Merge branch 'master' into alsa

1 files changed, 349 insertions, 349 deletions

diff --git a/aac-enc-dabplus.c b/aac-enc-dabplus.c
index 7516a15..cd86053 100644
--- a/aac-enc-dabplus.c
+++ b/aac-enc-dabplus.c
@@ -5,7 +5,7 @@
  * you may not use this file except in compliance with the License.
  * You may obtain a copy of the License at
  *
- *	  http://www.apache.org/licenses/LICENSE-2.0
+ *    http://www.apache.org/licenses/LICENSE-2.0
  *
  * Unless required by applicable law or agreed to in writing, software
  * distributed under the License is distributed on an "AS IS" BASIS,
@@ -29,26 +29,26 @@
 #include <fec.h>
 
 void usage(const char* name) {
-	fprintf(stderr, "%s [OPTION...]\n", name);
-	fprintf(stderr,
-"	  -b, --bitrate={ 8, 16, ..., 192 }    Output bitrate in kbps. Must be 8 multiple.\n"
-//"	  -d, --data=FILENAME                  Set data filename.\n"
-//"	  -g, --fs-bug                         Turn on FS bug mitigation.\n"
-"	  -i, --input=FILENAME                 Input filename (default: stdin).\n"
-"	  -o, --output=FILENAME                Output filename (default: stdout).\n"
-"	  -a, --afterburner                    Turn on AAC encoder quality increaser.\n"
-//"	  -m, --message                        Turn on AAC frame messages.\n"
-//"	  -p, --pad=BYTES                      Set PAD size in bytes.\n"
-"	  -f, --format={ wav, raw }            Set input file format (default: wav).\n"
-"	  -c, --channels={ 1, 2 }              Nb of input channels for raw input (default: 2).\n"
-"	  -r, --rate={ 32000, 48000 }          Sample rate for raw input (default: 48000).\n"
-//"	  -t, --type=TYPE                      Set data type (dls|pad|packet|dg).\n"
-//"	  -v, --verbose=LEVEL                  Set verbosity level.\n"
-//"	  -V, --version                        Print version and exit.\n"
-//"	  --mi=[ 0, ... ]                      Set AAC frame messages interval in milliseconds.\n"
-//"	  --ma=[ 0, ... ]                      Set AAC frame messages attack time in milliseconds.\n"
-//"	  -t, --adts                           Set ADTS output format (for debugging).\n"
-//"	  -l, --lp                             Set frame size to 1024 instead of 960.\n"
+    fprintf(stderr, "%s [OPTION...]\n", name);
+    fprintf(stderr,
+"     -b, --bitrate={ 8, 16, ..., 192 }    Output bitrate in kbps. Must be 8 multiple.\n"
+//"   -d, --data=FILENAME                  Set data filename.\n"
+//"   -g, --fs-bug                         Turn on FS bug mitigation.\n"
+"     -i, --input=FILENAME                 Input filename (default: stdin).\n"
+"     -o, --output=FILENAME                Output filename (default: stdout).\n"
+"     -a, --afterburner                    Turn on AAC encoder quality increaser.\n"
+//"   -m, --message                        Turn on AAC frame messages.\n"
+//"   -p, --pad=BYTES                      Set PAD size in bytes.\n"
+"     -f, --format={ wav, raw }            Set input file format (default: wav).\n"
+"     -c, --channels={ 1, 2 }              Nb of input channels for raw input (default: 2).\n"
+"     -r, --rate={ 32000, 48000 }          Sample rate for raw input (default: 48000).\n"
+//"   -t, --type=TYPE                      Set data type (dls|pad|packet|dg).\n"
+//"   -v, --verbose=LEVEL                  Set verbosity level.\n"
+//"   -V, --version                        Print version and exit.\n"
+//"   --mi=[ 0, ... ]                      Set AAC frame messages interval in milliseconds.\n"
+//"   --ma=[ 0, ... ]                      Set AAC frame messages attack time in milliseconds.\n"
+//"   -t, --adts                           Set ADTS output format (for debugging).\n"
+//"   -l, --lp                             Set frame size to 1024 instead of 960.\n"
 
 );
 
@@ -71,8 +71,8 @@ int FindSRIndex(int sr)
 {
     int i;
     for (i = 0; i < 16; i++) {
-	if (sr == mpeg4audio_sample_rates[i])
-	    return i;
+    if (sr == mpeg4audio_sample_rates[i])
