From a8bd9b19bba683031f6c7a68e9e6ca653be18d6c Mon Sep 17 00:00:00 2001 From: "Matthias P. Braendli" Date: Fri, 25 Apr 2014 17:32:03 +0200 Subject: merge file and alsa encoders into dabplus-enc There was a lot of redundant code between the two --- .gitignore | 1 + Makefile.am | 42 +-- src/AlsaInput.h | 18 +- src/FileInput.cpp | 108 +++++++ src/FileInput.h | 61 ++++ src/dabplus-enc-alsa-zmq.cpp | 635 -------------------------------------- src/dabplus-enc-file-zmq.c | 620 ------------------------------------- src/dabplus-enc.cpp | 716 +++++++++++++++++++++++++++++++++++++++++++ 8 files changed, 910 insertions(+), 1291 deletions(-) create mode 100644 src/FileInput.cpp create mode 100644 src/FileInput.h delete mode 100644 src/dabplus-enc-alsa-zmq.cpp delete mode 100644 src/dabplus-enc-file-zmq.c create mode 100644 src/dabplus-enc.cpp diff --git a/.gitignore b/.gitignore index 2014f3a..094130a 100644 --- a/.gitignore +++ b/.gitignore @@ -30,3 +30,4 @@ mot-encoder dabplus-enc-file dabplus-enc-file-zmq dabplus-enc-alsa-zmq +dabplus-enc-zmq diff --git a/Makefile.am b/Makefile.am index 7801cc1..8662153 100644 --- a/Makefile.am +++ b/Makefile.am @@ -43,31 +43,22 @@ libfdk_aac_la_LDFLAGS = -version-info @FDK_AAC_VERSION@ -no-undefined \ #aac_enc_SOURCES = src/aac-enc.c \ # src/wavreader.c -dabplus_enc_file_zmq_LDADD = libfdk-aac.la -lfec -lzmq -dabplus_enc_file_zmq_CFLAGS = $(AM_CPPFLAGS) $(GITVERSION_FLAGS) -dabplus_enc_file_zmq_SOURCES = src/dabplus-enc-file-zmq.c \ - src/wavreader.c \ - src/encryption.c \ - src/encryption.h \ - src/utils.c \ - src/utils.h \ - contrib/lib_crc.h \ - contrib/lib_crc.c - -dabplus_enc_alsa_zmq_LDFLAGS = -no-install -dabplus_enc_alsa_zmq_LDADD = libfdk-aac.la -lfec -lzmq -lasound \ +dabplus_enc_LDFLAGS = -no-install +dabplus_enc_LDADD = libfdk-aac.la -lfec -lzmq -lasound \ -lrt -lboost_thread -dabplus_enc_alsa_zmq_CPPFLAGS = $(AM_CPPFLAGS) $(GITVERSION_FLAGS) -dabplus_enc_alsa_zmq_SOURCES = src/dabplus-enc-alsa-zmq.cpp \ - src/AlsaInput.cpp \ - src/AlsaInput.h \ - src/SampleQueue.h \ - src/encryption.c \ - src/encryption.h \ - src/zmq.hpp \ - src/utils.c \ - src/utils.h - +dabplus_enc_CPPFLAGS = $(AM_CPPFLAGS) $(GITVERSION_FLAGS) +dabplus_enc_SOURCES = src/dabplus-enc.cpp \ + src/FileInput.cpp \ + src/FileInput.h \ + src/AlsaInput.cpp \ + src/AlsaInput.h \ + src/SampleQueue.h \ + src/encryption.c \ + src/encryption.h \ + src/zmq.hpp \ + src/utils.c \ + src/utils.h \ + src/wavreader.c mot_encoder_CFLAGS = $(GITVERSION_FLAGS) @MAGICKWAND_CFLAGS@ -Icontrib mot_encoder_LDADD = @MAGICKWAND_LDADD@ @@ -75,8 +66,7 @@ mot_encoder_SOURCES = src/mot-encoder.c \ contrib/lib_crc.h \ contrib/lib_crc.c -bin_PROGRAMS = dabplus-enc-file-zmq$(EXEEXT) \ - dabplus-enc-alsa-zmq$(EXEEXT) \ +bin_PROGRAMS = dabplus-enc$(EXEEXT) \ mot-encoder$(EXEEXT) noinst_HEADERS = src/wavreader.h diff --git a/src/AlsaInput.h b/src/AlsaInput.h index 0d454a3..bd3a800 100644 --- a/src/AlsaInput.h +++ b/src/AlsaInput.h @@ -27,18 +27,19 @@ #include "SampleQueue.h" -using namespace std; - // 16 bits per sample is fine for now #define BYTES_PER_SAMPLE 2 // How many samples we insert into the queue each call #define NUM_SAMPLES_PER_CALL 10 // 10 samples @ 32kHz = 3.125ms +/* Common functionality for the direct alsa input and the + * threaded alsa input + */ class AlsaInput { public: - AlsaInput(const string& alsa_dev, + AlsaInput(const std::string& alsa_dev, unsigned int channels, unsigned int rate) : m_alsa_dev(alsa_dev), @@ -57,12 +58,10 @@ class AlsaInput /* Prepare the audio input */ int prepare(); - virtual void start() = 0; - protected: ssize_t m_read(uint8_t* buf, snd_pcm_uframes_t length); - string m_alsa_dev; + std::string m_alsa_dev; unsigned int m_channels; unsigned int m_rate; @@ -75,13 +74,11 @@ class AlsaInput class AlsaInputDirect : public AlsaInput { public: - AlsaInputDirect(const string& alsa_dev, + AlsaInputDirect(const std::string& alsa_dev, unsigned int channels, unsigned int rate) : AlsaInput(alsa_dev, channels, rate) { } - virtual void start() { }; - /* Read length Bytes from from the alsa device. * length must be a multiple of channels * bytes_per_sample. * @@ -97,7 +94,7 @@ class AlsaInputDirect : public AlsaInput class AlsaInputThreaded : public AlsaInput { public: - AlsaInputThreaded(const string& alsa_dev, + AlsaInputThreaded(const std::string& alsa_dev, unsigned int channels, unsigned int rate, SampleQueue& queue) : @@ -115,6 +112,7 @@ class AlsaInputThreaded : public AlsaInput } } + /* Start the ALSA thread that fills the queue */ virtual void start(); bool fault_detected() { return m_fault; }; diff --git a/src/FileInput.cpp b/src/FileInput.cpp new file mode 100644 index 0000000..c8023dd --- /dev/null +++ b/src/FileInput.cpp @@ -0,0 +1,108 @@ +/* ------------------------------------------------------------------ + * Copyright (C) 2014 Matthias P. Braendli + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either + * express or implied. + * See the License for the specific language governing permissions + * and limitations under the License. + * ------------------------------------------------------------------- + */ + +#include "FileInput.h" +#include "wavreader.h" +#include +#include + +#include + +int FileInput::prepare(void) +{ + if (m_raw_input) { + if (m_filename && strcmp(m_filename, "-")) { + m_in_fh = fopen(m_filename, "rb"); + if (!m_in_fh) { + fprintf(stderr, "Can't open input file!\n"); + return 1; + } + } + else { + m_in_fh = stdin; + } + } + else { + int bits_per_sample = 0; + int channels = 0; + int wav_format = 0; + int sample_rate = 0; + + m_wav = wav_read_open(m_filename); + if (!m_wav) { + fprintf(stderr, "Unable to open wav file %s\n", m_filename); + return 1; + } + if (!wav_get_header(m_wav, &wav_format, &channels, &sample_rate, + &bits_per_sample, NULL)) { + fprintf(stderr, "Bad wav file %s\n", m_filename); + return 1; + } + if (wav_format != 1) { + fprintf(stderr, "Unsupported WAV format %d\n", wav_format); + return 1; + } + if (bits_per_sample != 16) { + fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample); + return 1; + } + if (channels != 2) { + fprintf(stderr, "Unsupported WAV channels %d\n", channels); + return 1; + } + if (m_sample_rate != sample_rate) { + fprintf(stderr, + "WAV sample rate %d doesn't correspond to desired sample rate %d\n", + sample_rate, m_sample_rate); + return 1; + } + } + + return 0; +} + +ssize_t FileInput::read(uint8_t* buf, size_t length) +{ + ssize_t pcmread; + + if (m_raw_input) { + if (fread(buf, length, 1, m_in_fh) == 1) { + pcmread = length; + } + else { + fprintf(stderr, "Unable to read from input!