mirror of https://github.com/mpv-player/mpv
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@5357 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
atmos4
parent
9ef25a9736
commit
96b58cfa58
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@ -3,7 +3,7 @@ include ../config.mak
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LIBNAME = libmpcodecs.a
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AUDIO_SRCS=dec_audio.c ad.c ad_a52.c ad_acm.c ad_alaw.c ad_dk3adpcm.c ad_dk4adpcm.c ad_dshow.c ad_dvdpcm.c ad_ffmpeg.c ad_hwac3.c ad_imaadpcm.c ad_mp3.c ad_msadpcm.c ad_pcm.c ad_roqaudio.c ad_msgsm.c
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AUDIO_SRCS=dec_audio.c ad.c ad_a52.c ad_acm.c ad_alaw.c ad_dk3adpcm.c ad_dk4adpcm.c ad_dshow.c ad_dvdpcm.c ad_ffmpeg.c ad_hwac3.c ad_imaadpcm.c ad_mp3.c ad_msadpcm.c ad_pcm.c ad_roqaudio.c ad_msgsm.c ad_faad.c
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VIDEO_SRCS=dec_video.c vd.c vd_null.c vd_cinepak.c vd_qtrpza.c vd_ffmpeg.c vd_dshow.c vd_vfw.c vd_odivx.c vd_divx4.c vd_raw.c vd_xanim.c vd_msvidc.c vd_fli.c vd_qtrle.c vd_qtsmc.c vd_roqvideo.c vd_cyuv.c vd_nuv.c vd_libmpeg2.c vd_msrle.c vd_huffyuv.c vd_zlib.c
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ifeq ($(PNG),yes)
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@ -31,6 +31,7 @@ extern ad_functions_t mpcodecs_ad_roqaudio;
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extern ad_functions_t mpcodecs_ad_dshow;
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extern ad_functions_t mpcodecs_ad_acm;
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extern ad_functions_t mpcodecs_ad_msgsm;
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extern ad_functions_t mpcodecs_ad_faad;
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ad_functions_t* mpcodecs_ad_drivers[] =
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{
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@ -55,6 +56,9 @@ ad_functions_t* mpcodecs_ad_drivers[] =
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&mpcodecs_ad_dshow,
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#endif
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&mpcodecs_ad_acm,
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#endif
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#ifdef HAVE_FAAD
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&mpcodecs_ad_faad,
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#endif
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NULL
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};
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@ -0,0 +1,204 @@
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/* ad_faad.c - MPlayer AAC decoder using libfaad2
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* This file is part of MPlayer, see http://mplayerhq.hu/ for info.
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* (c)2002 by Felix Buenemann <atmosfear at users.sourceforge.net>
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* File licensed under the GPL, see http://www.fsf.org/ for more info.
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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 "config.h"
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#include "ad_internal.h"
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#ifdef HAVE_FAAD
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static ad_info_t info =
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{
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"AAC (MPEG2/4 Advanced Audio Coding)",
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"libfaad2",
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AFM_AAC,
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"Felix Buenemann",
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"faad2",
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"Under development!"
