mirror of
https://github.com/mpv-player/mpv
synced 2024-12-16 03:45:23 +00:00
777f684d5c
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@4209 b3059339-0415-0410-9bf9-f77b7e298cf2
1376 lines
46 KiB
C
1376 lines
46 KiB
C
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#define USE_G72X
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//#define USE_LIBAC3
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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 "config.h"
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#include "mp_msg.h"
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#include "help_mp.h"
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extern int verbose; // defined in mplayer.c
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#include "stream.h"
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#include "demuxer.h"
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#include "codec-cfg.h"
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#include "stheader.h"
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#include "dec_audio.h"
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//==========================================================================
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#include "libao2/afmt.h"
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#include "dll_init.h"
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#include "mp3lib/mp3.h"
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#ifdef USE_LIBAC3
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#include "libac3/ac3.h"
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#endif
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#include "liba52/a52.h"
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#include "liba52/mm_accel.h"
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static sample_t * a52_samples;
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static a52_state_t a52_state;
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static uint32_t a52_accel=0;
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static uint32_t a52_flags=0;
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#ifdef USE_G72X
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#include "g72x/g72x.h"
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static G72x_DATA g72x_data;
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#endif
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#include "alaw.h"
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#include "xa/xa_gsm.h"
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#include "ac3-iec958.h"
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#include "adpcm.h"
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#include "cpudetect.h"
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/* used for ac3surround decoder - set using -channels option */
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int audio_output_channels = 2;
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#ifdef USE_FAKE_MONO
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int fakemono=0;
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#endif
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#ifdef USE_DIRECTSHOW
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#include "loader/dshow/DS_AudioDecoder.h"
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static DS_AudioDecoder* ds_adec=NULL;
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#endif
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#ifdef HAVE_OGGVORBIS
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/* XXX is math.h really needed? - atmos */
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#include <math.h>
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#include <vorbis/codec.h>
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typedef struct ov_struct_st {
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ogg_sync_state oy; /* sync and verify incoming physical bitstream */
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ogg_stream_state os; /* take physical pages, weld into a logical
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stream of packets */
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ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
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ogg_packet op; /* one raw packet of data for decode */
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vorbis_info vi; /* struct that stores all the static vorbis bitstream
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settings */
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vorbis_comment vc; /* struct that stores all the bitstream user comments */
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vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
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vorbis_block vb; /* local working space for packet->PCM decode */
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} ov_struct_t;
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#endif
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#ifdef USE_LIBAVCODEC
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#ifdef USE_LIBAVCODEC_SO
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#include <libffmpeg/avcodec.h>
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#else
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#include "libavcodec/avcodec.h"
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#endif
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static AVCodec *lavc_codec=NULL;
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static AVCodecContext lavc_context;
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extern int avcodec_inited;
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#endif
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#ifdef USE_LIBMAD
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#include <mad.h>
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#define MAD_SINGLE_BUFFER_SIZE 8192
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#define MAD_TOTAL_BUFFER_SIZE ((MAD_SINGLE_BUFFER_SIZE)*3)
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static struct mad_stream mad_stream;
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static struct mad_frame mad_frame;
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static struct mad_synth mad_synth;
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static char* mad_in_buffer = 0; /* base pointer of buffer */
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// ensure buffer is filled with some data
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static void mad_prepare_buffer(sh_audio_t* sh_audio, struct mad_stream* ms, int length)
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{
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if(sh_audio->a_in_buffer_len < length) {
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int len = demux_read_data(sh_audio->ds, sh_audio->a_in_buffer+sh_audio->a_in_buffer_len, length-sh_audio->a_in_buffer_len);
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sh_audio->a_in_buffer_len += len;
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// printf("mad_prepare_buffer: read %d bytes\n", len);
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}
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}
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static void mad_postprocess_buffer(sh_audio_t* sh_audio, struct mad_stream* ms)
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{
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/* rotate buffer while possible, in order to reduce the overhead of endless memcpy */
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int delta = (unsigned char*)ms->next_frame - (unsigned char *)sh_audio->a_in_buffer;
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if((unsigned long)(sh_audio->a_in_buffer) - (unsigned long)mad_in_buffer <
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(MAD_TOTAL_BUFFER_SIZE - MAD_SINGLE_BUFFER_SIZE - delta)) {
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sh_audio->a_in_buffer += delta;
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sh_audio->a_in_buffer_len -= delta;
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} else {
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sh_audio->a_in_buffer = mad_in_buffer;
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sh_audio->a_in_buffer_len -= delta;
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memcpy(sh_audio->a_in_buffer, ms->next_frame, sh_audio->a_in_buffer_len);
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}
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}
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static inline
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signed short mad_scale(mad_fixed_t sample)
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{
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/* round */
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sample += (1L << (MAD_F_FRACBITS - 16));
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/* clip */
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if (sample >= MAD_F_ONE)
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sample = MAD_F_ONE - 1;
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else if (sample < -MAD_F_ONE)
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sample = -MAD_F_ONE;
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/* quantize */
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return sample >> (MAD_F_FRACBITS + 1 - 16);
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}
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static void mad_sync(sh_audio_t* sh_audio, struct mad_stream* ms)
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{
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int len;
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#if 1
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int skipped = 0;
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// printf("buffer len: %d\n", sh_audio->a_in_buffer_len);
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while(sh_audio->a_in_buffer_len - skipped)
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{
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len = mp_decode_mp3_header(sh_audio->a_in_buffer+skipped);
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if (len != -1)
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{
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// printf("Frame len=%d\n", len);
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break;
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}
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else
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skipped++;
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}
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if (skipped)
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{
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printf("Audio synced, skipped bytes: %d\n", skipped);
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// ms->skiplen += skipped;
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// printf("skiplen: %d (skipped: %d)\n", ms->skiplen, skipped);
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// if (sh_audio->a_in_buffer_len - skipped < MAD_BUFFER_GUARD)
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// printf("Mad reports: too small buffer\n");
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// mad_stream_buffer(ms, sh_audio->a_in_buffer+skipped, sh_audio->a_in_buffer_len-skipped);
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// mad_prepare_buffer(sh_audio, ms, sh_audio->a_in_buffer_len-skipped);
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/* move frame to the beginning of the buffer and fill up to a_in_buffer_size */
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sh_audio->a_in_buffer_len -= skipped;
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memcpy(sh_audio->a_in_buffer, sh_audio->a_in_buffer+skipped, sh_audio->a_in_buffer_len);
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mad_prepare_buffer(sh_audio, ms, sh_audio->a_in_buffer_size);
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mad_stream_buffer(ms, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len);
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// printf("bufflen: %d\n", sh_audio->a_in_buffer_len);
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// len = mp_decode_mp3_header(sh_audio->a_in_buffer);
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// printf("len: %d\n", len);
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ms->md_len = len;
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}
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#else
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len = mad_stream_sync(&ms);
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if (len == -1)
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{
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printf("Mad sync failed\n");
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}
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#endif
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}
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static void mad_print_error(struct mad_stream *mad_stream)
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{
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printf("error (0x%x): ", mad_stream->error);
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switch(mad_stream->error)
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{
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case MAD_ERROR_BUFLEN: printf("buffer too small"); break;
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case MAD_ERROR_BUFPTR: printf("invalid buffer pointer"); break;
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case MAD_ERROR_NOMEM: printf("not enought memory"); break;
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case MAD_ERROR_LOSTSYNC: printf("lost sync"); break;
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case MAD_ERROR_BADLAYER: printf("bad layer"); break;
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case MAD_ERROR_BADBITRATE: printf("bad bitrate"); break;
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case MAD_ERROR_BADSAMPLERATE: printf("bad samplerate"); break;
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case MAD_ERROR_BADEMPHASIS: printf("bad emphasis"); break;
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case MAD_ERROR_BADCRC: printf("bad crc"); break;
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case MAD_ERROR_BADBITALLOC: printf("forbidden bit alloc val"); break;
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case MAD_ERROR_BADSCALEFACTOR: printf("bad scalefactor index"); break;
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case MAD_ERROR_BADFRAMELEN: printf("bad frame length"); break;
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case MAD_ERROR_BADBIGVALUES: printf("bad bigvalues count"); break;
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case MAD_ERROR_BADBLOCKTYPE: printf("reserved blocktype"); break;
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case MAD_ERROR_BADSCFSI: printf("bad scalefactor selinfo"); break;
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case MAD_ERROR_BADDATAPTR: printf("bad maindatabegin ptr"); break;
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case MAD_ERROR_BADPART3LEN: printf("bad audio data len"); break;
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case MAD_ERROR_BADHUFFTABLE: printf("bad huffman table sel"); break;
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case MAD_ERROR_BADHUFFDATA: printf("huffman data overrun"); break;
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case MAD_ERROR_BADSTEREO: printf("incomp. blocktype for JS"); break;
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default:
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printf("unknown error");
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}
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printf("\n");
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}
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#endif
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static int a52_fillbuff(sh_audio_t *sh_audio){
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int length=0;
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int flags=0;
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int sample_rate=0;
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int bit_rate=0;
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sh_audio->a_in_buffer_len=0;
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// sync frame:
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while(1){
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while(sh_audio->a_in_buffer_len<7){
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int c=demux_getc(sh_audio->ds);
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if(c<0) return -1; // EOF
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sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c;
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}
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length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
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if(length>=7 && length<=3840) break; // we're done.
