diff options
Diffstat (limited to 'src/dabplus-enc.cpp')
| -rw-r--r-- | src/dabplus-enc.cpp | 77 |
1 files changed, 73 insertions, 4 deletions
diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp index 5b2b403..953073d 100644 --- a/src/dabplus-enc.cpp +++ b/src/dabplus-enc.cpp @@ -17,6 +17,33 @@ * ------------------------------------------------------------------- */ +/*! \mainpage Introduction + * The ODR-mmbTools FDK-AAC-DABplus Audio encoder can encode audio for + * ODR-DabMux, both DAB and DAB+. The DAB encoder is based on toolame. The + * DAB+ encoder uses a modified library of the Fraunhofer FDK AAC code from + * Android, patched for 960-transform to do DAB+ broadcast encoding. + * + * This document describes some internals of the encoder, and is intended + * to help developers understand and improve the software package. + * + * User documentation is available in the README and in the ODR-mmbTools + * Guide, available on the www.opendigitalradio.org website. + * + * The readme for the whole package is \ref md_README + * + * Interesting starting points for the encoder + * - \ref dabplus-enc.cpp + * - \ref VLC Input + * - \ref Alsa Input + * - \ref JACK Input + * - \ref SampleQueue + * - \ref charset.h + * - \ref libtoolame API + * + * For the mot-encoder: + * - \ref mot-encoder.cpp + */ + #include "config.h" #include "AlsaInput.h" #include "FileInput.h" @@ -56,7 +83,7 @@ extern "C" { -// Enumerate which encoder we can use +//! Enumeration of encoders we can use enum class encoder_selection_t { fdk_dabplus, toolame_dab @@ -166,6 +193,10 @@ void usage(const char* name) { } +/*! Setup the FDK AAC encoder + * + * \return 0 on success + */ int prepare_aac_encoder( HANDLE_AACENCODER *encoder, int subchannel_index, @@ -262,6 +293,9 @@ int prepare_aac_encoder( chrono::steady_clock::time_point timepoint_last_compensation; +/*! Wait the proper amount of time to throttle down to nominal encoding + * rate, if drift compensation is enabled. + */ void drift_compensation_delay(int sample_rate, int channels, size_t bytes) { const size_t bytes_per_second = sample_rate * BYTES_PER_SAMPLE * channels; @@ -726,6 +760,9 @@ int main(int argc, char *argv[]) int max_size = 8*input_buf.size() + NUM_SAMPLES_PER_CALL; + /*! The SampleQueue \c queue is given to the inputs, so that they + * can fill it. + */ SampleQueue<uint8_t> queue(BYTES_PER_SAMPLE, channels, max_size); /* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */ @@ -867,6 +904,18 @@ int main(int argc, char *argv[]) memset(&outbuf[0], 0x00, outbuf_size); memset(&input_buf[0], 0x00, input_buf.size()); + /*! \section Data input + * We read data input either in a blocking way (file input, VLC or ALSA + * without drift compensation) or in a non-blocking way (VLC or ALSA + * with drift compensation, JACK). + * + * The file input doesn't need the queue at all. But the other inputs + * do, and either use \c pop() or \c pop_wait() depending on if it's blocking or not + * + * In non-blocking, the \c queue makes the data available without delay, and the + * \c drift_compensation_delay() function handles rate throttling. + */ + if (infile) { read_bytes = file_in.read(&input_buf[0], input_buf.size()); if (read_bytes < 0) { @@ -911,6 +960,10 @@ int main(int argc, char *argv[]) else { const int timeout_ms = 1000; read_bytes = input_buf.size(); + + /*! pop_wait() must return after a timeout, otherwise the silence detector cannot do + * its job. + */ size_t bytes_from_queue = queue.pop_wait(&input_buf[0], read_bytes, timeout_ms); // returns bytes if (bytes_from_queue < read_bytes) { @@ -960,6 +1013,12 @@ int main(int argc, char *argv[]) #endif } + /*! Audio level measurement is always done assuming we have two + * channels, and is formally wrong in mono, but still gives + * numbers one can use. + * + * \todo fix level measurement in mono + */ for (int i = 0; i < read_bytes; i+=4) { int16_t l = input_buf[i] | (input_buf[i+1] << 8); int16_t r = input_buf[i+2] | (input_buf[i+3] << 8); @@ -967,7 +1026,12 @@ int main(int argc, char *argv[]) peak_right = MAX(peak_right, r); } - /* Silence detection */ + /*! Silence detection, looks at the audio level and is + * only useful if the connection dropped, or if no data is available. It is not + * useful if the source is nearly silent (some noise present), because the + * threshold is 0, and not configurable. The rationale is that we want to + * guard against connection issues, not source level issues + */ if (die_on_silence && MAX(peak_left, peak_right) == 0) { const unsigned int frame_time_msec = 1000ul * read_bytes / (BYTES_PER_SAMPLE * channels * sample_rate); @@ -1050,6 +1114,9 @@ int main(int argc, char *argv[]) } } + /*! toolame expects the audio to be in another shape as + * we have in input_buf, and we need to convert first + */ short input_buffers[2][1152]; if (channels == 1) { @@ -1076,7 +1143,10 @@ int main(int argc, char *argv[]) } } - /* Check if the encoder has generated output data */ + /*! Check if the encoder has generated output data. + * DAB+ requires RS encoding, which is not done in ODR-DabMux and not necessary + * for DAB. + */ if (numOutBytes != 0 and selected_encoder == encoder_selection_t::fdk_dabplus) { @@ -1088,7 +1158,6 @@ int main(int argc, char *argv[]) } calls = 0; - // ----------- RS encoding int row, col; unsigned char buf_to_rs_enc[110]; unsigned char rs_enc[10]; |