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mpv/audio/decode/ad_spdif.c
wm4 4b5cee4617 core: use channel map on demuxer level too 
This helps passing the channel layout correctly from decoder to audio
filter chain. (Because that part "reuses" the demuxer level codec
parameters, which is very disgusting.)

Note that ffmpeg stuff already passed the channel layout via
mp_copy_lav_codec_headers(). So other than easier dealing with the
demuxer/decoder parameters mess, there's no real advantage to doing
this.

Make the --channels option accept a channel map. Since simple numbers
map to standard layouts with the given number of channels, this is
downwards compatible. Likewise for demux_rawaudio.
2013-05-12 21:24:55 +02:00

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/*
* 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 "core/options.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[] = {
AV_CODEC_ID_AAC,
AV_CODEC_ID_AC3,
AV_CODEC_ID_DTS,
AV_CODEC_ID_EAC3,
AV_CODEC_ID_MP3,
AV_CODEC_ID_TRUEHD,
AV_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;
int num_channels = 0;
switch (lavf_ctx->streams[0]->codec->codec_id) {
case AV_CODEC_ID_AAC:
spdif_ctx->iec61937_packet_size = 16384;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = srate;
num_channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
sh->sample_format = AF_FORMAT_AC3_LE;
sh->samplerate = srate;
num_channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_DTS:
if(sh->opts->dtshd) {
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
num_channels = 2*4;
sh->i_bps = bps;
} else {
spdif_ctx->iec61937_packet_size = 32768;
sh->sample_format = AF_FORMAT_AC3_LE;
sh->samplerate = srate;
num_channels = 2;
sh->i_bps = bps;
}
break;
case AV_CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
num_channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_MP3:
spdif_ctx->iec61937_packet_size = 4608;
sh->sample_format = AF_FORMAT_MPEG2;
sh->samplerate = srate;
num_channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
num_channels = 8;
sh->i_bps = bps;
break;
default:
break;
}
if (num_channels)
mp_chmap_from_channels(&sh->channels, num_channels);
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] != AV_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");
}
}
}
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