+        return i;
     }
     return 16 - 1;
 }
@@ -84,341 +84,341 @@ void adts_hdr_up(char *buff, int size)
 
     /* frame length, 13 bits */
     buff[3] &= 0xFC;
-    buff[3] |= ((len >> 11) & 0x03);	/* 2b: aac_frame_length */
-    buff[4] = len >> 3;			/* 8b: aac_frame_length */
-    buff[5] = (len << 5) & 0xE0;	/* 3b: aac_frame_length */
+    buff[3] |= ((len >> 11) & 0x03);    /* 2b: aac_frame_length */
+    buff[4] = len >> 3;         /* 8b: aac_frame_length */
+    buff[5] = (len << 5) & 0xE0;    /* 3b: aac_frame_length */
     /* buffer fullness, 11 bits */
-    buff[5] |= ((buffer_fullness >> 6) & 0x1F);	/* 5b: adts_buffer_fullness */
-    buff[6] = (buffer_fullness << 2) & 0xFC;	/* 6b: adts_buffer_fullness */
-						/* 2b: num_raw_data_blocks */
+    buff[5] |= ((buffer_fullness >> 6) & 0x1F); /* 5b: adts_buffer_fullness */
+    buff[6] = (buffer_fullness << 2) & 0xFC;    /* 6b: adts_buffer_fullness */
+                        /* 2b: num_raw_data_blocks */
 }
 
 int main(int argc, char *argv[]) {
-	int subchannel_index = 8; //64kbps subchannel
-	int ch=0;
-	const char *infile, *outfile;
-	FILE *in_fh, *out_fh;
-	void *wav;
-	int wav_format, bits_per_sample, sample_rate=48000, channels=2;
-	uint8_t* input_buf;
-	int16_t* convert_buf;
-	void *rs_handler = NULL;
-	int aot = AOT_DABPLUS_AAC_LC;
-	int afterburner = 0, raw_input=0;
-	HANDLE_AACENCODER handle;
-	CHANNEL_MODE mode;
-	AACENC_InfoStruct info = { 0 };
-
-	const struct option longopts[] = {
-		{"bitrate",     required_argument,  0, 'b'},
-	    {"input",       required_argument,  0, 'i'},
-	    {"output",      required_argument,  0, 'o'},
-	    {"format",      required_argument,  0, 'f'},
-	    {"rate",        required_argument,  0, 'r'},
-	    {"channels",    required_argument,  0, 'c'},
-	    //{"lp",          no_argument,        0, 'l'},
-	    //{"adts",        no_argument,        0, 't'},
-	    {"afterburner", no_argument,        0, 'a'},
-	    {"help",        no_argument,        0, 'h'},
-	    {0,0,0,0},
-	};
-
-	int index;
-	while(ch != -1) {
-		ch = getopt_long(argc, argv, "tlhab:c:i:o:r:f:", longopts, &index);
-		switch (ch) {
-		case 'f':
-			if(strcmp(optarg, "raw")==0) {
-				raw_input = 1;
-			} else if(strcmp(optarg, "wav")!=0)
-				usage(argv[0]);
-			break;
-		case 'a':
-			afterburner = 1;
-			break;
-		case 'b':
-			subchannel_index = atoi(optarg) / 8;
-			break;
-		case 'c':
-			channels = atoi(optarg);
-			break;
-		case 'r':
-			sample_rate = atoi(optarg);
-			break;
-		case 'i':
-			infile = optarg;
-			break;
-		case 'o':
-			outfile = optarg;
-			break;
-		case '?':
-		case 'h':
-			usage(argv[0]);
-			return 1;
-		}
-	}
-
-	if(subchannel_index < 1 || subchannel_index > 24) {
-		fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n", subchannel_index);
-		return 1;
-	}
-
-	if(raw_input) {
-		if(infile && strcmp(infile, "-")) {
-			in_fh = fopen(infile, "rb");
-			if(!in_fh) {
-				fprintf(stderr, "Can't open input file!\n");
-				return 1;
-			}
-		} else {
-			in_fh = stdin;
-		}
-	} else {
-		wav = wav_read_open(infile);
-		if (!wav) {
-			fprintf(stderr, "Unable to open wav file %s\n", infile);
-			return 1;
-		}
-		if (!wav_get_header(wav, &wav_format, &channels, &sample_rate, &bits_per_sample, NULL)) {
-			fprintf(stderr, "Bad wav file %s\n", infile);
-			return 1;
-		}
-		if (wav_format != 1) {
-			fprintf(stderr, "Unsupported WAV format %d\n", wav_format);
-			return 1;
-		}
-		if (bits_per_sample != 16) {
-			fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
-			return 1;
-		}
-		if (channels > 2) {
-			fprintf(stderr, "Unsupported WAV channels %d\n", channels);
-			return 1;
-		}
-	}
-
-	if(outfile && strcmp(outfile, "-")) {
-		out_fh = fopen(outfile, "wb");
-		if(!out_fh) {
-			fprintf(stderr, "Can't open output file!\n");
-			return 1;
-		}
-	} else {
-		out_fh = stdout;
-	}
-
-
-	switch (channels) {
-	case 1: mode = MODE_1;       break;
-	case 2: mode = MODE_2;       break;
