/*
 * This file is part of mpv.
 *
 * mpv is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * mpv is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
 */

#include <stddef.h>
#include <stdbool.h>
#include <inttypes.h>
#include <limits.h>
#include <math.h>
#include <assert.h>

#include "config.h"
#include "mpv_talloc.h"

#include "common/msg.h"
#include "common/encode.h"
#include "options/options.h"
#include "common/common.h"
#include "osdep/timer.h"

#include "audio/format.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
#include "filters/f_async_queue.h"
#include "filters/f_decoder_wrapper.h"
#include "filters/filter_internal.h"

#include "core.h"
#include "command.h"

enum {
    AD_OK = 0,
    AD_EOF = -2,
    AD_WAIT = -4,
};

static void ao_process(struct mp_filter *f);

static void update_speed_filters(struct MPContext *mpctx)
{
    struct ao_chain *ao_c = mpctx->ao_chain;
    if (!ao_c)
        return;

    double speed = mpctx->opts->playback_speed;
    double resample = mpctx->speed_factor_a;
    double drop = 1.0;

    if (!mpctx->opts->pitch_correction) {
        resample *= speed;
        speed = 1.0;
    }

    if (mpctx->display_sync_active && mpctx->opts->video_sync == VS_DISP_ADROP) {
        drop *= speed * resample;
        resample = speed = 1.0;
    }

    mp_output_chain_set_audio_speed(ao_c->filter, speed, resample, drop);
}

static int recreate_audio_filters(struct MPContext *mpctx)
{
    struct ao_chain *ao_c = mpctx->ao_chain;
    assert(ao_c);

    if (!mp_output_chain_update_filters(ao_c->filter, mpctx->opts->af_settings))
        goto fail;

    update_speed_filters(mpctx);

    mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);

    return 0;

fail:
    MP_ERR(mpctx, "Audio filter initialized failed!\n");
    return -1;
}

int reinit_audio_filters(struct MPContext *mpctx)
{
    struct ao_chain *ao_c = mpctx->ao_chain;
    if (!ao_c)
        return 0;

    double delay = mp_output_get_measured_total_delay(ao_c->filter);

    if (recreate_audio_filters(mpctx) < 0)
        return -1;

    double ndelay = mp_output_get_measured_total_delay(ao_c->filter);

    // Only force refresh if the amount of dropped buffered data is going to
    // cause "issues" for the A/V sync logic.
    if (mpctx->audio_status == STATUS_PLAYING && delay - ndelay >= 0.2)
        issue_refresh_seek(mpctx, MPSEEK_EXACT);
    return 1;
}

static double db_gain(double db)
{
    return pow(10.0, db/20.0);
}

static float compute_replaygain(struct MPContext *mpctx)
{
    struct MPOpts *opts = mpctx->opts;

    float rgain = 1.0;

    struct replaygain_data *rg = NULL;
    struct track *track = mpctx->current_track[0][STREAM_AUDIO];
    if (track)
        rg = track->stream->codec->replaygain_data;
    if (opts->rgain_mode && rg) {
        MP_VERBOSE(mpctx, "Replaygain: Track=%f/%f Album=%f/%f\n",
                   rg->track_gain, rg->track_peak,
                   rg->album_gain, rg->album_peak);

        float gain, peak;
        if (opts->rgain_mode == 1) {
            gain = rg->track_gain;
            peak = rg->track_peak;
        } else {
            gain = rg->album_gain;
            peak = rg->album_peak;
        }

        gain += opts->rgain_preamp;
        rgain = db_gain(gain);

        MP_VERBOSE(mpctx, "Applying replay-gain: %f\n", rgain);

        if (!opts->rgain_clip) { // clipping prevention
            rgain = MPMIN(rgain, 1.0 / peak);
            MP_VERBOSE(mpctx, "...with clipping prevention: %f\n", rgain);
        }
    } else if (opts->rgain_fallback) {
        rgain = db_gain(opts->rgain_fallback);
        MP_VERBOSE(mpctx, "Applying fallback gain: %f\n", rgain);
    }

    return rgain;
}

// Called when opts->softvol_volume or opts->softvol_mute were changed.
void audio_update_volume(struct MPContext *mpctx)
{
    struct MPOpts *opts = mpctx->opts;
    struct ao_chain *ao_c = mpctx->ao_chain;
    if (!ao_c || !ao_c->ao)
        return;