\n"); + return 0; + } + } + else { + pcmread = wav_read_data(m_wav, buf, length); + } + + return pcmread; +} + +FileInput::~FileInput() +{ + if (m_raw_input && m_in_fh) { + fclose(m_in_fh); + } + else if (m_wav) { + wav_read_close(m_wav); + } +} + diff --git a/src/FileInput.h b/src/FileInput.h new file mode 100644 index 0000000..9330577 --- /dev/null +++ b/src/FileInput.h @@ -0,0 +1,61 @@ +/* ------------------------------------------------------------------ + * Copyright (C) 2014 Matthias P. Braendli + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either + * express or implied. + * See the License for the specific language governing permissions + * and limitations under the License. + * ------------------------------------------------------------------- + */ + +#ifndef _FILE_INPUT_H_ +#define _FILE_INPUT_H_ + +#include +#include + +class FileInput +{ + public: + FileInput(const char* filename, + bool raw_input, + int sample_rate) : + m_filename(filename), + m_raw_input(raw_input), + m_sample_rate(sample_rate) { } + + ~FileInput(); + + /* Open the file and prepare the wav decoder. + * + * Returns nonzero on error + */ + int prepare(void); + + /* Read length bytes into buf. + * + * Returns the number of bytes read. + */ + ssize_t read(uint8_t* buf, size_t length); + + protected: + const char* m_filename; + bool m_raw_input; + int m_sample_rate; + + /* handle to the wav reader */ + void *m_wav; + + FILE* m_in_fh; +}; + +#endif + diff --git a/src/dabplus-enc-alsa-zmq.cpp b/src/dabplus-enc-alsa-zmq.cpp deleted file mode 100644 index 699a256..0000000 --- a/src/dabplus-enc-alsa-zmq.cpp +++ /dev/null @@ -1,635 +0,0 @@ -/* ------------------------------------------------------------------ - * Copyright (C) 2011 Martin Storsjo - * Copyright (C) 2013,2014 Matthias P. Braendli - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either - * express or implied. - * See the License for the specific language governing permissions - * and limitations under the License. - * ------------------------------------------------------------------- - */ - -#include "AlsaInput.h" -#include "SampleQueue.h" -#include "zmq.hpp" - -extern "C" { -#include "encryption.h" -#include "utils.h" -} - - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "libAACenc/include/aacenc_lib.h" - -extern "C" { -#include -} - -using namespace std; - -void usage(const char* name) { - fprintf(stderr, - "dabplus-enc-alsa-zmq %s is a HE-AACv2 encoder for DAB+\n" - "based on fdk-aac-dabplus that can read from a ALSA source\n" - "and encode to a ZeroMQ output for ODR-DabMux.\n" - "\n" - "The -D option enables experimental sound card clock drift compensation.\n" - "A consumer sound card has a clock that is always a bit imprecise, and\n" - "would drift off after some time. ODR-DabMux cannot handle such drift\n" - "because it would have to throw away or insert a full DAB+ superframe,\n" - "which would create audible artifacts. This drift compensation can\n" - "make sure that the encoding rate is correct by inserting or deleting\n" - "audio samples.\n" - "\n" - "When this option is enabled, you will see U and O printed in\n" - "the console. These correspond to audio underruns and overruns caused\n" - "by sound card clock drift. When sparse, they should not create audible\n" - "artifacts.\n" - "\n" - "This encoder includes PAD (DLS and MOT Slideshow) support by\n" - "http://rd.csp.it to be used with mot-encoder\n" - "\n" - " http://opendigitalradio.org\n" - "\nUsage:\n" - "%s [OPTION...]\n", -#if defined(GITVERSION) - GITVERSION -#else - PACKAGE_VERSION -#endif - , name); - fprintf(stderr, - " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n" - " -D, --drift-comp Enable ALSA sound card drift compensation.\n" - //" -i, --input=FILENAME Input filename (default: stdin).\n" - " -o, --output=URI Output zmq uri. (e.g. 'tcp://*:9000')\n" - " -a, --afterburner Turn on AAC encoder quality increaser.\n" - " -p, --pad=BYTES Set PAD size in bytes.\n" - " -P, --pad-fifo=FILENAME Set PAD data input fifo name (default: /tmp/pad.fifo).\n" - " -d, --device=alsa_device Set ALSA input device (default: \"default\").\n" - " -c, --channels={ 1, 2 } Nb of input channels for raw input (default: 2).\n" - " -r, --rate={ 32000, 48000 } Sample rate for raw input (default: 48000).\n" - " -k, --secret-key=FILE Set the secret key for encryption.\n" - " -l, --level Show level indication.\n" - //" -V, --version Print version and exit.\n" - "\n" - "Only the tcp:// zeromq transport has been tested until now.\n" - ); - -} - -int prepare_aac_encoder( - HANDLE_AACENCODER *encoder, - int subchannel_index, - int channels, - int sample_rate, - int afterburner) -{ - HANDLE_AACENCODER handle = *encoder; - - int aot = AOT_DABPLUS_AAC_LC; - - CHANNEL_MODE mode; - switch (channels) { - case 1: mode = MODE_1; break; - case 2: mode = MODE_2; break; - default: - fprintf(stderr, "Unsupported channels number %d\n", channels); - return 1; - } - - - if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) { - fprintf(stderr, "Unable to open encoder\n"); - return 1; - } - - *encoder = handle; - - if(channels == 2 && subchannel_index <= 6) - aot = AOT_DABPLUS_PS; - else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10) - aot = AOT_DABPLUS_SBR; - - fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n", - subchannel_index, - aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "", - aot == AOT_DABPLUS_SBR ? "HE-AAC" : "", - aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "", - channels, sample_rate); - - if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) { - fprintf(stderr, "Unable to set the AOT\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) { - fprintf(stderr, "Unable to set the sample rate\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) { - fprintf(stderr, "Unable to set the channel mode\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) { - fprintf(stderr, "Unable to set the wav channel order\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) { - fprintf(stderr, "Unable to set the granule length\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) { - fprintf(stderr, "Unable to set the RAW transmux\n"); - return 1; - } - - /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) - * != AACENC_OK) { - fprintf(stderr, "Unable to set the bitrate mode\n"); - return 1; - }*/ - - - fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000); - if (aacEncoder_SetParam(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) { - fprintf(stderr, "Unable to set the bitrate\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) { - fprintf(stderr, "Unable to set the afterburner mode\n"); - return 1; - } - if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) { - fprintf(stderr, "Unable to initialize the encoder\n"); - return 1; - } - return 0; -} - -#define no_argument 0 -#define required_argument 1 -#define optional_argument 2 - -#define STATUS_PAD_INSERTED 0x1 -#define STATUS_OVERRUN 0x2 -#define STATUS_UNDERRUN 0x4 - -int main(int argc, char *argv[]) -{ - int subchannel_index = 8; //64kbps subchannel - int ch=0; - const char *alsa_device = "default"; - const