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// bad file => resync
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memcpy(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,6);
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--sh_audio->a_in_buffer_len;
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}
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mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate);
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sh_audio->samplerate=sample_rate;
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sh_audio->i_bps=bit_rate/8;
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demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+7,length-7);
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if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0)
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mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed! \n");
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return length;
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}
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// returns: number of available channels
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static int a52_printinfo(sh_audio_t *sh_audio){
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int flags, sample_rate, bit_rate;
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char* mode="unknown";
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int channels=0;
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a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
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switch(flags&A52_CHANNEL_MASK){
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case A52_CHANNEL: mode="channel"; channels=2; break;
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case A52_MONO: mode="mono"; channels=1; break;
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case A52_STEREO: mode="stereo"; channels=2; break;
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case A52_3F: mode="3f";channels=3;break;
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case A52_2F1R: mode="2f+1r";channels=3;break;
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case A52_3F1R: mode="3f+1r";channels=4;break;
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case A52_2F2R: mode="2f+2r";channels=4;break;
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case A52_3F2R: mode="3f+2r";channels=5;break;
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case A52_CHANNEL1: mode="channel1"; channels=2; break;
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case A52_CHANNEL2: mode="channel2"; channels=2; break;
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case A52_DOLBY: mode="dolby"; channels=2; break;
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}
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mp_msg(MSGT_DECAUDIO,MSGL_INFO,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n",
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channels, (flags&A52_LFE)?1:0,
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mode, (flags&A52_LFE)?"+lfe":"",
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sample_rate, bit_rate*0.001f);
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return (flags&A52_LFE) ? (channels+1) : channels;
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}
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int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen);
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static sh_audio_t* dec_audio_sh=NULL;
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#ifdef USE_LIBAC3
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// AC3 decoder buffer callback:
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static void ac3_fill_buffer(uint8_t **start,uint8_t **end){
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int len=ds_get_packet(dec_audio_sh->ds,start);
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//printf("<ac3:%d>\n",len);
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if(len<0)
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*start = *end = NULL;
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else
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*end = *start + len;
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}
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#endif
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// MP3 decoder buffer callback:
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int mplayer_audio_read(char *buf,int size){
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int len;
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len=demux_read_data(dec_audio_sh->ds,buf,size);
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return len;
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}
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int init_audio(sh_audio_t *sh_audio){
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int driver=sh_audio->codec->driver;
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sh_audio->samplesize=2;
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#if WORDS_BIGENDIAN
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sh_audio->sample_format=AFMT_S16_BE;
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#else
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sh_audio->sample_format=AFMT_S16_LE;
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#endif
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sh_audio->samplerate=0;
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//sh_audio->pcm_bswap=0;
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sh_audio->o_bps=0;
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sh_audio->a_buffer_size=0;
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sh_audio->a_buffer=NULL;
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sh_audio->a_in_buffer_len=0;
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// setup required min. in/out buffer size:
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sh_audio->audio_out_minsize=8192;// default size, maybe not enough for Win32/ACM
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switch(driver){
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case AFM_ACM:
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#ifndef USE_WIN32DLL
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mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoACMSupport);
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driver=0;
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#else
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// Win32 ACM audio codec:
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if(init_acm_audio_codec(sh_audio)){
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sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
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sh_audio->channels=sh_audio->o_wf.nChannels;
