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/* -*- c++ -*- */
/*
* Copyright 2017 Moritz Luca Schmid, Communications Engineering Lab (CEL) / Karlsruhe Institute of Technology (KIT).
*
* GNU Radio block written for gr-dab including the following third party elements:
* -QT-DAB: classes mp4Processor and faad-decoder except the reed-solomon class
* Copyright (C) 2013
* Jan van Katwijk (J.vanKatwijk@gmail.com)
* Lazy Chair Computing
* -KA9Q: the char-sized Reed-Solomon encoder and decoder of the libfec library
* (details on the license see /fec/LICENCE)
*
* This 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, or (at your option)
* any later version.
*
* This software 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 software; see the file COPYING. If not, write to
* the Free Software Foundation, Inc., 51 Franklin Street,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gnuradio/io_signature.h>
#include "mp4_decode_bs_impl.h"
#include <stdexcept>
#include <stdio.h>
#include <sstream>
#include <boost/format.hpp>
#include "neaacdec.h"
using namespace boost;
namespace gr {
namespace dab {
mp4_decode_bs::sptr
mp4_decode_bs::make(int bit_rate_n)
{
return gnuradio::get_initial_sptr
(new mp4_decode_bs_impl(bit_rate_n));
}
/*
* The private constructor
*/
mp4_decode_bs_impl::mp4_decode_bs_impl(int bit_rate_n)
: gr::block("mp4_decode_bs",
gr::io_signature::make(1, 1, sizeof(unsigned char)),
gr::io_signature::make(2, 2, sizeof(int16_t))),
d_bit_rate_n(bit_rate_n)
{
d_superframe_size = bit_rate_n * 110;
d_aacInitialized = false;
baudRate = 48000;
set_output_multiple(960 * 4); //TODO: right? baudRate*0.12 for output of one superframe
aacHandle = NeAACDecOpen();
//memset(d_aac_frame, 0, 960);
d_sample_rate = -1;
d_first = true;
}
/*
* Our virtual destructor.
*/
mp4_decode_bs_impl::~mp4_decode_bs_impl()
{
}
void
mp4_decode_bs_impl::forecast(int noutput_items, gr_vector_int &ninput_items_required)
{
ninput_items_required[0] = noutput_items; //TODO: how to calculate actual rate?
}
// returns aac channel configuration
int mp4_decode_bs_impl::get_aac_channel_configuration(int16_t m_mpeg_surround_config, uint8_t aacChannelMode)
{
switch (m_mpeg_surround_config) {
case 0: // no surround
return aacChannelMode ? 2 : 1;
case 1: // 5.1
return 6;
case 2: // 7.1
return 7;
default:
return -1;
}
}
bool mp4_decode_bs_impl::initialize(uint8_t dacRate,
uint8_t sbrFlag,
int16_t mpegSurround,
uint8_t aacChannelMode)
{
long unsigned int sample_rate;
uint8_t channels;
/* 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 =
dacRate ? (sbrFlag ? 6 : 3) :
(sbrFlag ? 8 : 5); // 24/48/16/32 kHz
int core_ch_config = get_aac_channel_configuration(mpegSurround,
aacChannelMode);
if (core_ch_config == -1) {
GR_LOG_ERROR(d_logger, "Unrecognized mpeg surround config (ignored)");
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 int init_result = NeAACDecInit2(aacHandle,
asc,
sizeof(asc),
&sample_rate,
&channels);
if (init_result != 0) {
/* If some error initializing occured, skip the file */
GR_LOG_ERROR(d_logger, "Error initializing decoding library");
NeAACDecClose(aacHandle);
return false;
}
return true;
}
void mp4_decode_bs_impl::handle_aac_frame(const uint8_t *v,
int16_t frame_length,
uint8_t dacRate,
uint8_t sbrFlag,
uint8_t mpegSurround,
uint8_t aacChannelMode,
int16_t *out_sample1,
int16_t *out_sample2)
{
// copy AU to process it
uint8_t au[2 * 960 + 10]; // sure, large enough
memcpy(au, v, frame_length);
memset(&au[frame_length], 0, 10);
if (((au[0] >> 5) & 07) == 4) {
int16_t count = au[1];
uint8_t buffer[count];
memcpy(buffer, &au[2], count);
uint8_t L0 = buffer[count - 1];
uint8_t L1 = buffer[count - 2];
// TODO: handle PADs
}
int tmp = MP42PCM(dacRate,
sbrFlag,
mpegSurround,