-	default:
-		fprintf(stderr, "Unsupported channels number %d\n", channels);
-		return 1;
-	}
-
-
-	if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) {
-		fprintf(stderr, "Unable to open encoder\n");
-		return 1;
-	}
-
-
-	if(channels == 2 && subchannel_index <= 6)
-		aot = AOT_DABPLUS_PS;
-	else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10)
-		aot = AOT_DABPLUS_SBR;
-
-	fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n",
-			subchannel_index,
-			aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "",
-			aot == AOT_DABPLUS_SBR ? "HE-AAC" : "",
-			aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "",
-			channels, sample_rate);
-
-	if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) {
-		fprintf(stderr, "Unable to set the AOT\n");
-		return 1;
-	}
-	if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
-		fprintf(stderr, "Unable to set the AOT\n");
-		return 1;
-	}
-	if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) {
-		fprintf(stderr, "Unable to set the channel mode\n");
-		return 1;
-	}
-	if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) {
-		fprintf(stderr, "Unable to set the wav channel order\n");
-		return 1;
-	}
-	if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
-		fprintf(stderr, "Unable to set the AOT\n");
-		return 1;
-	}
-	if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
-		fprintf(stderr, "Unable to set the RAW transmux\n");
-		return 1;
-	}
-
-	/*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) != AACENC_OK) {
-		fprintf(stderr, "Unable to set the bitrate mode\n");
-		return 1;
-	}*/
-
-
-	fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000);
-	if (aacEncoder_SetParam(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) {
-		fprintf(stderr, "Unable to set the bitrate\n");
-		return 1;
-	}
-	if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
-		fprintf(stderr, "Unable to set the afterburner mode\n");
-		return 1;
-	}
-	if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
-		fprintf(stderr, "Unable to initialize the encoder\n");
-		return 1;
-	}
-	if (aacEncInfo(handle, &info) != AACENC_OK) {
-		fprintf(stderr, "Unable to get the encoder info\n");
-		return 1;
-	}
-
-	fprintf(stderr, "DAB+ Encoding: framelen=%d\n", info.frameLength);
-
-	int input_size = channels*2*info.frameLength;
-	input_buf = (uint8_t*) malloc(input_size);
-	convert_buf = (int16_t*) malloc(input_size);
-
-	/* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */
-	rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135);
-	if (rs_handler == NULL) {
-		perror("init_rs_char failed");
-		return 0;
-	}
+    int subchannel_index = 8; //64kbps subchannel
+    int ch=0;
+    const char *infile, *outfile;
+    FILE *in_fh, *out_fh;
+    void *wav;
+    int wav_format, bits_per_sample, sample_rate=48000, channels=2;
+    uint8_t* input_buf;
+    int16_t* convert_buf;
+    void *rs_handler = NULL;
+    int aot = AOT_DABPLUS_AAC_LC;
+    int afterburner = 0, raw_input=0;
+    HANDLE_AACENCODER handle;
+    CHANNEL_MODE mode;
+    AACENC_InfoStruct info = { 0 };
+
+    const struct option longopts[] = {
+        {"bitrate",     required_argument,  0, 'b'},
+        {"input",       required_argument,  0, 'i'},
+        {"output",      required_argument,  0, 'o'},
+        {"format",      required_argument,  0, 'f'},
+        {"rate",        required_argument,  0, 'r'},
+        {"channels",    required_argument,  0, 'c'},
+        //{"lp",          no_argument,        0, 'l'},
+        //{"adts",        no_argument,        0, 't'},
+        {"afterburner", no_argument,        0, 'a'},
+        {"help",        no_argument,        0, 'h'},
+        {0,0,0,0},
+    };
+
+    int index;
+    while(ch != -1) {
+        ch = getopt_long(argc, argv, "tlhab:c:i:o:r:f:", longopts, &index);
+        switch (ch) {
+        case 'f':
+            if(strcmp(optarg, "raw")==0) {
+                raw_input = 1;