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sh_audio->samplerate=sh_audio->o_wf.nSamplesPerSec;
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// if(sh_audio->audio_out_minsize>16384) sh_audio->audio_out_minsize=16384;
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// sh_audio->a_buffer_size=sh_audio->audio_out_minsize;
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// if(sh_audio->a_buffer_size<sh_audio->audio_out_minsize+MAX_OUTBURST)
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// sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST;
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} else {
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mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_ACMiniterror);
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driver=0;
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}
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#endif
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break;
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case AFM_DSHOW:
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#ifndef USE_DIRECTSHOW
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mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoDShowAudio);
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driver=0;
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#else
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// Win32 DShow audio codec:
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// printf("DShow_audio: channs=%d rate=%d\n",sh_audio->channels,sh_audio->samplerate);
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if(!(ds_adec=DS_AudioDecoder_Open(sh_audio->codec->dll,&sh_audio->codec->guid,sh_audio->wf))){
|
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mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingDLLcodec,sh_audio->codec->dll);
|
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driver=0;
|
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} else {
|
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sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
|
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sh_audio->channels=sh_audio->wf->nChannels;
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sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
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sh_audio->audio_in_minsize=2*sh_audio->wf->nBlockAlign;
|
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if(sh_audio->audio_in_minsize<8192) sh_audio->audio_in_minsize=8192;
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sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize;
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sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size);
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sh_audio->a_in_buffer_len=0;
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sh_audio->audio_out_minsize=16384;
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}
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#endif
|
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break;
|
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case AFM_VORBIS:
|
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#ifndef HAVE_OGGVORBIS
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mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoOggVorbis);
|
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driver=0;
|
||
#else
|
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/* OggVorbis audio via libvorbis, compatible with files created by nandub and zorannt codec */
|
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sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame
|
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#endif
|
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break;
|
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case AFM_PCM:
|
||
case AFM_DVDPCM:
|
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case AFM_ALAW:
|
||
// PCM, aLaw
|
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sh_audio->audio_out_minsize=2048;
|
||
break;
|
||
case AFM_AC3:
|
||
case AFM_A52:
|
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// Dolby AC3 audio:
|
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// however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame
|
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sh_audio->audio_out_minsize=audio_output_channels*2*256*6;
|
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break;
|
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case AFM_HWAC3:
|
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// Dolby AC3 audio:
|
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sh_audio->audio_out_minsize=4*256*6;
|
||
sh_audio->sample_format = AFMT_AC3;
|
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sh_audio->channels=1;
|
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break;
|
||
case AFM_GSM:
|
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// MS-GSM audio codec:
|
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sh_audio->audio_out_minsize=4*320;
|
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break;
|
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case AFM_IMAADPCM:
|
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sh_audio->audio_out_minsize=4096;
|
||
sh_audio->ds->ss_div=IMA_ADPCM_SAMPLES_PER_BLOCK;
|
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sh_audio->ds->ss_mul=IMA_ADPCM_BLOCK_SIZE;
|
||
break;
|
||
case AFM_MSADPCM:
|
||
sh_audio->audio_out_minsize=sh_audio->wf->nBlockAlign * 8;
|
||
sh_audio->ds->ss_div = MS_ADPCM_SAMPLES_PER_BLOCK;
|
||
sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign;
|
||
break;
|
||
case AFM_FOX61ADPCM:
|
||
sh_audio->audio_out_minsize=FOX61_ADPCM_SAMPLES_PER_BLOCK * 4;
|
||
sh_audio->ds->ss_div=FOX61_ADPCM_SAMPLES_PER_BLOCK;
|
||
sh_audio->ds->ss_mul=FOX61_ADPCM_BLOCK_SIZE;
|
||
break;
|
||
case AFM_FOX62ADPCM:
|
||
sh_audio->audio_out_minsize=FOX62_ADPCM_SAMPLES_PER_BLOCK * 4;
|
||
sh_audio->ds->ss_div=FOX62_ADPCM_SAMPLES_PER_BLOCK;
|
||
sh_audio->ds->ss_mul=FOX62_ADPCM_BLOCK_SIZE;
|
||
break;
|
||
case AFM_MPEG:
|
||
// MPEG Audio:
|
||
sh_audio->audio_out_minsize=4608;
|
||
break;
|
||
#ifdef USE_G72X
|
||
case AFM_G72X:
|
||
// g72x_reader_init(&g72x_data,G723_16_BITS_PER_SAMPLE);
|
||
g72x_reader_init(&g72x_data,G723_24_BITS_PER_SAMPLE);
|
||
// g72x_reader_init(&g72x_data,G721_32_BITS_PER_SAMPLE);
|
||
// g72x_reader_init(&g72x_data,G721_40_BITS_PER_SAMPLE);
|
||
sh_audio->audio_out_minsize=g72x_data.samplesperblock*4;
|
||
break;
|
||
#endif
|
||
case AFM_FFMPEG:
|
||
#ifndef USE_LIBAVCODEC
|
||
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoLAVCsupport);
|
||
return 0;
|
||
#else
|
||
// FFmpeg Audio:
|
||
sh_audio->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
|
||
break;
|
||
#endif
|
||
|
||
#ifdef USE_LIBMAD
|
||
case AFM_MAD:
|
||
printf(__FILE__ ":%d:mad: setting minimum outputsize\n", __LINE__);
|
||
sh_audio->audio_out_minsize=4608;
|
||
if(sh_audio->audio_in_minsize<MAD_SINGLE_BUFFER_SIZE) sh_audio->audio_in_minsize=MAD_SINGLE_BUFFER_SIZE;