    float gain = MPMAX(opts->softvol_volume / 100.0, 0);
    gain = pow(gain, 3);
    gain *= compute_replaygain(mpctx);
    if (opts->softvol_mute == 1)
        gain = 0.0;

    ao_set_gain(ao_c->ao, gain);
}

// Call this if opts->playback_speed or mpctx->speed_factor_* change.
void update_playback_speed(struct MPContext *mpctx)
{
    mpctx->audio_speed = mpctx->opts->playback_speed * mpctx->speed_factor_a;
    mpctx->video_speed = mpctx->opts->playback_speed * mpctx->speed_factor_v;

    update_speed_filters(mpctx);
}

static void ao_chain_reset_state(struct ao_chain *ao_c)
{
    ao_c->last_out_pts = MP_NOPTS_VALUE;
    ao_c->out_eof = false;
    ao_c->underrun = false;
    ao_c->start_pts_known = false;
    ao_c->start_pts = MP_NOPTS_VALUE;
    ao_c->untimed_throttle = false;
    ao_c->underrun = false;
}

void reset_audio_state(struct MPContext *mpctx)
{
    if (mpctx->ao_chain) {
        ao_chain_reset_state(mpctx->ao_chain);
        struct track *t = mpctx->ao_chain->track;
        if (t && t->dec)
            mp_decoder_wrapper_set_play_dir(t->dec, mpctx->play_dir);
    }
    mpctx->audio_status = mpctx->ao_chain ? STATUS_SYNCING : STATUS_EOF;
    mpctx->delay = 0;
    mpctx->logged_async_diff = -1;
}

void uninit_audio_out(struct MPContext *mpctx)
{
    struct ao_chain *ao_c = mpctx->ao_chain;
    if (ao_c) {
        ao_c->ao_queue = NULL;
        TA_FREEP(&ao_c->queue_filter);
        ao_c->ao = NULL;
    }
    if (mpctx->ao) {
        // Note: with gapless_audio, stop_play is not correctly set
        if ((mpctx->opts->gapless_audio || mpctx->stop_play == AT_END_OF_FILE) &&
            ao_is_playing(mpctx->ao) && !get_internal_paused(mpctx))
        {
            MP_VERBOSE(mpctx, "draining left over audio\n");
            ao_drain(mpctx->ao);
        }
        ao_uninit(mpctx->ao);

        mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
    }
    mpctx->ao = NULL;
    TA_FREEP(&mpctx->ao_filter_fmt);
}

static void ao_chain_uninit(struct ao_chain *ao_c)
{
    struct track *track = ao_c->track;
    if (track) {
        assert(track->ao_c == ao_c);
        track->ao_c = NULL;
        if (ao_c->dec_src)
            assert(track->dec->f->pins[0] == ao_c->dec_src);
        talloc_free(track->dec->f);
        track->dec = NULL;
    }

    if (ao_c->filter_src)
        mp_pin_disconnect(ao_c->filter_src);

    talloc_free(ao_c->filter->f);
    talloc_free(ao_c->ao_filter);
    talloc_free(ao_c);
}

void uninit_audio_chain(struct MPContext *mpctx)
{
    if (mpctx->ao_chain) {
        ao_chain_uninit(mpctx->ao_chain);
        mpctx->ao_chain = NULL;

        mpctx->audio_status = STATUS_EOF;

        mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
    }
}

static char *audio_config_to_str_buf(char *buf, size_t buf_sz, int rate,
                                     int format, struct mp_chmap channels)
{
    char ch[128];
    mp_chmap_to_str_buf(ch, sizeof(ch), &channels);
    char *hr_ch = mp_chmap_to_str_hr(&channels);
    if (strcmp(hr_ch, ch) != 0)
        mp_snprintf_cat(ch, sizeof(ch), " (%s)", hr_ch);
    snprintf(buf, buf_sz, "%dHz %s %dch %s", rate,
             ch, channels.num, af_fmt_to_str(format));
    return buf;
}

