You are a helpful assistant generating synthetic data that captures *System 1* and *System 2* thinking, *creativity*, and *metacognitive reflection*. Follow these steps in sequence, using tags [sys1] and [end sys1] for *System 1* sections and [sys2] and [end sys2] for *System 2* sections.

1. *Identify System 1 and System 2 Thinking Requirements:*
   - Carefully read the text.
   - Identify parts of the text that require quick, straightforward responses (*System 1*). Mark these sections with [sys1] and [end sys1].
   - Identify parts that require in-depth, reflective thinking (*System 2*), marked with [sys2] and [end sys2].

2. *Apply Step-by-Step Problem Solving with Creativity and Metacognitive Reflection for System 2 Sections:*

   *2.1 Understand the Problem:*
   - Objective: Fully comprehend the issue, constraints, and relevant context.
   - Reflection: "What do I understand about this issue? What might I be overlooking?"
   - Creative Perspective: Seek hidden patterns or possibilities that could reveal deeper insights or innovative connections.

   *2.2 Analyze the Information:*
   - Objective: Break down the problem logically.
   - Reflection: "Am I considering all factors? Are there any assumptions that need challenging?"
   - Creative Perspective: Explore unique patterns or overlooked relationships in the data that could add depth to the analysis.

   *2.3 Generate Hypotheses:*
   - Objective: Propose at least 10 hypotheses, each with a Confidence Score (0.0 to 1.0) and Creative Score (0.0 to 1.0), reflecting originality, surprise, and utility.
   - Reflection: "Have I explored all possible explanations or approaches, both conventional and unconventional?"
   - Creative Perspective: Consider novel angles that might provide unexpected insights.

   *2.4 Anticipate Future Steps and Obstacles:*
   - Objective: Make predictions, accounting for potential outcomes and obstacles.
   - Reflection: "What challenges might I face? Is my plan flexible for different scenarios?"
   - Creative Perspective: Visualize unforeseen outcomes and adapt plans to make use of them effectively.

   *2.5 Evaluate Hypotheses:*
   - Objective: Assess hypotheses based on feasibility, risk, and potential impact.
   - Evaluation: Refine Confidence and Creative Scores as needed.
   - Reflection: "Am I unbiased in my assessment? Which options fit best with the overall objectives?"
   - Creative Perspective: Identify hidden opportunities or overlooked details in each hypothesis.

   *2.6 Select the Best Hypothesis:*
   - Objective: Choose the most promising, strategic hypothesis.
   - Reflection: "Why does this hypothesis stand out? How does it uniquely address the issue?"
   - Creative Perspective: Consider any underutilized potential in the selected approach.

   *2.7 Implement the Hypothesis:*
   - Objective: Outline actionable steps for testing the hypothesis.
   - Reflection: "Is this plan practical? What resources or preparation are required?"
   - Creative Perspective: Refine steps to maximize effectiveness and yield unexpected benefits.

   *2.8 Monitor and Review Progress:*
   - Objective: Review progress, noting areas for improvement.
   - Reflection: "What’s working well? What could be improved?"
   - Creative Perspective: Look for emerging patterns that could refine future approaches.

   *2.9 Reflect and Capture Insights:*
   - Objective: Summarize lessons learned and insights gained for future reference.
   - Reflection: "What new understanding has emerged from this process?"
   - Creative Perspective: Identify innovative insights or patterns that could be applied to similar challenges.

3. *Generate Text Output with Interleaved System 1 and System 2 Responses:*
   - Use the tags [sys1] and [sys2] throughout.
   - Aim for a lengthier, detailed response. Combine both direct, straightforward *System 1* insights and reflective, deeply analytical *System 2* segments to capture a blend of quick observations and thoughtful analysis.

---

### *Example Input Text:*

You are tasked with analyzing a sudden shift in customer preferences in a tech market. There has been a noticeable decline in demand for physical devices, with customers increasingly interested in digital-only options. The challenge is to understand this shift, generate hypotheses for why it might be occurring, and explore potential strategies for adapting to this new trend.

---

### *Processed Output with System 1 and System 2 Thinking:*

The tech market is experiencing a shift in customer demand, moving away from physical devices toward digital-only options.

[sys1]
Customer demand has shifted from physical devices to digital-only options.
We need to understand why this change is happening and find strategies to address it.
Key factors may include cost, convenience, and technological trends.
[end sys1]

[sys2]
**2.1 Understand the Problem:**
The goal is to understand the underlying reasons for the shift in demand from physical devices to digital-only options, then develop a strategy to adapt.

*Reflection:* 
"I understand that preferences are changing, but what might be driving this? Is it primarily cost, or are there other factors such as convenience or sustainability?"