|
||
sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize;
|
||
mad_in_buffer = sh_audio->a_in_buffer = malloc(MAD_TOTAL_BUFFER_SIZE);
|
||
sh_audio->a_in_buffer_len=0;
|
||
break;
|
||
#endif
|
||
}
|
||
|
||
if(!driver) return 0;
|
||
|
||
// allocate audio out buffer:
|
||
sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST; // worst case calc.
|
||
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"dec_audio: Allocating %d + %d = %d bytes for output buffer\n",
|
||
sh_audio->audio_out_minsize,MAX_OUTBURST,sh_audio->a_buffer_size);
|
||
|
||
sh_audio->a_buffer=malloc(sh_audio->a_buffer_size);
|
||
if(!sh_audio->a_buffer){
|
||
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_CantAllocAudioBuf);
|
||
return 0;
|
||
}
|
||
memset(sh_audio->a_buffer,0,sh_audio->a_buffer_size);
|
||
sh_audio->a_buffer_len=0;
|
||
|
||
switch(driver){
|
||
#ifdef USE_WIN32DLL
|
||
case AFM_ACM: {
|
||
int ret=acm_decode_audio(sh_audio,sh_audio->a_buffer,4096,sh_audio->a_buffer_size);
|
||
if(ret<0){
|
||
mp_msg(MSGT_DECAUDIO,MSGL_INFO,"ACM decoding error: %d\n",ret);
|
||
driver=0;
|
||
}
|
||
sh_audio->a_buffer_len=ret;
|
||
break;
|
||
}
|
||
#endif
|
||
case AFM_PCM: {
|
||
// AVI PCM Audio:
|
||
WAVEFORMATEX *h=sh_audio->wf;
|
||
sh_audio->i_bps=h->nAvgBytesPerSec;
|
||
sh_audio->channels=h->nChannels;
|
||
sh_audio->samplerate=h->nSamplesPerSec;
|
||
sh_audio->samplesize=(h->wBitsPerSample+7)/8;
|
||
switch(sh_audio->format){ // hardware formats:
|
||
case 0x6: sh_audio->sample_format=AFMT_A_LAW;break;
|
||
case 0x7: sh_audio->sample_format=AFMT_MU_LAW;break;
|
||
case 0x11: sh_audio->sample_format=AFMT_IMA_ADPCM;break;
|
||
case 0x50: sh_audio->sample_format=AFMT_MPEG;break;
|
||
case 0x736F7774: sh_audio->sample_format=AFMT_S16_LE;sh_audio->codec->driver=AFM_DVDPCM;break;
|
||
// case 0x2000: sh_audio->sample_format=AFMT_AC3;
|
||
default: sh_audio->sample_format=(sh_audio->samplesize==2)?AFMT_S16_LE:AFMT_U8;
|
||
}
|
||
break;
|
||
}
|
||
case AFM_DVDPCM: {
|
||
// DVD PCM Audio:
|
||
sh_audio->channels=2;
|
||
sh_audio->samplerate=48000;
|
||
sh_audio->i_bps=2*2*48000;
|
||
// sh_audio->pcm_bswap=1;
|
||
break;
|
||
}
|
||
case AFM_AC3: {
|
||
#ifndef USE_LIBAC3
|
||
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"WARNING: libac3 support is disabled. (hint: upgrade codecs.conf)\n");
|
||
driver=0;
|
||
#else
|
||
// Dolby AC3 audio:
|
||
dec_audio_sh=sh_audio; // save sh_audio for the callback:
|
||
ac3_config.fill_buffer_callback = ac3_fill_buffer;
|
||
ac3_config.num_output_ch = audio_output_channels;
|
||
ac3_config.flags = 0;
|
||
if(gCpuCaps.hasMMX){
|
||
ac3_config.flags |= AC3_MMX_ENABLE;
|
||
}
|
||
if(gCpuCaps.has3DNow){
|
||
ac3_config.flags |= AC3_3DNOW_ENABLE;
|
||
}
|
||
ac3_init();
|
||
sh_audio->ac3_frame = ac3_decode_frame();
|
||
if(sh_audio->ac3_frame){
|
||
ac3_frame_t* fr=(ac3_frame_t*)sh_audio->ac3_frame;
|
||
sh_audio->samplerate=fr->sampling_rate;
|
||
sh_audio->channels=ac3_config.num_output_ch;
|
||
// 1 frame: 6*256 samples 1 sec: sh_audio->samplerate samples
|
||
//sh_audio->i_bps=fr->frame_size*fr->sampling_rate/(6*256);
|
||
sh_audio->i_bps=fr->bit_rate*(1000/8);
|
||
} else {
|
||
driver=0; // bad frame -> disable audio
|
||
}
|
||
#endif
|
||
break;
|
||
}
|
||
case AFM_A52: {
|
||
sample_t level=1, bias=384;
|
||
int flags=0;
|
||
// Dolby AC3 audio:
|
||
if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE;
|
||
if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX;
|
||
if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT;
|
||
if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW;
|
||
a52_samples=a52_init (a52_accel);
|
||
if (a52_samples == NULL) {
|
||
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
|
||
driver=0;break;
|
||
}
|
||
sh_audio->a_in_buffer_size=3840;
|
||
sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size);
|
||
sh_audio->a_in_buffer_len=0;
|
||
if(a52_fillbuff(sh_audio)<0){
|
||
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
|
||
driver=0;break;
|
||
}
|
||
// 'a52 cannot upmix' hotfix:
|
||
a52_printinfo(sh_audio);
|
||
// if(audio_output_channels<sh_audio->channels)
|
||
// sh_audio->channels=audio_output_channels;
|
||
// channels setup:
|
||
sh_audio->channels=audio_output_channels;
|
||
while(sh_audio->channels>0){
|
||
switch(sh_audio->channels){
|
||
case 1: a52_flags=A52_MONO; break;
|
||
// case 2: a52_flags=A52_STEREO; break;
|
||
case 2: a52_flags=A52_DOLBY; break;
|
||
// case 3: a52_flags=A52_3F; break;
|
||
case 3: a52_flags=A52_2F1R; break;
|
||
case 4: a52_flags=A52_2F2R; break; // 2+2
|
||
case 5: a52_flags=A52_3F2R; break;
|
||
case 6: a52_flags=A52_3F2R|A52_LFE; break; // 5.1
|
||
}
|
||
// test:
|
||
flags=a52_flags|A52_ADJUST_LEVEL;
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags);
|
||
if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
|
||
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n");
|
||
driver=0;break;
|
||
}
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
|
||
// frame decoded, let's init resampler:
|
||
if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break;
|
||
--sh_audio->channels; // try to decrease no. of channels
|
||
}
|
||
if(sh_audio->channels<=0){
|
||
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n");
|
||
driver=0;break;
|
||
}
|
||
break;
|
||
}
|
||
case AFM_HWAC3: {
|
||
// Dolby AC3 passthrough:
|
||
a52_samples=a52_init (a52_accel);
|
||
if (a52_samples == NULL) {
|
||
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
|
||
driver=0;break;
|
||
}
|
||
sh_audio->a_in_buffer_size=3840;
|
||
sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size);
|
||
sh_audio->a_in_buffer_len=0;
|
||
if(a52_fillbuff(sh_audio)<0) {
|
||
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
|
||
driver=0;break;
|
||
}
|
||
|
||
//sh_audio->samplerate=ai.samplerate; // SET by a52_fillbuff()
|
||
//sh_audio->samplesize=ai.framesize;
|
||
//sh_audio->i_bps=ai.bitrate*(1000/8); // SET by a52_fillbuff()
|
||
//sh_audio->ac3_frame=malloc(6144);
|
||
//sh_audio->o_bps=sh_audio->i_bps; // XXX FIXME!!! XXX
|
||
|
||
// o_bps is calculated from samplesize*channels*samplerate
|
||
// a single ac3 frame is always translated to 6144 byte packet. (zero padding)
|
||
sh_audio->channels=1;
|
||
sh_audio->samplesize=4; // 1*4*(6*256) = 6144 (very TRICKY!)
|
||
|
||
break;
|
||
}
|
||
case AFM_ALAW: {
|
||
// aLaw audio codec:
|
||
sh_audio->channels=sh_audio->wf->nChannels;
|
||
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
|
||
sh_audio->i_bps=sh_audio->channels*sh_audio->samplerate;
|
||
break;
|
||
}
|
||
#ifdef USE_G72X
|
||
case AFM_G72X: {
|
||
// GSM 723 audio codec:
|
||
sh_audio->channels=sh_audio->wf->nChannels;
|
||
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
|
||
sh_audio->i_bps=(sh_audio->samplerate/g72x_data.samplesperblock)*g72x_data.blocksize;
|
||
break;
|
||
}
|
||
#endif
|
||
#ifdef USE_LIBAVCODEC
|
||
case AFM_FFMPEG: {
|
||
int x;
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
|
||
if(!avcodec_inited){
|
||
avcodec_init();
|
||
avcodec_register_all();
|
||
avcodec_inited=1;
