mpv/audio/out/ao_lavc.c

/*
 * audio encoding using libavformat
 *
 * Copyright (C) 2011-2012 Rudolf Polzer <divVerent@xonotic.org>
 * NOTE: this file is partially based on ao_pcm.c by Atmosfear
 *
 * 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 <stdio.h>
#include <stdlib.h>
#include <assert.h>
#include <limits.h>

#include <libavutil/common.h>

#include "config.h"
#include "options/options.h"
#include "common/common.h"
#include "audio/format.h"
#include "audio/fmt-conversion.h"
#include "mpv_talloc.h"
#include "ao.h"
#include "internal.h"
#include "common/msg.h"

#include "common/encode_lavc.h"

struct priv {
    struct encoder_context *enc;

    int pcmhack;
    int aframesize;
    int aframecount;
    int64_t savepts;
    int framecount;
    int64_t lastpts;
    int sample_size;
    const void *sample_padding;
    double expected_next_pts;

    AVRational worst_time_base;

    bool shutdown;
};

static void encode(struct ao *ao, double apts, void **data);

static bool supports_format(const AVCodec *codec, int format)
{
    for (const enum AVSampleFormat *sampleformat = codec->sample_fmts;
         sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE;
         sampleformat++)
    {
        if (af_from_avformat(*sampleformat) == format)
            return true;
    }
    return false;
}

static void select_format(struct ao *ao, const AVCodec *codec)
{
    int formats[AF_FORMAT_COUNT + 1];
    af_get_best_sample_formats(ao->format, formats);

    for (int n = 0; formats[n]; n++) {
        if (supports_format(codec, formats[n])) {
            ao->format = formats[n];
            break;
        }
    }
}

static void on_ready(void *ptr)
{
    struct ao *ao = ptr;
    struct priv *ac = ao->priv;

    ac->worst_time_base = encoder_get_mux_timebase_unlocked(ac->enc);

    ao_add_events(ao, AO_EVENT_INITIAL_UNBLOCK);
}

// open & setup audio device
static int init(struct ao *ao)
{
    struct priv *ac = ao->priv;

    ac->enc = encoder_context_alloc(ao->encode_lavc_ctx, STREAM_AUDIO, ao->log);
    if (!ac->enc)
        return -1;
    talloc_steal(ac, ac->enc);

    AVCodecContext *encoder = ac->enc->encoder;
    const AVCodec *codec = encoder->codec;

    int samplerate = af_select_best_samplerate(ao->samplerate,
                                               codec->supported_samplerates);
    if (samplerate > 0)
        ao->samplerate = samplerate;

    encoder->time_base.num = 1;
    encoder->time_base.den = ao->samplerate;

    encoder->sample_rate = ao->samplerate;

    struct mp_chmap_sel sel = {0};
    mp_chmap_sel_add_any(&sel);
    if (!ao_chmap_sel_adjust2(ao, &sel, &ao->channels, false))
        goto fail;
    mp_chmap_reorder_to_lavc(&ao->channels);
    encoder->channels = ao->channels.num;
    encoder->channel_layout = mp_chmap_to_lavc(&ao->channels);

    encoder->sample_fmt = AV_SAMPLE_FMT_NONE;

    select_format(ao, codec);

    ac->sample_size = af_fmt_to_bytes(ao->format);
    encoder->sample_fmt = af_to_avformat(ao->format);
    encoder->bits_per_raw_sample = ac->sample_size * 8;

    if (!encoder_init_codec_and_muxer(ac->enc, on_ready, ao))
        goto fail;

    ac->pcmhack = 0;
    if (encoder->frame_size <= 1)
        ac->pcmhack = av_get_bits_per_sample(encoder->codec_id) / 8;

    if (ac->pcmhack) {
        ac->aframesize = 16384; // "enough"
    } else {
        ac->aframesize = encoder->frame_size;
    }

    // enough frames for at least 0.25 seconds
    ac->framecount = ceil(ao->samplerate * 0.25 / ac->aframesize);
    // but at least one!
    ac->framecount = MPMAX(ac->framecount, 1);

    ac->savepts = AV_NOPTS_VALUE;
    ac->lastpts = AV_NOPTS_VALUE;

    ao->untimed = true;

    ao->period_size = ac->aframesize * ac->framecount;

    if (ao->channels.num > AV_NUM_DATA_POINTERS)
        goto fail;

    return 0;

fail:
    pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
    ac->shutdown = true;
    return -1;
}

// close audio device
static void uninit(struct ao *ao)
{
    struct priv *ac = ao->priv;
    struct encode_lavc_context *ectx = ao->encode_lavc_ctx;

    if (!ac->shutdown) {
        double outpts = ac->expected_next_pts;

        pthread_mutex_lock(&ectx->lock);
        if (!ac->enc->options->rawts)
            outpts += ectx->discontinuity_pts_offset;
        pthread_mutex_unlock(&ectx->lock);

        outpts += encoder_get_offset(ac->enc);
        encode(ao, outpts, NULL);
    }
}

// return: how many samples can be played without blocking
static int get_space(struct ao *ao)
{
    struct priv *ac = ao->priv;

    return ac->aframesize * ac->framecount;
}

// must get exactly ac->aframesize amount of data
static void encode(struct ao *ao, double apts, void **data)
{
    struct priv *ac = ao->priv;
    struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
    AVCodecContext *encoder = ac->enc->encoder;
    double realapts = ac->aframecount * (double) ac->aframesize /
                      ao->samplerate;

    ac->aframecount++;