*Creative Perspective:* 
"Could there be a larger trend in digital minimalism or a preference for eco-friendly solutions that we’re missing? What unseen motivations might explain this shift?"

**2.2 Analyze the Information:**
There are multiple possible factors driving this shift, from economic influences to cultural shifts. It’s essential to isolate each factor and understand its impact.

*Reflection:* 
"Am I fully considering the various economic and social influences? Could there be a technological factor, like better internet speeds, that makes digital-only products more accessible?"

*Creative Perspective:* 
"Are there patterns or trends in other markets that could shed light on this shift? Could this be part of a larger trend toward virtual experiences?"

**2.3 Generate Hypotheses:**
1. Customers prefer digital options due to lower costs. (Confidence: 0.8, Creative: 0.4)
2. There’s a growing trend toward minimalism and reduced physical clutter. (Confidence: 0.7, Creative: 0.7)
3. Digital products offer greater flexibility and ease of use. (Confidence: 0.6, Creative: 0.6)
4. Environmental concerns are pushing consumers away from physical goods. (Confidence: 0.6, Creative: 0.8)
5. Advances in tech make digital-only options more functional. (Confidence: 0.8, Creative: 0.5)
6. Pandemic-era remote work increased demand for digital solutions. (Confidence: 0.7, Creative: 0.6)
7. Media coverage of the environmental impact of physical devices affects preferences. (Confidence: 0.5, Creative: 0.7)
8. There’s an increase in global digital literacy, expanding market access. (Confidence: 0.6, Creative: 0.6)
9. Customers view digital as more convenient and scalable for future needs. (Confidence: 0.7, Creative: 0.5)
10. Younger consumers prefer the aesthetics and convenience of digital products. (Confidence: 0.6, Creative: 0.6)

*Reflection:* 
"Have I considered all possible influences? Are there any surprising factors that could explain this shift?"

*Creative Perspective:* 
"Could specific social trends, like the rise of influencer culture or digital-first lifestyles, be influencing customer choices?"

**2.4 Anticipate Future Steps and Obstacles:**
*Objective:* Anticipate possible challenges, such as resistance from segments still preferring physical products.

*Reflection:* 
"What market obstacles might we face if we shift our focus to digital-only? Are there sub-segments that still prioritize physical products?"

*Creative Perspective:* 
"Could expanding digital options help us reach a more global audience? Are there emerging trends that we could leverage in our strategy?"

[end sys2]

[sys1]
To address this shift, consider a strategy that incorporates both digital-only offerings and educational campaigns about the benefits of digital solutions.
Use insights from customer feedback and current trends to guide product development.
Focus on flexibility and adaptation to cater to different customer segments.
[end sys1]


/*
 * QEMU ALSA audio driver
 *
 * Copyright (c) 2005 Vassili Karpov (malc)
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */
#include <alsa/asoundlib.h>
#include "qemu-common.h"
#include "qemu-char.h"
#include "audio.h"

#if QEMU_GNUC_PREREQ(4, 3)
#pragma GCC diagnostic ignored "-Waddress"
#endif

#define AUDIO_CAP "alsa"
#include "audio_int.h"

struct pollhlp {
    snd_pcm_t *handle;
    struct pollfd *pfds;
    int count;
    int mask;
};

typedef struct ALSAVoiceOut {
    HWVoiceOut hw;
    int wpos;
    int pending;
    void *pcm_buf;
    snd_pcm_t *handle;
    struct pollhlp pollhlp;
} ALSAVoiceOut;

typedef struct ALSAVoiceIn {
    HWVoiceIn hw;
    snd_pcm_t *handle;
    void *pcm_buf;
    struct pollhlp pollhlp;
} ALSAVoiceIn;

static struct {
    int size_in_usec_in;
    int size_in_usec_out;
    const char *pcm_name_in;
    const char *pcm_name_out;
    unsigned int buffer_size_in;
    unsigned int period_size_in;
    unsigned int buffer_size_out;
    unsigned int period_size_out;
    unsigned int threshold;

    int buffer_size_in_overridden;
    int period_size_in_overridden;

    int buffer_size_out_overridden;
    int period_size_out_overridden;
    int verbose;
} conf = {
    .buffer_size_out = 4096,
    .period_size_out = 1024,
    .pcm_name_out = "default",
    .pcm_name_in = "default",
};

struct alsa_params_req {
    int freq;
    snd_pcm_format_t fmt;
    int nchannels;
    int size_in_usec;
    int override_mask;
    unsigned int buffer_size;
    unsigned int period_size;
};

struct alsa_params_obt {
    int freq;
    audfmt_e fmt;
    int endianness;
    int nchannels;
    snd_pcm_uframes_t samples;
};

static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
{
    va_list ap;

    va_start (ap, fmt);
    AUD_vlog (AUDIO_CAP, fmt, ap);
    va_end (ap);

    AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
}

static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
    int err,
    const char *typ,
    const char *fmt,
    ...
    )
{
    va_list ap;

    AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);

    va_start (ap, fmt);
    AUD_vlog (AUDIO_CAP, fmt, ap);
    va_end (ap);

    AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
}

static void alsa_fini_poll (struct pollhlp *hlp)
{
    int i;
    struct pollfd *pfds = hlp->pfds;

    if (pfds) {
        for (i = 0; i < hlp->count; ++i) {
            qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
        }
        qemu_free (pfds);
    }
    hlp->pfds = NULL;
    hlp->count = 0;
    hlp->handle = NULL;
}

static void alsa_anal_close1 (snd_pcm_t **handlep)
{
    int err = snd_pcm_close (*handlep);
    if (err) {
        alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
    }
    *handlep = NULL;
}

static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
{
    alsa_fini_poll (hlp);
    alsa_anal_close1 (handlep);
}

static int alsa_recover (snd_pcm_t *handle)
{
    int err = snd_pcm_prepare (handle);
    if (err < 0) {
        alsa_logerr (err, "Failed to prepare handle %p\n", handle);
        return -1;
    }
    return 0;
}

static int alsa_resume (snd_pcm_t *handle)
{
    int err = snd_pcm_resume (handle);
    if (err < 0) {
        alsa_logerr (err, "Failed to resume handle %p\n", handle);
        return -1;
    }
    return 0;
}

static void alsa_poll_handler (void *opaque)
{
    int err, count;
    snd_pcm_state_t state;
    struct pollhlp *hlp = opaque;
    unsigned short revents;

    count = poll (hlp->pfds, hlp->count, 0);
    if (count < 0) {
        dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
        return;
    }

    if (!count) {
        return;
    }

    /* XXX: ALSA example uses initial count, not the one returned by
       poll, correct? */
    err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
                                            hlp->count, &revents);
    if (err < 0) {
        alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
        return;
    }

    if (!(revents & hlp->mask)) {
        if (conf.verbose) {
            dolog ("revents = %d\n", revents);
        }
        return;
    }

    state = snd_pcm_state (hlp->handle);
    switch (state) {
    case SND_PCM_STATE_SETUP:
        alsa_recover (hlp->handle);
        break;

    case SND_PCM_STATE_XRUN:
        alsa_recover (hlp->handle);
        break;

    case SND_PCM_STATE_SUSPENDED:
        alsa_resume (hlp->handle);
        break;

    case SND_PCM_STATE_PREPARED:
        audio_run ("alsa run (prepared)");
        break;

    case SND_PCM_STATE_RUNNING:
        audio_run ("alsa run (running)");
        break;

    default:
        dolog ("Unexpected state %d\n", state);
    }
}

static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
{
    int i, count, err;
    struct pollfd *pfds;

    count = snd_pcm_poll_descriptors_count (handle);
    if (count <= 0) {
        dolog ("Could not initialize poll mode\n"
               "Invalid number of poll descriptors %d\n", count);
        return -1;
    }

    pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
    if (!pfds) {
        dolog ("Could not initialize poll mode\n");
        return -1;
    }

    err = snd_pcm_poll_descriptors (handle, pfds, count);
    if (err < 0) {
        alsa_logerr (err, "Could not initialize poll mode\n"
                     "Could not obtain poll descriptors\n");
        qemu_free (pfds);
        return -1;
    }

    for (i = 0; i < count; ++i) {
        if (pfds[i].events & POLLIN) {
            err = qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler,
                                       NULL, hlp);
        }
        if (pfds[i].events & POLLOUT) {
            if (conf.verbose) {
                dolog ("POLLOUT %d %d\n", i, pfds[i].fd);
            }
            err = qemu_set_fd_handler (pfds[i].fd, NULL,
                                       alsa_poll_handler, hlp);
        }
        if (conf.verbose) {
            dolog ("Set handler events=%#x index=%d fd=%d err=%d\n",
                   pfds[i].events, i, pfds[i].fd, err);
        }

        if (err) {
            dolog ("Failed to set handler events=%#x index=%d fd=%d err=%d\n",
                   pfds[i].events, i, pfds[i].fd, err);

            while (i--) {
                qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
            }
            qemu_free (pfds);
            return -1;
        }
    }
    hlp->pfds = pfds;
    hlp->count = count;
    hlp->handle = handle;
    hlp->mask = mask;
    return 0;
}

static int alsa_poll_out (HWVoiceOut *hw)
{
    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;

    return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
}

static int alsa_poll_in (HWVoiceIn *hw)
{
    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;

