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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>
#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->device_buffer = ac->aframesize * ac->framecount;
ao->period_size = ao->device_buffer;
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);
}
}
// 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);
}
}
// Note: currently relies on samples aligned to period sizes - will not work
// in the future.
static bool audio_write(struct ao *ao, void **data, int samples)
{
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 (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;
MP_VERBOSE(ao, "padding final frame with silence\n");
}
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);
return true;
}
static void get_state(struct ao *ao, struct mp_pcm_state *state)
{
state->free_samples = ao->device_buffer;
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,
.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
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