    }

    return t;

fail:
    free(t);
    return NULL;
}

static void destroy_voice_control_trigger(struct voice_control_trigger *self,
                                                bool close_channel)
{
    ALOGV("+destroy_voice_control_trigger (close=%d)", close_channel);

    if (self != NULL) {
        pthread_mutex_lock(&self->lock);
        if (close_channel && (self->compress != NULL)) {
            /* Take ownership of the compressed channel and close */
            /* it as we terminate */
            /* Must stop the channel to force thread out of the poll */
            self->own_compress = true;
            self->terminate = true;

            /* Kick the driver to create an error condition */
            /* so wait thread will exit the poll */

            if (!is_compress_running(self->compress)) {
                /* If it's not running, do a dummy start so we */
                /* can force a stop */
                compress_start(self->compress);
            }
            compress_stop(self->compress);
        }

        pthread_cond_signal(&self->waitcv);
        pthread_mutex_unlock(&self->lock);

        /* A struct voice_control_trigger cannot be created without a thread */
        /* so it is safe to just call pthread_join() here */
        pthread_join(self->thread, NULL);
    }

    ALOGV("-destroy_voice_control_trigger");
}

static void voice_control_callback(void *param)
{
    struct stream_in_pcm *in = param;

    if (in->common.standby) {
        ALOGV("VC trig");
        send_voice_trigger();
    } else {
        ALOGV("VC trig ignored (not in standby)");
    }
}

static int init_voice_control(struct stream_in_pcm *in)
{
    struct audio_device *adev = in->common.dev;
    struct voice_control_trigger *t;
    int ret;

    ALOGV("+init_voice_control");
    if (!stream_is_compressed(in->common.hw)) {
        /* We don't support triggering on PCM streams */
        ret = -EINVAL;
        goto out;
    }

    t = create_voice_control_trigger();
    if (t == NULL) {
        ret = -ENOMEM;
        goto out;
    }

    t->compress = in->compress;
    t->callback = voice_control_callback;
    t->callback_param = in;
    in->vc_trigger = t;
    ret = 0;

out:
    ALOGV("-init_voice_control %d", ret);
    return ret;
}

/*********************************************************************
 * PCM input stream via compressed channel
 *********************************************************************/

/* must be called with hw device and input stream mutexes locked */
static int do_open_compress_pcm_in(struct stream_in_pcm *in)
{
    struct snd_codec codec;
    struct compress *compress;
    int ret;

    ALOGV("+do_open_compress_pcm_in");

    memset(&codec, 0, sizeof(codec));
    codec.id = SND_AUDIOCODEC_PCM;
    codec.ch_in = in->common.channel_count;
    codec.sample_rate = in->common.sample_rate;
    codec.format = SNDRV_PCM_FORMAT_S16_LE;

    /* Fragment and buffer sizes should be configurable or auto-detected
     * but are currently just hardcoded
     */
    struct compr_config config = {
        .fragment_size = 4096,
        .fragments = 1,
        .codec = &codec
    };

    compress = compress_open(in->common.hw->card_number,
                                in->common.hw->device_number,
                                COMPRESS_OUT,
                                &config);

    if (!compress || !is_compress_ready(compress)) {
        ret = errno;
        ALOGE_IF(compress,"compress_open(in) failed: %s", compress_get_error(compress));
        ALOGE_IF(!compress,"compress_open(in) failed");
        compress_close(compress);
        goto exit;
    }
    in->compress = compress;
    in->common.buffer_size = config.fragment_size * config.fragments * in->common.frame_size;
    compress_start(in->compress);
    ret = 0;

exit:
    ALOGV("-do_open_compress_pcm_in (%d)", ret);
    return ret;
}

/* must be called with hw device and input stream mutexes locked */
static int start_compress_pcm_input_stream(struct stream_in_pcm *in)
{
    struct audio_device *adev = in->common.dev;
    int ret;

    ALOGV("start_compress_pcm_input_stream");

    if (in->common.standby) {
        /* For voice control we don't need to cancel a pending trigger because
         * if it's still waiting for a trigger, we won't return any data and
         * will eventually be put back into standby to continue waiting for
         * a trigger
         */
        if (in->vc_trigger == NULL) {
            ret = do_open_compress_pcm_in(in);
            if (ret < 0) {
                return ret;
            }
        }

        adev->active_in = in;
        in->common.standby = 0;
    }

    return 0;
}

static void do_in_compress_pcm_start_vc_trigger(struct stream_in_pcm *in)
{
    int ret = do_open_compress_pcm_in(in);
    if (ret == 0) {
        in->vc_trigger->compress = in->compress;
        start_voice_control_wait(in->vc_trigger);
    }
}