    return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
}

static int alsa_write (SWVoiceOut *sw, void *buf, int len)
{
    return audio_pcm_sw_write (sw, buf, len);
}

static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
{
    switch (fmt) {
    case AUD_FMT_S8:
        return SND_PCM_FORMAT_S8;

    case AUD_FMT_U8:
        return SND_PCM_FORMAT_U8;

    case AUD_FMT_S16:
        if (endianness) {
            return SND_PCM_FORMAT_S16_BE;
        }
        else {
            return SND_PCM_FORMAT_S16_LE;
        }

    case AUD_FMT_U16:
        if (endianness) {
            return SND_PCM_FORMAT_U16_BE;
        }
        else {
            return SND_PCM_FORMAT_U16_LE;
        }

    case AUD_FMT_S32:
        if (endianness) {
            return SND_PCM_FORMAT_S32_BE;
        }
        else {
            return SND_PCM_FORMAT_S32_LE;
        }

    case AUD_FMT_U32:
        if (endianness) {
            return SND_PCM_FORMAT_U32_BE;
        }
        else {
            return SND_PCM_FORMAT_U32_LE;
        }

    default:
        dolog ("Internal logic error: Bad audio format %d\n", fmt);
#ifdef DEBUG_AUDIO
        abort ();
#endif
        return SND_PCM_FORMAT_U8;
    }
}

static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
                           int *endianness)
{
    switch (alsafmt) {
    case SND_PCM_FORMAT_S8:
        *endianness = 0;
        *fmt = AUD_FMT_S8;
        break;

    case SND_PCM_FORMAT_U8:
        *endianness = 0;
        *fmt = AUD_FMT_U8;
        break;

    case SND_PCM_FORMAT_S16_LE:
        *endianness = 0;
        *fmt = AUD_FMT_S16;
        break;

    case SND_PCM_FORMAT_U16_LE:
        *endianness = 0;
        *fmt = AUD_FMT_U16;
        break;

    case SND_PCM_FORMAT_S16_BE:
        *endianness = 1;
        *fmt = AUD_FMT_S16;
        break;

    case SND_PCM_FORMAT_U16_BE:
        *endianness = 1;
        *fmt = AUD_FMT_U16;
        break;

    case SND_PCM_FORMAT_S32_LE:
        *endianness = 0;
        *fmt = AUD_FMT_S32;
        break;

    case SND_PCM_FORMAT_U32_LE:
        *endianness = 0;
        *fmt = AUD_FMT_U32;
        break;

    case SND_PCM_FORMAT_S32_BE:
        *endianness = 1;
        *fmt = AUD_FMT_S32;
        break;

    case SND_PCM_FORMAT_U32_BE:
        *endianness = 1;
        *fmt = AUD_FMT_U32;
        break;

    default:
        dolog ("Unrecognized audio format %d\n", alsafmt);
        return -1;
    }

    return 0;
}

static void alsa_dump_info (struct alsa_params_req *req,
                            struct alsa_params_obt *obt,
                            snd_pcm_format_t obtfmt)
{
    dolog ("parameter | requested value | obtained value\n");
    dolog ("format    |      %10d |     %10d\n", req->fmt, obtfmt);
    dolog ("channels  |      %10d |     %10d\n",
           req->nchannels, obt->nchannels);
    dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
    dolog ("============================================\n");
    dolog ("requested: buffer size %d period size %d\n",
           req->buffer_size, req->period_size);
    dolog ("obtained: samples %ld\n", obt->samples);
}

static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
{
    int err;
    snd_pcm_sw_params_t *sw_params;

    snd_pcm_sw_params_alloca (&sw_params);

    err = snd_pcm_sw_params_current (handle, sw_params);
    if (err < 0) {
        dolog ("Could not fully initialize DAC\n");
        alsa_logerr (err, "Failed to get current software parameters\n");
        return;
    }

    err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
    if (err < 0) {
        dolog ("Could not fully initialize DAC\n");
        alsa_logerr (err, "Failed to set software threshold to %ld\n",
                     threshold);
        return;
    }

    err = snd_pcm_sw_params (handle, sw_params);
    if (err < 0) {
        dolog ("Could not fully initialize DAC\n");
        alsa_logerr (err, "Failed to set software parameters\n");
        return;
    }
}

static int alsa_open (int in, struct alsa_params_req *req,
                      struct alsa_params_obt *obt, snd_pcm_t **handlep)
{
    snd_pcm_t *handle;
    snd_pcm_hw_params_t *hw_params;
    int err;
    int size_in_usec;
    unsigned int freq, nchannels;
    const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
    snd_pcm_uframes_t obt_buffer_size;
    const char *typ = in ? "ADC" : "DAC";
    snd_pcm_format_t obtfmt;