// Decide whether on a format change, we should reinit the AO.
static bool keep_weak_gapless_format(struct mp_aframe *old, struct mp_aframe* new)
{
    bool res = false;
    struct mp_aframe *new_mod = mp_aframe_new_ref(new);
    if (!new_mod)
        abort();

    // If the sample formats are compatible (== libswresample generally can
    // convert them), keep the AO. On other changes, recreate it.

    int old_fmt = mp_aframe_get_format(old);
    int new_fmt = mp_aframe_get_format(new);

    if (af_format_conversion_score(old_fmt, new_fmt) == INT_MIN)
        goto done; // completely incompatible formats

    if (!mp_aframe_set_format(new_mod, old_fmt))
        goto done;

    res = mp_aframe_config_equals(old, new_mod);

done:
    talloc_free(new_mod);
    return res;
}

static void ao_chain_set_ao(struct ao_chain *ao_c, struct ao *ao)
{
    if (ao_c->ao != ao) {
        assert(!ao_c->ao);
        ao_c->ao = ao;
        ao_c->ao_queue = ao_get_queue(ao_c->ao);
        ao_c->queue_filter = mp_async_queue_create_filter(ao_c->ao_filter,
                                                          MP_PIN_IN, ao_c->ao_queue);
        mp_async_queue_set_notifier(ao_c->queue_filter, ao_c->ao_filter);
        // Make sure filtering never stops with frames stuck in access filter.
        mp_filter_set_high_priority(ao_c->queue_filter, true);
    }

    if (ao_c->filter->ao_needs_update)
        mp_output_chain_set_ao(ao_c->filter, ao_c->ao);

    mp_filter_wakeup(ao_c->ao_filter);
}

static void reinit_audio_filters_and_output(struct MPContext *mpctx)
{
    struct MPOpts *opts = mpctx->opts;
    struct ao_chain *ao_c = mpctx->ao_chain;
    assert(ao_c);
    struct track *track = ao_c->track;

    assert(ao_c->filter->ao_needs_update);

    // The "ideal" filter output format
    struct mp_aframe *out_fmt = mp_aframe_new_ref(ao_c->filter->output_aformat);
    if (!out_fmt)
        abort();

    if (!mp_aframe_config_is_valid(out_fmt)) {
        talloc_free(out_fmt);
        goto init_error;
    }

    if (af_fmt_is_pcm(mp_aframe_get_format(out_fmt))) {
        if (opts->force_srate)
            mp_aframe_set_rate(out_fmt, opts->force_srate);
        if (opts->audio_output_format)
            mp_aframe_set_format(out_fmt, opts->audio_output_format);
        if (opts->audio_output_channels.num_chmaps == 1)
            mp_aframe_set_chmap(out_fmt, &opts->audio_output_channels.chmaps[0]);
    }

    // Weak gapless audio: if the filter output format is the same as the
    // previous one, keep the AO and don't reinit anything.
    // Strong gapless: always keep the AO
    if ((mpctx->ao_filter_fmt && mpctx->ao && opts->gapless_audio < 0 &&
         keep_weak_gapless_format(mpctx->ao_filter_fmt, out_fmt)) ||
        (mpctx->ao && opts->gapless_audio > 0))
    {
        ao_chain_set_ao(ao_c, mpctx->ao);
        talloc_free(out_fmt);
        return;
    }

    // Wait until all played.
    if (mpctx->ao && ao_is_playing(mpctx->ao)) {
        talloc_free(out_fmt);
        return;
    }
    // Format change during syncing. Force playback start early, then wait.
    if (ao_c->ao_queue && mp_async_queue_get_frames(ao_c->ao_queue) &&
        mpctx->audio_status == STATUS_SYNCING)
    {
        mpctx->audio_status = STATUS_READY;
        mp_wakeup_core(mpctx);
        talloc_free(out_fmt);
        return;
    }
    if (mpctx->audio_status == STATUS_READY) {
        talloc_free(out_fmt);
        return;
    }

    uninit_audio_out(mpctx);

    int out_rate = mp_aframe_get_rate(out_fmt);
    int out_format = mp_aframe_get_format(out_fmt);
    struct mp_chmap out_channels = {0};
    mp_aframe_get_chmap(out_fmt, &out_channels);

    int ao_flags = 0;
    bool spdif_fallback = af_fmt_is_spdif(out_format) &&
                          ao_c->spdif_passthrough;

