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author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2016-02-15 04:32:00 +0100 |
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committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2016-02-15 04:32:00 +0100 |
commit | ba346d2469facf500cbcaa9cf9117ce04ea0b6da (patch) | |
tree | fa1e026da22296f8b044c8960c5860377c4fd6e2 /src | |
parent | e65ff4adee3a806881e3c1ebe1b273b9b664eb26 (diff) | |
download | fdk-aac-dabplus-ba346d2469facf500cbcaa9cf9117ce04ea0b6da.tar.gz fdk-aac-dabplus-ba346d2469facf500cbcaa9cf9117ce04ea0b6da.tar.bz2 fdk-aac-dabplus-ba346d2469facf500cbcaa9cf9117ce04ea0b6da.zip |
Use libtoolame-dab in dabplus-enc
Diffstat (limited to 'src')
-rw-r--r-- | src/dabplus-enc.cpp | 287 | ||||
-rw-r--r-- | src/utils.h | 1 |
2 files changed, 191 insertions, 97 deletions
diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp index d0130fd..8abc82a 100644 --- a/src/dabplus-enc.cpp +++ b/src/dabplus-enc.cpp @@ -46,16 +46,23 @@ extern "C" { extern "C" { #include <fec.h> +#include "libtoolame-dab/toolame.h" } +// Enumerate which encoder we can use +enum class encoder_selection_t { + fdk_dabplus, + toolame_dab +}; + using namespace std; void usage(const char* name) { fprintf(stderr, "dabplus-enc %s is a HE-AACv2 encoder for DAB+\n" - "based on fdk-aac-dabplus that can read from" - "JACK, ALSA or a file source\n" - "and encode to a ZeroMQ output for ODR-DabMux.\n" + "based on fdk-aac-dabplus and a Toolame-based MPEG\n" + "encoder for DAB that can read from JACK, ALSA or\n" + "a file source and encode to a ZeroMQ output for ODR-DabMux.\n" "(Experimental!)It can also use libvlc as an input.\n" "\n" "The -D option enables experimental sound card clock drift compensation.\n" @@ -110,6 +117,7 @@ void usage(const char* name) { " Drift compensation\n" " -D, --drift-comp Enable ALSA/VLC sound card drift compensation.\n" " Encoder parameters:\n" + " -a, --dab Encode in DAB and not in DAB+.\n" " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be a multiple of 8.\n" " -A, --no-afterburner Disable AAC encoder quality increaser.\n" " -c, --channels={ 1, 2 } Nb of input channels (default: 2).\n" @@ -240,9 +248,11 @@ int prepare_aac_encoder( int main(int argc, char *argv[]) { - int subchannel_index = 8; //64kbps subchannel + int bitrate = 64; //64kbps subchannel int ch=0; + encoder_selection_t selected_encoder = encoder_selection_t::fdk_dabplus; + // For the ALSA input const char *alsa_device = NULL; @@ -314,7 +324,7 @@ int main(int argc, char *argv[]) {"vlc-uri", required_argument, 0, 'v'}, {"write-icy-text", required_argument, 0, 'w'}, {"aaclc", no_argument, 0, 0 }, - {"afterburner", no_argument, 0, 'a'}, + {"dab", no_argument, 0, 'a'}, {"drift-comp", no_argument, 0, 'D'}, {"fifo-silence", no_argument, 0, 3 }, {"help", no_argument, 0, 'h'}, @@ -361,13 +371,13 @@ int main(int argc, char *argv[]) inFifoSilence = true; break; case 'a': - fprintf(stderr, "Warning, -a option does not exist anymore!\n"); + selected_encoder = encoder_selection_t::toolame_dab; break; case 'A': afterburner = false; break; case 'b': - subchannel_index = atoi(optarg) / 8; + bitrate = atoi(optarg); break; case 'c': channels = atoi(optarg); @@ -464,15 +474,25 @@ int main(int argc, char *argv[]) return 1; } - if (subchannel_index < 1 || subchannel_index > 24) { - fprintf(stderr, "Bad subchannel index: %d, must be between 1 and 24. Try other bitrate.\n", - subchannel_index); - return 1; - } + if (selected_encoder == encoder_selection_t::fdk_dabplus) { + int subchannel_index = bitrate / 8; - if ( ! (sample_rate == 32000 || sample_rate == 48000)) { - fprintf(stderr, "Invalid sample rate. Possible values are: 32000, 48000.