/* Copyright (C) 2014 Matthias P. Braendli (http://www.opendigitalradio.org) This program is free software: you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation, either version 3 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License along with this program. If not, see . faad_decoder.cpp Use libfaad to decode the AAC content of the DAB+ subchannel Authors: Matthias P. Braendli */ #include "faad_decoder.h" #include "wavfile.h" #include "utils.h" #include #include #include #include #include #include #include #include using namespace std; FaadDecoder::FaadDecoder() : m_data_len(0), m_fd(NULL), m_aac(NULL), m_initialised(false) { } void FaadDecoder::open(string filename, bool ps_flag, bool aac_channel_mode, bool dac_rate, bool sbr_flag, int mpeg_surround_config) { m_filename = filename; m_ps_flag = ps_flag; m_aac_channel_mode = aac_channel_mode; m_dac_rate = dac_rate; m_sbr_flag = sbr_flag; m_mpeg_surround_config = mpeg_surround_config; stringstream ss; ss << filename << ".aac"; m_aac = fopen(ss.str().c_str(), "w"); } bool FaadDecoder::decode(vector > aus) { /* ADTS header creation taken from SDR-J */ adts_fixed_header fh; adts_variable_header vh; fh.syncword = 0xfff; fh.id = 0; fh.layer = 0; fh.protection_absent = 1; fh.profile_objecttype = 0; // aac main - 1 fh.private_bit = 0; // ignored when decoding fh.original_copy = 0; fh.home = 0; vh.copyright_id_bit = 0; vh.copyright_id_start = 0; vh.adts_buffer_fullness = 1999; // ? according to OpenDab vh.no_raw_data_blocks = 0; uint8_t d_header[10]; d_header[0] = fh.syncword >> 4; d_header[1] = (fh.syncword & 0xf) << 4; d_header[1] |= fh.id << 3; d_header[1] |= fh.layer << 1; d_header[1] |= fh.protection_absent; d_header[2] = fh.profile_objecttype << 6; // sampling frequency index filled in dynamically d_header[2] |= fh.private_bit << 1; // channel configuration filled in dynamically d_header[3] = fh.original_copy << 5; d_header[3] |= fh.home << 4; d_header[3] |= vh.copyright_id_bit << 3; d_header[3] |= vh.copyright_id_start << 2; // framelength filled in dynamically d_header[4] = 0; d_header[5] = vh.adts_buffer_fullness >> 6; d_header[6] = (vh.adts_buffer_fullness & 0x3f) << 2; d_header[6] |= vh.no_raw_data_blocks; if (!m_dac_rate && m_sbr_flag) fh.sampling_freq_idx = 8; // AAC core sampling rate 16 kHz else if (m_dac_rate && m_sbr_flag) fh.sampling_freq_idx = 6; // AAC core sampling rate 24 kHz else if (!m_dac_rate && !m_sbr_flag) fh.sampling_freq_idx = 5; // AAC core sampling rate 32 kHz else if (m_dac_rate && !m_sbr_flag) fh.sampling_freq_idx = 3; // AAC core sampling rate 48 kHz setBits (&d_header[2], fh.sampling_freq_idx, 2, 4); if (m_mpeg_surround_config == 0) { if (m_sbr_flag && !m_aac_channel_mode && m_ps_flag) fh.channel_conf = 2; else fh.channel_conf = 1 << (m_aac_channel_mode ? 1 : 0); } else if (m_mpeg_surround_config == 1) { fh.channel_conf = 6; } else { printf("Unrecognized mpeg surround config (ignored)\n"); return false; } setBits (&d_header[2], fh.channel_conf, 7, 3); for (size_t au_ix = 0; au_ix < aus.size(); au_ix++) { vector& au = aus[au_ix]; uint8_t helpBuffer[960]; memset(helpBuffer, 0, sizeof(helpBuffer)); // Set length in header (header + au) vh.aac_frame_length = 7 + au.size(); setBits(&d_header[3], vh.aac_frame_length, 6, 13); memcpy(helpBuffer, d_header, 7 * sizeof(uint8_t)); memcpy(&helpBuffer[7], &au[0], au.size() * sizeof (uint8_t)); fwrite(helpBuffer, 1, vh.aac_frame_length, m_aac); NeAACDecFrameInfo hInfo; int16_t* outBuffer; if (!m_initialised) { long unsigned samplerate; unsigned char channels; int len; if ((len = NeAACDecInit(m_faad_handle.decoder, helpBuffer, vh.aac_frame_length, &samplerate, &channels)) < 0) { /* If some error initializing occured, skip the file */ printf("Error initializing decoder library (%d).\n", len); NeAACDecClose(m_faad_handle.decoder); return false; } m_initialised = true; outBuffer = (int16_t *)NeAACDecDecode( m_faad_handle.decoder, &hInfo, helpBuffer + len, vh.aac_frame_length - len ); } else { outBuffer = (int16_t *)NeAACDecDecode( m_faad_handle.decoder, &hInfo, helpBuffer, vh.aac_frame_length ); } assert(outBuffer != NULL); m_sample_rate = hInfo.samplerate; m_channels = hInfo.channels; size_t samples = hInfo.samples; #if 0 printf("bytes consumed %d\n", (int)(hInfo.bytesconsumed)); printf("samplerate = %d, samples = %zu, channels = %d," " error = %d, sbr = %d\n", m_sample_rate, samples, m_channels, hInfo.error, hInfo.sbr); printf("header = %d\n", hInfo.header_type); #endif if (hInfo.error != 0) { printf("FAAD Warning: %s\n", faacDecGetErrorMessage(hInfo.error)); return false; } if (m_fd == NULL) { stringstream ss; ss << m_filename << ".wav"; m_fd = wavfile_open(ss.str().c_str(), m_sample_rate); } if (samples) { if (m_channels == 1) { int16_t *buffer = (int16_t *)alloca (2 * samples); size_t i; for (i = 0; i < samples; i ++) { buffer [2 * i] = ((int16_t *)outBuffer) [i]; buffer [2 * i + 1] = buffer [2 * i]; } wavfile_write(m_fd, buffer, 2*samples); } else if (m_channels == 2) { wavfile_write(m_fd, outBuffer, samples); } else { printf("Cannot handle %d channels\n", m_channels); } } } return true; } void FaadDecoder::close() { if (m_initialised) { wavfile_close(m_fd); } }