    if (opts->ao_null_fallback && !spdif_fallback)
        ao_flags |= AO_INIT_NULL_FALLBACK;

    if (opts->audio_stream_silence)
        ao_flags |= AO_INIT_STREAM_SILENCE;

    if (opts->audio_exclusive)
        ao_flags |= AO_INIT_EXCLUSIVE;

    if (af_fmt_is_pcm(out_format)) {
        if (!opts->audio_output_channels.set ||
            opts->audio_output_channels.auto_safe)
            ao_flags |= AO_INIT_SAFE_MULTICHANNEL_ONLY;

        mp_chmap_sel_list(&out_channels,
                          opts->audio_output_channels.chmaps,
                          opts->audio_output_channels.num_chmaps);
    }

    mpctx->ao_filter_fmt = out_fmt;

    mpctx->ao = ao_init_best(mpctx->global, ao_flags, mp_wakeup_core_cb,
                             mpctx, mpctx->encode_lavc_ctx, out_rate,
                             out_format, out_channels);

    int ao_rate = 0;
    int ao_format = 0;
    struct mp_chmap ao_channels = {0};
    if (mpctx->ao)
        ao_get_format(mpctx->ao, &ao_rate, &ao_format, &ao_channels);

    // Verify passthrough format was not changed.
    if (mpctx->ao && af_fmt_is_spdif(out_format)) {
        if (out_rate != ao_rate || out_format != ao_format ||
            !mp_chmap_equals(&out_channels, &ao_channels))
        {
            MP_ERR(mpctx, "Passthrough format unsupported.\n");
            ao_uninit(mpctx->ao);
            mpctx->ao = NULL;
        }
    }

    if (!mpctx->ao) {
        // If spdif was used, try to fallback to PCM.
        if (spdif_fallback && ao_c->track && ao_c->track->dec) {
            MP_VERBOSE(mpctx, "Falling back to PCM output.\n");
            ao_c->spdif_passthrough = false;
            ao_c->spdif_failed = true;
            mp_decoder_wrapper_set_spdif_flag(ao_c->track->dec, false);
            if (!mp_decoder_wrapper_reinit(ao_c->track->dec))
                goto init_error;
            reset_audio_state(mpctx);
            mp_output_chain_reset_harder(ao_c->filter);
            mp_wakeup_core(mpctx); // reinit with new format next time
            return;
        }

        MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
        mpctx->error_playing = MPV_ERROR_AO_INIT_FAILED;
        goto init_error;
    }

    char tmp[192];
    MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(mpctx->ao),
            audio_config_to_str_buf(tmp, sizeof(tmp), ao_rate, ao_format,
                                    ao_channels));
    MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(mpctx->ao));
    update_window_title(mpctx, true);

    ao_c->ao_resume_time =
        opts->audio_wait_open > 0 ? mp_time_sec() + opts->audio_wait_open : 0;

    ao_set_paused(mpctx->ao, get_internal_paused(mpctx));

    ao_chain_set_ao(ao_c, mpctx->ao);

    audio_update_volume(mpctx);

    // Almost nonsensical hack to deal with certain format change scenarios.
    if (mpctx->audio_status == STATUS_PLAYING)
        ao_start(mpctx->ao);

    mp_wakeup_core(mpctx);
    mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);

    return;

init_error:
    uninit_audio_chain(mpctx);
    uninit_audio_out(mpctx);
    error_on_track(mpctx, track);
}

int init_audio_decoder(struct MPContext *mpctx, struct track *track)
{
    assert(!track->dec);
    if (!track->stream)
        goto init_error;

    track->dec = mp_decoder_wrapper_create(mpctx->filter_root, track->stream);
    if (!track->dec)
        goto init_error;

    if (track->ao_c)
        mp_decoder_wrapper_set_spdif_flag(track->dec, true);

    if (!mp_decoder_wrapper_reinit(track->dec))
        goto init_error;

    return 1;

init_error:
    if (track->sink)
        mp_pin_disconnect(track->sink);
    track->sink = NULL;
    error_on_track(mpctx, track);
    return 0;
}

void reinit_audio_chain(struct MPContext *mpctx)
{
    struct track *track = NULL;
    track = mpctx->current_track[0][STREAM_AUDIO];
    if (!track || !track->stream) {
        uninit_audio_out(mpctx);
        error_on_track(mpctx, track);
        return;
    }
    reinit_audio_chain_src(mpctx, track);
}

static const struct mp_filter_info ao_filter = {
    .name = "ao",
    .process = ao_process,
};