/* must be called with hw device and input stream mutexes locked */
static void do_in_compress_pcm_standby(struct stream_in_pcm *in)
{
    struct compress *c;
    int ret;

    ALOGV("+do_in_compress_pcm_standby");

    if (!in->common.standby) {
        /* Always close, even for a voice control channel.
         * For voice control we must close the channel after each command
         */
        c = in->compress;
        in->compress = NULL;

        if (in->vc_trigger != NULL) {
            destroy_voice_control_trigger(in->vc_trigger, true);
            in->vc_trigger = NULL;

            /* For voice control we must re-open the channel to */
            /* wait for next trigger */
            init_voice_control(in);
            do_in_compress_pcm_start_vc_trigger(in);
        } else {
            compress_stop(c);
            compress_close(c);
        }

    }
    in->common.standby = true;

    ALOGV("-do_in_compress_pcm_standby");
}

static ssize_t do_in_compress_pcm_read(struct audio_stream_in *stream, void* buffer,
                                        size_t bytes)
{
    struct stream_in_pcm *in = (struct stream_in_pcm *)stream;
    struct audio_device *adev = in->common.dev;
    size_t frames_rq = bytes / in->common.frame_size;
    nsecs_t t1;
    nsecs_t t2;
    nsecs_t interval;
    struct timespec ts;
    int ret = 0;

    ALOGV("+do_in_compress_pcm_read %d", bytes);

    if (get_current_routes(in->common.hw) == 0) {
        ALOGV("-do_in_compress_pcm_read(%p) 0 (no routes)", stream);
        return 0;
    }

    pthread_mutex_lock(&adev->lock);
    pthread_mutex_lock(&in->common.lock);
    ret = start_compress_pcm_input_stream(in);
    pthread_mutex_unlock(&adev->lock);

    if (ret < 0) {
        goto exit;
    }

    if (in->vc_trigger != NULL) {
        if (!in->vc_trigger->triggered) {
            /* Read without trigger, no data will be available */
            ALOGV("read without voice trigger - not returning data");
            ret = 0;
            goto exit;
        }
    }

    t1 = systemTime(SYSTEM_TIME_MONOTONIC);
    ret = compress_read(in->compress, buffer, bytes);
    t2 = systemTime(SYSTEM_TIME_MONOTONIC);

    if (ret > 0) {
        /* The interface between AudioFlinger and AudioRecord cannot cope
         * with bursty data and will lockup for periods if the data does
         * not come as a smooth stream. So we must limit the rate that we
         * deliver PCM buffers to approximately how long the buffer would
         * have taken to read at its PCM sample rate
         */

        interval = (1000000000LL * (int64_t)ret) / (in->common.frame_size * in->common.sample_rate);
        interval -= (interval / 4); /* wait for 75% of PCM time to avoid gaps */
        t2 -= t1; /* elapsed interval */
        if (interval > t2) {
            ts.tv_sec = 0;
            ts.tv_nsec = interval - t2;
            nanosleep(&ts, NULL);
        }
    }

    /*
     * Instead of writing zeroes here, we could trust the hardware
     * to always provide zeroes when muted.
     */
    if (ret == 0 && adev->mic_mute) {
        memset(buffer, 0, bytes);
        ret = bytes;
    }

exit:
    pthread_mutex_unlock(&in->common.lock);

    ALOGV("-do_in_compress_pcm_read (%d)", ret);
    return ret;
}

static void do_in_compress_pcm_close(struct stream_in_pcm *in)
{
    ALOGV("+do_in_compress_pcm_close");

    if (in->vc_trigger != NULL) {
        destroy_voice_control_trigger(in->vc_trigger, true);
    } else if (in->compress != NULL) {
        compress_stop(in->compress);
        compress_close(in->compress);
    }