aacChannelMode,
au,
frame_length,
out_sample1,
out_sample2);
}
int16_t mp4_decode_bs_impl::MP42PCM(uint8_t dacRate,
uint8_t sbrFlag,
int16_t mpegSurround,
uint8_t aacChannelMode,
uint8_t buffer[],
int16_t bufferLength,
int16_t *out_sample1,
int16_t *out_sample2)
{
int16_t samples;
long unsigned int sample_rate;
int16_t *outBuffer;
NeAACDecFrameInfo hInfo;
uint8_t dummy[10000];
uint8_t channels;
// initialize AAC decoder at the beginning
if (!d_aacInitialized) {
if (!initialize(dacRate, sbrFlag, mpegSurround, aacChannelMode))
return 0;
d_aacInitialized = true;
GR_LOG_DEBUG(d_logger, "AAC initialized");
}
outBuffer = (int16_t *) NeAACDecDecode(aacHandle, &hInfo, buffer, bufferLength);
sample_rate = hInfo.samplerate;
samples = hInfo.samples;
if ((sample_rate == 24000) ||
(sample_rate == 32000) ||
(sample_rate == 48000) ||
(sample_rate != (long unsigned) baudRate)) {
baudRate = sample_rate;
}
d_sample_rate = sample_rate;
GR_LOG_DEBUG(d_logger, format("bytes consumed %d") % (int) (hInfo.bytesconsumed));
GR_LOG_DEBUG(d_logger,
format("sample_rate = %d, samples = %d, channels = %d, error = %d, sbr = %d") % sample_rate %
samples %
(int) (hInfo.channels) % (int) (hInfo.error) % (int) (hInfo.sbr));
channels = hInfo.channels;
if (hInfo.error != 0) {
fprintf(stderr, "Warning: %s\n",
faacDecGetErrorMessage(hInfo.error));
return 0;
}
// write samples to output buffer
if (channels == 2) {
// the 2 channels are transmitted intereleaved; each channel gets samples/2 PCM samples
for (int n = 0; n < samples / 2; n++) {
out_sample1[n + d_nsamples_produced] = (int16_t) outBuffer[n * 2];
out_sample2[n + d_nsamples_produced] = (int16_t) outBuffer[n * 2 + 1];
}
} else if (channels == 1) {
int16_t *buffer = (int16_t *) alloca(2 * samples);
int16_t i;
for (int n = 0; n < samples / 2; n++) {
// only 1 channel -> reproduce each sample to send it to a stereo output anyway
out_sample1[n + d_nsamples_produced] = (int16_t) outBuffer[n * 2];
out_sample2[n + d_nsamples_produced] = (int16_t) outBuffer[n * 2 + 1];
}
} else
GR_LOG_ERROR(d_logger, "Cannot handle these channels -> dump samples");
GR_LOG_DEBUG(d_logger, format("Produced %d PCM samples (for each channel)") % (samples / 2));
d_nsamples_produced += samples / 2;
return samples / 2;
}
/*! \brief CRC16 check
* CRC16 check according to ETSI EN 300 401
* @param msg data to check
* @param len length of dataword without the 2 bytes crc at the end
* @return true if CRC passed
*/
bool mp4_decode_bs_impl::crc16(const uint8_t *msg, int16_t len)
{
int i, j;
uint16_t accumulator = 0xFFFF;
uint16_t crc;
uint16_t genpoly = 0x1021;
for (i = 0; i < len; i++) {
int16_t data = msg[i] << 8;
for (j = 8; j > 0; j--) {
if ((data ^ accumulator) & 0x8000)
accumulator = ((accumulator << 1) ^ genpoly) & 0xFFFF;
else
accumulator = (accumulator << 1) & 0xFFFF;
data = (data << 1) & 0xFFFF;
}
}
// compare calculated CRC with CRC in the AU
crc = ~((msg[len] << 8) | msg[len + 1]) & 0xFFFF;
return (crc ^ accumulator) == 0;
}
uint16_t mp4_decode_bs_impl::BinToDec(const uint8_t *data, size_t offset, size_t length)
{
uint32_t output = (*(data + offset / 8) << 16) | ((*(data + offset / 8 + 1)) << 8) |
(*(data + offset / 8 + 2)); // should be big/little endian save
output >>= 24 - length - offset % 8;
output &= (0xFFFF >> (16 - length));
return static_cast<uint16_t>(output);
}
int
mp4_decode_bs_impl::general_work(int noutput_items,
gr_vector_int &ninput_items,
gr_vector_const_void_star &input_items,
gr_vector_void_star &output_items)
{
const unsigned char *in = (const unsigned char *) input_items[0] + d_superframe_size;
int16_t *out1 = (int16_t *) output_items[0];
int16_t *out2 = (int16_t *) output_items[1];
d_nsamples_produced = 0;
if (d_first) {
add_item_tag(0, nitems_written(0), pmt::mp("audio_start"), pmt::PMT_NIL);