char *outuri = NULL; - int sample_rate=48000, channels=2; - const int bytes_per_sample = 2; - void *rs_handler = NULL; - bool afterburner = false; - bool drift_compensation = false; - AACENC_InfoStruct info = { 0 }; - - /* Keep track of peaks */ - int peak_left = 0; - int peak_right = 0; - - - const char* pad_fifo = "/tmp/pad.fifo"; - int pad_fd; - unsigned char pad_buf[128]; - int padlen; - - int status = 0; - int show_level = 0; - - /* Data for ZMQ CURVE authentication */ - char* keyfile = NULL; - char secretkey[CURVE_KEYLEN+1]; - - const struct option longopts[] = { - {"bitrate", required_argument, 0, 'b'}, - {"output", required_argument, 0, 'o'}, - {"device", required_argument, 0, 'd'}, - {"rate", required_argument, 0, 'r'}, - {"channels", required_argument, 0, 'c'}, - {"pad", required_argument, 0, 'p'}, - {"pad-fifo", required_argument, 0, 'P'}, - {"secret-key", required_argument, 0, 'k'}, - {"drift-comp", no_argument, 0, 'D'}, - {"afterburner", no_argument, 0, 'a'}, - {"help", no_argument, 0, 'h'}, - {"level", no_argument, 0, 'l'}, - {0,0,0,0}, - }; - - if (argc < 2) { - usage(argv[0]); - return 1; - } - - int index; - while(ch != -1) { - ch = getopt_long(argc, argv, "ahDlb:c:k:o:r:d:p:P:", longopts, &index); - switch (ch) { - case 'd': - alsa_device = optarg; - break; - case 'a': - afterburner = true; - break; - case 'b': - subchannel_index = atoi(optarg) / 8; - break; - case 'c': - channels = atoi(optarg); - break; - case 'r': - sample_rate = atoi(optarg); - break; - case 'o': - outuri = optarg; - break; - case 'k': - keyfile = optarg; - break; - case 'D': - drift_compensation = true; - break; - case 'p': - padlen = atoi(optarg); - break; - case 'P': - pad_fifo = optarg; - break; - case 'l': - show_level = 1; - break; - case '?': - case 'h': - usage(argv[0]); - return 1; - } - } - - if(subchannel_index < 1 || subchannel_index > 24) { - fprintf(stderr, "Bad subchannels number: %d, 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; - } - - const int enc_calls_per_output = sample_rate / 16000; - - if (!outuri) { - fprintf(stderr, "ZeroMQ output URI not defined\n"); - return 1; - } - - if (padlen != 0) { - int flags; - if (mkfifo(pad_fifo, S_IWUSR | S_IRUSR | S_IRGRP | S_IROTH) != 0) { - if (errno != EEXIST) { - fprintf(stderr, "Can't create pad file: %d!\n", errno); - return 1; - } - } - pad_fd = open(pad_fifo, O_RDONLY | O_NONBLOCK); - if (pad_fd == -1) { - fprintf(stderr, "Can't open pad file!\n"); - return 1; - } - flags = fcntl(pad_fd, F_GETFL, 0); - if (fcntl(pad_fd, F_SETFL, flags | O_NONBLOCK)) { - fprintf(stderr, "Can't set non-blocking mode in pad file!\n"); - return 1; - } - } - - zmq::context_t zmq_ctx; - zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB); - - if (keyfile) { - fprintf(stderr, "Enabling encryption\n"); - - int rc = readkey(keyfile, secretkey); - if (rc) { - fprintf(stderr, "Error reading secret key\n"); - return 2; - } - - const int yes = 1; - zmq_sock.setsockopt(ZMQ_CURVE_SERVER, - &yes, sizeof(yes)); - - zmq_sock.setsockopt(ZMQ_CURVE_SECRETKEY, - secretkey, CURVE_KEYLEN); - } - zmq_sock.connect(outuri); - - HANDLE_AACENCODER encoder; - - if (prepare_aac_encoder(&encoder, subchannel_index, channels, - sample_rate, afterburner) != 0) { - fprintf(stderr, "Encoder preparation failed\n"); - return 2; - } - - 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); - - uint8_t input_buf[input_size]; - - int max_size = 2*input_size + NUM_SAMPLES_PER_CALL; - - SampleQueue queue(BYTES_PER_SAMPLE, channels, max_size); - - /* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */ - rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135); - if (rs_handler == NULL) { - perror("init_rs_char failed"); - return 1; - } - - // We'll use either of the two possible alsa inputs. - AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue); - AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate); - - if (drift_compensation) { - if (alsa_in_threaded.prepare() != 0) { - fprintf(stderr, "Alsa preparation failed\n"); - return 1; - } - - fprintf(stderr, "Start ALSA capture thread\n"); - alsa_in_threaded.start(); - } - else { - if (alsa_in_direct.prepare() != 0) { - fprintf(stderr, "Alsa preparation failed\n"); - return 1; - } - } - - int outbuf_size = subchannel_index*120; - uint8_t zmqframebuf[2048]; - zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)zmqframebuf; - - uint8_t outbuf[2048]; - - if(outbuf_size % 5 != 0) { - fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5); - } - - fprintf(stderr, "Starting encoding\n"); - - int send_error_count = 0; - struct timespec tp_next; - clock_gettime(CLOCK_MONOTONIC, &tp_next); - - int calls = 0; // for checking - while (1) { - int in_identifier[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA}; - 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; - - - // -------------- wait the right amount of time - if (drift_compensation) { - struct timespec tp_now; - clock_gettime(CLOCK_MONOTONIC, &tp_now); - - unsigned long time_now = (1000000000ul * tp_now.tv_sec) + - tp_now.tv_nsec; - unsigned long time_next = (1000000000ul * tp_next.tv_sec) + - tp_next.tv_nsec; - - const unsigned long dabplus_superframe_nsec = 120000000ul; - - const unsigned long wait_time = - dabplus_superframe_nsec / enc_calls_per_output; - - unsigned long waiting = wait_time - (time_now - time_next); - if ((time_now - time_next) < wait_time) { - //printf("Sleep %zuus\n", waiting / 1000); - usleep(waiting / 1000); - } - - // Move our time_counter 60ms into the future. - // The encoder needs two calls for one frame - tp_next.tv_nsec += wait_time; - if (tp_next.tv_nsec > 1000000000L) { - tp_next.tv_nsec -= 1000000000L; - tp_next.tv_sec += 1; - } - } - - // --------------- Read data from the PAD fifo - int ret; - if (padlen != 0) { - ret = read(pad_fd, pad_buf, padlen); - } - else { - ret = 0; - } - - - if(ret < 0 && errno == EAGAIN) { - // If this condition passes, there is no data to be read - in_buf.numBufs = 1; // Samples; - } - else if(ret >= 0) { - // Otherwise, you're good to go and buffer should contain "count" bytes. - in_buf.numBufs = 2; // Samples + Data; - if (ret > 0) - status |= STATUS_PAD_INSERTED; - } - else { - // Some other error occurred during read. - fprintf(stderr, "Unable to read from PAD!\n"); - break; - } - - // -------------- Read Data - memset(outbuf, 0x00, outbuf_size); - - ssize_t read; - if (drift_compensation) { - if (alsa_in_threaded.fault_detected()) { - fprintf(stderr, "Detected fault in alsa input!\n"); - break; - } - - size_t overruns; - read = queue.pop(input_buf, input_size, &overruns); // returns bytes - - if (read != input_size) { - status |= STATUS_UNDERRUN; - } - - if (overruns) { - status |= STATUS_OVERRUN; - } - } - else { - read = alsa_in_direct.read(input_buf, input_size); - if (read < 0) { - break; - } - else if (read != input_size) { - fprintf(stderr, "Short alsa read !