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mpv/audio/decode/ad_lavc.c
wm4 5fd8a1e04c audio: make decoders output refcounted frames
This rewrites the audio decode loop to some degree. Audio filters don't
do refcounted frames yet, so af.c contains a hacky "emulation".

Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of
estimating how much audio we need to filter, we always filter full
frames. Maybe this should be adjusted later: in case filtering increases
the volume of the audio data, we should try not to buffer too much
filter output by reducing the input that is fed at once.

For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it
doesn't seem worth the trouble.
2014-11-10 22:02:05 +01:00

294 lines
8.4 KiB
C

/*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <stdbool.h>
#include <assert.h>
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include <libavutil/common.h>
#include <libavutil/intreadwrite.h>
#include "talloc.h"
#include "config.h"
#include "common/av_common.h"
#include "common/codecs.h"
#include "common/msg.h"
#include "options/options.h"
#include "ad.h"
#include "audio/fmt-conversion.h"
struct priv {
AVCodecContext *avctx;
AVFrame *avframe;
struct mp_audio frame;
bool force_channel_map;
struct demux_packet *packet;
uint32_t skip_samples;
};
static void uninit(struct dec_audio *da);
#define OPT_BASE_STRUCT struct ad_lavc_params
struct ad_lavc_params {
float ac3drc;
int downmix;
int threads;
char **avopts;
};
const struct m_sub_options ad_lavc_conf = {
.opts = (const m_option_t[]) {
OPT_FLOATRANGE("ac3drc", ac3drc, 0, 0, 2),
OPT_FLAG("downmix", downmix, 0),
OPT_INTRANGE("threads", threads, 0, 1, 16),
OPT_KEYVALUELIST("o", avopts, 0),
{0}
},
.size = sizeof(struct ad_lavc_params),
.defaults = &(const struct ad_lavc_params){
.ac3drc = 1.,
.downmix = 1,
.threads = 1,
},
};
static int init(struct dec_audio *da, const char *decoder)
{
struct MPOpts *mpopts = da->opts;
struct ad_lavc_params *opts = mpopts->ad_lavc_params;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
struct sh_stream *sh = da->header;
struct sh_audio *sh_audio = sh->audio;
struct priv *ctx = talloc_zero(NULL, struct priv);
da->priv = ctx;
ctx->force_channel_map = sh_audio->force_channels;
lavc_codec = avcodec_find_decoder_by_name(decoder);
if (!lavc_codec) {
MP_ERR(da, "Cannot find codec '%s' in libavcodec...\n", decoder);
uninit(da);
return 0;
}
lavc_context = avcodec_alloc_context3(lavc_codec);
ctx->avctx = lavc_context;
ctx->avframe = av_frame_alloc();
lavc_context->refcounted_frames = 1;
lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id;
if (opts->downmix) {
lavc_context->request_channel_layout =
mp_chmap_to_lavc(&mpopts->audio_output_channels);
}
// Always try to set - option only exists for AC3 at the moment
av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc,
AV_OPT_SEARCH_CHILDREN);
#if HAVE_AVFRAME_SKIP_SAMPLES
// Let decoder add AV_FRAME_DATA_SKIP_SAMPLES.
av_opt_set(lavc_context, "flags2", "+skip_manual", AV_OPT_SEARCH_CHILDREN);
#endif
mp_set_avopts(da->log, lavc_context, opts->avopts);
lavc_context->codec_tag = sh->format;
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->bitrate;
lavc_context->block_align = sh_audio->block_align;
lavc_context->bits_per_coded_sample = sh_audio->bits_per_coded_sample;
lavc_context->channels = sh_audio->channels.num;
if (!mp_chmap_is_unknown(&sh_audio->channels))
lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);
// demux_mkv
if (sh_audio->codecdata_len && sh_audio->codecdata &&
!lavc_context->extradata) {
mp_lavc_set_extradata(lavc_context, sh_audio->codecdata,
sh_audio->codecdata_len);
}
if (sh->lav_headers)