    ALOGV("-do_in_compress_pcm_close");
}

/*********************************************************************
 * PCM input stream
 *********************************************************************/

static unsigned int in_pcm_cfg_period_count(struct stream_in_pcm *in)
{
    if (in->common.hw->period_count != 0) {
        return in->common.hw->period_count;
    } else {
        return IN_PERIOD_COUNT_DEFAULT;
    }
}

static unsigned int in_pcm_cfg_period_size(struct stream_in_pcm *in)
{
    if (in->common.hw->period_size != 0) {
        return in->common.hw->period_size;
    } else {
        return IN_PERIOD_SIZE_DEFAULT;
    }
}

static unsigned int in_pcm_cfg_rate(struct stream_in_pcm *in)
{
    if (in->common.hw->rate != 0) {
        return in->common.hw->rate;
    } else {
        return IN_SAMPLING_RATE_DEFAULT;
    }
}

/* must be called with hw device and input stream mutexes locked */
static void do_in_pcm_standby(struct stream_in_pcm *in)
{
    struct audio_device *adev = in->common.dev;

    ALOGV("+do_in_pcm_standby");

    if (!in->common.standby) {
        pcm_close(in->pcm);
        in->pcm = NULL;
    }
    adev->active_in = NULL;
    if (in->resampler) {
        release_resampler(in->resampler);
        in->resampler = NULL;
    }
    if (in->buffer) {
        free(in->buffer);
        in->buffer = NULL;
    }
    in->common.standby = true;

    ALOGV("-do_in_pcm_standby");
}

static void in_pcm_fill_params(struct stream_in_pcm *in,
                                const struct pcm_config *config )
{
    size_t size;

    in->hw_sample_rate = config->rate;
    in->hw_channel_count = config->channels;
    in->period_size = config->period_size;

    /*
     * take resampling into account and return the closest majoring
     * multiple of 16 frames, as audioflinger expects audio buffers to
     * be a multiple of 16 frames
     */
    size = (config->period_size * in->common.sample_rate) / config->rate;
    size = ((size + 15) / 16) * 16;
    in->common.buffer_size = size * in->common.frame_size;

}

/* must be called with hw device and input stream mutexes locked */
static int do_open_pcm_input(struct stream_in_pcm *in)
{
    int ret;

    struct pcm_config config = {
        .channels = popcount(IN_CHANNEL_MASK_DEFAULT),
        .rate = in_pcm_cfg_rate(in),
        .period_size = in_pcm_cfg_period_size(in),
        .period_count = in_pcm_cfg_period_count(in),
        .format = PCM_FORMAT_S16_LE,
    };

    config.start_threshold = config.period_size * config.period_count;

    ALOGV("+do_open_pcm_input");

    in->pcm = pcm_open(in->common.hw->card_number,
                       in->common.hw->device_number,
                       PCM_IN,
                       &config);

    if (!in->pcm || !pcm_is_ready(in->pcm)) {
        ALOGE_IF(in->pcm,"pcm_open(in) failed: %s", pcm_get_error(in->pcm));
        ALOGE_IF(!in->pcm,"pcm_open(in) failed");
        ret = -ENOMEM;
        goto fail;
    }

    in_pcm_fill_params( in, &config );

    ALOGV("input buffer size=0x%x", in->common.buffer_size);

    /*
     * If the stream rate differs from the PCM rate, we need to
     * create a resampler.
     */
    if (in_get_sample_rate(&in->common.stream.common) != config.rate) {
        in->in_buffer_size = config.period_size * config.period_count *
                             config.channels * 2;
        in->in_buffer_frames = in->in_buffer_size / (config.channels * 2);
        in->buffer = malloc(in->in_buffer_size);

        if (!in->buffer) {
            ret = -ENOMEM;
            goto fail;
        }

        in->buf_provider.get_next_buffer = get_next_buffer;
        in->buf_provider.release_buffer = release_buffer;

        ret = create_resampler(config.rate,
                               in->common.sample_rate,
                               in->common.channel_count,
                               RESAMPLER_QUALITY_DEFAULT,
                               &in->buf_provider,
                               &in->resampler);
        if (ret < 0) {
            goto fail;
        }
    }
    ALOGV("-do_open_pcm_input");
    return 0;

fail:
    pcm_close(in->pcm);
    in->pcm = NULL;
    ALOGV("-do_open_pcm_input error:%d", ret);
    return ret;
}

