mirror of https://github.com/mpv-player/mpv
311 lines
9.7 KiB
C
311 lines
9.7 KiB
C
/*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <string.h>
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#include <libavformat/avformat.h>
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#include <libavcodec/avcodec.h>
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#include <libavutil/opt.h>
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#include "config.h"
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#include "mp_msg.h"
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#include "ad_internal.h"
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static const ad_info_t info = {
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"libavformat/spdifenc audio pass-through decoder.",
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"spdif",
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"Naoya OYAMA",
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"Naoya OYAMA",
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"For ALL hardware decoders"
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};
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LIBAD_EXTERN(spdif)
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#define FILENAME_SPDIFENC "spdif"
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#define OUTBUF_SIZE 65536
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struct spdifContext {
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AVFormatContext *lavf_ctx;
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int iec61937_packet_size;
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int out_buffer_len;
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int out_buffer_size;
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uint8_t *out_buffer;
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uint8_t pb_buffer[OUTBUF_SIZE];
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};
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static int read_packet(void *p, uint8_t *buf, int buf_size)
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{
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// spdifenc does not use read callback.
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return 0;
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}
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static int write_packet(void *p, uint8_t *buf, int buf_size)
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{
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int len;
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struct spdifContext *ctx = p;
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len = FFMIN(buf_size, ctx->out_buffer_size -ctx->out_buffer_len);
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memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, len);
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ctx->out_buffer_len += len;
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return len;
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}
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static int64_t seek(void *p, int64_t offset, int whence)
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{
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// spdifenc does not use seek callback.
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return 0;
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}
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static int preinit(sh_audio_t *sh)
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{
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sh->samplesize = 2;
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return 1;
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}
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static int init(sh_audio_t *sh)
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{
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int i, x, in_size, srate, bps, *dtshd_rate;
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unsigned char *start;
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double pts;
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static const struct {
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const char *name; enum CodecID id;
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} fmt_id_type[] = {
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{ "aac" , CODEC_ID_AAC },
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{ "ac3" , CODEC_ID_AC3 },
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{ "dca" , CODEC_ID_DTS },
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{ "eac3", CODEC_ID_EAC3 },
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{ "mpa" , CODEC_ID_MP3 },
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{ "thd" , CODEC_ID_TRUEHD },
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{ NULL , 0 }
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};
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AVFormatContext *lavf_ctx = NULL;
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AVStream *stream = NULL;
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const AVOption *opt = NULL;
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struct spdifContext *spdif_ctx = NULL;
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spdif_ctx = av_mallocz(sizeof(*spdif_ctx));
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if (!spdif_ctx)
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goto fail;
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spdif_ctx->lavf_ctx = avformat_alloc_context();
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if (!spdif_ctx->lavf_ctx)
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goto fail;
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sh->context = spdif_ctx;
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lavf_ctx = spdif_ctx->lavf_ctx;
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lavf_ctx->oformat = av_guess_format(FILENAME_SPDIFENC, NULL, NULL);
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if (!lavf_ctx->oformat)
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goto fail;
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lavf_ctx->priv_data = av_mallocz(lavf_ctx->oformat->priv_data_size);
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if (!lavf_ctx->priv_data)
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goto fail;
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lavf_ctx->pb = avio_alloc_context(spdif_ctx->pb_buffer, OUTBUF_SIZE, 1, spdif_ctx,
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read_packet, write_packet, seek);
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if (!lavf_ctx->pb)
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goto fail;
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stream = avformat_new_stream(lavf_ctx, 0);
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if (!stream)
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goto fail;
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lavf_ctx->duration = AV_NOPTS_VALUE;
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lavf_ctx->start_time = AV_NOPTS_VALUE;
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for (i = 0; fmt_id_type[i].name; i++) {
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if (!strcmp(sh->codec->dll, fmt_id_type[i].name)) {
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lavf_ctx->streams[0]->codec->codec_id = fmt_id_type[i].id;
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break;
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}
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}
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lavf_ctx->raw_packet_buffer_remaining_size = RAW_PACKET_BUFFER_SIZE;
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if (AVERROR_PATCHWELCOME == lavf_ctx->oformat->write_header(lavf_ctx)) {
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mp_msg(MSGT_DECAUDIO,MSGL_INFO,
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"This codec is not supported by spdifenc.\n");
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goto fail;
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}
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// get sample_rate & bitrate from parser
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bps = srate = 0;
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x = ds_get_packet_pts(sh->ds, &start, &pts);
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in_size = x;
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if (x <= 0) {
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pts = MP_NOPTS_VALUE;
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x = 0;
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}
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ds_parse(sh->ds, &start, &x, pts, 0);
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if (x == 0) { // not enough buffer
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srate = 48000; //fake value
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bps = 768000/8; //fake value
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} else if (sh->avctx) {
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if (sh->avctx->sample_rate < 44100) {
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mp_msg(MSGT_DECAUDIO,MSGL_INFO,
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"This stream sample_rate[%d Hz] may be broken. "
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"Force reset 48000Hz.\n",
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sh->avctx->sample_rate);
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srate = 48000; //fake value
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} else
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srate = sh->avctx->sample_rate;
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bps = sh->avctx->bit_rate/8;
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}
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sh->ds->buffer_pos -= in_size;
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switch (lavf_ctx->streams[0]->codec->codec_id) {
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case CODEC_ID_AAC:
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spdif_ctx->iec61937_packet_size = 16384;