|
||
}
|
||
lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
|
||
if(!lavc_codec){
|
||
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll);
|
||
return 0;
|
||
}
|
||
memset(&lavc_context, 0, sizeof(lavc_context));
|
||
/* open it */
|
||
if (avcodec_open(&lavc_context, lavc_codec) < 0) {
|
||
mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec);
|
||
return 0;
|
||
}
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n");
|
||
|
||
// Decode at least 1 byte: (to get header filled)
|
||
x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
|
||
if(x>0) sh_audio->a_buffer_len=x;
|
||
|
||
#if 1
|
||
sh_audio->channels=lavc_context.channels;
|
||
sh_audio->samplerate=lavc_context.sample_rate;
|
||
sh_audio->i_bps=lavc_context.bit_rate/8;
|
||
#else
|
||
sh_audio->channels=sh_audio->wf->nChannels;
|
||
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
|
||
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
|
||
#endif
|
||
break;
|
||
}
|
||
#endif
|
||
case AFM_GSM: {
|
||
// MS-GSM audio codec:
|
||
GSM_Init();
|
||
sh_audio->channels=sh_audio->wf->nChannels;
|
||
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
|
||
// decodes 65 byte -> 320 short
|
||
// 1 sec: sh_audio->channels*sh_audio->samplerate samples
|
||
// 1 frame: 320 samples
|
||
sh_audio->i_bps=65*(sh_audio->channels*sh_audio->samplerate)/320; // 1:10
|
||
break;
|
||
}
|
||
case AFM_IMAADPCM:
|
||
// IMA-ADPCM 4:1 audio codec:
|
||
sh_audio->channels=sh_audio->wf->nChannels;
|
||
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
|
||
// decodes 34 byte -> 64 short
|
||
sh_audio->i_bps=IMA_ADPCM_BLOCK_SIZE*(sh_audio->channels*sh_audio->samplerate)/IMA_ADPCM_SAMPLES_PER_BLOCK; // 1:4
|
||
break;
|
||
case AFM_MSADPCM:
|
||
sh_audio->channels=sh_audio->wf->nChannels;
|
||
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
|
||
sh_audio->i_bps = sh_audio->wf->nBlockAlign *
|
||
(sh_audio->channels*sh_audio->samplerate) / MS_ADPCM_SAMPLES_PER_BLOCK;
|
||
break;
|
||
case AFM_FOX61ADPCM:
|
||
sh_audio->channels=sh_audio->wf->nChannels;
|
||
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
|
||
sh_audio->i_bps=FOX61_ADPCM_BLOCK_SIZE*
|
||
(sh_audio->channels*sh_audio->samplerate) / FOX61_ADPCM_SAMPLES_PER_BLOCK;
|
||
break;
|
||
case AFM_FOX62ADPCM:
|
||
sh_audio->channels=sh_audio->wf->nChannels;
|
||
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
|
||
sh_audio->i_bps=FOX62_ADPCM_BLOCK_SIZE*
|
||
(sh_audio->channels*sh_audio->samplerate) / FOX62_ADPCM_SAMPLES_PER_BLOCK;
|
||
break;
|
||
case AFM_MPEG: {
|
||
// MPEG Audio:
|
||
dec_audio_sh=sh_audio; // save sh_audio for the callback:
|
||
#ifdef USE_FAKE_MONO
|
||
MP3_Init(fakemono);
|
||
#else
|
||
MP3_Init();
|
||
#endif
|
||
MP3_samplerate=MP3_channels=0;
|
||
sh_audio->a_buffer_len=MP3_DecodeFrame(sh_audio->a_buffer,-1);
|
||
sh_audio->channels=2; // hack
|
||
sh_audio->samplerate=MP3_samplerate;
|
||
sh_audio->i_bps=MP3_bitrate*(1000/8);
|
||
MP3_PrintHeader();
|
||
break;
|
||
}
|
||
#ifdef HAVE_OGGVORBIS
|
||
case AFM_VORBIS: {
|
||
// OggVorbis Audio:
|
||
#if 0 /* just here for reference - atmos */
|
||
ogg_sync_state oy; /* sync and verify incoming physical bitstream */
|
||
ogg_stream_state os; /* take physical pages, weld into a logical
|
||
stream of packets */
|
||
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
|
||
ogg_packet op; /* one raw packet of data for decode */
|
||
|
||
vorbis_info vi; /* struct that stores all the static vorbis bitstream
|
||
settings */
|
||
vorbis_comment vc; /* struct that stores all the bitstream user comments */
|
||
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
|
||
vorbis_block vb; /* local working space for packet->PCM decode */
|
||
#else
|
||
/* nix, nada, rien, nothing, nem, n<>x */
|
||
#endif
|
||
|
||
uint32_t hdrsizes[3];/* stores vorbis header sizes from AVI audio header,
|
||
maybe use ogg_uint32_t */
|
||
//int i;
|
||
int ret;
|
||
char *buffer;
|
||
ogg_packet hdr;
|
||
//ov_struct_t *s=&sh_audio->ov;
|
||
sh_audio->ov=malloc(sizeof(ov_struct_t));
|
||
//s=&sh_audio->ov;
|
||
|
||
vorbis_info_init(&sh_audio->ov->vi);
|
||
vorbis_comment_init(&sh_audio->ov->vc);
|
||
|
||
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: cbsize: %i\n", sh_audio->wf->cbSize);
|
||
memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX), 3*sizeof(uint32_t));
|
||
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: Read header sizes: initial: %i comment: %i codebook: %i\n", hdrsizes[0], hdrsizes[1], hdrsizes[2]);
|
||
/*for(i=12; i <= 40; i+=2) { // header bruteforce :)
|
||
memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+i, 3*sizeof(uint32_t));
|
||
printf("OggVorbis: Read header sizes (%i): %ld %ld %ld\n", i, hdrsizes[0], hdrsizes[1], hdrsizes[2]);
|
||
}*/
|
||
|
||
/* read headers */ // FIXME disable sound on errors here, we absolutely need this headers! - atmos
|
||
hdr.packet=NULL;
|
||
hdr.b_o_s = 1; /* beginning of stream for first packet */
|
||
hdr.bytes = hdrsizes[0];
|
||
hdr.packet = realloc(hdr.packet,hdr.bytes);
|
||
memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t),hdr.bytes);
|
||
if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
|
||
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: initial (identification) header broken!\n");
|
||
hdr.b_o_s = 0;
|
||
hdr.bytes = hdrsizes[1];
|
||
hdr.packet = realloc(hdr.packet,hdr.bytes);
|
||
memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0],hdr.bytes);
|
||
if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
|
||
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: comment header broken!\n");
|
||
hdr.bytes = hdrsizes[2];
|
||
hdr.packet = realloc(hdr.packet,hdr.bytes);
|
||
memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0]+hdrsizes[1],hdr.bytes);
|
||
if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
|
||
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: codebook header broken!\n");
|
||
hdr.bytes=0;
|
||
hdr.packet = realloc(hdr.packet,hdr.bytes); /* free */
|
||
/* done with the headers */
|
||
|
||
|
||
/* Throw the comments plus a few lines about the bitstream we're
|
||
decoding */
|
||
{
|
||
char **ptr=sh_audio->ov->vc.user_comments;
|
||
while(*ptr){
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr);
|
||
++ptr;
|
||
}
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel, %ldHz, %ldkbit/s %cBR\n",sh_audio->ov->vi.channels,sh_audio->ov->vi.rate,sh_audio->ov->vi.bitrate_nominal/1000, (sh_audio->ov->vi.bitrate_lower!=sh_audio->ov->vi.bitrate_nominal)||(sh_audio->ov->vi.bitrate_upper!=sh_audio->ov->vi.bitrate_nominal)?'V':'C');
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",sh_audio->ov->vc.vendor);
|
||
}
|
||
sh_audio->channels=sh_audio->ov->vi.channels;
|
||
sh_audio->samplerate=sh_audio->ov->vi.rate;
|
||
sh_audio->i_bps=sh_audio->ov->vi.bitrate_nominal/8;
|
||
|
||
// printf("[\n");
|
||
// sh_audio->a_buffer_len=sh_audio->audio_out_minsize;///ov->vi.channels;
|
||
// printf("]\n");
|
||
|
||
/* OK, got and parsed all three headers. Initialize the Vorbis
|
||
packet->PCM decoder. */
|
||
vorbis_synthesis_init(&sh_audio->ov->vd,&sh_audio->ov->vi); /* central decode state */
