mirror of
https://github.com/mpv-player/mpv
synced 2025-01-05 03:06:28 +01:00
a47ef5fed6
Change ad_pcm to copy input packet data into an internal buffer instead of relying on the packet still existing at the next decode() call. The extra memcpy could be avoided by improving the demuxer API a bit but I don't feel like implementing that now. Also add a ADCTRL_RESYNC_STREAM handler to drop buffered data from previous stream position.
208 lines
6.9 KiB
C
208 lines
6.9 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 <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <stdbool.h>
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#include "talloc.h"
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#include "config.h"
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#include "ad_internal.h"
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#include "libaf/af_format.h"
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#include "libaf/reorder_ch.h"
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static const ad_info_t info = {
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"Uncompressed PCM audio decoder",
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"pcm",
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"Nick Kurshev",
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"A'rpi",
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""
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};
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struct ad_pcm_context {
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unsigned char *buffer;
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int buffer_pos;
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int buffer_len;
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int buffer_size;
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};
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LIBAD_EXTERN(pcm)
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static int init(sh_audio_t * sh_audio)
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{
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WAVEFORMATEX *h = sh_audio->wf;
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if (!h)
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return 0;
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sh_audio->i_bps = h->nAvgBytesPerSec;
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sh_audio->channels = h->nChannels;
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sh_audio->samplerate = h->nSamplesPerSec;
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sh_audio->samplesize = (h->wBitsPerSample + 7) / 8;
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sh_audio->sample_format = AF_FORMAT_S16_LE; // default
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switch (sh_audio->format) { /* hardware formats: */
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case 0x0:
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case 0x1: // Microsoft PCM
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case 0xfffe: // Extended
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switch (sh_audio->samplesize) {
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case 1: sh_audio->sample_format = AF_FORMAT_U8; break;
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case 2: sh_audio->sample_format = AF_FORMAT_S16_LE; break;
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case 3: sh_audio->sample_format = AF_FORMAT_S24_LE; break;
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case 4: sh_audio->sample_format = AF_FORMAT_S32_LE; break;
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}
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break;
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case 0x3: // IEEE float
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sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
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break;
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case 0x6: sh_audio->sample_format = AF_FORMAT_A_LAW; break;
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case 0x7: sh_audio->sample_format = AF_FORMAT_MU_LAW; break;
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case 0x11: sh_audio->sample_format = AF_FORMAT_IMA_ADPCM; break;
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case 0x50: sh_audio->sample_format = AF_FORMAT_MPEG2; break;
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/* case 0x2000: sh_audio->sample_format=AFMT_AC3; */
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case 0x20776172: // 'raw '
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sh_audio->sample_format = AF_FORMAT_S16_BE;
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if (sh_audio->samplesize == 1)
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sh_audio->sample_format = AF_FORMAT_U8;
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break;
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case 0x736F7774: // 'twos'
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sh_audio->sample_format = AF_FORMAT_S16_BE;
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// intended fall-through
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case 0x74776F73: // 'sowt'
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if (sh_audio->samplesize == 1)
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sh_audio->sample_format = AF_FORMAT_S8;
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break;
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case 0x32336c66: // 'fl32', bigendian float32
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sh_audio->sample_format = AF_FORMAT_FLOAT_BE;
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sh_audio->samplesize = 4;
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break;
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case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC
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sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
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sh_audio->samplesize = 4;
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break;
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/* case 0x34366c66: // 'fl64', bigendian float64
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sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
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sh_audio->samplesize=8;
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break;
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case 0x666c3634: // '46lf', little endian float64, MPlayer internal fourCC
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sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
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sh_audio->samplesize=8;
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break;*/
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case 0x34326e69: // 'in24', bigendian int24
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sh_audio->sample_format = AF_FORMAT_S24_BE;
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sh_audio->samplesize = 3;
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break;
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case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC
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sh_audio->sample_format = AF_FORMAT_S24_LE;
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sh_audio->samplesize = 3;
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break;
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case 0x32336e69: // 'in32', bigendian int32
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sh_audio->sample_format = AF_FORMAT_S32_BE;
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sh_audio->samplesize = 4;
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break;
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case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC
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sh_audio->sample_format = AF_FORMAT_S32_LE;
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sh_audio->samplesize = 4;
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break;
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default:
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if (sh_audio->samplesize != 2)
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sh_audio->sample_format = AF_FORMAT_U8;
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}
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if (!sh_audio->samplesize) // this would cause MPlayer to hang later
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sh_audio->samplesize = 2;
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sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
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return 1;
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}
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static int preinit(sh_audio_t * sh)
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{
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sh->audio_out_minsize = 2048;
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return 1;
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}
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static void uninit(sh_audio_t * sh)
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{
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talloc_free(sh->context);
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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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struct ad_pcm_context *ctx = sh->context;
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int skip;
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switch (cmd) {
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case ADCTRL_RESYNC_STREAM:
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ctx->buffer_len = 0;
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return true;
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case ADCTRL_SKIP_FRAME:
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skip = sh->i_bps / 16;
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skip = skip & (~3);
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demux_read_data(sh->ds, NULL, skip);
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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 int decode_audio(sh_audio_t * sh_audio, unsigned char *buf, int minlen,
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int maxlen)
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{
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int len = sh_audio->channels * sh_audio->samplesize;
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minlen = (minlen + len - 1) / len * len;
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if (minlen > maxlen)
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// if someone needs hundreds of channels adjust audio_out_minsize
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// based on channels in preinit()
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return -1;
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len = 0;
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struct ad_pcm_context *ctx = sh_audio->context;
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while (len < minlen) {
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if (ctx->buffer_len - ctx->buffer_pos <= 0) {
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double pts;
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unsigned char *ptr;
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int plen = ds_get_packet_pts(sh_audio->ds, &ptr, &pts);
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if (plen < 0)
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break;
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if (ctx->buffer_size < plen) {
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talloc_free(ctx->buffer);
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ctx->buffer = talloc_size(ctx, plen);
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ctx->buffer_size = plen;
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}
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memcpy(ctx->buffer, ptr, plen);
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ctx->buffer_len = plen;
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ctx->buffer_pos = 0;
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if (pts != MP_NOPTS_VALUE) {
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sh_audio->pts = pts;
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sh_audio->pts_bytes = 0;
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}
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}
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int from_stored = ctx->buffer_len - ctx->buffer_pos;
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if (from_stored > minlen - len)
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from_stored = minlen - len;
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memcpy(buf + len, ctx->buffer + ctx->buffer_pos, from_stored);
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ctx->buffer_pos += from_stored;
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sh_audio->pts_bytes += from_stored;
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len += from_stored;
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}
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if (len == 0)
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len = -1; // The loop above only exits at error/EOF
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if (len > 0 && sh_audio->channels >= 5) {
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reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
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AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
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sh_audio->channels, len / sh_audio->samplesize,
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sh_audio->samplesize);
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}
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return len;
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}
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