changeset 5356:739cad21c32c

Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.

--- a/libmpcodecs/Makefile	Tue Mar 26 12:43:01 2002 +0000
+++ b/libmpcodecs/Makefile	Tue Mar 26 15:53:18 2002 +0000
@@ -3,7 +3,7 @@
 
 LIBNAME = libmpcodecs.a
 
-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
+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
 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
 
 ifeq ($(PNG),yes)

--- a/libmpcodecs/ad.c	Tue Mar 26 12:43:01 2002 +0000
+++ b/libmpcodecs/ad.c	Tue Mar 26 15:53:18 2002 +0000
@@ -31,6 +31,7 @@
 extern ad_functions_t mpcodecs_ad_dshow;
 extern ad_functions_t mpcodecs_ad_acm;
 extern ad_functions_t mpcodecs_ad_msgsm;
+extern ad_functions_t mpcodecs_ad_faad;
 
 ad_functions_t* mpcodecs_ad_drivers[] =
 {
@@ -56,5 +57,8 @@
 #endif
   &mpcodecs_ad_acm,
 #endif
+#ifdef HAVE_FAAD
+ &mpcodecs_ad_faad,
+#endif
   NULL
 };

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/libmpcodecs/ad_faad.c	Tue Mar 26 15:53:18 2002 +0000
@@ -0,0 +1,204 @@
+/* ad_faad.c - MPlayer AAC decoder using libfaad2
+ * This file is part of MPlayer, see http://mplayerhq.hu/ for info.  
+ * (c)2002 by Felix Buenemann <atmosfear at users.sourceforge.net>
+ * File licensed under the GPL, see http://www.fsf.org/ for more info.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "ad_internal.h"
+
+#ifdef HAVE_FAAD
+
+static ad_info_t info = 
+{
+	"AAC (MPEG2/4 Advanced Audio Coding)",
+	"libfaad2",
+	AFM_AAC,
+	"Felix Buenemann",
+	"faad2",
+	"Under development!"
+};
+
+LIBAD_EXTERN(faad)
+
+#include <faad.h>
+
+/* configure maximum supported channels, *
+ * this is theoretically max. 64 chans   */
+#define FAAD_MAX_CHANNELS 6
+#define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS)		       
+
+//#define AAC_DUMP_COMPRESSED  
+
+static faacDecHandle faac_hdec;
+static faacDecFrameInfo faac_finfo;
+
+static int preinit(sh_audio_t *sh)
+{
+  sh->audio_in_minsize=FAAD_BUFFLEN;
+  sh->audio_out_minsize=2048*FAAD_MAX_CHANNELS;
+
+  // XXX: why is this duplicated in sh struct? ::atmos
+  sh->a_in_buffer_size=sh->audio_in_minsize;
+  sh->a_in_buffer=malloc(sh->a_in_buffer_size);
+  sh->a_in_buffer_len=0;
+  return 1;
+}
+
+static int init(sh_audio_t *sh)
+{
+  unsigned long faac_samplerate, faac_channels;
+  int faac_init;
+  faac_hdec = faacDecOpen();
+
+  sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
+
+  // If we don't get the ES descriptor, try manual config
+  if(!sh->codecdata_len) {
+#if 1
+    faacDecConfigurationPtr faac_conf;
+    /* Set the default object type and samplerate */
+    /* This is useful for RAW AAC files */
+    faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
+    if(sh->samplerate)
+      faac_conf->defSampleRate = sh->samplerate;
+    /* XXX: FAAD support FLOAT output, how do we handle
+      * that (FAAD_FMT_FLOAT)? ::atmos
+      */
+    if(sh->samplesize)
+      switch(sh->samplesize){
+	case 1: // 8Bit
+	  mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
+	default:
+	case 2: // 16Bit
+	  faac_conf->outputFormat = FAAD_FMT_16BIT;
+	  break;
+	case 3: // 24Bit
+	  faac_conf->outputFormat = FAAD_FMT_24BIT;
+	  break;
+	case 4: // 32Bit
+	  faac_conf->outputFormat = FAAD_FMT_32BIT;
+	  break;
+      }
+    //faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
+
+    faacDecSetConfiguration(faac_hdec, faac_conf);
+#endif
+
+    /* init the codec */
+    faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
+       &faac_samplerate, &faac_channels);
+    sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed
+
+  } else { // We have ES DS in codecdata
+    /*int i;
+    for(i = 0; i < sh_audio->codecdata_len; i++)
+      printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
+
+    faac_init = faacDecInit2(faac_hdec, sh->codecdata,
+       sh->codecdata_len,	&faac_samplerate, &faac_channels);
+  }
+  if(faac_init < 0) {
+    mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
+    faacDecClose(faac_hdec);
+    // XXX: free a_in_buffer here or in uninit?
+    return 0;
+  } else {
+    mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
+    mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz  channels: %d\n", faac_samplerate, faac_channels);
+    sh->channels = faac_channels;
+    sh->samplerate = faac_samplerate;
+    //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
+    if(!sh->i_bps) {
+      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
+      sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
+    } else 
+      mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
+  }  
+  return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+  mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n");
+  faacDecClose(faac_hdec);
+  if(sh->a_in_buffer_size) {
+    free(sh->a_in_buffer);
+    sh->a_in_buffer = NULL;
+  } 
+}
+
+static int control(sh_audio_t *sh,int cmd,void* arg, ...)
+{
+    switch(cmd)
+    {
+#if 0      
+      case ADCTRL_RESYNC_STREAM:
+	  return CONTROL_TRUE;
+      case ADCTRL_SKIP_FRAME:
+	  return CONTROL_TRUE;
+#endif
+    }
+  return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
+{
+	int j = 0, len = 0;	      
+	void *faac_sample_buffer;
+	//memset(sh->a_buffer,0,sh->a_buffer_size); // XXX: check if needed.
+
+	while(len < minlen) {
+	  /* update buffer */
+          if(sh->a_in_buffer_len < sh->a_in_buffer_size){
+            sh->a_in_buffer_len +=
+              demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
+              sh->a_in_buffer_size - sh->a_in_buffer_len);
+          }
+	  
+#ifdef DUMP_AAC_COMPRESSED
+	  {int i;
+	  for (i = 0; i < 16; i++)
+	    printf ("%02X ", sh->a_in_buffer[i]);
+	  printf ("\n");}
+#endif
+	  do {
+	    faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer+j);
+	    /* update buffer index after faacDecDecode */
+            if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len){
+              sh->a_in_buffer_len=0;
+	    } else {
+	      sh->a_in_buffer_len-=faac_finfo.bytesconsumed;
+	      memcpy(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len);
+	    }
+
+	    if(faac_finfo.error > 0) {
+	      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Trying to resync!\n");
+	      j++;
+	    } else
+	      break;
+	  } while(j < FAAD_BUFFLEN);	  
+
+	  if(faac_finfo.error > 0) {
+	    mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n",
+	      faacDecGetErrorMessage(faac_finfo.error));
+	  } else if (faac_finfo.samples == 0)
+	    mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n");
+	  else {
+	    /* XXX: samples already multiplied by channels! */
+	    mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%d Bytes)!\n",
+	       sh->samplesize*faac_finfo.samples);
+	    memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples);
+	    len += sh->samplesize*faac_finfo.samples;
+	    //printf("FAAD: buffer: %d bytes  consumed: %d \n", k, faac_finfo.bytesconsumed);
+	  }
+	}
+   return len;
+}
+
+#endif /* !HAVE_FAAD */
+