initial commit from tarball

frontend/Makefile.am

@@ -0,0 +1,9 @@
+bin_PROGRAMS = aacplusenc
+man_MANS = ../docs/aacplusenc.1
+
+aacplusenc_SOURCES = main.c
+
+INCLUDES = -I$(top_srcdir)/include  -I$(top_srcdir)/src
+LDADD = $(top_builddir)/src/libaacplus.la -lm
+
+aacplusenc_LDFLAGS = @FFTW3_LDFLAGS@

frontend/au_channel.h

@@ -0,0 +1,66 @@
+#include <stdio.h>
+#include <string.h>
+
+#define WAV_HEADER_SIZE 64
+
+#define WAV_FORMAT_FLOAT 0xFFFE
+
+typedef struct {
+	int	sampleRate;
+	int	nChannels;
+	long	nSamples;
+	int	aFmt;
+} WavInfo;
+
+inline FILE* AuChannelOpen (const char* filename, WavInfo* info)
+{
+	unsigned char header[12];
+	unsigned char data[WAV_HEADER_SIZE];
+	FILE *handle;
+	unsigned int   chunksize;
+	
+	
+	if (!strcmp(filename,"-"))
+		handle = stdin;
+	else
+		handle = fopen(filename, "rb");
+
+	if(!handle) return NULL;
+
+	if(fread(header, 1, 12, handle) != 12) return NULL;
+	info->nSamples		= (header[4] | (header[5] << 8) | (header[6] << 16) | (header[7] << 24)) + 8;
+
+	while (memcmp(header, "data", 4) != 0){
+	    if(fread(header, 1, 8, handle) != 8) return NULL;
+	    chunksize             = (header[4] | (header[5] << 8) | (header[6] << 16) | (header[7] << 24));
+	    //fprintf(stderr, "%c%c%c%c %d", header[0],  header[1], header[2], header[3], chunksize);
+            if(!memcmp(header, "fmt ", 4)) {
+                if(chunksize > WAV_HEADER_SIZE) return NULL;
+	        if(fread(data, 1, chunksize, handle) != chunksize) return NULL;
+                info->aFmt		= data[0] | data[1] << 8;
+                info->nChannels		= data[2] | data[3] << 8;
+                info->sampleRate	= data[4] | data[5] << 8 | data[6] << 12 | data[7] << 16;
+	    } else if(memcmp(header, "data", 4) != 0) {
+                if(fseek(handle, chunksize, SEEK_CUR) != 0) return NULL;
+	    }
+	}
+
+	return handle;
+}
+
+inline void AuChannelClose (FILE *audioChannel)
+{
+	fclose(audioChannel);
+}
+
+inline size_t AuChannelReadShort(FILE *audioChannel, short *samples, int nSamples, int *readed)
+{
+	*readed = fread(samples, 2, nSamples, audioChannel);
+	return *readed <= 0;
+}
+
+inline size_t AuChannelReadFloat(FILE *audioChannel, float *samples, int nSamples, int *readed)
+{
+	*readed = fread(samples, 4, nSamples, audioChannel);
+	return *readed <= 0;
+}