|
||
vorbis_block_init(&sh_audio->ov->vd,&sh_audio->ov->vb); /* local state for most of the decode
|
||
so multiple block decodes can
|
||
proceed in parallel. We could init
|
||
multiple vorbis_block structures
|
||
for vd here */
|
||
//printf("OggVorbis: synthesis and block init done.\n");
|
||
ogg_sync_init(&sh_audio->ov->oy); /* Now we can read pages */
|
||
|
||
while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) {
|
||
if(ret == -1)
|
||
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n");
|
||
else
|
||
if(ret == 0) {
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: need more data to verify page, reading more data.\n");
|
||
/* submit a a_buffer_len block to libvorbis' Ogg layer */
|
||
buffer=ogg_sync_buffer(&sh_audio->ov->oy,256);
|
||
ogg_sync_wrote(&sh_audio->ov->oy,demux_read_data(sh_audio->ds,buffer,256));
|
||
}
|
||
}
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: successfull.\n");
|
||
ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og); /* we can ignore any errors here
|
||
as they'll also become apparent
|
||
at packetout */
|
||
|
||
/* Get the serial number and set up the rest of decode. */
|
||
/* serialno first; use it to set up a logical stream */
|
||
ogg_stream_init(&sh_audio->ov->os,ogg_page_serialno(&sh_audio->ov->og));
|
||
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n");
|
||
|
||
break;
|
||
}
|
||
#endif
|
||
|
||
#ifdef USE_LIBMAD
|
||
case AFM_MAD:
|
||
{
|
||
printf("%s %s %s (%s)\n", mad_version, mad_copyright, mad_author, mad_build);
|
||
|
||
printf(__FILE__ ":%d:mad: initialising\n", __LINE__);
|
||
mad_frame_init(&mad_frame);
|
||
mad_stream_init(&mad_stream);
|
||
|
||
printf(__FILE__ ":%d:mad: preparing buffer\n", __LINE__);
|
||
mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size);
|
||
mad_stream_buffer(&mad_stream, (unsigned char*)(sh_audio->a_in_buffer), sh_audio->a_in_buffer_len);
|
||
// mad_stream_sync(&mad_stream);
|
||
mad_sync(sh_audio, &mad_stream);
|
||
mad_synth_init(&mad_synth);
|
||
|
||
if(mad_frame_decode(&mad_frame, &mad_stream) == 0)
|
||
{
|
||
printf(__FILE__ ":%d:mad: post processing buffer\n", __LINE__);
|
||
mad_postprocess_buffer(sh_audio, &mad_stream);
|
||
}
|
||
else
|
||
{
|
||
printf(__FILE__ ":%d:mad: frame decoding failed\n", __LINE__);
|
||
mad_print_error(&mad_stream);
|
||
}
|
||
|
||
switch (mad_frame.header.mode)
|
||
{
|
||
case MAD_MODE_SINGLE_CHANNEL:
|
||
sh_audio->channels=1;
|
||
break;
|
||
case MAD_MODE_DUAL_CHANNEL:
|
||
case MAD_MODE_JOINT_STEREO:
|
||
case MAD_MODE_STEREO:
|
||
sh_audio->channels=2;
|
||
break;
|
||
default:
|
||
mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "mad: unknown number of channels\n");
|
||
}
|
||
mp_msg(MSGT_DECAUDIO, MSGL_HINT, "mad: channels: %d (mad channel mode: %d)\n",
|
||
sh_audio->channels, mad_frame.header.mode);
|
||
/* var. name changed in 0.13.0 (beta) (libmad/CHANGES) -- alex */
|
||
#if (MAD_VERSION_MAJOR >= 0) && (MAD_VERSION_MINOR >= 13)
|
||
sh_audio->samplerate=mad_frame.header.samplerate;
|
||
#else
|
||
sh_audio->samplerate=mad_frame.header.sfreq;
|
||
#endif
|
||
sh_audio->i_bps=mad_frame.header.bitrate;
|
||
printf(__FILE__ ":%d:mad: continuing\n", __LINE__);
|
||
break;
|
||
}
|
||
#endif
|
||
}
|
||
|
||
if(!sh_audio->channels || !sh_audio->samplerate){
|
||
mp_msg(MSGT_DECAUDIO,MSGL_WARN,MSGTR_UnknownAudio);
|
||
driver=0;
|
||
}
|
||
|
||
if(!driver){
|
||
if(sh_audio->a_buffer) free(sh_audio->a_buffer);
|
||
sh_audio->a_buffer=NULL;
|
||
return 0;
|
||
}
|
||
|
||
if(!sh_audio->o_bps)
|
||
sh_audio->o_bps=sh_audio->channels*sh_audio->samplerate*sh_audio->samplesize;
|
||
return driver;
|
||
}
|
||
|
||
// Audio decoding:
|
||
|
||
// Decode a single frame (mp3,acm etc) or 'minlen' bytes (pcm/alaw etc)
|
||
// buffer length is 'maxlen' bytes, it shouldn't be exceeded...
|
||
|
||
int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){
|
||
int len=-1;
|
||
switch(sh_audio->codec->driver){
|
||
#ifdef USE_LIBAVCODEC
|
||
case AFM_FFMPEG: {
|
||
unsigned char *start=NULL;
|
||
int y;
|
||
while(len<minlen){
|
||
int len2=0;
|
||
int x=ds_get_packet(sh_audio->ds,&start);
|
||
if(x<=0) break; // error
|
||
y=avcodec_decode_audio(&lavc_context,(INT16*)buf,&len2,start,x);
|
||
if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
|
||
if(y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
|
||
if(len2>0){
|
||
//len=len2;break;
|
||
if(len<0) len=len2; else len+=len2;
|
||
buf+=len2;
|
||
}
|
||
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
|
||
}
|
||
}
|
||
break;
|
||
#endif
|
||
case AFM_MPEG: // MPEG layer 2 or 3
|
||
len=MP3_DecodeFrame(buf,-1);
|
||
// len=MP3_DecodeFrame(buf,3);
|
||
break;
|
||
#ifdef HAVE_OGGVORBIS
|
||
case AFM_VORBIS: { // OggVorbis
|
||
/* note: good minlen would be 4k or 8k IMHO - atmos */
|
||
int ret;
|
||
char *buffer;
|
||
int bytes;
|
||
int samples;
|
||
float **pcm;
|
||
//ogg_int16_t convbuffer[4096];
|
||
// int convsize;
|
||
int readlen=1024;
|
||
len=0;
|
||
// convsize=minlen/sh_audio->ov->vi.channels;
|
||
|
||
while(len < minlen) { /* double loop allows for break in inner loop */
|
||
while(len < minlen) { /* without aborting the outer loop - atmos */
|
||
ret=ogg_stream_packetout(&sh_audio->ov->os,&sh_audio->ov->op);
|
||
if(ret==0) {
|
||
int xxx=0;
|
||
//printf("OggVorbis: Packetout: need more data, paging!\n");
|
||
while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) {
|
||
if(ret == -1)
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n");
|
||
else
|
||
if(ret == 0) {
|
||
//printf("OggVorbis: Pageout: need more data to verify page, reading more data.\n");
|
||
/* submit a readlen k block to libvorbis' Ogg layer */
|
||
buffer=ogg_sync_buffer(&sh_audio->ov->oy,readlen);
|
||
bytes=demux_read_data(sh_audio->ds,buffer,readlen);
|
||
xxx+=bytes;
|
||
ogg_sync_wrote(&sh_audio->ov->oy,bytes);
|
||
if(bytes==0)
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: 0Bytes written, possible End of Stream\n");
|
||
}
|
||
}
|
||
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[sync: %d ]\n",xxx);
|
||
//printf("OggVorbis: Pageout: successfull, pagin in.\n");
|
||
if(ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og)<0)
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pagein failed!\n");
|
||
break;
|
||
} else if(ret<0) {
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Packetout: missing or corrupt data, skipping packet!\n");
|
||
break;
|
||
} else {
|
||
|
||
/* we have a packet. Decode it */
|
||
|
||
if(vorbis_synthesis(&sh_audio->ov->vb,&sh_audio->ov->op)==0) /* test for success! */
|
||
vorbis_synthesis_blockin(&sh_audio->ov->vd,&sh_audio->ov->vb);
|
||
|
||
/* **pcm is a multichannel float vector. In stereo, for
|
||
example, pcm[0] is left, and pcm[1] is right. samples is
|
||
the size of each channel. Convert the float values
|
||
(-1.<=range<=1.) to whatever PCM format and write it out */
|
||
|
||
while((samples=vorbis_synthesis_pcmout(&sh_audio->ov->vd,&pcm))>0){
|
||
int i,j;
|
||
int clipflag=0;
|
||
int convsize=(maxlen-len)/(2*sh_audio->ov->vi.channels); // max size!