\n"); - } - } - - for (int i = 0; i < read; 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); - peak_left = MAX(peak_left, l); - peak_right = MAX(peak_right, r); - } - - // -------------- AAC Encoding - - in_ptr[0] = input_buf; - in_ptr[1] = pad_buf; - in_size[0] = read; - in_size[1] = 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 = 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; - 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; - - 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); - break; - } - calls++; - - /* Check if the encoder has generated output data */ - if (out_args.numOutBytes != 0) - { - // Our timing code depends on this - if (! ((sample_rate == 32000 && calls == 2) || - (sample_rate == 48000 && calls == 3)) ) { - fprintf(stderr, "INTERNAL ERROR! sample rate %d, calls %d\n", - sample_rate, calls); - } - calls = 0; - - // ----------- RS encoding - int row, col; - unsigned char buf_to_rs_enc[110]; - unsigned char rs_enc[10]; - for(row=0; row < subchannel_index; row++) { - for(col=0;col < 110; col++) { - buf_to_rs_enc[col] = outbuf[subchannel_index * col + row]; - } - - encode_rs_char(rs_handler, buf_to_rs_enc, rs_enc); - - for(col=110; col<120; col++) { - outbuf[subchannel_index * col + row] = rs_enc[col-110]; - assert(subchannel_index * col + row < outbuf_size); - } - } - - // ------------ ZeroMQ transmit - try { - zmq_frame_header->version = 1; - zmq_frame_header->encoder = ZMQ_ENCODER_FDK; - zmq_frame_header->datasize = outbuf_size; - zmq_frame_header->audiolevel_left = peak_left; - zmq_frame_header->audiolevel_right = peak_right; - - memcpy(ZMQ_FRAME_DATA(zmq_frame_header), - outbuf, outbuf_size); - - zmq_sock.send(zmqframebuf, ZMQ_FRAME_SIZE(zmq_frame_header), - ZMQ_DONTWAIT); - } - catch (zmq::error_t& e) { - fprintf(stderr, "ZeroMQ send error !\n"); - send_error_count ++; - } - - if (send_error_count > 10) - { - fprintf(stderr, "ZeroMQ send failed ten times, aborting!\n"); - break; - } - - if (show_level && out_args.numOutBytes + row*10 == outbuf_size) { - fprintf(stderr, "\rIn: [%6s|%-6s] %1s %1s %1s", - level(0, &peak_left), - level(1, &peak_right), - status & STATUS_PAD_INSERTED ? "P" : " ", - status & STATUS_UNDERRUN ? "U" : " ", - status & STATUS_OVERRUN ? "O" : " "); - } - - status = 0; - } - - fflush(stdout); - } - fprintf(stderr, "\n"); - - zmq_sock.close(); - free_rs_char(rs_handler); - - aacEncClose(&encoder); -} - diff --git a/src/dabplus-enc-file-zmq.c b/src/dabplus-enc-file-zmq.c deleted file mode 100644 index 1bfae12..0000000 --- a/src/dabplus-enc-file-zmq.c +++ /dev/null @@ -1,620 +0,0 @@ -/* ------------------------------------------------------------------ - * Copyright (C) 2011 Martin Storsjo - * Copyright (C) 2013,2014 Matthias P. Braendli - * Copyright (C) 2014 CSP Innovazione nelle ICT s.c.a r.l. - * http://rd.csp.it/ - * - * http://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 . - * ------------------------------------------------------------------- - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include "libAACenc/include/aacenc_lib.h" -#include "wavreader.h" -#include "encryption.h" -#include "utils.h" - -#include -#include -#include -#include -#include - -#include "contrib/lib_crc.h" - -void usage(const char* name) { - fprintf(stderr, - "dabplus-enc-file-zmq %s is a HE-AACv2 encoder for DAB+\n" - "based on fdk-aac-dabplus that can read from a file\n" - "or pipe source and encode to a ZeroMQ output for ODR-DabMux,\n" - "a file or standard output.\n" - "\n" - "It includes PAD (DLS and MOT Slideshow) support by http://rd.csp.it\n" - "to be used with mot-encoder\n" - "\n" - " http://opendigitalradio.org\n" - "\nUsage:\n" - "%s [OPTION...]\n", -#if defined(GITVERSION) - GITVERSION -#else - PACKAGE_VERSION -#endif - , name); - - fprintf(stderr, - " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n" - " -i, --input=FILENAME Input filename (default: stdin).\n" - " -o, --output=URI Output zmq uri or filename.\n" - " Use '-' for standard output,\n" - " 'tcp://odr-dabmux-host:9000' for ZeroMQ.\n" - " Protocols supported: tcp, pgm, epgm\n" - " -a, --afterburner Turn on AAC encoder quality increaser.\n" - " -p, --pad=BYTES Set PAD size in bytes.\n" - " -P, --pad-fifo=FILENAME Set PAD data input fifo name (default: /tmp/pad.fifo).\n" - " -f, --format={ wav, raw } Set input file format (default: wav).\n" - " -c, --channels={ 1, 2 } Nb of input channels for raw input (default: 2).\n" - " -r, --rate={ 32000, 48000 } Sample rate for raw input (default: 48000).\n" - " -k, --secret-key=FILE Set the secret key for encryption.\n" - " -l, --level Show level indication.\n" - //" -v, --verbose=LEVEL Set verbosity level.\n" - "\n" - "Only the tcp:// zeromq transport has been tested until now.\n" - - ); - -} - - -#define no_argument 0 -#define required_argument 1 -#define optional_argument 2 - -int main(int argc, char *argv[]) { - int subchannel_index = 8; //64kbps subchannel - int ch=0; - const char *infile = NULL; - const char *outuri = NULL; - FILE *in_fh; - FILE *out_fh; - void *wav; - int wav_format, bits_per_sample, sample_rate=48000, channels=2; - uint8_t* input_buf; - int16_t* convert_buf; - - /* Keep track of peaks */ - int peak_left = 0; - int peak_right = 0; - - void *rs_handler = NULL; - int aot = AOT_DABPLUS_AAC_LC; - int afterburner = 0, raw_input=0; - HANDLE_AACENCODER handle; - CHANNEL_MODE mode; - AACENC_InfoStruct info = { 0 }; - - char* pad_fifo = "/tmp/pad.fifo"; - int pad_fd; - unsigned char pad_buf[128]; - int padlen; - - void *zmq_context = zmq_ctx_new(); - void *zmq_sock = NULL; - - int show_level = 0; - - /* Data for ZMQ CURVE authentication */ - char* keyfile = NULL; - char secretkey[CURVE_KEYLEN+1]; - - const struct option longopts[] = { - {"bitrate", required_argument, 0, 'b'}, - {"input", required_argument, 0, 'i'}, - {"output", required_argument, 0, 'o'}, - {"format", required_argument, 0, 'f'}, - {"rate", required_argument, 0, 'r'}, - {"channels", required_argument, 0, 'c'}, - {"pad", required_argument, 0, 'p'}, - {"pad-fifo", required_argument, 0, 'P'}, - {"secret-key", required_argument, 0, 'k'}, - {"afterburner", no_argument, 0, 'a'}, - {"help", no_argument, 0, 'h'}, - {"level", no_argument, 0, 'l'}, - {0,0,0,0}, - }; - - if (argc == 1) { - usage(argv[0]); - return 0; - } - - int index; - while(ch != -1) { - ch = getopt_long(argc, argv, "tlhab:c:i:k:o:r:f:p:P:", longopts, &index); - switch (ch) { - case 'f': - if(strcmp(optarg, "raw")==0) { - raw_input = 1; - } else if(strcmp(optarg, "wav")!