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};
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LIBAD_EXTERN(faad)
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#include <faad.h>
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/* configure maximum supported channels, *
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* this is theoretically max. 64 chans */
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#define FAAD_MAX_CHANNELS 6
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#define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS)
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//#define AAC_DUMP_COMPRESSED
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static faacDecHandle faac_hdec;
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static faacDecFrameInfo faac_finfo;
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static int preinit(sh_audio_t *sh)
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{
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sh->audio_in_minsize=FAAD_BUFFLEN;
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sh->audio_out_minsize=2048*FAAD_MAX_CHANNELS;
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// XXX: why is this duplicated in sh struct? ::atmos
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sh->a_in_buffer_size=sh->audio_in_minsize;
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sh->a_in_buffer=malloc(sh->a_in_buffer_size);
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sh->a_in_buffer_len=0;
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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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unsigned long faac_samplerate, faac_channels;
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int faac_init;
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faac_hdec = faacDecOpen();
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sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
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// If we don't get the ES descriptor, try manual config
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if(!sh->codecdata_len) {
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#if 1
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faacDecConfigurationPtr faac_conf;
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/* Set the default object type and samplerate */
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/* This is useful for RAW AAC files */
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faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
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if(sh->samplerate)
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faac_conf->defSampleRate = sh->samplerate;
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/* XXX: FAAD support FLOAT output, how do we handle
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* that (FAAD_FMT_FLOAT)? ::atmos
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*/
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if(sh->samplesize)
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switch(sh->samplesize){
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case 1: // 8Bit
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mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
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default:
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case 2: // 16Bit
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faac_conf->outputFormat = FAAD_FMT_16BIT;
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break;
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case 3: // 24Bit
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faac_conf->outputFormat = FAAD_FMT_24BIT;
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break;
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case 4: // 32Bit
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faac_conf->outputFormat = FAAD_FMT_32BIT;
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break;
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}
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//faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
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faacDecSetConfiguration(faac_hdec, faac_conf);
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#endif
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/* init the codec */
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faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
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&faac_samplerate, &faac_channels);
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sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed
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} else { // We have ES DS in codecdata
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/*int i;
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for(i = 0; i < sh_audio->codecdata_len; i++)
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printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
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faac_init = faacDecInit2(faac_hdec, sh->codecdata,
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sh->codecdata_len, &faac_samplerate, &faac_channels);
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}
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if(faac_init < 0) {
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mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
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faacDecClose(faac_hdec);
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// XXX: free a_in_buffer here or in uninit?
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return 0;
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} else {
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mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
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mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz channels: %d\n", faac_samplerate, faac_channels);
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sh->channels = faac_channels;
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sh->samplerate = faac_samplerate;
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//sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
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if(!sh->i_bps) {
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mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
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sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
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} else
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mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
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}
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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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mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n");
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faacDecClose(faac_hdec);
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if(sh->a_in_buffer_size) {
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free(sh->a_in_buffer);
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sh->a_in_buffer = NULL;
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}
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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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switch(cmd)
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{
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#if 0
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case ADCTRL_RESYNC_STREAM:
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return CONTROL_TRUE;
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case ADCTRL_SKIP_FRAME:
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return CONTROL_TRUE;
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#endif
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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,unsigned char *buf,int minlen,int maxlen)
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{
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int j = 0, len = 0;
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void *faac_sample_buffer;
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//memset(sh->a_buffer,0,sh->a_buffer_size); // XXX: check if needed.
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while(len < minlen) {
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/* update buffer */
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if(sh->a_in_buffer_len < sh->a_in_buffer_size){
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sh->a_in_buffer_len +=
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demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
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sh->a_in_buffer_size - sh->a_in_buffer_len);
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}
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#ifdef DUMP_AAC_COMPRESSED
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{int i;
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for (i = 0; i < 16; i++)
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printf ("%02X ", sh->a_in_buffer[i]);
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printf ("\n");}
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#endif
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do {
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faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer+j);
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/* update buffer index after faacDecDecode */
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if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len){
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sh->a_in_buffer_len=0;
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} else {
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sh->a_in_buffer_len-=faac_finfo.bytesconsumed;
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memcpy(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len);
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}
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if(faac_finfo.error > 0) {
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mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Trying to resync!\n");
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j++;
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} else
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break;
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} while(j < FAAD_BUFFLEN);
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if(faac_finfo.error > 0) {
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mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n",
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faacDecGetErrorMessage(faac_finfo.error));
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} else if (faac_finfo.samples == 0)
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mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n");
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else {
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/* XXX: samples already multiplied by channels! */
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mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%d Bytes)!\n",
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sh->samplesize*faac_finfo.samples);
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memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples);
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len += sh->samplesize*faac_finfo.samples;
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//printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed);
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}
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}
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return len;
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}
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#endif /* !HAVE_FAAD */
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