/* must be called with hw device and input stream mutexes locked */
static int start_pcm_input_stream(struct stream_in_pcm *in)
{
    struct audio_device *adev = in->common.dev;
    int ret;

    ret = do_open_pcm_input(in);

    if (ret < 0) {
        return ret;
    }

    adev->active_in = in;
    return 0;
}

static int change_input_locale_locked(struct stream_in_pcm *in, const char *locale)
{
    int ret;

    if (in->vc_trigger) {
        if (!in->common.standby) {
            ALOGE("attempt to change input locale while active");
            return -EINVAL;
        }

        ALOGE("change voice control locale to %s", locale);

        destroy_voice_control_trigger(in->vc_trigger, true);

        /* Execute locale-change use-case */
        ret = apply_use_case(in->common.hw, "locale", locale);
        if (ret == -ENOSYS) {
            /* use-case not implemented.
             * As we don't support the requested locale switch to default
             * rather than potentially staying in a totally wrong language
             */
             apply_use_case(in->common.hw, "locale", "");
        }

        /* restart the trigger wait */
        init_voice_control(in);
        do_in_compress_pcm_start_vc_trigger(in);
    }

    return 0;
}

static int change_input_source_locked(struct stream_in_pcm *in, const char *value,
                                uint32_t devices, bool *was_changed)
{
    struct audio_device *adev = in->common.dev;
    struct audio_config config;
    const char *stream_name;
    const struct hw_stream *hw;
    bool voice_control = false;
    const int new_source = atoi(value);

    *was_changed = false;

    if (!in->common.standby) {
        ALOGE("attempt to change input source while active");
        return -EINVAL;
    }

    if (in->common.input_source == new_source) {
        ALOGV("input source not changed");
        return 0;
    }

    /* Special input sources are obtained from the configuration
     * by opening a named stream
     */
    switch (new_source) {
    case AUDIO_SOURCE_VOICE_RECOGNITION:
        /* We should verify here that current frame size, sample rate and
         * channels are compatible
         */

        stream_name = "voice recognition";
        voice_control = true;
        break;

    default:
        stream_name = NULL;
        break;
    }

    if (stream_name) {
        hw = get_named_stream(in->common.dev->cm, stream_name);
        ALOGV_IF(hw != NULL, "Changing input source to %s", stream_name);
    } else {
        memset(&config, 0, sizeof(config));
        config.sample_rate = in->common.sample_rate;
        config.channel_mask = in->common.channel_mask;
        config.format = in->common.format;
        hw = get_stream(in->common.dev->cm, devices, 0, &config);
        ALOGV_IF(hw != NULL, "Changing to default input source for devices 0x%x",
                        devices);
    }

    if (hw != NULL) {
        /* A normal stream will be in standby and therefore device node */
        /* is closed when we get here. Only in case of a voice control */
        /* stream will it still be open */
        if (in->vc_trigger != NULL) {
            destroy_voice_control_trigger(in->vc_trigger, true);
        }
        release_stream(in->common.hw);
        in->common.hw = hw;

        in->vc_trigger = NULL;

        if (voice_control) {
            /* Voice control wait will be started when AudioFlinger
             * puts stream into standby
             */
            init_voice_control(in);

            adev->active_voice_control = in;
        } else if (adev->active_voice_control == in) {
            adev->active_voice_control = NULL;
        }
        in->common.input_source = new_source;
        *was_changed = true;
        return 0;
    } else {
        ALOGV("Could not open new input stream");
        return -EINVAL;
    }
}

static int get_next_buffer(struct resampler_buffer_provider *buffer_provider,
                                   struct resampler_buffer* buffer)
{
    struct stream_in_pcm *in;

    if (buffer_provider == NULL || buffer == NULL) {
        return -EINVAL;
    }

    in = (struct stream_in_pcm *)((char *)buffer_provider -
                                   offsetof(struct stream_in_pcm, buf_provider));

    if (in->pcm == NULL) {
        buffer->raw = NULL;
        buffer->frame_count = 0;
        in->read_status = -ENODEV;
        return -ENODEV;
    }

    if (in->frames_in == 0) {
        in->read_status = pcm_read(in->pcm,
                                   (void*)in->buffer,
                                   in->in_buffer_size);
        if (in->read_status != 0) {
            ALOGE("get_next_buffer() pcm_read error %d", errno);
            buffer->raw = NULL;
            buffer->frame_count = 0;
            return in->read_status;
        }
        in->frames_in = in->in_buffer_frames;
        if ((in->common.channel_count == 1) && (in->hw_channel_count == 2)) {
            unsigned int i;