|
||
int bout=(samples<convsize?samples:convsize);
|
||
|
||
if(bout<=0) break;
|
||
|
||
/* convert floats to 16 bit signed ints (host order) and
|
||
interleave */
|
||
for(i=0;i<sh_audio->ov->vi.channels;i++){
|
||
ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]);
|
||
ogg_int16_t *ptr=convbuffer+i;
|
||
float *mono=pcm[i];
|
||
for(j=0;j<bout;j++){
|
||
#if 1
|
||
int val=mono[j]*32767.f;
|
||
#else /* optional dither */
|
||
int val=mono[j]*32767.f+drand48()-0.5f;
|
||
#endif
|
||
/* might as well guard against clipping */
|
||
if(val>32767){
|
||
val=32767;
|
||
clipflag=1;
|
||
}
|
||
if(val<-32768){
|
||
val=-32768;
|
||
clipflag=1;
|
||
}
|
||
*ptr=val;
|
||
ptr+=sh_audio->ov->vi.channels;
|
||
}
|
||
}
|
||
|
||
if(clipflag)
|
||
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(sh_audio->ov->vd.sequence));
|
||
|
||
//fwrite(convbuffer,2*sh_audio->ov->vi.channels,bout,stderr); //dump pcm to file for debugging
|
||
//memcpy(buf+len,convbuffer,2*sh_audio->ov->vi.channels*bout);
|
||
len+=2*sh_audio->ov->vi.channels*bout;
|
||
|
||
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples);
|
||
|
||
vorbis_synthesis_read(&sh_audio->ov->vd,bout); /* tell libvorbis how
|
||
many samples we
|
||
actually consumed */
|
||
}
|
||
} // from else, packetout ok
|
||
} // while len
|
||
} // outer while len
|
||
if(ogg_page_eos(&sh_audio->ov->og))
|
||
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: End of Stream reached!\n"); // FIXME clearup decoder, notify mplayer - atmos
|
||
|
||
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[len: %d ]\n",len);
|
||
|
||
break;
|
||
}
|
||
#endif
|
||
case AFM_PCM: // AVI PCM
|
||
len=demux_read_data(sh_audio->ds,buf,minlen);
|
||
break;
|
||
case AFM_DVDPCM: // DVD PCM
|
||
{ int j;
|
||
len=demux_read_data(sh_audio->ds,buf,minlen);
|
||
//if(i&1){ printf("Warning! pcm_audio_size&1 !=0 (%d)\n",i);i&=~1; }
|
||
// swap endian:
|
||
for(j=0;j<len;j+=2){
|
||
char x=buf[j];
|
||
buf[j]=buf[j+1];
|
||
buf[j+1]=x;
|
||
}
|
||
break;
|
||
}
|
||
case AFM_ALAW: // aLaw decoder
|
||
{ int l=demux_read_data(sh_audio->ds,buf,minlen/2);
|
||
unsigned short *d=(unsigned short *) buf;
|
||
unsigned char *s=buf;
|
||
len=2*l;
|
||
if(sh_audio->format==6){
|
||
// aLaw
|
||
while(l>0){ --l; d[l]=alaw2short[s[l]]; }
|
||
} else {
|
||
// uLaw
|
||
while(l>0){ --l; d[l]=ulaw2short[s[l]]; }
|
||
}
|
||
break;
|
||
}
|
||
case AFM_GSM: // MS-GSM decoder
|
||
{ unsigned char ibuf[65]; // 65 bytes / frame
|
||
if(demux_read_data(sh_audio->ds,ibuf,65)!=65) break; // EOF
|
||
XA_MSGSM_Decoder(ibuf,(unsigned short *) buf); // decodes 65 byte -> 320 short
|
||
// XA_GSM_Decoder(buf,(unsigned short *) &sh_audio->a_buffer[sh_audio->a_buffer_len]); // decodes 33 byte -> 160 short
|
||
len=2*320;
|
||
break;
|
||
}
|
||
#ifdef USE_G72X
|
||
case AFM_G72X: // GSM 723 decoder
|
||
{ if(demux_read_data(sh_audio->ds,g72x_data.block, g72x_data.blocksize)!=g72x_data.blocksize) break; // EOF
|
||
g72x_decode_block(&g72x_data);
|
||
len=2*g72x_data.samplesperblock;
|
||
memcpy(buf,g72x_data.samples,len);
|
||
break;
|
||
}
|
||
#endif
|
||
case AFM_IMAADPCM:
|
||
{ unsigned char ibuf[IMA_ADPCM_BLOCK_SIZE * 2]; // bytes / stereo frame
|
||
if (demux_read_data(sh_audio->ds, ibuf,
|
||
IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) !=
|
||
IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels)
|
||
break; // EOF
|
||
len=2*ima_adpcm_decode_block((unsigned short*)buf,ibuf, sh_audio->wf->nChannels);
|
||
break;
|
||
}
|
||
case AFM_MSADPCM:
|
||
{ static unsigned char *ibuf = NULL;
|
||
if (!ibuf)
|
||
ibuf = (unsigned char *)malloc
|
||
(sh_audio->wf->nBlockAlign * sh_audio->wf->nChannels);
|
||
if (demux_read_data(sh_audio->ds, ibuf,
|
||
sh_audio->wf->nBlockAlign) !=
|
||
sh_audio->wf->nBlockAlign)
|
||
break; // EOF
|
||
len= 2 * ms_adpcm_decode_block(
|
||
(unsigned short*)buf,ibuf, sh_audio->wf->nChannels,
|
||
sh_audio->wf->nBlockAlign);
|
||
break;
|
||
}
|
||
case AFM_FOX61ADPCM:
|
||
{ unsigned char ibuf[FOX61_ADPCM_BLOCK_SIZE]; // bytes / stereo frame
|
||
if (demux_read_data(sh_audio->ds, ibuf, FOX61_ADPCM_BLOCK_SIZE) !=
|
||
FOX61_ADPCM_BLOCK_SIZE)
|
||
break; // EOF
|
||
len=2*fox61_adpcm_decode_block((unsigned short*)buf,ibuf);
|
||
break;
|
||
}
|
||
case AFM_FOX62ADPCM:
|
||
{ unsigned char ibuf[FOX62_ADPCM_BLOCK_SIZE * 2]; // bytes / stereo frame
|
||
if (demux_read_data(sh_audio->ds, ibuf,
|
||
FOX62_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) !=
|
||
FOX62_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels)
|
||
break; // EOF
|
||
len = 2 * fox62_adpcm_decode_block(
|
||
(unsigned short*)buf,ibuf);
|
||
break;
|
||
}
|
||
#ifdef USE_LIBAC3
|
||
case AFM_AC3: // AC3 decoder
|
||
//printf("{1:%d}",avi_header.idx_pos);fflush(stdout);
|
||
if(!sh_audio->ac3_frame) sh_audio->ac3_frame=ac3_decode_frame();
|
||
//printf("{2:%d}",avi_header.idx_pos);fflush(stdout);
|
||
if(sh_audio->ac3_frame){
|
||
len = 256 * 6 *sh_audio->channels*sh_audio->samplesize;
|
||
memcpy(buf,((ac3_frame_t*)sh_audio->ac3_frame)->audio_data,len);
|
||
sh_audio->ac3_frame=NULL;
|
||
}
|
||
//printf("{3:%d}",avi_header.idx_pos);fflush(stdout);
|
||
break;
|
||
#endif
|
||
case AFM_A52: { // AC3 decoder
|
||
sample_t level=1, bias=384;
|
||
int flags=a52_flags|A52_ADJUST_LEVEL;
|
||
int i;
|
||
if(!sh_audio->a_in_buffer_len)
|
||
if(a52_fillbuff(sh_audio)<0) break; // EOF
|
||
sh_audio->a_in_buffer_len=0;
|
||
if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
|
||
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n");
|
||
break;
|
||
}
|
||
// a52_dynrng (&state, NULL, NULL); // disable dynamic range compensation
|
||
|
||
// frame decoded, let's resample:
|
||
//a52_resample_init(a52_accel,flags,sh_audio->channels);
|
||
len=0;
|
||
for (i = 0; i < 6; i++) {
|
||
if (a52_block (&a52_state, a52_samples)){
|
||
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n");
|
||
break;
|
||
}
|
||
len+=2*a52_resample(a52_samples,&buf[len]);
|
||
}
|
||
// printf("len = %d \n",len); // 6144 on all vobs I tried so far... (5.1 and 2.0) ::atmos
|
||
break;
|
||
}
|
||
case AFM_HWAC3: // AC3 through SPDIF
|
||
if(!sh_audio->a_in_buffer_len)
|
||
if((len=a52_fillbuff(sh_audio))<0) break; //EOF
|
||
sh_audio->a_in_buffer_len=0;
|
||
len = ac3_iec958_build_burst(len, 0x01, 1, sh_audio->a_in_buffer, buf);
|
||
// len = 6144 = 4*(6*256)
|
||
break;
|
||
#ifdef USE_WIN32DLL
|
||
case AFM_ACM:
|
||
// len=sh_audio->audio_out_minsize; // optimal decoded fragment size
|
||
// if(len<minlen) len=minlen; else
|
||
// if(len>maxlen) len=maxlen;
|
||
// len=acm_decode_audio(sh_audio,buf,len);
|
||
len=acm_decode_audio(sh_audio,buf,minlen,maxlen);
|
||
break;
|
||
#endif
|
||
|
||
#ifdef USE_DIRECTSHOW
|
||
case AFM_DSHOW: // DirectShow
|
||
{ int size_in=0;
|
||
int size_out=0;
|
||
int srcsize=DS_AudioDecoder_GetSrcSize(ds_adec, maxlen);
|
||
mp_msg(MSGT_DECAUDIO,MSGL_DBG3,"DShow says: srcsize=%d (buffsize=%d) out_size=%d\n",srcsize,sh_audio->a_in_buffer_size,maxlen);
|
||
if(srcsize>sh_audio->a_in_buffer_size) srcsize=sh_audio->a_in_buffer_size; // !!!!!!