=0) - usage(argv[0]); - break; - case 'a': - afterburner = 1; - break; - case 'b': - subchannel_index = atoi(optarg) / 8; - break; - case 'c': - channels = atoi(optarg); - break; - case 'r': - sample_rate = atoi(optarg); - break; - case 'i': - infile = optarg; - break; - case 'k': - keyfile = optarg; - break; - case 'o': - outuri = optarg; - break; - case 'p': - padlen = atoi(optarg); - break; - case 'P': - pad_fifo = optarg; - break; - case 'l': - show_level = 1; - break; - case '?': - case 'h': - usage(argv[0]); - return 1; - } - } - - if(subchannel_index < 1 || subchannel_index > 24) { - fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n", - subchannel_index); - return 1; - } - if(padlen != 0) { - int flags; - if (mkfifo(pad_fifo, S_IWUSR | S_IRUSR | S_IRGRP | S_IROTH) != 0) { - if (errno != EEXIST) { - fprintf(stderr, "Can't create pad file: %d!\n", errno); - return 1; - } - } - pad_fd = open(pad_fifo, O_RDONLY | O_NONBLOCK); - if (pad_fd == -1) { - fprintf(stderr, "Can't open pad file!\n"); - return 1; - } - flags = fcntl(pad_fd, F_GETFL, 0); - if (fcntl(pad_fd, F_SETFL, flags | O_NONBLOCK)) { - fprintf(stderr, "Can't set non-blocking mode in pad file!\n"); - return 1; - } - } - - if(raw_input) { - if(infile && strcmp(infile, "-")) { - in_fh = fopen(infile, "rb"); - if(!in_fh) { - fprintf(stderr, "Can't open input file!\n"); - return 1; - } - } else { - in_fh = stdin; - } - } else { - wav = wav_read_open(infile); - if (!wav) { - fprintf(stderr, "Unable to open wav file %s\n", infile); - return 1; - } - if (!wav_get_header(wav, &wav_format, &channels, &sample_rate, &bits_per_sample, NULL)) { - fprintf(stderr, "Bad wav file %s\n", infile); - return 1; - } - if (wav_format != 1) { - fprintf(stderr, "Unsupported WAV format %d\n", wav_format); - return 1; - } - if (bits_per_sample != 16) { - fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample); - return 1; - } - if (channels > 2) { - fprintf(stderr, "Unsupported WAV channels %d\n", channels); - return 1; - } - } - - if (outuri) { - if (strcmp(outuri, "-") == 0) { - out_fh = stdout; - } - else if (strncmp(outuri, "file://", 7) == 0) { - out_fh = fopen(&outuri[7], "wb"); - - if(!out_fh) { - fprintf(stderr, "Can't open output file!\n"); - return 1; - } - } - else if ((strncmp(outuri, "tcp://", 6) == 0) || - (strncmp(outuri, "pgm://", 6) == 0) || - (strncmp(outuri, "epgm://", 7) == 0)) { - zmq_sock = zmq_socket(zmq_context, ZMQ_PUB); - if (zmq_sock == NULL) { - fprintf(stderr, "Error occurred during zmq_socket: %s\n", - zmq_strerror(errno)); - return 2; - } - if (keyfile) { - fprintf(stderr, "Enabling encryption\n"); - - int rc = readkey(keyfile, secretkey); - if (rc) { - fprintf(stderr, "Error reading secret key\n"); - return 2; - } - - const int yes = 1; - rc = zmq_setsockopt(zmq_sock, ZMQ_CURVE_SERVER, - &yes, sizeof(yes)); - if (rc) { - fprintf(stderr, "Error: %s\n", zmq_strerror(errno)); - return 2; - } - - rc = zmq_setsockopt(zmq_sock, ZMQ_CURVE_SECRETKEY, - secretkey, CURVE_KEYLEN); - if (rc) { - fprintf(stderr, "Error: %s\n", zmq_strerror(errno)); - return 2; - } - } - if (zmq_connect(zmq_sock, outuri) != 0) { - fprintf(stderr, "Error occurred during zmq_connect: %s\n", - zmq_strerror(errno)); - return 2; - } - } - else { - out_fh = fopen(outuri, "wb"); - - if(!out_fh) { - fprintf(stderr, "Can't open output file!\n"); - return 1; - } - } - } else { - fprintf(stderr, "Output URI not defined\n"); - return 1; - } - - - switch (channels) { - case 1: mode = MODE_1; break; - case 2: mode = MODE_2; break; - default: - fprintf(stderr, "Unsupported channels number %d\n", channels); - return 1; - } - - - if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) { - fprintf(stderr, "Unable to open encoder\n"); - return 1; - } - - - if(channels == 2 && subchannel_index <= 6) - aot = AOT_DABPLUS_PS; - else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10) - aot = AOT_DABPLUS_SBR; - - fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n", - subchannel_index, - aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "", - aot == AOT_DABPLUS_SBR ? "HE-AAC" : "", - aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "", - channels, sample_rate); - - if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) { - fprintf(stderr, "Unable to set the AOT\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) { - fprintf(stderr, "Unable to set the samplerate\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) { - fprintf(stderr, "Unable to set the channel mode\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) { - fprintf(stderr, "Unable to set the wav channel order\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) { - fprintf(stderr, "Unable to set the granule length\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) { - fprintf(stderr, "Unable to set the RAW transmux\n"); - return 1; - } - - /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) != AACENC_OK) { - fprintf(stderr, "Unable to set the bitrate mode\n"); - return 1; - }*/ - - - fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000); - if (aacEncoder_SetParam(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) { - fprintf(stderr, "Unable to set the bitrate\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) { - fprintf(stderr, "Unable to set the afterburner mode\n"); - return 1; - } - if (aacEncoder_SetParam(handle, AACENC_ANCILLARY_BITRATE, 0) != AACENC_OK) { - fprintf(stderr, "Unable to set the ancillary bitrate\n"); - return 1; - } - if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) { - fprintf(stderr, "Unable to initialize the encoder\n"); - return 1; - } - if (aacEncInfo(handle, &info) != AACENC_OK) { - fprintf(stderr, "Unable to get the encoder info\n"); - return 1; - } - - fprintf(stderr, "DAB+ Encoding: framelen=%d\n", info.frameLength); - - int input_size = channels*2*info.frameLength; - input_buf = (uint8_t*) malloc(input_size); - convert_buf = (int16_t*) malloc(input_size); - - /* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */ - rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135); - if (rs_handler == NULL) { - perror("init_rs_char failed"); - return 0; - } - - int outbuf_size = subchannel_index*120; - uint8_t zmqframebuf[2048]; - struct zmq_frame_header_t *zmq_frame_header = - (struct zmq_frame_header_t*)zmqframebuf; - - uint8_t outbuf[2048]; - - if(outbuf_size % 5 != 0) { - fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5); - } - - fprintf(stderr, "outbuf_size: %d\n", outbuf_size); - //outbuf_size += (4 * subchannel_index * (8*8)/8) - outbuf_size/5; - fprintf(stderr, "outbuf_size: %d\n", outbuf_size); - - int frame=0; - int send_error_count = 0; - while (1) { - memset(outbuf, 0x00, outbuf_size); - - AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 }; - AACENC_InArgs in_args = { 0 }; - AACENC_OutArgs out_args = { 0 }; - int in_identifier[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA}; - int in_size[2], in_elem_size[2]; - int out_identifier = OUT_BITSTREAM_DATA; - int out_size, out_elem_size; - int pcmread=0, i, ret; - int send_error; - void *in_ptr[2], *out_ptr; - AACENC_ERROR err; - - // Read data from the PAD fifo - if (padlen != 0) { - ret = read(pad_fd, pad_buf, padlen); - } - else { - ret = 0; - } - - - if(ret < 0 && errno == EAGAIN) { - // If this condition passes, there is no data to be read - in_buf.numBufs = 1; // Samples; - } - else if(ret >= 0) { - // Otherwise, you're good to go and buffer should contain "count" bytes. - in_buf.numBufs = 2; // Samples + Data; - if (ret > 0) - fprintf(stderr, "p"); - } - else { - // Some other error occurred during read. - fprintf(stderr, "Unable to read from PAD!