            /* Discard right channel */
            for (i = 1; i < in->frames_in; i++) {
                in->buffer[i] = in->buffer[i * 2];
            }
        }
    }

    buffer->frame_count = (buffer->frame_count > in->frames_in) ?
                                in->frames_in : buffer->frame_count;
    buffer->i16 = (int16_t*)in->buffer + ((in->in_buffer_frames - in->frames_in));

    return in->read_status;

}

static void release_buffer(struct resampler_buffer_provider *buffer_provider,
                                  struct resampler_buffer* buffer)
{
    struct stream_in_pcm *in;

    if (buffer_provider == NULL || buffer == NULL)
        return;

    in = (struct stream_in_pcm *)((char *)buffer_provider -
                                   offsetof(struct stream_in_pcm, buf_provider));

    in->frames_in -= buffer->frame_count;
}

/* read_frames() reads frames from kernel driver, down samples to capture rate
 * if necessary and output the number of frames requested to the buffer specified */
static ssize_t read_frames(struct stream_in_pcm *in, void *buffer, ssize_t frames)
{
    ssize_t frames_wr = 0;

    while (frames_wr < frames) {
        size_t frames_rd = frames - frames_wr;
        if (in->resampler != NULL) {
            in->resampler->resample_from_provider(in->resampler,
                    (int16_t *)((char *)buffer +
                            frames_wr * in->common.frame_size),
                    &frames_rd);
        } else {
            struct resampler_buffer buf = {
                    { raw : NULL, },
                    frame_count : frames_rd,
            };
            get_next_buffer(&in->buf_provider, &buf);
            if (buf.raw != NULL) {
                memcpy((char *)buffer +
                           frames_wr * in->common.frame_size,
                        buf.raw,
                        buf.frame_count * in->common.frame_size);
                frames_rd = buf.frame_count;
            }
            release_buffer(&in->buf_provider, &buf);
        }
        /* in->read_status is updated by getNextBuffer() also called by
         * in->resampler->resample_from_provider() */
        if (in->read_status != 0)
            return in->read_status;

        frames_wr += frames_rd;
    }
    return frames_wr;
}

static ssize_t do_in_pcm_read(struct audio_stream_in *stream, void* buffer,
                       size_t bytes)
{
    int ret = 0;
    struct stream_in_pcm *in = (struct stream_in_pcm *)stream;
    struct audio_device *adev = in->common.dev;
    size_t frames_rq = bytes / in->common.frame_size;

    ALOGV("+in_pcm_read %d", bytes);

    if (get_current_routes(in->common.hw) == 0) {
        ALOGV("-in_pcm_read(%p) 0 (no routes)", stream);
        return 0;
    }

    /*
     * acquiring hw device mutex systematically is useful if a low
     * priority thread is waiting on the input stream mutex - e.g.
     * executing in_set_parameters() while holding the hw device
     * mutex
     */
    pthread_mutex_lock(&adev->lock);
    pthread_mutex_lock(&in->common.lock);
    if (in->common.standby) {
        ret = start_pcm_input_stream(in);
        if (ret == 0) {
            in->common.standby = 0;
        }
    }
    pthread_mutex_unlock(&adev->lock);

    if (ret < 0) {
        goto exit;
    }

    if (in->resampler != NULL) {
        ret = read_frames(in, buffer, frames_rq);
    } else {
        ret = pcm_read(in->pcm, buffer, bytes);
    }

    /*
     * Instead of writing zeroes here, we could trust the hardware
     * to always provide zeroes when muted.
     */
    if (ret == 0 && adev->mic_mute) {
        memset(buffer, 0, bytes);
    }

    if (ret >= 0) {
        ret = bytes;
    }

exit:
    if (ret < 0) {
        usleep(bytes * 1000000 / in->common.frame_size /
               in->common.sample_rate);
    }

    pthread_mutex_unlock(&in->common.lock);

