mpv/audio/out/ao_lavc.c

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/*
* 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>
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#include <assert.h>
#include <limits.h>
#include <libavutil/common.h>
#include <libavutil/samplefmt.h>
#include "config.h"
#include "options/options.h"
#include "common/common.h"
#include "audio/aframe.h"
#include "audio/chmap_avchannel.h"
#include "audio/format.h"
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#include "audio/fmt-conversion.h"
#include "filters/filter_internal.h"
#include "filters/f_utils.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 framecount;
int64_t lastpts;
int sample_size;
double expected_next_pts;
struct mp_filter *filter_root;
struct mp_filter *fix_frame_size;
AVRational worst_time_base;
bool shutdown;
};
static bool write_frame(struct ao *ao, struct mp_frame frame);
static bool supports_format(const AVCodec *codec, int format)
{
for (const enum AVSampleFormat *sampleformat = codec->sample_fmts;
sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE;
sampleformat++)
{
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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);
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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);
mp_chmap_to_av_layout(&encoder->ch_layout, &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->lastpts = AV_NOPTS_VALUE;
ao->untimed = true;
ao->device_buffer = ac->aframesize * ac->framecount;
ac->filter_root = mp_filter_create_root(ao->global);
ac->fix_frame_size = mp_fixed_aframe_size_create(ac->filter_root,
ac->aframesize, true);
MP_HANDLE_OOM(ac->fix_frame_size);
return 0;
fail:
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mp_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;
if (!ac->shutdown) {
if (!write_frame(ao, MP_EOF_FRAME))
MP_WARN(ao, "could not flush last frame\n");
encoder_encode(ac->enc, NULL);
}
talloc_free(ac->filter_root);
}
// must get exactly ac->aframesize amount of data
static void encode(struct ao *ao, struct mp_aframe *af)
{
struct priv *ac = ao->priv;
AVCodecContext *encoder = ac->enc->encoder;
double outpts = mp_aframe_get_pts(af);
AVFrame *frame = mp_aframe_to_avframe(af);
MP_HANDLE_OOM(frame);
frame->pts = rint(outpts * av_q2d(av_inv_q(encoder->time_base)));
int64_t frame_pts = av_rescale_q(frame->pts, encoder->time_base,
ac->worst_time_base);
if (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);
}
static bool write_frame(struct ao *ao, struct mp_frame frame)
{
struct priv *ac = ao->priv;
// Can't push in frame if it doesn't want it output one.
mp_pin_out_request_data(ac->fix_frame_size->pins[1]);
if (!mp_pin_in_write(ac->fix_frame_size->pins[0], frame))
return false; // shouldn't happen™
while (1) {
struct mp_frame fr = mp_pin_out_read(ac->fix_frame_size->pins[1]);
if (!fr.type)
break;
if (fr.type != MP_FRAME_AUDIO)
continue;
struct mp_aframe *af = fr.data;
encode(ao, af);
mp_frame_unref(&fr);
}
return true;
}
static bool audio_write(struct ao *ao, void **data, int samples)
{
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
// See ao_driver.write_frames.
struct mp_aframe *af = mp_aframe_new_ref(*(struct mp_aframe **)data);
double nextpts;
double pts = mp_aframe_get_pts(af);
double outpts = pts;
// for ectx PTS fields
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mp_mutex_lock(&ectx->lock);
if (!ectx->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;
}
// Calculate expected pts of next audio frame (input side).
ac->expected_next_pts = pts + mp_aframe_get_size(af) / (double) ao->samplerate;
// Set next allowed input pts value (input side).
if (!ectx->options->rawts) {
nextpts = ac->expected_next_pts + ectx->discontinuity_pts_offset;
if (nextpts > ectx->next_in_pts)
ectx->next_in_pts = nextpts;
}
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mp_mutex_unlock(&ectx->lock);
mp_aframe_set_pts(af, outpts);
return write_frame(ao, MAKE_FRAME(MP_FRAME_AUDIO, af));
}
static void get_state(struct ao *ao, struct mp_pcm_state *state)
{
state->free_samples = 1;
state->queued_samples = 0;
state->delay = 0;
}
static bool set_pause(struct ao *ao, bool paused)
{
return true; // signal support so common code doesn't write silence
}
static void start(struct ao *ao)
{
// we use data immediately
}
static void reset(struct ao *ao)
{
}
const struct ao_driver audio_out_lavc = {
.encode = true,
.description = "audio encoding using libavcodec",
.name = "lavc",
.initially_blocked = true,
.write_frames = true,
.priv_size = sizeof(struct priv),
.init = init,
.uninit = uninit,
.get_state = get_state,
.set_pause = set_pause,
.write = audio_write,
.start = start,
.reset = reset,
};
// vim: sw=4 ts=4 et tw=80