    pthread_mutex_lock(&ectx->lock);
    if (data)
        ectx->audio_pts_offset = realapts - apts;
    pthread_mutex_unlock(&ectx->lock);

    if(data) {
        AVFrame *frame = av_frame_alloc();
        frame->format = af_to_avformat(ao->format);
        frame->nb_samples = ac->aframesize;

        size_t num_planes = af_fmt_is_planar(ao->format) ? ao->channels.num : 1;
        assert(num_planes <= AV_NUM_DATA_POINTERS);
        for (int n = 0; n < num_planes; n++)
            frame->extended_data[n] = data[n];

        frame->linesize[0] = frame->nb_samples * ao->sstride;

        frame->pts = rint(apts * av_q2d(av_inv_q(encoder->time_base)));

        int64_t frame_pts = av_rescale_q(frame->pts, encoder->time_base,
                                         ac->worst_time_base);
        while (ac->lastpts != AV_NOPTS_VALUE && frame_pts <= ac->lastpts) {
            // whatever the fuck this code does?
            MP_WARN(ao, "audio frame pts went backwards (%d <- %d), autofixed\n",
                    (int)frame->pts, (int)ac->lastpts);
            frame_pts = ac->lastpts + 1;
            ac->lastpts = frame_pts;
            frame->pts = av_rescale_q(frame_pts, ac->worst_time_base,
                                      encoder->time_base);
            frame_pts = av_rescale_q(frame->pts, encoder->time_base,
                                     ac->worst_time_base);
        }
        ac->lastpts = frame_pts;

        frame->quality = encoder->global_quality;
        encoder_encode(ac->enc, frame);
        av_frame_free(&frame);
    } else {
        encoder_encode(ac->enc, NULL);
    }
}

// this should round samples down to frame sizes
// return: number of samples played
static int play(struct ao *ao, void **data, int samples, int flags)
{
    struct priv *ac = ao->priv;
    struct encoder_context *enc = ac->enc;
    struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
    int bufpos = 0;
    double nextpts;
    int orig_samples = samples;

    // for ectx PTS fields
    pthread_mutex_lock(&ectx->lock);

    double pts = ectx->last_audio_in_pts;
    pts += ectx->samples_since_last_pts / (double)ao->samplerate;

    size_t num_planes = af_fmt_is_planar(ao->format) ? ao->channels.num : 1;

    void *tempdata = NULL;
    void *padded[MP_NUM_CHANNELS];

    if ((flags & AOPLAY_FINAL_CHUNK) && (samples % ac->aframesize)) {
       tempdata = talloc_new(NULL);
       size_t bytelen = samples * ao->sstride;
       size_t extralen = (ac->aframesize - 1) * ao->sstride;
       for (int n = 0; n < num_planes; n++) {
           padded[n] = talloc_size(tempdata, bytelen + extralen);
           memcpy(padded[n], data[n], bytelen);
           af_fill_silence((char *)padded[n] + bytelen, extralen, ao->format);
       }
       data = padded;
       samples = (bytelen + extralen) / ao->sstride;
    }

    double outpts = pts;
    if (!enc->options->rawts) {
        // Fix and apply the discontinuity pts offset.
        nextpts = pts;
        if (ectx->discontinuity_pts_offset == MP_NOPTS_VALUE) {
            ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts;
        } else if (fabs(nextpts + ectx->discontinuity_pts_offset -
                        ectx->next_in_pts) > 30)
        {
            MP_WARN(ao, "detected an unexpected discontinuity (pts jumped by "
                    "%f seconds)\n",
                    nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts);
            ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts;
        }

        outpts = pts + ectx->discontinuity_pts_offset;
    }

    pthread_mutex_unlock(&ectx->lock);

    // Shift pts by the pts offset first.
    outpts += encoder_get_offset(enc);

    while (samples - bufpos >= ac->aframesize) {
        void *start[MP_NUM_CHANNELS] = {0};
        for (int n = 0; n < num_planes; n++)
            start[n] = (char *)data[n] + bufpos * ao->sstride;
        encode(ao, outpts + bufpos / (double) ao->samplerate, start);
        bufpos += ac->aframesize;
    }

    // Calculate expected pts of next audio frame (input side).
    ac->expected_next_pts = pts + bufpos / (double) ao->samplerate;

    pthread_mutex_lock(&ectx->lock);

    // Set next allowed input pts value (input side).
    if (!enc->options->rawts) {
        nextpts = ac->expected_next_pts + ectx->discontinuity_pts_offset;
        if (nextpts > ectx->next_in_pts)
            ectx->next_in_pts = nextpts;
    }

    talloc_free(tempdata);

    int taken = MPMIN(bufpos, orig_samples);
    ectx->samples_since_last_pts += taken;

    pthread_mutex_unlock(&ectx->lock);

    if (flags & AOPLAY_FINAL_CHUNK) {
        if (bufpos < orig_samples)
            MP_ERR(ao, "did not write enough data at the end\n");
    } else {
        if (bufpos > orig_samples)
            MP_ERR(ao, "audio buffer overflow (should never happen)\n");
    }

    return taken;
}

static void drain(struct ao *ao)
{
    // pretend we support it, so generic code doesn't force a wait
}

const struct ao_driver audio_out_lavc = {
    .encode = true,
    .description = "audio encoding using libavcodec",
    .name      = "lavc",
    .initially_blocked = true,
    .reports_underruns = true, // not a thing
    .priv_size = sizeof(struct priv),
    .init      = init,
    .uninit    = uninit,
    .get_space = get_space,
    .play      = play,
    .drain     = drain,
};

// vim: sw=4 ts=4 et tw=80