// (track=NULL creates a blank chain, used for lavfi-complex)
void reinit_audio_chain_src(struct MPContext *mpctx, struct track *track)
{
    assert(!mpctx->ao_chain);

    mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);

    struct ao_chain *ao_c = talloc_zero(NULL, struct ao_chain);
    mpctx->ao_chain = ao_c;
    ao_c->mpctx = mpctx;
    ao_c->log = mpctx->log;
    ao_c->filter =
        mp_output_chain_create(mpctx->filter_root, MP_OUTPUT_CHAIN_AUDIO);
    ao_c->spdif_passthrough = true;
    ao_c->last_out_pts = MP_NOPTS_VALUE;
    ao_c->delay = mpctx->opts->audio_delay;

    ao_c->ao_filter = mp_filter_create(mpctx->filter_root, &ao_filter);
    if (!ao_c->filter || !ao_c->ao_filter)
        goto init_error;
    ao_c->ao_filter->priv = ao_c;

    mp_filter_add_pin(ao_c->ao_filter, MP_PIN_IN, "in");
    mp_pin_connect(ao_c->ao_filter->pins[0], ao_c->filter->f->pins[1]);

    if (track) {
        ao_c->track = track;
        track->ao_c = ao_c;
        if (!init_audio_decoder(mpctx, track))
            goto init_error;
        ao_c->dec_src = track->dec->f->pins[0];
        mp_pin_connect(ao_c->filter->f->pins[0], ao_c->dec_src);
    }

    reset_audio_state(mpctx);

    if (recreate_audio_filters(mpctx) < 0)
        goto init_error;

    if (mpctx->ao)
        audio_update_volume(mpctx);

    mp_wakeup_core(mpctx);
    return;

init_error:
    uninit_audio_chain(mpctx);
    uninit_audio_out(mpctx);
    error_on_track(mpctx, track);
}

// Return pts value corresponding to the start point of audio written to the
// ao queue so far.
double written_audio_pts(struct MPContext *mpctx)
{
    return mpctx->ao_chain ? mpctx->ao_chain->last_out_pts : MP_NOPTS_VALUE;
}

// Return pts value corresponding to currently playing audio.
double playing_audio_pts(struct MPContext *mpctx)
{
    double pts = written_audio_pts(mpctx);
    if (pts == MP_NOPTS_VALUE || !mpctx->ao)
        return pts;
    return pts - mpctx->audio_speed * ao_get_delay(mpctx->ao);
}

// This garbage is needed for untimed AOs. These consume audio infinitely fast,
// so try keeping approximate A/V sync by blocking audio transfer as needed.
static void update_throttle(struct MPContext *mpctx)
{
    struct ao_chain *ao_c = mpctx->ao_chain;
    bool new_throttle = mpctx->audio_status == STATUS_PLAYING &&
                        mpctx->delay > 0 && ao_c && ao_c->ao &&
                        ao_untimed(ao_c->ao) &&
                        mpctx->video_status != STATUS_EOF;
    if (ao_c && new_throttle != ao_c->untimed_throttle) {
        ao_c->untimed_throttle = new_throttle;
        mp_wakeup_core(mpctx);
        mp_filter_wakeup(ao_c->ao_filter);
    }
}

static void ao_process(struct mp_filter *f)
{
    struct ao_chain *ao_c = f->priv;
    struct MPContext *mpctx = ao_c->mpctx;

    if (!ao_c->queue_filter) {
        // This will eventually lead to the creation of the AO + queue, due
        // to how f_output_chain and AO management works.
        mp_pin_out_request_data(f->ppins[0]);
        // Check for EOF with no data case, which is a mess because everything
        // hates us.
        struct mp_frame frame = mp_pin_out_read(f->ppins[0]);
        if (frame.type == MP_FRAME_EOF) {
            MP_VERBOSE(mpctx, "got EOF with no data before it\n");
            ao_c->out_eof = true;
            mpctx->audio_status = STATUS_DRAINING;
            mp_wakeup_core(mpctx);
        } else if (frame.type) {
            mp_pin_out_unread(f->ppins[0], frame);
        }
        return;
    }