\n"); - break; - } - - if(raw_input) { - if(fread(input_buf, input_size, 1, in_fh) == 1) { - pcmread = input_size; - } else { - fprintf(stderr, "Unable to read from input!\n"); - break; - } - } else { - pcmread = wav_read_data(wav, input_buf, input_size); - } - - for (i = 0; i < pcmread/2; i++) { - const uint8_t* in = &input_buf[2*i]; - convert_buf[i] = in[0] | (in[1] << 8); - } - - for (i = 0; i < pcmread/2; i+=2) { - peak_left = MAX(peak_left, convert_buf[i]); - peak_right = MAX(peak_right, convert_buf[i+1]); - } - - if (pcmread <= 0) { - in_args.numInSamples = -1; - } else { - in_ptr[0] = convert_buf; - in_ptr[1] = pad_buf; - in_size[0] = pcmread; - in_size[1] = padlen; - - in_elem_size[0] = 2; - in_elem_size[1] = sizeof(UCHAR); - - in_args.numInSamples = pcmread/2; - in_args.numAncBytes = padlen; - - //in_buf.numBufs = 2; // Samples + Data - - 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; - 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; - - if ((err = aacEncEncode(handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) { - if (err == AACENC_ENCODE_EOF) - break; - fprintf(stderr, "Encoding failed\n"); - return 1; - } - if (out_args.numOutBytes == 0) - continue; -#if 0 - unsigned char au_start[6]; - unsigned char* sfbuf = outbuf; - au_start[0] = 6; - au_start[1] = (*(sfbuf + 3) << 4) + ((*(sfbuf + 4)) >> 4); - au_start[2] = ((*(sfbuf + 4) & 0x0f) << 8) + *(sfbuf + 5); - fprintf (stderr, "au_start[0] = %d\n", au_start[0]); - fprintf (stderr, "au_start[1] = %d\n", au_start[1]); - fprintf (stderr, "au_start[2] = %d\n", au_start[2]); -#endif - - int row, col; - unsigned char buf_to_rs_enc[110]; - unsigned char rs_enc[10]; - for(row=0; row < subchannel_index; row++) { - for(col=0;col < 110; col++) { - buf_to_rs_enc[col] = outbuf[subchannel_index * col + row]; - } - - encode_rs_char(rs_handler, buf_to_rs_enc, rs_enc); - - for(col=110; col<120; col++) { - outbuf[subchannel_index * col + row] = rs_enc[col-110]; - assert(subchannel_index * col + row < outbuf_size); - } - } - - if (zmq_sock) { - zmq_frame_header->version = 1; - zmq_frame_header->encoder = ZMQ_ENCODER_FDK; - zmq_frame_header->datasize = outbuf_size; - zmq_frame_header->audiolevel_left = peak_left; - zmq_frame_header->audiolevel_right = peak_right; - - memcpy(ZMQ_FRAME_DATA(zmq_frame_header), - outbuf, outbuf_size); - - send_error = zmq_send(zmq_sock, zmqframebuf, - ZMQ_FRAME_SIZE(zmq_frame_header), ZMQ_DONTWAIT); - if (send_error < 0) { - fprintf(stderr, "ZeroMQ send failed! %s\n", zmq_strerror(errno)); - send_error_count ++; - } - } - else { - fwrite(outbuf, 1, /*out_args.numOutBytes*/ outbuf_size, out_fh); - } - - if (send_error_count > 10) - { - fprintf(stderr, "ZeroMQ send failed ten times, aborting!\n"); - break; - } - //fwrite(outbuf, 1, /*out_args.numOutBytes*/ outbuf_size, out_fh); - //fprintf(stderr, "Written %d/%d bytes!\n", out_args.numOutBytes + row*10, outbuf_size); - - if (show_level && out_args.numOutBytes + row*10 == outbuf_size) { - fprintf(stderr, "\rIn: [%6s|%-6s]", - level(0, &peak_left), - level(1, &peak_right)); - } - - frame++; - } - fprintf(stderr, "\n"); - - free(input_buf); - free(convert_buf); - if(raw_input) { - fclose(in_fh); - } else { - wav_read_close(wav); - } - - if (zmq_sock) { - zmq_close(zmq_sock); - zmq_ctx_term(zmq_context); - } - else { - fclose(out_fh); - } - - free_rs_char(rs_handler); - - aacEncClose(&handle); - - return 0; -} - diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp new file mode 100644 index 0000000..82780d5 --- /dev/null +++ b/src/dabplus-enc.cpp @@ -0,0 +1,716 @@ +/* ------------------------------------------------------------------ + * Copyright (C) 2011 Martin Storsjo + * Copyright (C) 2013,2014 Matthias P. Braendli + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either + * express or implied. + * See the License for the specific language governing permissions + * and limitations under the License. + * ------------------------------------------------------------------- + */ + +#include "AlsaInput.h" +#include "FileInput.h" +#include "SampleQueue.h" +#include "zmq.hpp" + +extern "C" { +#include "encryption.h" +#include "utils.h" +#include "wavreader.h" +} + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "libAACenc/include/aacenc_lib.h" + +extern "C" { +#include +} + +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 a ALSA or file source\n" + "and encode to a ZeroMQ output for ODR-DabMux.\n" + "\n" + "The -D option enables experimental sound card clock drift compensation.\n" + "A consumer sound card has a clock that is always a bit imprecise, and\n" + "would drift off after some time. ODR-DabMux cannot handle such drift\n" + "because it would have to throw away or insert a full DAB+ superframe,\n" + "which would create audible artifacts. This drift compensation can\n" + "make sure that the encoding rate is correct by inserting or deleting\n" + "audio samples.\n" + "\n" + "When this option is enabled, you will see U and O printed in\n" + "the console. These correspond to audio underruns and overruns caused\n" + "by sound card clock drift. When sparse, they should not create audible\n" + "artifacts.\n" + "\n" + "This encoder includes PAD (DLS and MOT Slideshow) support by\n" + "http://rd.csp.it to be used with mot-encoder\n" + "\n" + " http://opendigitalradio.org\n" + "\nUsage:\n" + "%s (-i file|-d alsa_device) [OPTION...]\n", +#if defined(GITVERSION) + GITVERSION +#else + PACKAGE_VERSION +#endif + , name); + fprintf(stderr, + " For the alsa input:\n" + " -d, --device=alsa_device Set ALSA input device (default: \"default\").\n" + " -D, --drift-comp Enable ALSA sound card drift compensation.\n" + " For the file input:\n" + " -i, --input=FILENAME Input filename (default: stdin).\n" + " -f, --format={ wav, raw } Set input file format (default: wav).\n" + " Encoder parameters:\n" + " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n" + " -o, --output=URI Output zmq uri. (e.g. 'tcp://*:9000')\n" + " -or- Output file uri. (e.g. 'file.dab')\n" + " -or- a single dash '-' to denote stdout\n" + " -a, --afterburner Turn on AAC encoder quality increaser.\n" + " -p, --pad=BYTES Set PAD size in bytes.\n" + " -P, --pad-fifo=FILENAME Set PAD data input fifo name (default: /tmp/pad.fifo).\n" + " -c, --channels={ 1, 2 } Nb of input channels (default: 2).\n" + " -r, --rate={ 32000, 48000 } Input sample rate (default: 48000).