/* ------------------------------------------------------------------ * Copyright (C) 2011 Martin Storsjo * * Licensed under the Apache License, Version 2.0 (the "License"); * 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 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either * express or implied. * See the License for the specific language governing permissions * and limitations under the License. * ------------------------------------------------------------------- */ #include #include #include #include #include #include #include "libAACenc/include/aacenc_lib.h" #include "wavreader.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" ); } #define no_argument 0 #define required_argument 1 #define optional_argument 2 const int bitrate_table[4][24] = { // kbps 8 16 24 32 40 48 56 64 72 80 88 96 104 112 120 128 136 144 152 160 168 176 184 192 /*16kHz*/ {6733, 14067, 21400, 28733, 36067, 43400, 50733, 58067, 65400, 72733, 80067, 87400, 94733, 102067, 109400, 116733, 124067, 131400, 138733, 146067, 153400, 160733, 168067, 175400}, /*24kHz*/ {6533, 13867, 21200, 28533, 35867, 43200, 50533, 57867, 65200, 72533, 79867, 87200, 94533, 101867, 109200, 116533, 123867, 131200, 138533, 145867, 153200, 160533, 167867, 175200}, /*32kHz*/ {6267, 13600, 20933, 28267, 35600, 42933, 50267, 57600, 64933, 72267, 79600, 86933, 94267, 101600, 108933, 116267, 123600, 130933, 138267, 145600, 152933, 160267, 167600, 174933}, /*48kHz*/ {5800, 13133, 20467, 27800, 35133, 42467, 49800, 57133, 64467, 71800, 79133, 86467, 93800, 101133, 108467, 115800, 123133, 130467, 137800, 145133, 152467, 159800, 167133, 174467} }; #define ADTS_HEADER_SIZE 7 #define ADTS_MPEG_ID 1 /* 0: MPEG-4, 1: MPEG-2 */ #define ADTS_MPEG_PROFILE 1 const int mpeg4audio_sample_rates[16] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350 }; int FindSRIndex(int sr) { int i; for (i = 0; i < 16; i++) { if (sr == mpeg4audio_sample_rates[i]) return i; } return 16 - 1; } void adts_hdr_up(char *buff, int size) { unsigned short len = size + ADTS_HEADER_SIZE; unsigned short buffer_fullness = 0x07FF; /* 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 */ /* 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 */ } 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, lp=0; uint8_t* input_buf; int16_t* convert_buf; int aot = AOT_DABPLUS_AAC_LC; int afterburner = 0, adts_output=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 't': adts_output = 1; break; 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 'l': lp = 1; 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; } int sr_idx = 3; switch (sample_rate) { case 16000: sr_idx=0; break; case 24000: sr_idx=1; break; case 32000: sr_idx=2; break; case 48000: sr_idx=3; break; } int bitrate = bitrate_table[sr_idx][subchannel_index]; if(channels == 2 && bitrate <= 44000) aot = AOT_DABPLUS_PS; else if((channels == 1 && bitrate <= 64000) || bitrate <= 96000) aot = AOT_DABPLUS_SBR; fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s\n", subchannel_index, aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "", aot == AOT_DABPLUS_SBR ? "HE-AAC" : "", aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : ""); if(lp) aot = AOT_MP2_AAC_LC; 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, (lp ? 1024 : 960)) != AACENC_OK) { fprintf(stderr, "Unable to set the GRANULE\n"); return 1; } // if (vbr) { // if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, vbr) != AACENC_OK) { // fprintf(stderr, "Unable to set the VBR bitrate mode\n"); // return 1; // } // } else { fprintf(stderr, "AAC bitrate set to: %d\n", bitrate); if (aacEncoder_SetParam(handle, AACENC_BITRATE, bitrate) != AACENC_OK) { fprintf(stderr, "Unable to set the bitrate\n"); return 1; } if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_MP4_RAW) != AACENC_OK) { fprintf(stderr, "Unable to set the RAW transmux\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, "AAC frame length: %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); unsigned char adtsbuf[ADTS_HEADER_SIZE]; if(adts_output) { int mpeg_id = ADTS_MPEG_ID; int profile = ADTS_MPEG_PROFILE; int srate_idx = 0; if(aot == AOT_DABPLUS_PS || aot == AOT_DABPLUS_SBR) { srate_idx = FindSRIndex(sample_rate/2); } else { srate_idx = FindSRIndex(sample_rate); } adtsbuf[0] = 0xFF; /* 8bits: syncword */ adtsbuf[1] = 0xF0; /* 4bits: syncword */ adtsbuf[1] |= mpeg_id << 3; /* 1bit: mpeg id = 0 */ /* 2bits: layer = 00 */ adtsbuf[1] |= 1; /* 1bit: protection absent (1 - means "no protection")*/ adtsbuf[2] = ((profile << 6) & 0xC0); /* 2bits: profile */ adtsbuf[2] |= ((srate_idx << 2) & 0x3C); /* 4b: sampling_frequency_index */ /* 1b: private = 0 */ adtsbuf[2] |= ((channels >> 2) & 0x1); /* 1b: channel_configuration */ adtsbuf[3] = ((channels << 6) & 0xC0); /* 2b: channel_configuration */ } int loops = 0; while (1) { 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; uint8_t outbuf[20480]; 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(adts_output) { adts_hdr_up(adtsbuf, out_args.numOutBytes); fwrite(adtsbuf, 1, 7, out_fh); } fwrite(outbuf, 1, out_args.numOutBytes, out_fh); fprintf(stderr, "Written %d bytes!\n", out_args.numOutBytes); } free(input_buf); free(convert_buf); fclose(in_fh); fclose(out_fh); aacEncClose(&handle); return 0; }