+            } else if(strcmp(optarg, "wav")!=0)
+                usage(argv[0]);
+            break;
+        case 'a':
+            afterburner = 1;
+            break;
+        case 'b':
+            subchannel_index = atoi(optarg) / 8;
+            break;
+        case 'c':
+            channels = atoi(optarg);
+            break;
+        case 'r':
+            sample_rate = atoi(optarg);
+            break;
+        case 'i':
+            infile = optarg;
+            break;
+        case 'o':
+            outfile = optarg;
+            break;
+        case '?':
+        case 'h':
+            usage(argv[0]);
+            return 1;
+        }
+    }
+
+    if(subchannel_index < 1 || subchannel_index > 24) {
+        fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n", subchannel_index);
+        return 1;
+    }
+
+    if(raw_input) {
+        if(infile && strcmp(infile, "-")) {
+            in_fh = fopen(infile, "rb");
+            if(!in_fh) {
+                fprintf(stderr, "Can't open input file!\n");
+                return 1;
+            }
+        } else {
+            in_fh = stdin;
+        }
+    } else {
+        wav = wav_read_open(infile);
+        if (!wav) {
+            fprintf(stderr, "Unable to open wav file %s\n", infile);
+            return 1;
+        }
+        if (!wav_get_header(wav, &wav_format, &channels, &sample_rate, &bits_per_sample, NULL)) {
+            fprintf(stderr, "Bad wav file %s\n", infile);
+            return 1;
+        }
+        if (wav_format != 1) {
+            fprintf(stderr, "Unsupported WAV format %d\n", wav_format);
+            return 1;
+        }
+        if (bits_per_sample != 16) {
+            fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
+            return 1;
+        }
+        if (channels > 2) {
+            fprintf(stderr, "Unsupported WAV channels %d\n", channels);
+            return 1;
+        }
+    }
+
+    if(outfile && strcmp(outfile, "-")) {
+        out_fh = fopen(outfile, "wb");
+        if(!out_fh) {
+            fprintf(stderr, "Can't open output file!\n");
+            return 1;
+        }
+    } else {
+        out_fh = stdout;
+    }
+
+
+    switch (channels) {
+    case 1: mode = MODE_1;       break;
+    case 2: mode = MODE_2;       break;
+    default:
+        fprintf(stderr, "Unsupported channels number %d\n", channels);
+        return 1;
+    }
+
+
+    if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) {
+        fprintf(stderr, "Unable to open encoder\n");
+        return 1;
+    }
+
+
+    if(channels == 2 && subchannel_index <= 6)
+        aot = AOT_DABPLUS_PS;
+    else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10)
+        aot = AOT_DABPLUS_SBR;
+
+    fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n",
+            subchannel_index,
+            aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "",
+            aot == AOT_DABPLUS_SBR ? "HE-AAC" : "",
+            aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "",
+            channels, sample_rate);
+
+    if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the AOT\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the AOT\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the channel mode\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the wav channel order\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the AOT\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the RAW transmux\n");
+        return 1;
+    }
+
+    /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the bitrate mode\n");
+        return 1;
+    }*/
+
+
+    fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000);
+    if (aacEncoder_SetParam(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the bitrate\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the afterburner mode\n");
+        return 1;
+    }
+    if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
