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mirror of https://github.com/mpv-player/mpv synced 2024-11-18 21:16:10 +01:00
mpv/audio/decode/ad_spdif.c
wm4 4d016a92c8 core: redo how codecs are mapped, remove codecs.conf
Use codec names instead of FourCCs to identify codecs. Rewrite how
codecs are selected and initialized. Now each decoder exports a list
of decoders (and the codec it supports) via add_decoders(). The order
matters, and the first decoder for a given decoder is preferred over
the other decoders. E.g. all ad_mpg123 decoders are preferred over
ad_lavc, because it comes first in the mpcodecs_ad_drivers array.
Likewise, decoders within ad_lavc that are enumerated first by
libavcodec (using av_codec_next()) are preferred. (This is actually
critical to select h264 software decoding by default instead of vdpau.
libavcodec and ffmpeg/avconv use the same method to select decoders by
default, so we hope this is sane.)

The codec names follow libavcodec's codec names as defined by
AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders
have names different from the canonical codec name. The AVCodecDescriptor
API is relatively new, so we need a compatibility layer for older
libavcodec versions for codec names that are referenced internally,
and which are different from the decoder name. (Add a configure check
for that, because checking versions is getting way too messy.)

demux/codec_tags.c is generated from the former codecs.conf (minus
"special" decoders like vdpau, and excluding the mappings that are the
same as the mappings libavformat's exported RIFF tables). It contains
all the mappings from FourCCs to codec name. This is needed for
demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the
codec as determined by libavformat, while the other demuxers have to do
this on their own, using the mp_set_audio/video_codec_from_tag()
functions. Note that the sh_audio/video->format members don't uniquely
identify the codec anymore, and sh->codec takes over this role.

Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which
provide cover the functionality of the removed switched.

Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure
container/video combinations (e.g. the sample Film_200_zygo_pro.mov)
are played flipped. ffplay/avplay doesn't handle this properly either,
so we don't care and blame ffmeg/libav instead.
2013-02-10 17:25:56 +01:00