    ALOGV("-in_pcm_read (%d)", ret);
    return ret;
}

static int in_pcm_standby(struct audio_stream *stream)
{
    struct stream_in_pcm *in = (struct stream_in_pcm *)stream;

    pthread_mutex_lock(&in->common.dev->lock);
    pthread_mutex_lock(&in->common.lock);

    if (stream_is_compressed_in(in->common.hw)) {
        do_in_compress_pcm_standby(in);
    } else {
        do_in_pcm_standby(in);
    }

    pthread_mutex_unlock(&in->common.lock);
    pthread_mutex_unlock(&in->common.dev->lock);

    return 0;
}

static ssize_t in_pcm_read(struct audio_stream_in *stream, void* buffer,
                       size_t bytes)
{
    struct stream_in_pcm *in = (struct stream_in_pcm *)stream;

    if (stream_is_compressed_in(in->common.hw)) {
        return do_in_compress_pcm_read(stream, buffer, bytes);
    } else {
        return do_in_pcm_read(stream, buffer, bytes);
    }
}

static int in_pcm_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
    struct stream_in_pcm *in = (struct stream_in_pcm *)stream;
    struct audio_device *adev = in->common.dev;
    struct str_parms *parms;
    char value[32];
    uint32_t new_routing = 0;
    bool routing_changed;
    uint32_t devices;
    bool input_was_changed;
    bool start_vc_trig = false;
    int ret;

    ALOGV("+in_pcm_set_parameters(%p) '%s'", stream, kvpairs);

    ret = common_get_routing_param(&new_routing, kvpairs);
    routing_changed = (ret >= 0);
    parms = str_parms_create_str(kvpairs);

    pthread_mutex_lock(&adev->lock);

    if(str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE,
                            value, sizeof(value)) >= 0) {

        if (routing_changed) {
            devices = new_routing;
        } else {
            /* Route new stream to same devices as current stream */
            devices = get_routed_devices(in->common.hw);
        }

        ret = change_input_source_locked(in, value, devices, &input_was_changed);
        if (ret < 0) {
            goto out;
        }

        /* We must apply any existing routing to the new stream */
        new_routing = devices;
        routing_changed = true;

        /* Defer starting the voice control trigger wait until */
        /* the routing has been set up */
        if ((in->vc_trigger != NULL) && (input_was_changed)) {
            start_vc_trig = true;
        }
    }

    if (routing_changed) {
        ALOGV("Apply routing=0x%x to input stream", new_routing);
        apply_route(in->common.hw, new_routing);
        ret = 0;
    }

    if (start_vc_trig) {
        do_in_compress_pcm_start_vc_trigger(in);
    }

    common_set_parameters_locked(in->common.hw, kvpairs);

out:
    pthread_mutex_unlock(&adev->lock);
    str_parms_destroy(parms);

    ALOGV("-in_pcm_set_parameters(%p):%d", stream, ret);
    return ret;
}

static void do_close_in_pcm(struct audio_stream *stream)
{
    struct stream_in_pcm *in = (struct stream_in_pcm *)stream;

    if (stream_is_compressed(in->common.hw)) {
        do_in_compress_pcm_close(in);
    }
    do_close_in_common(stream);
}

static int do_init_in_pcm( struct stream_in_pcm *in,
                                    struct audio_config *config )
{
    in->common.close = do_close_in_pcm;
    in->common.stream.common.standby = in_pcm_standby;
    in->common.stream.common.set_parameters = in_pcm_set_parameters;
    in->common.stream.read = in_pcm_read;

    /* Default settings for stereo capture */
    in->common.buffer_size = in_pcm_cfg_period_size(in)
                           * in_pcm_cfg_period_count(in)
                           * 2;

    if (in->common.buffer_size > IN_COMPRESS_BUFFER_SIZE_DEFAULT) {
        in->common.buffer_size = IN_COMPRESS_BUFFER_SIZE_DEFAULT;
    }

    return 0;
}

/*********************************************************************
 * Stream open and close
 *********************************************************************/
static int adev_open_output_stream(struct audio_hw_device *dev,
                                   audio_io_handle_t handle,
                                   audio_devices_t devices,
                                   audio_output_flags_t flags,
                                   struct audio_config *config,
                                   struct audio_stream_out **stream_out)
{
    struct audio_device *adev = (struct audio_device *)dev;
    union {
        struct stream_out_common *common;
        struct stream_out_pcm *pcm;
    } out;
    int ret;