\n" + " -k, --secret-key=FILE Enable ZMQ encryption with the given secret key.\n" + " -l, --level Show peak audio level indication.\n" + "\n" + "Only the tcp:// zeromq transport has been tested until now,\n" + " but epgm:// and pgm:// are also accepted\n" + ); + +} + +int prepare_aac_encoder( + HANDLE_AACENCODER *encoder, + int subchannel_index, + int channels, + int sample_rate, + int afterburner) +{ + HANDLE_AACENCODER handle = *encoder; + + int aot = AOT_DABPLUS_AAC_LC; + + CHANNEL_MODE mode; + switch (channels) { + case 1: mode = MODE_1; break; + case 2: mode = MODE_2; break; + default: + fprintf(stderr, "Unsupported channels number %d\n", channels); + return 1; + } + + + if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) { + fprintf(stderr, "Unable to open encoder\n"); + return 1; + } + + *encoder = handle; + + if(channels == 2 && subchannel_index <= 6) + aot = AOT_DABPLUS_PS; + else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10) + aot = AOT_DABPLUS_SBR; + + fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n", + subchannel_index, + aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "", + aot == AOT_DABPLUS_SBR ? "HE-AAC" : "", + aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "", + channels, sample_rate); + + if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) { + fprintf(stderr, "Unable to set the AOT\n"); + return 1; + } + if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) { + fprintf(stderr, "Unable to set the sample rate\n"); + return 1; + } + if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) { + fprintf(stderr, "Unable to set the channel mode\n"); + return 1; + } + if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) { + fprintf(stderr, "Unable to set the wav channel order\n"); + return 1; + } + if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) { + fprintf(stderr, "Unable to set the granule length\n"); + return 1; + } + if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) { + fprintf(stderr, "Unable to set the RAW transmux\n"); + return 1; + } + + /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) + * != AACENC_OK) { + fprintf(stderr, "Unable to set the bitrate mode\n"); + return 1; + }*/ + + + fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000); + if (aacEncoder_SetParam(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) { + fprintf(stderr, "Unable to set the bitrate\n"); + return 1; + } + if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) { + fprintf(stderr, "Unable to set the afterburner mode\n"); + return 1; + } + if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) { + fprintf(stderr, "Unable to initialize the encoder\n"); + return 1; + } + return 0; +} + +#define no_argument 0 +#define required_argument 1 +#define optional_argument 2 + +#define STATUS_PAD_INSERTED 0x1 +#define STATUS_OVERRUN 0x2 +#define STATUS_UNDERRUN 0x4 + +int main(int argc, char *argv[]) +{ + int subchannel_index = 8; //64kbps subchannel + int ch=0; + + // For the ALSA input + const char *alsa_device = NULL; + + // For the file input + const char *infile = NULL; + int raw_input = 0; + + // For the file output + FILE *out_fh; + + const char *outuri = NULL; + int sample_rate=48000, channels=2; + const int bytes_per_sample = 2; + void *rs_handler = NULL; + bool afterburner = false; + bool drift_compensation = false; + AACENC_InfoStruct info = { 0 }; + + /* Keep track of peaks */ + int peak_left = 0; + int peak_right = 0; + + /* For MOT Slideshow and DLS insertion */ + const char* pad_fifo = "/tmp/pad.fifo"; + int pad_fd; + unsigned char pad_buf[128]; + int padlen; + + /* Encoder status, see the above STATUS macros */ + int status = 0; + + /* Whether to show the 'sox'-like measurement */ + int show_level = 0; + + /* Data for ZMQ CURVE authentication */ + char* keyfile = NULL; + char secretkey[CURVE_KEYLEN+1]; + + const struct option longopts[] = { + {"bitrate", required_argument, 0, 'b'}, + {"channels", required_argument, 0, 'c'}, + {"device", required_argument, 0, 'd'}, + {"format", required_argument, 0, 'f'}, + {"input", required_argument, 0, 'i'}, + {"output", required_argument, 0, 'o'}, + {"pad", required_argument, 0, 'p'}, + {"pad-fifo", required_argument, 0, 'P'}, + {"rate", required_argument, 0, 'r'}, + {"secret-key", required_argument, 0, 'k'}, + {"afterburner", no_argument, 0, 'a'}, + {"drift-comp", no_argument, 0, 'D'}, + {"help", no_argument, 0, 'h'}, + {"level", no_argument, 0, 'l'}, + {0,0,0,0}, + }; + + if (argc < 2) { + usage(argv[0]); + return 1; + } + + int index; + while(ch != -1) { + ch = getopt_long(argc, argv, "ahDlb:c:f:i:k:o:r:d:p:P:", longopts, &index); + switch (ch) { + case 'a': + afterburner = true; + break; + case 'b': + subchannel_index = atoi(optarg) / 8; + break; + case 'c': + channels = atoi(optarg); + break; + case 'd': + alsa_device = optarg; + break; + case 'D': + drift_compensation = true; + break; + case 'f': + if(strcmp(optarg, "raw")==0) { + raw_input = 1; + } else if(strcmp(optarg, "wav")!=0) + usage(argv[0]); + break; + case 'i': + infile = optarg; + break; + case 'k': + keyfile = optarg; + break; + case 'l': + show_level = 1; + break; + case 'o': + outuri = optarg; + break; + case 'p': + padlen = atoi(optarg); + break; + case 'P': + pad_fifo = optarg; + break; + case 'r': + sample_rate = atoi(optarg); + break; + case '?': + case 'h': + usage(argv[0]); + return 1; + } + } + + if (alsa_device && infile) { + fprintf(stderr, "You must define either alsa or file input, not both\n"); + return 1; + } + + if (subchannel_index < 1 || subchannel_index > 24) { + fprintf(stderr, "Bad subchannels number: %d, 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; + } + + /* We assume that we need to call the encoder + * enc_calls_per_output before it gives us one encoded audio + * frame. This information is used when the alsa drift compensation + * is active + */ + const int enc_calls_per_output = sample_rate / 16000; + + zmq::context_t zmq_ctx; + zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB); + + if (outuri) { + if (strcmp(outuri, "-") == 0) { + out_fh = stdout; + } + else if ((strncmp(outuri, "tcp://", 6) == 0) || + (strncmp(outuri, "pgm://", 6) == 0) || + (strncmp(outuri, "epgm://", 7) == 0)) { + if (keyfile) { + fprintf(stderr, "Enabling encryption\n"); + + int rc = readkey(keyfile, secretkey); + if (rc) { + fprintf(stderr, "Error reading secret key\n"); + return 2; + } + + const int yes = 1; + zmq_sock.setsockopt(ZMQ_CURVE_SERVER, + &yes, sizeof(yes)); + + zmq_sock.setsockopt(ZMQ_CURVE_SECRETKEY, + secretkey, CURVE_KEYLEN); + } + zmq_sock.connect(outuri); + } + else { // We assume it's a file name + out_fh = fopen(outuri, "wb"); + + if (!out_fh) { + fprintf(stderr, "Can't open output file!