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sh->sample_format = AF_FORMAT_IEC61937_LE;
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sh->samplerate = srate;
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sh->channels = 2;
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sh->i_bps = bps;
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break;
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case CODEC_ID_AC3:
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spdif_ctx->iec61937_packet_size = 6144;
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sh->sample_format = AF_FORMAT_IEC61937_LE;
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sh->samplerate = srate;
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sh->channels = 2;
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sh->i_bps = bps;
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break;
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case CODEC_ID_DTS: // FORCE USE DTS-HD
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opt = av_opt_find(&lavf_ctx->oformat->priv_class,
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"dtshd_rate", NULL, 0, 0);
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if (!opt)
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goto fail;
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dtshd_rate = (int*)(((uint8_t*)lavf_ctx->priv_data) +
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opt->offset);
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*dtshd_rate = 192000*4;
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spdif_ctx->iec61937_packet_size = 32768;
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sh->sample_format = AF_FORMAT_IEC61937_LE;
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sh->samplerate = 192000; // DTS core require 48000
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sh->channels = 2*4;
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sh->i_bps = bps;
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break;
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case CODEC_ID_EAC3:
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spdif_ctx->iec61937_packet_size = 24576;
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sh->sample_format = AF_FORMAT_IEC61937_LE;
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sh->samplerate = 192000;
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sh->channels = 2;
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sh->i_bps = bps;
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break;
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case CODEC_ID_MP3:
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spdif_ctx->iec61937_packet_size = 4608;
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sh->sample_format = AF_FORMAT_MPEG2;
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sh->samplerate = srate;
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sh->channels = 2;
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sh->i_bps = bps;
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break;
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case CODEC_ID_TRUEHD:
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spdif_ctx->iec61937_packet_size = 61440;
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sh->sample_format = AF_FORMAT_IEC61937_LE;
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sh->samplerate = 192000;
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sh->channels = 8;
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sh->i_bps = bps;
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break;
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default:
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break;
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}
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return 1;
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fail:
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uninit(sh);
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return 0;
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}
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static int decode_audio(sh_audio_t *sh, unsigned char *buf,
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int minlen, int maxlen)
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{
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struct spdifContext *spdif_ctx = sh->context;
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AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
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AVPacket pkt;
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double pts;
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int ret, in_size, consumed, x;
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unsigned char *start = NULL;
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consumed = spdif_ctx->out_buffer_len = 0;
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spdif_ctx->out_buffer_size = maxlen;
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spdif_ctx->out_buffer = buf;
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while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen
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&& spdif_ctx->out_buffer_len < minlen) {
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if (sh->ds->eof)
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break;
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x = ds_get_packet_pts(sh->ds, &start, &pts);
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if (x <= 0) {
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x = 0;
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ds_parse(sh->ds, &start, &x, MP_NOPTS_VALUE, 0);
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if (x == 0)
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continue; // END_NOT_FOUND
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in_size = x;
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} else {
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in_size = x;
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consumed = ds_parse(sh->ds, &start, &x, pts, 0);
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if (x == 0) {
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mp_msg(MSGT_DECAUDIO,MSGL_V,
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"start[%p] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n",
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start, pkt.size, in_size, consumed, x);
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continue; // END_NOT_FOUND
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}
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sh->ds->buffer_pos -= in_size - consumed;
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}
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av_init_packet(&pkt);
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pkt.data = start;
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pkt.size = x;
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mp_msg(MSGT_DECAUDIO,MSGL_V,
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"start[%p] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n",
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start, pkt.size, in_size, consumed, x);
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if (pts != MP_NOPTS_VALUE) {
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sh->pts = pts;
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sh->pts_bytes = 0;
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}
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ret = lavf_ctx->oformat->write_packet(lavf_ctx, &pkt);
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if (ret < 0)
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break;
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}
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sh->pts_bytes += spdif_ctx->out_buffer_len;
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return spdif_ctx->out_buffer_len;
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}
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static int control(sh_audio_t *sh, int cmd, void* arg, ...)
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{
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unsigned char *start;
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double pts;
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switch (cmd) {
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case ADCTRL_RESYNC_STREAM:
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case ADCTRL_SKIP_FRAME:
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ds_get_packet_pts(sh->ds, &start, &pts);
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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static void uninit(sh_audio_t *sh)
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{
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struct spdifContext *spdif_ctx = sh->context;
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AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
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if (lavf_ctx) {
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if (lavf_ctx->oformat)
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lavf_ctx->oformat->write_trailer(lavf_ctx);
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av_freep(&lavf_ctx->pb);
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if (lavf_ctx->streams) {
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av_freep(&lavf_ctx->streams[0]->codec);
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av_freep(&lavf_ctx->streams[0]->info);
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av_freep(&lavf_ctx->streams[0]);
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}
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av_freep(&lavf_ctx->streams);
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av_freep(&lavf_ctx->priv_data);
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}
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av_freep(&lavf_ctx);
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av_freep(&spdif_ctx);
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}
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