+        fprintf(stderr, "Unable to initialize the encoder\n");
+        return 1;
+    }
+    if (aacEncInfo(handle, &info) != AACENC_OK) {
+        fprintf(stderr, "Unable to get the encoder info\n");
+        return 1;
+    }
+
+    fprintf(stderr, "DAB+ Encoding: framelen=%d\n", info.frameLength);
+
+    int input_size = channels*2*info.frameLength;
+    input_buf = (uint8_t*) malloc(input_size);
+    convert_buf = (int16_t*) malloc(input_size);
+
+    /* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */
+    rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135);
+    if (rs_handler == NULL) {
+        perror("init_rs_char failed");
+        return 0;
+    }
 
     int loops = 0;
     int outbuf_size = subchannel_index*120;
-	uint8_t outbuf[20480];
-
-	if(outbuf_size % 5 != 0) {
-		fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5);
-	}
-
-	fprintf(stderr, "outbuf_size: %d\n", outbuf_size);
-	//outbuf_size += (4 * subchannel_index * (8*8)/8) - outbuf_size/5;
-	fprintf(stderr, "outbuf_size: %d\n", outbuf_size);
-
-	int frame=0;
-	while (1) {
-		memset(outbuf, 0x00, outbuf_size);
-
-		AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
-		AACENC_InArgs in_args = { 0 };
-		AACENC_OutArgs out_args = { 0 };
-		int in_identifier = IN_AUDIO_DATA;
-		int in_size, in_elem_size;
-		int out_identifier = OUT_BITSTREAM_DATA;
-		int out_size, out_elem_size;
-		int read=0, i;
-		void *in_ptr, *out_ptr;
-		AACENC_ERROR err;
-
-		if(raw_input) {
-			if(fread(input_buf, input_size, 1, in_fh) == 1) {
-				read = input_size;
-			} else {
-				fprintf(stderr, "Unable to read from input!\n");
-				break;
-			}
-		} else {
-			read = wav_read_data(wav, input_buf, input_size);
-		}
-
-		for (i = 0; i < read/2; i++) {
-			const uint8_t* in = &input_buf[2*i];
-			convert_buf[i] = in[0] | (in[1] << 8);
-		}
-
-		if (read <= 0) {
-			in_args.numInSamples = -1;
-		} else {
-			in_ptr = convert_buf;
-			in_size = read;
-			in_elem_size = 2;
-
-			in_args.numInSamples = read/2;
-			in_buf.numBufs = 1;
-			in_buf.bufs = &in_ptr;
-			in_buf.bufferIdentifiers = &in_identifier;
-			in_buf.bufSizes = &in_size;
-			in_buf.bufElSizes = &in_elem_size;
-		}
-		out_ptr = outbuf;
-		out_size = sizeof(outbuf);
-		out_elem_size = 1;
-		out_buf.numBufs = 1;
-		out_buf.bufs = &out_ptr;
-		out_buf.bufferIdentifiers = &out_identifier;
-		out_buf.bufSizes = &out_size;
-		out_buf.bufElSizes = &out_elem_size;
-
-		if ((err = aacEncEncode(handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) {
-			if (err == AACENC_ENCODE_EOF)
-				break;
-			fprintf(stderr, "Encoding failed\n");
-			return 1;
-		}
-		if (out_args.numOutBytes == 0)
-			continue;
+    uint8_t outbuf[20480];
+
+    if(outbuf_size % 5 != 0) {
+        fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5);
+    }
+
+    fprintf(stderr, "outbuf_size: %d\n", outbuf_size);
+    //outbuf_size += (4 * subchannel_index * (8*8)/8) - outbuf_size/5;
+    fprintf(stderr, "outbuf_size: %d\n", outbuf_size);
+
+    int frame=0;
+    while (1) {
+        memset(outbuf, 0x00, outbuf_size);
+
+        AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
+        AACENC_InArgs in_args = { 0 };
+        AACENC_OutArgs out_args = { 0 };
+        int in_identifier = IN_AUDIO_DATA;
+        int in_size, in_elem_size;
+        int out_identifier = OUT_BITSTREAM_DATA;
+        int out_size, out_elem_size;
+        int read=0, i;
+        void *in_ptr, *out_ptr;
+        AACENC_ERROR err;
+
+        if(raw_input) {
+            if(fread(input_buf, input_size, 1, in_fh) == 1) {
+                read = input_size;
+            } else {
+                fprintf(stderr, "Unable to read from input!\n");
+                break;
+            }
+        } else {
+            read = wav_read_data(wav, input_buf, input_size);
+        }
+
+        for (i = 0; i < read/2; i++) {
+            const uint8_t* in = &input_buf[2*i];
+            convert_buf[i] = in[0] | (in[1] << 8);