d_first = false;
}
for (int n = 0; n < noutput_items / (960 * 4); n++) {
// process superframe header
// bits 0 .. 15 is firecode
// bit 16 is unused
d_dac_rate = (in[n * d_superframe_size + 2] >> 6) & 01; // bit 17
d_sbr_flag = (in[n * d_superframe_size + 2] >> 5) & 01; // bit 18
d_aac_channel_mode = (in[n * d_superframe_size + 2] >> 4) & 01; // bit 19
d_ps_flag = (in[n * d_superframe_size + 2] >> 3) & 01; // bit 20
d_mpeg_surround = (in[n * d_superframe_size + 2] & 07); // bits 21 .. 23
// log header information
GR_LOG_DEBUG(d_logger,
format("superframe header: dac_rate %d, sbr_flag %d, aac_mode %d, ps_flag %d, surround %d") %
(int) d_dac_rate % (int) d_sbr_flag % (int) d_aac_channel_mode % (int) d_ps_flag %
(int) d_mpeg_surround);
switch (2 * d_dac_rate + d_sbr_flag) {
default: // cannot happen
case 0:
d_num_aus = 4;
d_au_start[0] = 8;
d_au_start[1] = in[n * d_superframe_size + 3] * 16 + (in[n * d_superframe_size + 4] >> 4);
d_au_start[2] = (in[n * d_superframe_size + 4] & 0xf) * 256 + in[n * d_superframe_size + 5];
d_au_start[3] = in[n * d_superframe_size + 6] * 16 + (in[n * d_superframe_size + 7] >> 4);
d_au_start[4] = d_superframe_size;
break;
case 1:
d_num_aus = 2;
d_au_start[n * d_superframe_size + 0] = 5;
d_au_start[1] = in[n * d_superframe_size + 3] * 16 + (in[n * d_superframe_size + 4] >> 4);
d_au_start[2] = d_superframe_size;
break;
case 2:
d_num_aus = 6;
d_au_start[0] = 11;
d_au_start[1] = in[n * d_superframe_size + 3] * 16 + (in[n * d_superframe_size + 4] >> 4);
d_au_start[2] = (in[n * d_superframe_size + 4] & 0xf) * 256 + in[n * d_superframe_size + 5];
d_au_start[3] = in[n * d_superframe_size + 6] * 16 + (in[n * d_superframe_size + 7] >> 4);
d_au_start[4] = (in[n * d_superframe_size + 7] & 0xf) * 256 + in[8];
d_au_start[5] = in[n * d_superframe_size + 9] * 16 + (in[n * d_superframe_size + 10] >> 4);
d_au_start[6] = d_superframe_size;
break;
case 3:
d_num_aus = 3;
d_au_start[0] = 6;
d_au_start[1] = in[n * d_superframe_size + 3] * 16 + (in[n * d_superframe_size + 4] >> 4);
d_au_start[2] = (in[n * d_superframe_size + 4] & 0xf) * 256 + in[n * d_superframe_size + 5];
d_au_start[3] = d_superframe_size;
break;
}
// each of the d_num_aus AUs of each superframe (110 * d_bit_rate_n packed bytes) is now processed separately
for (int i = 0; i < d_num_aus; i++) {
int16_t aac_frame_length;
// sanity check for the address
if (d_au_start[i + 1] < d_au_start[i]) {
throw std::runtime_error("AU start address invalid");
// should not happen, the header is firecode checked
}
aac_frame_length = d_au_start[i + 1] - d_au_start[i] - 2;
// sanity check for the aac_frame_length // FIXME: Causes crash after running for a long time. Ignore and continue instead of throwing exception
if ((aac_frame_length >= 960) || (aac_frame_length < 0)) {
throw std::out_of_range((boost::format("aac frame length not in range (%d)") % aac_frame_length).str());
}
// CRC check of each AU (the 2 byte (16 bit) CRC word is excluded in aac_frame_length)
if (crc16(&in[n * d_superframe_size + d_au_start[i]], aac_frame_length)) {
GR_LOG_DEBUG(d_logger, format("CRC check of AU %d successful") % i);
// handle proper AU
handle_aac_frame(&in[n * d_superframe_size + d_au_start[i]],
aac_frame_length,
d_dac_rate,
d_sbr_flag,
d_mpeg_surround,
d_aac_channel_mode,
out1,
out2);
} else {
// dump corrupted AU
GR_LOG_DEBUG(d_logger, format("CRC failure with dab+ frame"));
}
}
}
// Tell runtime system how many input items we consumed on
// each input stream.
consume_each(noutput_items * d_superframe_size / (960 * 4));
// Tell runtime system how many output items we produced.
return d_nsamples_produced;
}
} /* namespace dab */
} /* namespace gr */
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