    // Due to mp_async_queue_set_notifier() thhis function is called when the
    // queue becomes full. This affects state changes in the normal playloop,
    // so wake it up. But avoid redundant wakeups during normal playback.
    if (mpctx->audio_status != STATUS_PLAYING &&
        mp_async_queue_is_full(ao_c->ao_queue))
        mp_wakeup_core(mpctx);

    if (mpctx->audio_status == STATUS_SYNCING && !ao_c->start_pts_known)
        return;

    if (ao_c->untimed_throttle)
        return;

    if (!mp_pin_can_transfer_data(ao_c->queue_filter->pins[0], f->ppins[0]))
        return;

    struct mp_frame frame = mp_pin_out_read(f->ppins[0]);
    if (frame.type == MP_FRAME_AUDIO) {
        struct mp_aframe *af = frame.data;

        double endpts = get_play_end_pts(mpctx);
        if (endpts != MP_NOPTS_VALUE) {
            endpts *= mpctx->play_dir;
            // Avoid decoding and discarding the entire rest of the file.
            if (mp_aframe_get_pts(af) >= endpts) {
                mp_pin_out_unread(f->ppins[0], frame);
                if (!ao_c->out_eof) {
                    ao_c->out_eof = true;
                    mp_pin_in_write(ao_c->queue_filter->pins[0], MP_EOF_FRAME);
                }
                return;
            }
        }
        double startpts = mpctx->audio_status == STATUS_SYNCING ?
                                            ao_c->start_pts : MP_NOPTS_VALUE;
        mp_aframe_clip_timestamps(af, startpts, endpts);

        int samples = mp_aframe_get_size(af);
        if (!samples) {
            mp_filter_internal_mark_progress(f);
            mp_frame_unref(&frame);
            return;
        }

        ao_c->out_eof = false;

        if (mpctx->audio_status == STATUS_DRAINING ||
            mpctx->audio_status == STATUS_EOF)
        {
            // If a new frame comes decoder/filter EOF, we should preferably
            // call get_sync_pts() again, which (at least in obscure situations)
            // may require us to wait a while until the sync PTS is known. Our
            // code sucks and can't deal with that, so jump through a hoop to
            // get things done in the correct order.
            mp_pin_out_unread(f->ppins[0], frame);
            ao_c->start_pts_known = false;
            mpctx->audio_status = STATUS_SYNCING;
            mp_wakeup_core(mpctx);
            MP_VERBOSE(mpctx, "new audio frame after EOF\n");
            return;
        }

        mpctx->shown_aframes += samples;
        double real_samplerate = mp_aframe_get_rate(af) / mpctx->audio_speed;
        mpctx->delay += samples / real_samplerate;
        ao_c->last_out_pts = mp_aframe_end_pts(af);
        update_throttle(mpctx);

        // Gapless case: the AO is still playing from previous file. It makes
        // no sense to wait, and in fact the "full queue" event we're waiting
        // for may never happen, so start immediately.
        // If the new audio starts "later" (big video sync offset), transfer
        // of data is stopped somewhere else.
        if (mpctx->audio_status == STATUS_SYNCING && ao_is_playing(ao_c->ao)) {
            mpctx->audio_status = STATUS_READY;
            mp_wakeup_core(mpctx);
            MP_VERBOSE(mpctx, "previous audio still playing; continuing\n");
        }

        mp_pin_in_write(ao_c->queue_filter->pins[0], frame);
    } else if (frame.type == MP_FRAME_EOF) {
        MP_VERBOSE(mpctx, "audio filter EOF\n");

        ao_c->out_eof = true;
        mp_wakeup_core(mpctx);

        mp_pin_in_write(ao_c->queue_filter->pins[0], frame);
        mp_filter_internal_mark_progress(f);
    } else {
        mp_frame_unref(&frame);
    }
}

void reload_audio_output(struct MPContext *mpctx)
{
    if (!mpctx->ao)
        return;

    ao_reset(mpctx->ao);
    uninit_audio_out(mpctx);
    reinit_audio_filters(mpctx); // mostly to issue refresh seek

    struct ao_chain *ao_c = mpctx->ao_chain;

    if (ao_c) {
        reset_audio_state(mpctx);
        mp_output_chain_reset_harder(ao_c->filter);
    }