|
||
if(sh_audio->a_in_buffer_len<srcsize){
|
||
sh_audio->a_in_buffer_len+=
|
||
demux_read_data(sh_audio->ds,&sh_audio->a_in_buffer[sh_audio->a_in_buffer_len],
|
||
srcsize-sh_audio->a_in_buffer_len);
|
||
}
|
||
DS_AudioDecoder_Convert(ds_adec, sh_audio->a_in_buffer,sh_audio->a_in_buffer_len,
|
||
buf,maxlen, &size_in,&size_out);
|
||
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"DShow: audio %d -> %d converted (in_buf_len=%d of %d) %d\n",size_in,size_out,sh_audio->a_in_buffer_len,sh_audio->a_in_buffer_size,ds_tell_pts(sh_audio->ds));
|
||
if(size_in>=sh_audio->a_in_buffer_len){
|
||
sh_audio->a_in_buffer_len=0;
|
||
} else {
|
||
sh_audio->a_in_buffer_len-=size_in;
|
||
memcpy(sh_audio->a_in_buffer,&sh_audio->a_in_buffer[size_in],sh_audio->a_in_buffer_len);
|
||
}
|
||
len=size_out;
|
||
break;
|
||
}
|
||
#endif
|
||
|
||
#ifdef USE_LIBMAD
|
||
case AFM_MAD:
|
||
{
|
||
mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size);
|
||
mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len);
|
||
// mad_stream_sync(&mad_stream);
|
||
mad_sync(sh_audio, &mad_stream);
|
||
if(mad_frame_decode(&mad_frame, &mad_stream) == 0)
|
||
{
|
||
mad_synth_frame(&mad_synth, &mad_frame);
|
||
mad_postprocess_buffer(sh_audio, &mad_stream);
|
||
|
||
/* and fill buffer */
|
||
|
||
{
|
||
int i;
|
||
int end_size = mad_synth.pcm.length;
|
||
signed short* samples = (signed short*)buf;
|
||
if(end_size > maxlen/4)
|
||
end_size=maxlen/4;
|
||
|
||
for(i=0; i<mad_synth.pcm.length; ++i) {
|
||
*samples++ = mad_scale(mad_synth.pcm.samples[0][i]);
|
||
*samples++ = mad_scale(mad_synth.pcm.samples[0][i]);
|
||
// *buf++ = mad_scale(mad_synth.pcm.sampAles[1][i]);
|
||
}
|
||
len = end_size*4;
|
||
}
|
||
}
|
||
else
|
||
{
|
||
printf(__FILE__ ":%d:mad: frame decoding failed (error: %d)\n", __LINE__,
|
||
mad_stream.error);
|
||
mad_print_error(&mad_stream);
|
||
}
|
||
|
||
break;
|
||
}
|
||
#endif
|
||
}
|
||
return len;
|
||
}
|
||
|
||
void resync_audio_stream(sh_audio_t *sh_audio){
|
||
switch(sh_audio->codec->driver){
|
||
case AFM_MPEG:
|
||
MP3_DecodeFrame(NULL,-2); // resync
|
||
MP3_DecodeFrame(NULL,-2); // resync
|
||
MP3_DecodeFrame(NULL,-2); // resync
|
||
break;
|
||
#ifdef HAVE_OGGVORBIS
|
||
case AFM_VORBIS:
|
||
//printf("OggVorbis: resetting stream.\n");
|
||
ogg_sync_reset(&sh_audio->ov->oy);
|
||
ogg_stream_reset(&sh_audio->ov->os);
|
||
break;
|
||
#endif
|
||
#ifdef USE_LIBAC3
|
||
case AFM_AC3:
|
||
ac3_bitstream_reset(); // reset AC3 bitstream buffer
|
||
// if(verbose){ printf("Resyncing AC3 audio...");fflush(stdout);}
|
||
sh_audio->ac3_frame=ac3_decode_frame(); // resync
|
||
// if(verbose) printf(" OK!\n");
|
||
break;
|
||
#endif
|
||
case AFM_A52:
|
||
case AFM_ACM:
|
||
case AFM_DSHOW:
|
||
case AFM_HWAC3:
|
||
sh_audio->a_in_buffer_len=0; // reset ACM/DShow audio buffer
|
||
break;
|
||
|
||
#ifdef USE_LIBMAD
|
||
case AFM_MAD:
|
||
mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size);
|
||
mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len);
|
||
// mad_stream_sync(&mad_stream);
|
||
mad_sync(sh_audio, &mad_stream);
|
||
mad_postprocess_buffer(sh_audio, &mad_stream);
|
||
break;
|
||
#endif
|
||
}
|
||
}
|
||
|
||
void skip_audio_frame(sh_audio_t *sh_audio){
|
||
switch(sh_audio->codec->driver){
|
||
case AFM_MPEG: MP3_DecodeFrame(NULL,-2);break; // skip MPEG frame
|
||
#ifdef USE_LIBAC3
|
||
case AFM_AC3: sh_audio->ac3_frame=ac3_decode_frame();break; // skip AC3 frame
|
||
#endif
|
||
case AFM_HWAC3:
|
||
case AFM_A52: a52_fillbuff(sh_audio);break; // skip AC3 frame
|
||
case AFM_ACM:
|
||
case AFM_DSHOW: {
|
||
int skip=sh_audio->wf->nBlockAlign;
|
||
if(skip<16){
|
||
skip=(sh_audio->wf->nAvgBytesPerSec/16)&(~7);
|
||
if(skip<16) skip=16;
|
||
}
|
||
demux_read_data(sh_audio->ds,NULL,skip);
|
||
break;
|
||
}
|
||
case AFM_PCM:
|
||
case AFM_DVDPCM:
|
||
case AFM_ALAW: {
|
||
int skip=sh_audio->i_bps/16;
|
||
skip=skip&(~3);
|
||
demux_read_data(sh_audio->ds,NULL,skip);
|
||
break;
|
||
}
|
||
#ifdef USE_LIBMAD
|
||
case AFM_MAD:
|
||
{
|
||
mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size);
|
||
mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len);
|
||
mad_stream_skip(&mad_stream, 2);
|
||
// mad_stream_sync(&mad_stream);
|
||
mad_sync(sh_audio, &mad_stream);
|
||
mad_postprocess_buffer(sh_audio, &mad_stream);
|
||
break;
|
||
}
|
||
#endif
|
||
|
||
default: ds_fill_buffer(sh_audio->ds); // skip PCM frame
|
||
}
|
||
}
|