\n"); - return 1; + if (subchannel_index < 1 || subchannel_index > 24) { + fprintf(stderr, "Bad subchannel index: %d, must be between 1 and 24. Try other bitrate.\n", + subchannel_index); + return 1; + } + + if ( ! (sample_rate == 32000 || sample_rate == 48000)) { + fprintf(stderr, "Invalid sample rate. Possible values are: 32000, 48000.\n"); + return 1; + } + } + else if (selected_encoder == encoder_selection_t::toolame_dab) { + if ( ! (sample_rate == 24000 || sample_rate == 48000)) { + fprintf(stderr, "Invalid sample rate. Possible values are: 24000, 48000.\n"); + return 1; + } } if (padlen < 0) { @@ -554,12 +574,46 @@ int main(int argc, char *argv[]) } + std::vector<uint8_t> input_buf; + HANDLE_AACENCODER encoder; - if (prepare_aac_encoder(&encoder, subchannel_index, channels, - sample_rate, afterburner, &aot) != 0) { - fprintf(stderr, "Encoder preparation failed\n"); - return 1; + if (selected_encoder == encoder_selection_t::fdk_dabplus) { + int subchannel_index = bitrate / 8; + if (prepare_aac_encoder(&encoder, subchannel_index, channels, + sample_rate, afterburner, &aot) != 0) { + fprintf(stderr, "Encoder preparation failed\n"); + return 1; + } + + if (aacEncInfo(encoder, &info) != AACENC_OK) { + fprintf(stderr, "Unable to get the encoder info\n"); + return 1; + } + + // Each DAB+ frame will need input_size audio bytes + const int input_size = channels * BYTES_PER_SAMPLE * info.frameLength; + fprintf(stderr, "DAB+ Encoding: framelen=%d (%dB)\n", + info.frameLength, + input_size); + + input_buf.resize(input_size); + } + else if (selected_encoder == encoder_selection_t::toolame_dab) { + int err = toolame_init(); + + if (err == 0) { + toolame_set_bitrate(bitrate); + } + + if (err) { + fprintf(stderr, "libtoolame-dab init failed: %d\n", err); + return err; + } + + // TODO int toolame_set_pad(int pad_len); + + input_buf.resize(2 * 1152); } /* We assume that we need to call the encoder @@ -567,24 +621,13 @@ int main(int argc, char *argv[]) * frame. This information is used when the alsa drift compensation * is active */ - const int enc_calls_per_output = - (aot == AOT_DABPLUS_AAC_LC) ? sample_rate / 8000 : sample_rate / 16000; - + int enc_calls_per_output = 1; // Valid for libtoolame-dab - if (aacEncInfo(encoder, &info) != AACENC_OK) { - fprintf(stderr, "Unable to get the encoder info\n"); - return 1; + if (selected_encoder == encoder_selection_t::fdk_dabplus) { + enc_calls_per_output = (aot == AOT_DABPLUS_AAC_LC) ? sample_rate / 8000 : sample_rate / 16000; } - // Each DAB+ frame will need input_size audio bytes - const int input_size = channels * BYTES_PER_SAMPLE * info.frameLength; - fprintf(stderr, "DAB+ Encoding: framelen=%d (%dB)\n", - info.frameLength, - input_size); - - uint8_t input_buf[input_size]; - - int max_size = 8*input_size + NUM_SAMPLES_PER_CALL; + int max_size = 8*input_buf.size() + NUM_SAMPLES_PER_CALL; SampleQueue<uint8_t> queue(BYTES_PER_SAMPLE, channels, max_size); @@ -661,11 +704,22 @@ int main(int argc, char *argv[]) } } - int outbuf_size = subchannel_index*120; - uint8_t zmqframebuf[ZMQ_HEADER_SIZE + 24*120]; - zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)zmqframebuf; + int outbuf_size; + std::vector<uint8_t> zmqframebuf; + std::vector<uint8_t> outbuf; + if (selected_encoder == encoder_selection_t::fdk_dabplus) { + outbuf_size = bitrate/8*120; + outbuf.resize(24*120); + zmqframebuf.resize(ZMQ_HEADER_SIZE + 24*120); + } + else if (selected_encoder == encoder_selection_t::toolame_dab) { + outbuf_size = 3 * bitrate; + outbuf.resize(outbuf_size); + zmqframebuf.resize(ZMQ_HEADER_SIZE + outbuf_size); + } + + zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)&zmqframebuf[0]; - uint8_t outbuf[24*120]; unsigned char pad_buf[padlen + 1]; @@ -686,8 +740,6 @@ int main(int argc, char *argv[]) int out_identifier = OUT_BITSTREAM_DATA; AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 }; - AACENC_InArgs in_args = { 0 }; - AACENC_OutArgs out_args = { 0 }; void *in_ptr[2], *out_ptr; int in_size[2], in_elem_size[2]; int out_size, out_elem_size; @@ -750,20 +802,20 @@ int main(int argc, char *argv[]) } // -------------- Read Data - memset(outbuf, 0x00, outbuf_size); - memset(input_buf, 0x00, input_size); + memset(&outbuf[0], 0x00, outbuf_size); + memset(&input_buf[0], 0x00, input_buf.size()); ssize_t read; if (infile) { - read = file_in.read(input_buf, input_size); + read = file_in.read(&input_buf[0], input_buf.size()); if (read < 0) { break; } - else if (read != input_size) { + else if (read != input_buf.size()) { if (inFifoSilence && file_in.eof()) { - memset(input_buf, 0, input_size); - read = input_size; - usleep((long)input_size * 1000000 / + memset(&input_buf[0], 0, input_buf.size()); + read = input_buf.size(); + usleep((long)input_buf.size() * 1000000 / (BYTES_PER_SAMPLE * channels * sample_rate)); } else { @@ -786,9 +838,9 @@ int main(int argc, char *argv[]) } size_t overruns; - read = queue.pop(input_buf, input_size, &overruns); // returns bytes + read = queue.pop(input_buf, input_buf.size(), &overruns); // returns bytes - if (read != input_size) { + if (read != input_buf.size()) { status |= STATUS_UNDERRUN; } @@ -799,12 +851,12 @@ int main(int argc, char *argv[]) else { vlc_in = &vlc_in_direct; - read = vlc_in_direct.read(input_buf, input_size); + read = vlc_in_direct.read(input_buf, input_buf.size()); if (read < 0) { fprintf(stderr, "Detected fault in VLC input!\n"); break; } - else if (read != input_size) { + else if (read != input_buf.size()) { fprintf(stderr, "Short VLC read !\n"); break; } @@ -823,9 +875,9 @@ int main(int argc, char *argv[]) } size_t overruns; - read = queue.pop(input_buf, input_size, &overruns); // returns bytes + read = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes - if (read != input_size) { + if (read != input_buf.size()) { status |= STATUS_UNDERRUN; } @@ -834,11 +886,11 @@ int main(int argc, char *argv[]) } } else { - read = alsa_in_direct.read(input_buf, input_size); + read = alsa_in_direct.read(&input_buf[0], input_buf.size()); if (read < 0) { break; } - else if (read != input_size) { + else if (read != input_buf.size()) { fprintf(stderr, "Short alsa read !\n"); } } @@ -869,50 +921,80 @@ int main(int argc, char *argv[]) measured_silence_ms = 0; } - // -------------- AAC Encoding - - int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0; - - - in_ptr[0] = input_buf; - in_ptr[1] = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes - in_size[0] = read; - in_size[1] = calculated_padlen; - in_elem_size[0] = BYTES_PER_SAMPLE; - in_elem_size[1] = sizeof(uint8_t); - in_args.numInSamples = input_size/BYTES_PER_SAMPLE; - in_args.numAncBytes = calculated_padlen; + int numOutBytes = 0; + if (selected_encoder == encoder_selection_t::fdk_dabplus) { + AACENC_InArgs in_args = { 0 }; + AACENC_OutArgs out_args = { 0 }; + // -------------- AAC Encoding + + const int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0; + const int subchannel_index = bitrate / 8; + + in_ptr[0] = &input_buf[0]; + in_ptr[1] = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes + in_size[0] = read; + in_size[1] = calculated_padlen; + in_elem_size[0] = BYTES_PER_SAMPLE; + in_elem_size[1] = sizeof(uint8_t); + in_args.numInSamples = input_buf.size()/BYTES_PER_SAMPLE; + in_args.numAncBytes = calculated_padlen; + + in_buf.bufs = (void**)&in_ptr; + in_buf.bufferIdentifiers = in_identifier; + in_buf.bufSizes = in_size; + in_buf.bufElSizes = in_elem_size; + + out_ptr = &outbuf[0]; + out_size = outbuf.size(); + 