    // Whether we can use spdif might have changed. If we failed to use spdif
    // in the previous initialization, try it with spdif again (we'll fallback
    // to PCM again if necessary).
    if (ao_c && ao_c->track) {
        struct mp_decoder_wrapper *dec = ao_c->track->dec;
        if (dec && ao_c->spdif_failed) {
            ao_c->spdif_passthrough = true;
            ao_c->spdif_failed = false;
            mp_decoder_wrapper_set_spdif_flag(ao_c->track->dec, true);
            if (!mp_decoder_wrapper_reinit(dec)) {
                MP_ERR(mpctx, "Error reinitializing audio.\n");
                error_on_track(mpctx, ao_c->track);
            }
        }
    }

    mp_wakeup_core(mpctx);
}

// Returns audio start pts for seeking or video sync.
// Returns false if PTS is not known yet.
static bool get_sync_pts(struct MPContext *mpctx, double *pts)
{
    struct MPOpts *opts = mpctx->opts;

    *pts = MP_NOPTS_VALUE;

    if (!opts->initial_audio_sync)
        return true;

    bool sync_to_video = mpctx->vo_chain && mpctx->video_status != STATUS_EOF &&
                         !mpctx->vo_chain->is_sparse;

    if (sync_to_video) {
        if (mpctx->video_status < STATUS_READY)
            return false; // wait until we know a video PTS
        if (mpctx->video_pts != MP_NOPTS_VALUE)
            *pts = mpctx->video_pts - opts->audio_delay;
    } else if (mpctx->hrseek_active) {
        *pts = mpctx->hrseek_pts;
    } else {
        // If audio-only is enabled mid-stream during playback, sync accordingly.
        *pts = mpctx->playback_pts;
    }

    return true;
}

// Look whether audio can be started yet - if audio has to start some time
// after video.
// Caller needs to ensure mpctx->restart_complete is OK
void audio_start_ao(struct MPContext *mpctx)
{
    struct ao_chain *ao_c = mpctx->ao_chain;
    if (!ao_c || !ao_c->ao || mpctx->audio_status != STATUS_READY)
        return;
    double pts = MP_NOPTS_VALUE;
    if (!get_sync_pts(mpctx, &pts))
        return;
    double apts = playing_audio_pts(mpctx); // (basically including mpctx->delay)
    if (pts != MP_NOPTS_VALUE && apts != MP_NOPTS_VALUE && pts < apts &&
        mpctx->video_status != STATUS_EOF)
    {
        double diff = (apts - pts) / mpctx->opts->playback_speed;
        if (!get_internal_paused(mpctx))
            mp_set_timeout(mpctx, diff);
        if (mpctx->logged_async_diff != diff) {
            MP_VERBOSE(mpctx, "delaying audio start %f vs. %f, diff=%f\n",
                       apts, pts, diff);
            mpctx->logged_async_diff = diff;
        }
        return;
    }

    MP_VERBOSE(mpctx, "starting audio playback\n");
    ao_start(ao_c->ao);
    mpctx->audio_status = STATUS_PLAYING;
    if (ao_c->out_eof)
        mpctx->audio_status = STATUS_DRAINING;
    ao_c->underrun = false;
    mpctx->logged_async_diff = -1;
    mp_wakeup_core(mpctx);
}

void fill_audio_out_buffers(struct MPContext *mpctx)
{
    struct MPOpts *opts = mpctx->opts;

    if (mpctx->ao && ao_query_and_reset_events(mpctx->ao, AO_EVENT_RELOAD))
        reload_audio_output(mpctx);

    if (mpctx->ao && ao_query_and_reset_events(mpctx->ao,
                                               AO_EVENT_INITIAL_UNBLOCK))
        ao_unblock(mpctx->ao);

    update_throttle(mpctx);

    struct ao_chain *ao_c = mpctx->ao_chain;
    if (!ao_c)
        return;

    if (ao_c->filter->failed_output_conversion) {
        error_on_track(mpctx, ao_c->track);
        return;
    }

    if (ao_c->filter->ao_needs_update)
        reinit_audio_filters_and_output(mpctx);

    if (mpctx->vo_chain && ao_c->track && ao_c->track->dec &&
        mp_decoder_wrapper_get_pts_reset(ao_c->track->dec))
    {
        MP_WARN(mpctx, "Reset playback due to audio timestamp reset.\n");
        reset_playback_state(mpctx);
        mp_wakeup_core(mpctx);
    }

    if (mpctx->audio_status == STATUS_SYNCING) {
        double pts;
        bool ok = get_sync_pts(mpctx, &pts);