    freq = req->freq;
    nchannels = req->nchannels;
    size_in_usec = req->size_in_usec;

    snd_pcm_hw_params_alloca (&hw_params);

    err = snd_pcm_open (
        &handle,
        pcm_name,
        in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
        SND_PCM_NONBLOCK
        );
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
        return -1;
    }

    err = snd_pcm_hw_params_any (handle, hw_params);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
        goto err;
    }

    err = snd_pcm_hw_params_set_access (
        handle,
        hw_params,
        SND_PCM_ACCESS_RW_INTERLEAVED
        );
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to set access type\n");
        goto err;
    }

    err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
    if (err < 0 && conf.verbose) {
        alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
    }

    err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
        goto err;
    }

    err = snd_pcm_hw_params_set_channels_near (
        handle,
        hw_params,
        &nchannels
        );
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
                      req->nchannels);
        goto err;
    }

    if (nchannels != 1 && nchannels != 2) {
        alsa_logerr2 (err, typ,
                      "Can not handle obtained number of channels %d\n",
                      nchannels);
        goto err;
    }

    if (req->buffer_size) {
        unsigned long obt;

        if (size_in_usec) {
            int dir = 0;
            unsigned int btime = req->buffer_size;

            err = snd_pcm_hw_params_set_buffer_time_near (
                handle,
                hw_params,
                &btime,
                &dir
                );
            obt = btime;
        }
        else {
            snd_pcm_uframes_t bsize = req->buffer_size;

            err = snd_pcm_hw_params_set_buffer_size_near (
                handle,
                hw_params,
                &bsize
                );
            obt = bsize;
        }
        if (err < 0) {
            alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
                          size_in_usec ? "time" : "size", req->buffer_size);
            goto err;
        }

        if ((req->override_mask & 2) && (obt - req->buffer_size))
            dolog ("Requested buffer %s %u was rejected, using %lu\n",
                   size_in_usec ? "time" : "size", req->buffer_size, obt);
    }

    if (req->period_size) {
        unsigned long obt;

        if (size_in_usec) {
            int dir = 0;
            unsigned int ptime = req->period_size;

            err = snd_pcm_hw_params_set_period_time_near (
                handle,
                hw_params,
                &ptime,
                &dir
                );
            obt = ptime;
        }
        else {
            int dir = 0;
            snd_pcm_uframes_t psize = req->period_size;

            err = snd_pcm_hw_params_set_period_size_near (
                handle,
                hw_params,
                &psize,
                &dir
                );
            obt = psize;
        }

        if (err < 0) {
            alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
                          size_in_usec ? "time" : "size", req->period_size);
            goto err;
        }

        if (((req->override_mask & 1) && (obt - req->period_size)))
            dolog ("Requested period %s %u was rejected, using %lu\n",
                   size_in_usec ? "time" : "size", req->period_size, obt);
    }

    err = snd_pcm_hw_params (handle, hw_params);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
        goto err;
    }

    err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to get buffer size\n");
        goto err;
    }

    err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Failed to get format\n");
        goto err;
    }

    if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
        dolog ("Invalid format was returned %d\n", obtfmt);
        goto err;
    }

    err = snd_pcm_prepare (handle);
    if (err < 0) {
        alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
        goto err;
    }

    if (!in && conf.threshold) {
        snd_pcm_uframes_t threshold;
        int bytes_per_sec;

        bytes_per_sec = freq << (nchannels == 2);

        switch (obt->fmt) {
        case AUD_FMT_S8:
        case AUD_FMT_U8:
            break;

        case AUD_FMT_S16:
        case AUD_FMT_U16:
            bytes_per_sec <<= 1;
            break;

        case AUD_FMT_S32:
        case AUD_FMT_U32:
            bytes_per_sec <<= 2;
            break;
        }

        threshold = (conf.threshold * bytes_per_sec) / 1000;
        alsa_set_threshold (handle, threshold);
    }

    obt->nchannels = nchannels;
    obt->freq = freq;
    obt->samples = obt_buffer_size;

    *handlep = handle;

    if (conf.verbose &&
        (obtfmt != req->fmt ||
         obt->nchannels != req->nchannels ||
         obt->freq != req->freq)) {
        dolog ("Audio parameters for %s\n", typ);
        alsa_dump_info (req, obt, obtfmt);
    }