frontend/main.c

@@ -0,0 +1,141 @@
+#include <stdio.h>
+#include <assert.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "au_channel.h"
+#include "aacplus.h"
+
+int main(int argc, char *argv[])
+{
+
+  WavInfo inputInfo;
+  FILE *inputFile = NULL;
+  FILE *hADTSFile;
+
+  int  error;
+  int  bEncodeMono = 0;
+  int frmCnt = 0;
+
+   /*
+   * parse command line arguments
+   */
+  if (argc != 5) {
+    fprintf(stderr, "\nUsage:   %s <wav_file> <bitstream_file> <bitrate> <(m)ono/(s)tereo>\n", argv[0]);
+    fprintf(stderr, "\nExample: %s input.wav out.aac 24000 s\n", argv[0]);
+    return 0;
+  }
+
+  if ( strcmp (argv[4],"m") == 0 ) {
+    bEncodeMono = 1;
+  }
+  else {
+    if ( strcmp (argv[4],"s") != 0 ) {
+      fprintf(stderr, "\nWrong mode %s, use either (m)ono or (s)tereo\n", argv[4]);
+      return 0;
+    }
+  }
+  fflush(stdout);
+
+  inputFile = AuChannelOpen (argv[1], &inputInfo);
+
+  if(inputFile == NULL){
+    fprintf(stderr,"could not open %s\n",argv[1]);
+    exit(10);
+  }
+
+  if (inputInfo.nChannels==1 && !bEncodeMono) {
+	  fprintf(stderr,"Need stereo input for stereo coding mode !\n");
+	  exit(10);
+  }
+
+  if (strcmp(argv[2],"-")==0)
+   hADTSFile=stdout;
+  else
+   hADTSFile = fopen(argv[2], "wb");
+
+  if(!hADTSFile) {
+   fprintf(stderr, "\nFailed to create ADTS file\n") ;
+      exit(10);
+    }
+
+  /*
+    Be verbose
+   */
+  unsigned long inputSamples=0;
+  unsigned long maxOutputBytes=0;
+  aacplusEncHandle hEncoder = aacplusEncOpen(inputInfo.sampleRate,
+		  	  inputInfo.nChannels,
+		  	  &inputSamples,
+		  	  &maxOutputBytes);
+
+  aacplusEncConfiguration *cfg = aacplusEncGetCurrentConfiguration(hEncoder);
+  cfg->bitRate = atoi(argv[3]);
+  cfg->bandWidth = 0;
+  cfg->outputFormat = 1;
+  cfg->nChannelsOut = bEncodeMono ? 1 : inputInfo.nChannels;
+  if(inputInfo.aFmt == WAV_FORMAT_FLOAT){
+    cfg->inputFormat = AACPLUS_INPUT_FLOAT;
+  }
+	
+  fprintf(stdout,"input file %s: \nsr = %d, nc = %d fmt = %d\n\n",
+          argv[1], inputInfo.sampleRate, inputInfo.nChannels, inputInfo.aFmt);
+  fprintf(stdout,"output file %s: \nbr = %d inputSamples = %lu  maxOutputBytes = %lu nc = %d m = %d\n\n",
+            argv[2], cfg->bitRate, inputSamples, maxOutputBytes, cfg->nChannelsOut, bEncodeMono);
+  fflush(stdout);
+
+  int ret = 0;
+  if((ret = aacplusEncSetConfiguration(hEncoder, cfg)) == 0) {
+      fprintf(stdout,"setting cfg failed\n", ret);
+      return -1;
+  }
+
+  uint8_t *outputBuffer = malloc(maxOutputBytes);
+  int32_t *TimeDataPcm;
+  if(inputInfo.aFmt == WAV_FORMAT_FLOAT) {
+    TimeDataPcm = calloc(inputSamples, sizeof(float));
+  } else {
+    TimeDataPcm = calloc(inputSamples, sizeof(short));
+  }
+
+  int stopLoop = 0;
+  int bytes = 0;
+  do {
+      int numSamplesRead = 0;
+      if(inputInfo.aFmt == WAV_FORMAT_FLOAT) {
+          if ( AuChannelReadFloat(inputFile, (float *) TimeDataPcm, inputSamples, &numSamplesRead) > 0) {
+                  stopLoop = 1;
+                  break;
+          }
+      } else {
+          if ( AuChannelReadShort(inputFile, (short *) TimeDataPcm, inputSamples, &numSamplesRead) > 0) {
+                  stopLoop = 1;
+                  break;
+          }
+      }
+
+      if(numSamplesRead < inputSamples) {
+          stopLoop = 1;
+          break;
+      }
+
+      bytes = aacplusEncEncode(hEncoder, (int32_t *) TimeDataPcm, numSamplesRead,
+              outputBuffer,
+              maxOutputBytes);
+
+      if(bytes > 0) fwrite(outputBuffer, bytes, 1, hADTSFile);
+
+      frmCnt++;
+      fprintf(stderr,"[%d]\r",frmCnt); fflush(stderr);
+  } while (!stopLoop && bytes >= 0);
+
+  fprintf(stderr,"\n");
+  fflush(stderr);
+
+  printf("\nencoding finished\n");
+  aacplusEncClose(hEncoder);
+  fclose(hADTSFile);
+  free(outputBuffer);
+  free(TimeDataPcm);
+  return 0;
+}