mp_copy_lav_codec_headers(lavc_context, sh->lav_headers);
mp_set_avcodec_threads(lavc_context, opts->threads);
/* open it */
if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
MP_ERR(da, "Could not open codec.\n");
uninit(da);
return 0;
}
if (lavc_context->bit_rate != 0)
da->bitrate = lavc_context->bit_rate;
return 1;
}
static void uninit(struct dec_audio *da)
{
struct priv *ctx = da->priv;
if (!ctx)
return;
AVCodecContext *lavc_context = ctx->avctx;
if (lavc_context) {
if (avcodec_close(lavc_context) < 0)
MP_ERR(da, "Could not close codec.\n");
av_freep(&lavc_context->extradata);
av_freep(&lavc_context);
}
av_frame_free(&ctx->avframe);
}
static int control(struct dec_audio *da, int cmd, void *arg)
{
struct priv *ctx = da->priv;
switch (cmd) {
case ADCTRL_RESET:
avcodec_flush_buffers(ctx->avctx);
talloc_free(ctx->packet);
ctx->packet = NULL;
ctx->skip_samples = 0;
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int decode_packet(struct dec_audio *da, struct mp_audio **out)
{
struct priv *priv = da->priv;
AVCodecContext *avctx = priv->avctx;
struct demux_packet *mpkt = priv->packet;
if (!mpkt) {
if (demux_read_packet_async(da->header, &mpkt) == 0)
return AD_WAIT;
}
priv->packet = talloc_steal(priv, mpkt);
int in_len = mpkt ? mpkt->len : 0;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, NULL);
// If we don't have a PTS yet, use the first packet PTS we can get.
if (da->pts == MP_NOPTS_VALUE && mpkt && mpkt->pts != MP_NOPTS_VALUE) {
da->pts = mpkt->pts;
da->pts_offset = 0;
}
int got_frame = 0;
av_frame_unref(priv->avframe);
int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
if (mpkt) {
// At least "shorten" decodes sub-frames, instead of the whole packet.
// At least "mpc8" can return 0 and wants the packet again next time.
if (ret >= 0) {
ret = FFMIN(ret, mpkt->len); // sanity check against decoder overreads
mpkt->buffer += ret;
mpkt->len -= ret;
mpkt->pts = MP_NOPTS_VALUE; // don't reset PTS next time
}
if (mpkt->len == 0 || ret < 0) {
talloc_free(mpkt);
priv->packet = NULL;
}
// LATM may need many packets to find mux info
if (ret == AVERROR(EAGAIN))
return AD_OK;
}
if (ret < 0) {
MP_ERR(da, "Error decoding audio.\n");
return AD_ERR;
}
if (!got_frame)
return mpkt ? AD_OK : AD_EOF;
double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, NULL);
if (out_pts != MP_NOPTS_VALUE) {
da->pts = out_pts;
da->pts_offset = 0;
}
struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe);
if (!mpframe)
return AD_ERR;
struct mp_chmap lavc_chmap = mpframe->channels;
if (lavc_chmap.num != avctx->channels)
mp_chmap_from_channels(&lavc_chmap, avctx->channels);
if (priv->force_channel_map) {
struct sh_audio *sh_audio = da->header->audio;
if (lavc_chmap.num == sh_audio->channels.num)
lavc_chmap = sh_audio->channels;
}
mp_audio_set_channels(mpframe, &lavc_chmap);
#if HAVE_AVFRAME_SKIP_SAMPLES
AVFrameSideData *sd =
av_frame_get_side_data(priv->avframe, AV_FRAME_DATA_SKIP_SAMPLES);
if (sd && sd->size >= 10) {
char *d = sd->data;
priv->skip_samples += AV_RL32(d + 0);
uint32_t pad = AV_RL32(d + 4);
uint32_t skip = MPMIN(priv->skip_samples, mpframe->samples);
if (skip) {
mp_audio_skip_samples(mpframe, skip);
da->pts_offset += skip;
priv->skip_samples -= skip;
}
if (pad <= mpframe->samples)
mpframe->samples -= pad;
}
#endif
*out = mpframe;
av_frame_unref(priv->avframe);
MP_DBG(da, "Decoded %d -> %d samples\n", in_len, mpframe->samples);
return 0;
}
static void add_decoders(struct mp_decoder_list *list)
{
mp_add_lavc_decoders(list, AVMEDIA_TYPE_AUDIO);
}
const struct ad_functions ad_lavc = {
.name = "lavc",
.add_decoders = add_decoders,
.init = init,
.uninit = uninit,
.control = control,
.decode_packet = decode_packet,
};