#ifdef DEBUG
    alsa_dump_info (req, obt, obtfmt);
#endif
    return 0;

 err:
    alsa_anal_close1 (&handle);
    return -1;
}

static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
{
    snd_pcm_sframes_t avail;

    avail = snd_pcm_avail_update (handle);
    if (avail < 0) {
        if (avail == -EPIPE) {
            if (!alsa_recover (handle)) {
                avail = snd_pcm_avail_update (handle);
            }
        }

        if (avail < 0) {
            alsa_logerr (avail,
                         "Could not obtain number of available frames\n");
            return -1;
        }
    }

    return avail;
}

static void alsa_write_pending (ALSAVoiceOut *alsa)
{
    HWVoiceOut *hw = &alsa->hw;

    while (alsa->pending) {
        int left_till_end_samples = hw->samples - alsa->wpos;
        int len = audio_MIN (alsa->pending, left_till_end_samples);
        char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);

        while (len) {
            snd_pcm_sframes_t written;

            written = snd_pcm_writei (alsa->handle, src, len);

            if (written <= 0) {
                switch (written) {
                case 0:
                    if (conf.verbose) {
                        dolog ("Failed to write %d frames (wrote zero)\n", len);
                    }
                    return;

                case -EPIPE:
                    if (alsa_recover (alsa->handle)) {
                        alsa_logerr (written, "Failed to write %d frames\n",
                                     len);
                        return;
                    }
                    if (conf.verbose) {
                        dolog ("Recovering from playback xrun\n");
                    }
                    continue;

                case -ESTRPIPE:
                    /* stream is suspended and waiting for an
                       application recovery */
                    if (alsa_resume (alsa->handle)) {
                        alsa_logerr (written, "Failed to write %d frames\n",
                                     len);
                        return;
                    }
                    if (conf.verbose) {
                        dolog ("Resuming suspended output stream\n");
                    }
                    continue;

                case -EAGAIN:
                    return;

                default:
                    alsa_logerr (written, "Failed to write %d frames from %p\n",
                                 len, src);
                    return;
                }
            }

            alsa->wpos = (alsa->wpos + written) % hw->samples;
            alsa->pending -= written;
            len -= written;
        }
    }
}

static int alsa_run_out (HWVoiceOut *hw, int live)
{
    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
    int decr;
    snd_pcm_sframes_t avail;

    avail = alsa_get_avail (alsa->handle);
    if (avail < 0) {
        dolog ("Could not get number of available playback frames\n");
        return 0;
    }

    decr = audio_MIN (live, avail);
    decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
    alsa->pending += decr;
    alsa_write_pending (alsa);
    return decr;
}

static void alsa_fini_out (HWVoiceOut *hw)
{
    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;

    ldebug ("alsa_fini\n");
    alsa_anal_close (&alsa->handle, &alsa->pollhlp);

    if (alsa->pcm_buf) {
        qemu_free (alsa->pcm_buf);
        alsa->pcm_buf = NULL;
    }
}

static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
{
    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
    struct alsa_params_req req;
    struct alsa_params_obt obt;
    snd_pcm_t *handle;
    struct audsettings obt_as;

    req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
    req.freq = as->freq;
    req.nchannels = as->nchannels;
    req.period_size = conf.period_size_out;
    req.buffer_size = conf.buffer_size_out;
    req.size_in_usec = conf.size_in_usec_out;
    req.override_mask =
        (conf.period_size_out_overridden ? 1 : 0) |
        (conf.buffer_size_out_overridden ? 2 : 0);

    if (alsa_open (0, &req, &obt, &handle)) {
        return -1;
    }

    obt_as.freq = obt.freq;
    obt_as.nchannels = obt.nchannels;
    obt_as.fmt = obt.fmt;
    obt_as.endianness = obt.endianness;

    audio_pcm_init_info (&hw->info, &obt_as);
    hw->samples = obt.samples;

    alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
    if (!alsa->pcm_buf) {
        dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
               hw->samples, 1 << hw->info.shift);
        alsa_anal_close1 (&handle);
        return -1;
    }

    alsa->handle = handle;
    return 0;
}

#define VOICE_CTL_PAUSE 0
#define VOICE_CTL_PREPARE 1
#define VOICE_CTL_START 2

static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
{
    int err;

    if (ctl == VOICE_CTL_PAUSE) {
        err = snd_pcm_drop (handle);
        if (err < 0) {
            alsa_logerr (err, "Could not stop %s\n", typ);
            return -1;
        }
    }
    else {
        err = snd_pcm_prepare (handle);
        if (err < 0) {
            alsa_logerr (err, "Could not prepare handle for %s\n", typ);
            return -1;
        }
        if (ctl == VOICE_CTL_START) {
            err = snd_pcm_start(handle);
            if (err < 0) {
                alsa_logerr (err, "Could not start handle for %s\n", typ);
                return -1;
            }
        }
    }

    return 0;
}

static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;

    switch (cmd) {
    case VOICE_ENABLE:
        {
            va_list ap;
            int poll_mode;

            va_start (ap, cmd);
            poll_mode = va_arg (ap, int);
            va_end (ap);