+        }
+
+        if (read <= 0) {
+            in_args.numInSamples = -1;
+        } else {
+            in_ptr = convert_buf;
+            in_size = read;
+            in_elem_size = 2;
+
+            in_args.numInSamples = read/2;
+            in_buf.numBufs = 1;
+            in_buf.bufs = &in_ptr;
+            in_buf.bufferIdentifiers = &in_identifier;
+            in_buf.bufSizes = &in_size;
+            in_buf.bufElSizes = &in_elem_size;
+        }
+        out_ptr = outbuf;
+        out_size = sizeof(outbuf);
+        out_elem_size = 1;
+        out_buf.numBufs = 1;
+        out_buf.bufs = &out_ptr;
+        out_buf.bufferIdentifiers = &out_identifier;
+        out_buf.bufSizes = &out_size;
+        out_buf.bufElSizes = &out_elem_size;
+
+        if ((err = aacEncEncode(handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) {
+            if (err == AACENC_ENCODE_EOF)
+                break;
+            fprintf(stderr, "Encoding failed\n");
+            return 1;
+        }
+        if (out_args.numOutBytes == 0)
+            continue;
 #if 0
-		unsigned char au_start[6];
-		unsigned char* sfbuf = outbuf;
-		au_start[0] = 6;
-		au_start[1] = (*(sfbuf + 3) << 4) + ((*(sfbuf + 4)) >> 4);
-		au_start[2] = ((*(sfbuf + 4) & 0x0f) << 8) + *(sfbuf + 5);
-		fprintf (stderr, "au_start[0] = %d\n", au_start[0]);
-		fprintf (stderr, "au_start[1] = %d\n", au_start[1]);
-		fprintf (stderr, "au_start[2] = %d\n", au_start[2]);
+        unsigned char au_start[6];
+        unsigned char* sfbuf = outbuf;
+        au_start[0] = 6;
+        au_start[1] = (*(sfbuf + 3) << 4) + ((*(sfbuf + 4)) >> 4);
+        au_start[2] = ((*(sfbuf + 4) & 0x0f) << 8) + *(sfbuf + 5);
+        fprintf (stderr, "au_start[0] = %d\n", au_start[0]);
+        fprintf (stderr, "au_start[1] = %d\n", au_start[1]);
+        fprintf (stderr, "au_start[2] = %d\n", au_start[2]);
 #endif
 
-		int row, col;
-		char buf_to_rs_enc[110];
-		char rs_enc[10];
-		for(row=0; row < subchannel_index; row++) {
-			for(col=0;col < 110; col++) {
-				buf_to_rs_enc[col] = outbuf[subchannel_index * col + row];
-			}
-
-			encode_rs_char(rs_handler, buf_to_rs_enc, rs_enc);
-
-			for(col=110; col<120; col++) {
-				outbuf[subchannel_index * col + row] = rs_enc[col-110];
-				assert(subchannel_index * col + row < outbuf_size);
-			}
-		}
-
-		fwrite(outbuf, 1, /*out_args.numOutBytes*/ outbuf_size, out_fh);
-		//fprintf(stderr, "Written %d/%d bytes!\n", out_args.numOutBytes + row*10, outbuf_size);
-		if(out_args.numOutBytes + row*10 == outbuf_size)
-			fprintf(stderr, ".");
-
-//		if(frame > 10)
-//			break;
-		frame++;
-	}
-	free(input_buf);
-	free(convert_buf);
-	if(raw_input) {
-		fclose(in_fh);
-	} else {
-		wav_read_close(wav);
-	}
-	fclose(out_fh);
-	free_rs_char(rs_handler);
-
-	aacEncClose(&handle);
-
-	return 0;
+        int row, col;
+        char buf_to_rs_enc[110];
+        char rs_enc[10];
+        for(row=0; row < subchannel_index; row++) {
+            for(col=0;col < 110; col++) {
+                buf_to_rs_enc[col] = outbuf[subchannel_index * col + row];
+            }
+
+            encode_rs_char(rs_handler, buf_to_rs_enc, rs_enc);
+
+            for(col=110; col<120; col++) {
+                outbuf[subchannel_index * col + row] = rs_enc[col-110];
+                assert(subchannel_index * col + row < outbuf_size);
+            }
+        }
+
+        fwrite(outbuf, 1, /*out_args.numOutBytes*/ outbuf_size, out_fh);
+        //fprintf(stderr, "Written %d/%d bytes!\n", out_args.numOutBytes + row*10, outbuf_size);
+        if(out_args.numOutBytes + row*10 == outbuf_size)
+            fprintf(stderr, ".");
+
+//      if(frame > 10)
+//          break;
+        frame++;
+    }
+    free(input_buf);
+    free(convert_buf);
+    if(raw_input) {
+        fclose(in_fh);
+    } else {
+        wav_read_close(wav);
+    }
+    fclose(out_fh);
+    free_rs_char(rs_handler);
+
+    aacEncClose(&handle);
+
+    return 0;
 }