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; + + AACENC_ERROR err; + if ((err = aacEncEncode(encoder, &in_buf, &out_buf, &in_args, &out_args)) + != AACENC_OK) { + if (err == AACENC_ENCODE_EOF) { + fprintf(stderr, "encoder error: EOF reached\n"); + break; + } + fprintf(stderr, "Encoding failed (%d)\n", err); + retval = 3; + break; + } + calls++; - in_buf.bufs = (void**)&in_ptr; - in_buf.bufferIdentifiers = in_identifier; - in_buf.bufSizes = in_size; - in_buf.bufElSizes = in_elem_size; + numOutBytes = out_args.numOutBytes; + } + else if (selected_encoder == encoder_selection_t::toolame_dab) { + const int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0; + uint8_t *xpad_data = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes - 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; + short input_buffers[2][1152]; - AACENC_ERROR err; - if ((err = aacEncEncode(encoder, &in_buf, &out_buf, &in_args, &out_args)) - != AACENC_OK) { - if (err == AACENC_ENCODE_EOF) { - fprintf(stderr, "encoder error: EOF reached\n"); - break; + if (channels == 1) { + memcpy(input_buffers[0], &input_buf[0], 1152); } - fprintf(stderr, "Encoding failed (%d)\n", err); - retval = 3; - break; + else if (channels == 2) { + for (int ch = 0; ch < 2; ch++) { + for (int i = 0; i < 1152; i++) { + input_buffers[ch][i] = input_buf[2*i + ch]; + } + } + } + else { + fprintf(stderr, "INTERNAL ERROR! invalid number of channels\n"); + } + + toolame_encode_frame(input_buffers, xpad_data, &outbuf[0]); } - calls++; /* Check if the encoder has generated output data */ - if (out_args.numOutBytes != 0) - { + if (numOutBytes != 0 and + selected_encoder == encoder_selection_t::fdk_dabplus) { + // Our timing code depends on this if (calls != enc_calls_per_output) { fprintf(stderr, "INTERNAL ERROR! calls=%d" @@ -925,6 +1007,7 @@ int main(int argc, char *argv[]) int row, col; unsigned char buf_to_rs_enc[110]; unsigned char rs_enc[10]; + const int subchannel_index = bitrate / 8; for(row=0; row < subchannel_index; row++) { for(col=0;col < 110; col++) { buf_to_rs_enc[col] = outbuf[subchannel_index * col + row]; @@ -937,25 +1020,32 @@ int main(int argc, char *argv[]) assert(subchannel_index * col + row < outbuf_size); } } + } + if (numOutBytes != 0) { if (out_fh) { - fwrite(outbuf, 1, outbuf_size, out_fh); + fwrite(&outbuf[0], 1, outbuf_size, out_fh); } else { // ------------ ZeroMQ transmit try { zmq_frame_header->version = 1; - zmq_frame_header->encoder = ZMQ_ENCODER_FDK; + if (selected_encoder == encoder_selection_t::fdk_dabplus) { + zmq_frame_header->encoder = ZMQ_ENCODER_FDK; + } + else if (selected_encoder == encoder_selection_t::toolame_dab) { + zmq_frame_header->encoder = ZMQ_ENCODER_TOOLAME; + } zmq_frame_header->datasize = outbuf_size; zmq_frame_header->audiolevel_left = peak_left; zmq_frame_header->audiolevel_right = peak_right; - assert(ZMQ_FRAME_SIZE(zmq_frame_header) <= NUMOF(zmqframebuf)); + assert(ZMQ_FRAME_SIZE(zmq_frame_header) <= zmqframebuf.size()); memcpy(ZMQ_FRAME_DATA(zmq_frame_header), - outbuf, outbuf_size); + &outbuf[0], outbuf_size); - zmq_sock.send(zmqframebuf, ZMQ_FRAME_SIZE(zmq_frame_header), + zmq_sock.send(&zmqframebuf[0], ZMQ_FRAME_SIZE(zmq_frame_header), ZMQ_DONTWAIT); } catch (zmq::error_t& e) { @@ -970,7 +1060,10 @@ int main(int argc, char *argv[]) break; } } + } + if (numOutBytes != 0) + { if (show_level) { if (channels == 1) { fprintf(stderr, "\rIn: [%-6s] %1s %1s %1s", diff --git a/src/utils.h b/src/utils.h index c75935f..a0ab1ae 100644 --- a/src/utils.h +++ b/src/utils.h @@ -35,6 +35,7 @@ struct zmq_frame_header_t } __attribute__ ((packed)); #define ZMQ_ENCODER_FDK 1 +#define ZMQ_ENCODER_TOOLAME 2 #define ZMQ_HEADER_SIZE sizeof(struct zmq_frame_header_t) |