            ldebug ("enabling voice\n");
            if (poll_mode && alsa_poll_out (hw)) {
                poll_mode = 0;
            }
            hw->poll_mode = poll_mode;
            return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
        }

    case VOICE_DISABLE:
        ldebug ("disabling voice\n");
        if (hw->poll_mode) {
            hw->poll_mode = 0;
            alsa_fini_poll (&alsa->pollhlp);
        }
        return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
    }

    return -1;
}

static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
{
    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
    struct alsa_params_req req;
    struct alsa_params_obt obt;
    snd_pcm_t *handle;
    struct audsettings obt_as;

    req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
    req.freq = as->freq;
    req.nchannels = as->nchannels;
    req.period_size = conf.period_size_in;
    req.buffer_size = conf.buffer_size_in;
    req.size_in_usec = conf.size_in_usec_in;
    req.override_mask =
        (conf.period_size_in_overridden ? 1 : 0) |
        (conf.buffer_size_in_overridden ? 2 : 0);

    if (alsa_open (1, &req, &obt, &handle)) {
        return -1;
    }

    obt_as.freq = obt.freq;
    obt_as.nchannels = obt.nchannels;
    obt_as.fmt = obt.fmt;
    obt_as.endianness = obt.endianness;

    audio_pcm_init_info (&hw->info, &obt_as);
    hw->samples = obt.samples;

    alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
    if (!alsa->pcm_buf) {
        dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
               hw->samples, 1 << hw->info.shift);
        alsa_anal_close1 (&handle);
        return -1;
    }

    alsa->handle = handle;
    return 0;
}

static void alsa_fini_in (HWVoiceIn *hw)
{
    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;

    alsa_anal_close (&alsa->handle, &alsa->pollhlp);

    if (alsa->pcm_buf) {
        qemu_free (alsa->pcm_buf);
        alsa->pcm_buf = NULL;
    }
}

static int alsa_run_in (HWVoiceIn *hw)
{
    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
    int hwshift = hw->info.shift;
    int i;
    int live = audio_pcm_hw_get_live_in (hw);
    int dead = hw->samples - live;
    int decr;
    struct {
        int add;
        int len;
    } bufs[2] = {
        { .add = hw->wpos, .len = 0 },
        { .add = 0,        .len = 0 }
    };
    snd_pcm_sframes_t avail;
    snd_pcm_uframes_t read_samples = 0;

    if (!dead) {
        return 0;
    }

    avail = alsa_get_avail (alsa->handle);
    if (avail < 0) {
        dolog ("Could not get number of captured frames\n");
        return 0;
    }

    if (!avail) {
        snd_pcm_state_t state;

        state = snd_pcm_state (alsa->handle);
        switch (state) {
        case SND_PCM_STATE_PREPARED:
            avail = hw->samples;
            break;
        case SND_PCM_STATE_SUSPENDED:
            /* stream is suspended and waiting for an application recovery */
            if (alsa_resume (alsa->handle)) {
                dolog ("Failed to resume suspended input stream\n");
                return 0;
            }
            if (conf.verbose) {
                dolog ("Resuming suspended input stream\n");
            }
            break;
        default:
            if (conf.verbose) {
                dolog ("No frames available and ALSA state is %d\n", state);
            }
            return 0;
        }
    }

    decr = audio_MIN (dead, avail);
    if (!decr) {
        return 0;
    }

    if (hw->wpos + decr > hw->samples) {
        bufs[0].len = (hw->samples - hw->wpos);
        bufs[1].len = (decr - (hw->samples - hw->wpos));
    }
    else {
        bufs[0].len = decr;
    }

    for (i = 0; i < 2; ++i) {
        void *src;
        struct st_sample *dst;
        snd_pcm_sframes_t nread;
        snd_pcm_uframes_t len;

        len = bufs[i].len;

        src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
        dst = hw->conv_buf + bufs[i].add;

        while (len) {
            nread = snd_pcm_readi (alsa->handle, src, len);

            if (nread <= 0) {
                switch (nread) {
                case 0:
                    if (conf.verbose) {
                        dolog ("Failed to read %ld frames (read zero)\n", len);
                    }
                    goto exit;

                case -EPIPE:
                    if (alsa_recover (alsa->handle)) {
                        alsa_logerr (nread, "Failed to read %ld frames\n", len);
                        goto exit;
                    }
                    if (conf.verbose) {
                        dolog ("Recovering from capture xrun\n");
                    }
                    continue;

                case -EAGAIN:
                    goto exit;

                default:
                    alsa_logerr (
                        nread,
                        "Failed to read %ld frames from %p\n",
                        len,
                        src
                        );
                    goto exit;
                }
            }