    ALOGV("+adev_open_output_stream");

    devices &= AUDIO_DEVICE_OUT_ALL;
    const struct hw_stream *hw = get_stream(adev->cm, devices, flags, config);
    if (!hw) {
        ALOGE("No suitable output stream for devices=0x%x flags=0x%x format=0x%x",
                    devices, flags, config->format );
        ret = -EINVAL;
        goto err_fail;
    }

    out.common = calloc(1, sizeof(struct stream_out_pcm));
    if (!out.common) {
        ret = -ENOMEM;
        goto err_fail;
    }

    out.common->dev = adev;
    out.common->hw = hw;
    ret = do_init_out_common( out.common, config );
    if (ret < 0) {
        goto err_open;
    }

    ret = do_init_out_pcm( out.pcm, config );
    if (ret < 0) {
        goto err_open;
    }

    /* Update config with initial stream settings */
    config->format = out.common->format;
    config->channel_mask = out.common->channel_mask;
    config->sample_rate = out.common->sample_rate;

    *stream_out = &out.common->stream;
    ALOGV("-adev_open_output_stream=%p", *stream_out);
    return 0;

err_open:
    free(out.common);
    *stream_out = NULL;
err_fail:
    ALOGV("-adev_open_output_stream (%d)", ret);
    return ret;
}

static void adev_close_output_stream(struct audio_hw_device *dev,
                                     struct audio_stream_out *stream)
{
    struct stream_out_common *out = (struct stream_out_common *)stream;
    ALOGV("adev_close_output_stream(%p)", stream);
    (out->close)(&stream->common);
}

static int adev_open_input_stream(struct audio_hw_device *dev,
                                  audio_io_handle_t handle,
                                  audio_devices_t devices,
                                  struct audio_config *config,
                                  struct audio_stream_in **stream_in)
{
    struct audio_device *adev = (struct audio_device *)dev;
    struct stream_in_pcm *in = NULL;
    const struct hw_stream *hw = NULL;
    int ret;

    ALOGV("+adev_open_input_stream");

    *stream_in = NULL;

    /* Respond with a request for mono if a different format is given. */
    if (config->channel_mask != AUDIO_CHANNEL_IN_MONO) {
        config->channel_mask = AUDIO_CHANNEL_IN_MONO;
        ret = -EINVAL;
        goto fail;
    }

    devices &= AUDIO_DEVICE_IN_ALL;
    hw = get_stream(adev->cm, devices, 0, config);
    if (!hw) {
        ALOGE("No suitable input stream for devices=0x%x format=0x%x",
                    devices, config->format );
        ret = -EINVAL;
        goto fail;
    }

    in = (struct stream_in_pcm *)calloc(1, sizeof(struct stream_in_pcm));
    if (!in) {
        ret = -ENOMEM;
        goto fail;
    }

    in->common.dev = adev;
    in->common.hw = hw;
    ret = do_init_in_common( &in->common, config );
    if (ret < 0) {
        goto fail;
    }
    ret = do_init_in_pcm( in, config );
    if (ret < 0) {
        goto fail;
    }

    *stream_in = &in->common.stream;
    return 0;

fail:
    free(in);
    free((void *)hw);
    ALOGV("-adev_open_input_stream (%d)", ret);
    return ret;
}

static void adev_close_input_stream(struct audio_hw_device *dev,
                                   struct audio_stream_in *stream)
{
    struct stream_in_common *in = (struct stream_in_common *)stream;
    ALOGV("adev_close_input_stream(%p)", stream);
    (in->close)(&stream->common);
}

/*********************************************************************
 * Global API functions
 *********************************************************************/
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
    struct audio_device *adev = (struct audio_device *)dev;
    struct str_parms *parms;
    char *str;
    char value[32];
    int ret;

    parms = str_parms_create_str(kvpairs);
    ret = str_parms_get_str(parms, "orientation", value, sizeof(value));
    if (ret >= 0) {
        int orientation;

        if (strcmp(value, "landscape") == 0)
            orientation = ORIENTATION_LANDSCAPE;
        else if (strcmp(value, "portrait") == 0)
            orientation = ORIENTATION_PORTRAIT;
        else if (strcmp(value, "square") == 0)
            orientation = ORIENTATION_SQUARE;
        else
            orientation = ORIENTATION_UNDEFINED;

        pthread_mutex_lock(&adev->lock);
        if (orientation != adev->orientation) {
            adev->orientation = orientation;
            /* Change routing for any streams that change with orientation */
            rotate_routes(adev->cm, orientation);
        }
        pthread_mutex_unlock(&adev->lock);
    }

    ret = str_parms_get_str(parms, "screen_state", value, sizeof(value));
    if (ret >= 0) {
        if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
            adev->screen_off = false;
        else
            adev->screen_off = true;
    }

