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Diffstat (limited to 'src/faad_decoder.cpp')
| -rw-r--r-- | src/faad_decoder.cpp | 177 |
1 files changed, 177 insertions, 0 deletions
diff --git a/src/faad_decoder.cpp b/src/faad_decoder.cpp new file mode 100644 index 0000000..73ea374 --- /dev/null +++ b/src/faad_decoder.cpp @@ -0,0 +1,177 @@ +/* + 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 <http://www.gnu.org/licenses/>. + + faad_decoder.cpp + Use libfaad to decode the AAC content of the DAB+ subchannel + + Authors: + Matthias P. Braendli <matthias@mpb.li> +*/ + +#include "faad_decoder.hpp" +extern "C" { +#include "wavfile.h" +} +#include <stdio.h> +#include <stdint.h> +#include <stdlib.h> +#include <string.h> +#include <cassert> +#include <string> +#include <sstream> +#include <vector> + +using namespace std; + +FaadDecoder::FaadDecoder() : + m_data_len(0), + m_fd(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; +} + +bool FaadDecoder::decode(vector<vector<uint8_t> > aus) +{ + for (size_t au_ix = 0; au_ix < aus.size(); au_ix++) { + + vector<uint8_t>& au = aus[au_ix]; + + NeAACDecFrameInfo hInfo; + int16_t* outBuffer; + + if (!m_initialised) { + /* AudioSpecificConfig structure (the only way to select 960 transform here!) + * + * 00010 = AudioObjectType 2 (AAC LC) + * xxxx = (core) sample rate index + * xxxx = (core) channel config + * 100 = GASpecificConfig with 960 transform + * + * SBR: implicit signaling sufficient - libfaad2 automatically assumes SBR on sample rates <= 24 kHz + * => explicit signaling works, too, but is not necessary here + * + * PS: implicit signaling sufficient - libfaad2 therefore always uses stereo output (if PS support was enabled) + * => explicit signaling not possible, as libfaad2 does not support AudioObjectType 29 (PS) + */ + + int core_sr_index = m_dac_rate ? (m_sbr_flag ? 6 : 3) : (m_sbr_flag ? 8 : 5); // 24/48/16/32 kHz + int core_ch_config = get_aac_channel_configuration(); + if(core_ch_config == -1) { + printf("Unrecognized mpeg surround config (ignored): %d\n", m_mpeg_surround_config); + return false; + } + + uint8_t asc[2]; + asc[0] = 0b00010 << 3 | core_sr_index >> 1; + asc[1] = (core_sr_index & 0x01) << 7 | core_ch_config << 3 | 0b100; + + + long unsigned samplerate; + unsigned char channels; + + long int init_result = NeAACDecInit2(m_faad_handle.decoder, asc, sizeof(asc), &samplerate, &channels); + if(init_result != 0) { + /* If some error initializing occured, skip the file */ + printf("Error initializing decoder library: %s\n", NeAACDecGetErrorMessage(-init_result)); + NeAACDecClose(m_faad_handle.decoder); + return false; + } + + m_initialised = true; + } + + outBuffer = (int16_t *)NeAACDecDecode(m_faad_handle.decoder, &hInfo, &au[0], au.size()); + 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); + } +} + +int FaadDecoder::get_aac_channel_configuration() +{ + switch(m_mpeg_surround_config) { + case 0: // no surround + return m_aac_channel_mode ? 2 : 1; + case 1: // 5.1 + return 6; + case 2: // 7.1 + return 7; + default: + return -1; + } +} + + |