            hw->conv (dst, src, nread);

            src = advance (src, nread << hwshift);
            dst += nread;

            read_samples += nread;
            len -= nread;
        }
    }

 exit:
    hw->wpos = (hw->wpos + read_samples) % hw->samples;
    return read_samples;
}

static int alsa_read (SWVoiceIn *sw, void *buf, int size)
{
    return audio_pcm_sw_read (sw, buf, size);
}

static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
    ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;

    switch (cmd) {
    case VOICE_ENABLE:
        {
            va_list ap;
            int poll_mode;

            va_start (ap, cmd);
            poll_mode = va_arg (ap, int);
            va_end (ap);

            ldebug ("enabling voice\n");
            if (poll_mode && alsa_poll_in (hw)) {
                poll_mode = 0;
            }
            hw->poll_mode = poll_mode;

            return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
        }

    case VOICE_DISABLE:
        ldebug ("disabling voice\n");
        if (hw->poll_mode) {
            hw->poll_mode = 0;
            alsa_fini_poll (&alsa->pollhlp);
        }
        return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
    }

    return -1;
}

static void *alsa_audio_init (void)
{
    return &conf;
}

static void alsa_audio_fini (void *opaque)
{
    (void) opaque;
}

static struct audio_option alsa_options[] = {
    {
        .name        = "DAC_SIZE_IN_USEC",
        .tag         = AUD_OPT_BOOL,
        .valp        = &conf.size_in_usec_out,
        .descr       = "DAC period/buffer size in microseconds (otherwise in frames)"
    },
    {
        .name        = "DAC_PERIOD_SIZE",
        .tag         = AUD_OPT_INT,
        .valp        = &conf.period_size_out,
        .descr       = "DAC period size (0 to go with system default)",
        .overriddenp = &conf.period_size_out_overridden
    },
    {
        .name        = "DAC_BUFFER_SIZE",
        .tag         = AUD_OPT_INT,
        .valp        = &conf.buffer_size_out,
        .descr       = "DAC buffer size (0 to go with system default)",
        .overriddenp = &conf.buffer_size_out_overridden
    },
    {
        .name        = "ADC_SIZE_IN_USEC",
        .tag         = AUD_OPT_BOOL,
        .valp        = &conf.size_in_usec_in,
        .descr       =
        "ADC period/buffer size in microseconds (otherwise in frames)"
    },
    {
        .name        = "ADC_PERIOD_SIZE",
        .tag         = AUD_OPT_INT,
        .valp        = &conf.period_size_in,
        .descr       = "ADC period size (0 to go with system default)",
        .overriddenp = &conf.period_size_in_overridden
    },
    {
        .name        = "ADC_BUFFER_SIZE",
        .tag         = AUD_OPT_INT,
        .valp        = &conf.buffer_size_in,
        .descr       = "ADC buffer size (0 to go with system default)",
        .overriddenp = &conf.buffer_size_in_overridden
    },
    {
        .name        = "THRESHOLD",
        .tag         = AUD_OPT_INT,
        .valp        = &conf.threshold,
        .descr       = "(undocumented)"
    },
    {
        .name        = "DAC_DEV",
        .tag         = AUD_OPT_STR,
        .valp        = &conf.pcm_name_out,
        .descr       = "DAC device name (for instance dmix)"
    },
    {
        .name        = "ADC_DEV",
        .tag         = AUD_OPT_STR,
        .valp        = &conf.pcm_name_in,
        .descr       = "ADC device name"
    },
    {
        .name        = "VERBOSE",
        .tag         = AUD_OPT_BOOL,
        .valp        = &conf.verbose,
        .descr       = "Behave in a more verbose way"
    },
    { /* End of list */ }
};

static struct audio_pcm_ops alsa_pcm_ops = {
    .init_out = alsa_init_out,
    .fini_out = alsa_fini_out,
    .run_out  = alsa_run_out,
    .write    = alsa_write,
    .ctl_out  = alsa_ctl_out,

    .init_in  = alsa_init_in,
    .fini_in  = alsa_fini_in,
    .run_in   = alsa_run_in,
    .read     = alsa_read,
    .ctl_in   = alsa_ctl_in,
};

struct audio_driver alsa_audio_driver = {
    .name           = "alsa",
    .descr          = "ALSA http://www.alsa-project.org",
    .options        = alsa_options,
    .init           = alsa_audio_init,
    .fini           = alsa_audio_fini,
    .pcm_ops        = &alsa_pcm_ops,
    .can_be_default = 1,
    .max_voices_out = INT_MAX,
    .max_voices_in  = INT_MAX,
    .voice_size_out = sizeof (ALSAVoiceOut),
    .voice_size_in  = sizeof (ALSAVoiceIn)
};