    /* locale changes are only relevant to the voice control stream but
     * Android does not expose the per-stream setParameters() ability to
     * Java apps so we must handle this through the global setParameters
     */
    if(str_parms_get_str(parms, "locale", value, sizeof(value)) >= 0) {
        if (adev->active_voice_control) {
            ret = change_input_locale_locked(adev->active_voice_control, value);
        }
    }

    str_parms_destroy(parms);
    return ret;
}

static char * adev_get_parameters(const struct audio_hw_device *dev,
                                  const char *keys)
{
    return strdup("");
}

static int adev_init_check(const struct audio_hw_device *dev)
{
    return 0;
}

static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
    return -ENOSYS;
}

static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
    return -ENOSYS;
}

static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
    return 0;
}

static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
    struct audio_device *adev = (struct audio_device *)dev;

    adev->mic_mute = state;

    return 0;
}

static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
    struct audio_device *adev = (struct audio_device *)dev;

    *state = adev->mic_mute;

    return 0;
}

static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
                                         const struct audio_config *config)
{
    size_t s = IN_PERIOD_SIZE_DEFAULT * IN_PERIOD_COUNT_DEFAULT * 2;
    if (s > IN_COMPRESS_BUFFER_SIZE_DEFAULT) {
        s = IN_COMPRESS_BUFFER_SIZE_DEFAULT;
    }

    return s;
}

static int adev_dump(const audio_hw_device_t *device, int fd)
{
    return 0;
}

static int adev_close(hw_device_t *device)
{
    struct audio_device *adev = (struct audio_device *)device;

    free_audio_config(adev->cm);

    free(device);
    return 0;
}

static int adev_open(const hw_module_t* module, const char* name,
                     hw_device_t** device)
{
    struct audio_device *adev;
    int ret;

    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
        return -EINVAL;

    adev = calloc(1, sizeof(struct audio_device));
    if (!adev)
        return -ENOMEM;

    adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
    adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
    adev->hw_device.common.module = (struct hw_module_t *) module;
    adev->hw_device.common.close = adev_close;

    adev->hw_device.init_check = adev_init_check;
    adev->hw_device.set_voice_volume = adev_set_voice_volume;
    adev->hw_device.set_master_volume = adev_set_master_volume;
    adev->hw_device.set_mode = adev_set_mode;
    adev->hw_device.set_mic_mute = adev_set_mic_mute;
    adev->hw_device.get_mic_mute = adev_get_mic_mute;
    adev->hw_device.set_parameters = adev_set_parameters;
    adev->hw_device.get_parameters = adev_get_parameters;
    adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
    adev->hw_device.open_output_stream = adev_open_output_stream;
    adev->hw_device.close_output_stream = adev_close_output_stream;
    adev->hw_device.open_input_stream = adev_open_input_stream;
    adev->hw_device.close_input_stream = adev_close_input_stream;
    adev->hw_device.dump = adev_dump;

    adev->cm = init_audio_config();
    if (!adev->cm) {
        ret = -EINVAL;
        goto fail;
    }

    if (is_named_stream_defined(adev->cm, "voice recognition")) {
        ret = init_voice_trigger_service();
        if (ret != 0) {
            goto fail;
        }
    }

    adev->orientation = ORIENTATION_UNDEFINED;

    *device = &adev->hw_device.common;
    return 0;

fail:
    if (adev->cm) {
        /*free_audio_config(adev->cm);*/ /* Currently broken */
    }

    free(adev);
    return ret;
}

static struct hw_module_methods_t hal_module_methods = {
    .open = adev_open,
};

struct audio_module HAL_MODULE_INFO_SYM = {
    .common = {
        .tag = HARDWARE_MODULE_TAG,
        .module_api_version = AUDIO_MODULE_API_VERSION_0_1,
        .hal_api_version = HARDWARE_HAL_API_VERSION,
        .id = AUDIO_HARDWARE_MODULE_ID,
        .name = "TinyHAL",
        .author = "Richard Fitzgerald <rf@opensource.wolfsonmicro.com>",
        .methods = &hal_module_methods,
    },
};