        // If the AO is still playing from the previous file (due to gapless),
        // but if video is active, this may not work if audio starts later than
        // video, and gapless has no advantages anyway. So block doing anything
        // until the old audio is fully played.
        // (Buggy if AO underruns.)
        if (mpctx->ao && ao_is_playing(mpctx->ao) &&
            mpctx->video_status != STATUS_EOF)
            ok = false;

        if (ao_c->start_pts_known != ok || ao_c->start_pts != pts) {
            ao_c->start_pts_known = ok;
            ao_c->start_pts = pts;
            mp_filter_wakeup(ao_c->ao_filter);
        }

        if (ao_c->ao && mp_async_queue_is_full(ao_c->ao_queue)) {
            mpctx->audio_status = STATUS_READY;
            mp_wakeup_core(mpctx);
            MP_VERBOSE(mpctx, "audio ready\n");
        } else if (ao_c->out_eof) {
            // Force playback start early.
            mpctx->audio_status = STATUS_READY;
            mp_wakeup_core(mpctx);
            MP_VERBOSE(mpctx, "audio ready (and EOF)\n");
        }
    }

    if (ao_c->ao && !ao_is_playing(ao_c->ao) && !ao_c->underrun &&
        (mpctx->audio_status == STATUS_PLAYING ||
         mpctx->audio_status == STATUS_DRAINING))
    {
        // Should be playing, but somehow isn't.

        if (ao_c->out_eof && !mp_async_queue_get_frames(ao_c->ao_queue)) {
            MP_VERBOSE(mpctx, "AO signaled EOF (while in state %s)\n",
                       mp_status_str(mpctx->audio_status));
            mpctx->audio_status = STATUS_EOF;
            mp_wakeup_core(mpctx);
            // stops untimed AOs, stops pull AOs from streaming silence
            ao_reset(ao_c->ao);
        } else {
            if (!ao_c->ao_underrun) {
                MP_WARN(mpctx, "Audio device underrun detected.\n");
                ao_c->ao_underrun = true;
                mp_wakeup_core(mpctx);
                ao_c->underrun = true;
            }

            // Wait until buffers are filled before recovering underrun.
            if (ao_c->out_eof || mp_async_queue_is_full(ao_c->ao_queue)) {
                MP_VERBOSE(mpctx, "restarting audio after underrun\n");
                ao_start(mpctx->ao_chain->ao);
                ao_c->ao_underrun = false;
                ao_c->underrun = false;
                mp_wakeup_core(mpctx);
            }
        }
    }

    if (mpctx->audio_status == STATUS_PLAYING && ao_c->out_eof) {
        mpctx->audio_status = STATUS_DRAINING;
        MP_VERBOSE(mpctx, "audio draining\n");
        mp_wakeup_core(mpctx);
    }

    if (mpctx->audio_status == STATUS_DRAINING) {
        // Wait until the AO has played all queued data. In the gapless case,
        // we trigger EOF immediately, and let it play asynchronously.
        if (!ao_c->ao || (!ao_is_playing(ao_c->ao) ||
                          (opts->gapless_audio && !ao_untimed(ao_c->ao))))
        {
            MP_VERBOSE(mpctx, "audio EOF reached\n");
            mpctx->audio_status = STATUS_EOF;
            mp_wakeup_core(mpctx);
        }
    }

    if (mpctx->restart_complete)
        audio_start_ao(mpctx); // in case it got delayed
}

// Drop data queued for output, or which the AO is currently outputting.
void clear_audio_output_buffers(struct MPContext *mpctx)
{
    if (mpctx->ao)
        ao_reset(mpctx->ao);
}