309 lines
9.6 KiB
C

/*
* This file is part of MPlayer.
*
* Copyright (C) 2012 Naoya OYAMA
*
* 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 <string.h>
#include <libavformat/avformat.h>
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include "config.h"
#include "core/mp_msg.h"
#include "core/av_common.h"
#include "ad_internal.h"
LIBAD_EXTERN(spdif)
#define FILENAME_SPDIFENC "spdif"
#define OUTBUF_SIZE 65536
struct spdifContext {
AVFormatContext *lavf_ctx;
int iec61937_packet_size;
int out_buffer_len;
int out_buffer_size;
uint8_t *out_buffer;
uint8_t pb_buffer[OUTBUF_SIZE];
};
static int read_packet(void *p, uint8_t *buf, int buf_size)
{
// spdifenc does not use read callback.
return 0;
}
static int write_packet(void *p, uint8_t *buf, int buf_size)
{
int len;
struct spdifContext *ctx = p;
len = FFMIN(buf_size, ctx->out_buffer_size -ctx->out_buffer_len);
memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, len);
ctx->out_buffer_len += len;
return len;
}
static int64_t seek(void *p, int64_t offset, int whence)
{
// spdifenc does not use seek callback.
return 0;
}
static int preinit(sh_audio_t *sh)
{
sh->samplesize = 2;
return 1;
}
static int codecs[] = {
CODEC_ID_AAC,
CODEC_ID_AC3,
CODEC_ID_DTS,
CODEC_ID_EAC3,
CODEC_ID_MP3,
CODEC_ID_TRUEHD,
CODEC_ID_NONE
};
static int init(sh_audio_t *sh, const char *decoder)
{
int x, in_size, srate, bps, *dtshd_rate;
unsigned char *start;
double pts;
AVFormatContext *lavf_ctx = NULL;
AVStream *stream = NULL;
const AVOption *opt = NULL;
struct spdifContext *spdif_ctx = NULL;
spdif_ctx = av_mallocz(sizeof(*spdif_ctx));
if (!spdif_ctx)
goto fail;
spdif_ctx->lavf_ctx = avformat_alloc_context();
if (!spdif_ctx->lavf_ctx)
goto fail;
sh->context = spdif_ctx;
lavf_ctx = spdif_ctx->lavf_ctx;
lavf_ctx->oformat = av_guess_format(FILENAME_SPDIFENC, NULL, NULL);
if (!lavf_ctx->oformat)
goto fail;
lavf_ctx->priv_data = av_mallocz(lavf_ctx->oformat->priv_data_size);
if (!lavf_ctx->priv_data)
goto fail;
lavf_ctx->pb = avio_alloc_context(spdif_ctx->pb_buffer, OUTBUF_SIZE, 1, spdif_ctx,
read_packet, write_packet, seek);
if (!lavf_ctx->pb)
goto fail;
stream = avformat_new_stream(lavf_ctx, 0);
if (!stream)
goto fail;
lavf_ctx->duration = AV_NOPTS_VALUE;
lavf_ctx->start_time = AV_NOPTS_VALUE;
lavf_ctx->streams[0]->codec->codec_id = mp_codec_to_av_codec_id(decoder);
lavf_ctx->raw_packet_buffer_remaining_size = RAW_PACKET_BUFFER_SIZE;
if (AVERROR_PATCHWELCOME == lavf_ctx->oformat->write_header(lavf_ctx)) {
mp_msg(MSGT_DECAUDIO,MSGL_INFO,
"This codec is not supported by spdifenc.\n");
goto fail;
}
// get sample_rate & bitrate from parser
x = ds_get_packet_pts(sh->ds, &start, &pts);
in_size = x;
if (x <= 0) {
pts = MP_NOPTS_VALUE;
x = 0;
}
ds_parse(sh->ds, &start, &x, pts, 0);
srate = 48000; //fake value
bps = 768000/8; //fake value
if (x && sh->avctx) { // we have parser and large enough buffer
if (sh->avctx->sample_rate < 44100) {
mp_msg(MSGT_DECAUDIO,MSGL_INFO,
"This stream sample_rate[%d Hz] may be broken. "
"Force reset 48000Hz.\n",
sh->avctx->sample_rate);
srate = 48000; //fake value
} else
srate = sh->avctx->sample_rate;
bps = sh->avctx->bit_rate/8;
}
sh->ds->buffer_pos -= in_size;
switch (lavf_ctx->streams[0]->codec->codec_id) {
case CODEC_ID_AAC:
spdif_ctx->iec61937_packet_size = 16384;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
sh->sample_format = AF_FORMAT_AC3_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case CODEC_ID_DTS: // FORCE USE DTS-HD
opt = av_opt_find(&lavf_ctx->oformat->priv_class,
"dtshd_rate", NULL, 0, 0);
if (!opt)
goto fail;
dtshd_rate = (int*)(((uint8_t*)lavf_ctx->priv_data) +
opt->offset);
*dtshd_rate = 192000*4;
spdif_ctx->iec61937_packet_size = 32768;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000; // DTS core require 48000
sh->channels = 2*4;
sh->i_bps = bps;
break;
case CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
sh->channels = 2;
sh->i_bps = bps;
break;
case CODEC_ID_MP3:
spdif_ctx->iec61937_packet_size = 4608;
sh->sample_format = AF_FORMAT_MPEG2;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
sh->channels = 8;
sh->i_bps = bps;
break;
default:
break;
}
return 1;
fail:
uninit(sh);
return 0;
}
static int decode_audio(sh_audio_t *sh, unsigned char *buf,
int minlen, int maxlen)
{
struct spdifContext *spdif_ctx = sh->context;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
AVPacket pkt;
double pts;
int ret, in_size, consumed, x;
unsigned char *start = NULL;
consumed = spdif_ctx->out_buffer_len = 0;
spdif_ctx->out_buffer_size = maxlen;
spdif_ctx->out_buffer = buf;
while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen
&& spdif_ctx->out_buffer_len < minlen) {
if (sh->ds->eof)
break;
x = ds_get_packet_pts(sh->ds, &start, &pts);
if (x <= 0) {
x = 0;
ds_parse(sh->ds, &start, &x, MP_NOPTS_VALUE, 0);
if (x == 0)
continue; // END_NOT_FOUND
in_size = x;
} else {
in_size = x;
consumed = ds_parse(sh->ds, &start, &x, pts, 0);
if (x == 0) {
mp_msg(MSGT_DECAUDIO,MSGL_V,
"start[%p] in_size[%d] consumed[%d] x[%d].\n",
start, in_size, consumed, x);
continue; // END_NOT_FOUND
}
sh->ds->buffer_pos -= in_size - consumed;
}
av_init_packet(&pkt);
pkt.data = start;
pkt.size = x;
mp_msg(MSGT_DECAUDIO,MSGL_V,
"start[%p] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n",
start, pkt.size, in_size, consumed, x);
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
ret = lavf_ctx->oformat->write_packet(lavf_ctx, &pkt);
if (ret < 0)
break;
}
sh->pts_bytes += spdif_ctx->out_buffer_len;
return spdif_ctx->out_buffer_len;
}
static int control(sh_audio_t *sh, int cmd, void* arg, ...)
{
unsigned char *start;
double pts;
switch (cmd) {
case ADCTRL_RESYNC_STREAM:
case ADCTRL_SKIP_FRAME:
ds_get_packet_pts(sh->ds, &start, &pts);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static void uninit(sh_audio_t *sh)
{
struct spdifContext *spdif_ctx = sh->context;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
if (lavf_ctx) {
if (lavf_ctx->oformat)
lavf_ctx->oformat->write_trailer(lavf_ctx);
av_freep(&lavf_ctx->pb);
if (lavf_ctx->streams) {
av_freep(&lavf_ctx->streams[0]->codec);
av_freep(&lavf_ctx->streams[0]->info);
av_freep(&lavf_ctx->streams[0]);
}
av_freep(&lavf_ctx->streams);
av_freep(&lavf_ctx->priv_data);
}
av_freep(&lavf_ctx);
av_freep(&spdif_ctx);
}
static void add_decoders(struct mp_decoder_list *list)
{
for (int n = 0; codecs[n] != CODEC_ID_NONE; n++) {
const char *format = mp_codec_from_av_codec_id(codecs[n]);
if (format) {
mp_add_decoder(list, "spdif", format, format,
"libavformat/spdifenc audio pass-through decoder");
}
}
}