\n"); + return 1; + } + } + } + else { + fprintf(stderr, "Output URI not defined\n"); + return 1; + } + + if (padlen != 0) { + int flags; + if (mkfifo(pad_fifo, S_IWUSR | S_IRUSR | S_IRGRP | S_IROTH) != 0) { + if (errno != EEXIST) { + fprintf(stderr, "Can't create pad file: %d!\n", errno); + return 1; + } + } + pad_fd = open(pad_fifo, O_RDONLY | O_NONBLOCK); + if (pad_fd == -1) { + fprintf(stderr, "Can't open pad file!\n"); + return 1; + } + flags = fcntl(pad_fd, F_GETFL, 0); + if (fcntl(pad_fd, F_SETFL, flags | O_NONBLOCK)) { + fprintf(stderr, "Can't set non-blocking mode in pad file!\n"); + return 1; + } + } + + + HANDLE_AACENCODER encoder; + + if (prepare_aac_encoder(&encoder, subchannel_index, channels, + sample_rate, afterburner) != 0) { + fprintf(stderr, "Encoder preparation failed\n"); + return 2; + } + + 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); + + uint8_t input_buf[input_size]; + + int max_size = 2*input_size + NUM_SAMPLES_PER_CALL; + + SampleQueue queue(BYTES_PER_SAMPLE, channels, max_size); + + /* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */ + rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135); + if (rs_handler == NULL) { + perror("init_rs_char failed"); + return 1; + } + + /* No input defined ? default to alsa "default" */ + if (!alsa_device) { + alsa_device = "default"; + } + + // We'll use one of the tree possible inputs + AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue); + AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate); + FileInput file_in(infile, raw_input, sample_rate); + + if (infile) { + if (file_in.prepare() != 0) { + fprintf(stderr, "File input preparation failed\n"); + return 1; + } + } + else if (drift_compensation) { + if (alsa_in_threaded.prepare() != 0) { + fprintf(stderr, "Alsa preparation failed\n"); + return 1; + } + + fprintf(stderr, "Start ALSA capture thread\n"); + alsa_in_threaded.start(); + } + else { + if (alsa_in_direct.prepare() != 0) { + fprintf(stderr, "Alsa preparation failed\n"); + return 1; + } + } + + int outbuf_size = subchannel_index*120; + uint8_t zmqframebuf[2048]; + zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)zmqframebuf; + + uint8_t outbuf[2048]; + + if(outbuf_size % 5 != 0) { + fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5); + } + + fprintf(stderr, "Starting encoding\n"); + + int send_error_count = 0; + struct timespec tp_next; + clock_gettime(CLOCK_MONOTONIC, &tp_next); + + int calls = 0; // for checking + while (1) { + int in_identifier[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA}; + 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; + + + // -------------- wait the right amount of time + if (drift_compensation) { + struct timespec tp_now; + clock_gettime(CLOCK_MONOTONIC, &tp_now); + + unsigned long time_now = (1000000000ul * tp_now.tv_sec) + + tp_now.tv_nsec; + unsigned long time_next = (1000000000ul * tp_next.tv_sec) + + tp_next.tv_nsec; + + const unsigned long dabplus_superframe_nsec = 120000000ul; + + const unsigned long wait_time = + dabplus_superframe_nsec / enc_calls_per_output; + + unsigned long waiting = wait_time - (time_now - time_next); + if ((time_now - time_next) < wait_time) { + //printf("Sleep %zuus\n", waiting / 1000); + usleep(waiting / 1000); + } + + // Move our time_counter 60ms into the future. + // The encoder needs two calls for one frame + tp_next.tv_nsec += wait_time; + if (tp_next.tv_nsec > 1000000000L) { + tp_next.tv_nsec -= 1000000000L; + tp_next.tv_sec += 1; + } + } + + // --------------- Read data from the PAD fifo + int ret; + if (padlen != 0) { + ret = read(pad_fd, pad_buf, padlen); + } + else { + ret = 0; + } + + + if(ret < 0 && errno == EAGAIN) { + // If this condition passes, there is no data to be read + in_buf.numBufs = 1; // Samples; + } + else if(ret >= 0) { + // Otherwise, you're good to go and buffer should contain "count" bytes. + in_buf.numBufs = 2; // Samples + Data; + if (ret > 0) + status |= STATUS_PAD_INSERTED; + } + else { + // Some other error occurred during read. + fprintf(stderr, "Unable to read from PAD!\n"); + break; + } + + // -------------- Read Data + memset(outbuf, 0x00, outbuf_size); + memset(input_buf, 0x00, input_size); + + ssize_t read; + if (infile) { + read = file_in.read(input_buf, input_size); + if (read < 0) { + break; + } + else if (read != input_size) { + fprintf(stderr, "Short file read !\n"); + break; + } + } + else if (drift_compensation) { + if (alsa_in_threaded.fault_detected()) { + fprintf(stderr, "Detected fault in alsa input!\n"); + break; + } + + size_t overruns; + read = queue.pop(input_buf, input_size, &overruns); // returns bytes + + if (read != input_size) { + status |= STATUS_UNDERRUN; + } + + if (overruns) { + status |= STATUS_OVERRUN; + } + } + else { + read = alsa_in_direct.read(input_buf, input_size); + if (read < 0) { + break; + } + else if (read != input_size) { + fprintf(stderr, "Short alsa read !\n"); + } + } + + for (int i = 0; i < read; 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); + peak_left = MAX(peak_left, l); + peak_right = MAX(peak_right, r); + } + + // -------------- AAC Encoding + + in_ptr[0] = input_buf; + in_ptr[1] = pad_buf; + in_size[0] = read; + in_size[1] = 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 = 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; + 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; + + 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); + break; + } + calls++; + + /* Check if the encoder has generated output data */ + if (out_args.numOutBytes != 0) + { + // Our timing code depends on this + if (! ((sample_rate == 32000 && calls == 2) || + (sample_rate == 48000 && calls == 3)) ) { + fprintf(stderr, "INTERNAL ERROR! sample rate %d, calls %d\n", + sample_rate, calls); + } + calls = 0; + + // ----------- RS encoding + int row, col; + unsigned char buf_to_rs_enc[110]; + unsigned char rs_enc[10]; + for(row=0; row < subchannel_index; row++) { + for(col=0;col < 110; col++) { + buf_to_rs_enc[col] = outbuf[subchannel_index * col + row]; + } + + encode_rs_char(rs_handler, buf_to_rs_enc, rs_enc); + + for(col=110; col<120; col++) { + outbuf[subchannel_index * col + row] = rs_enc[col-110]; + assert(subchannel_index * col + row < outbuf_size); + } + } + + // ------------ ZeroMQ transmit + try { + zmq_frame_header->version = 1; + zmq_frame_header->encoder = ZMQ_ENCODER_FDK; + zmq_frame_header->datasize = outbuf_size; + zmq_frame_header->audiolevel_left = peak_left; + zmq_frame_header->audiolevel_right = peak_right; + + memcpy(ZMQ_FRAME_DATA(zmq_frame_header), + outbuf, outbuf_size); + + zmq_sock.send(zmqframebuf, ZMQ_FRAME_SIZE(zmq_frame_header), + ZMQ_DONTWAIT); + } + catch (zmq::error_t& e) { + fprintf(stderr, "ZeroMQ send error !\n"); + send_error_count ++; + } + + if (send_error_count > 10) + { + fprintf(stderr, "ZeroMQ send failed ten times, aborting!\n"); + break; + } + + if (show_level && out_args.numOutBytes + row*10 == outbuf_size) { + fprintf(stderr, "\rIn: [%6s|%-6s] %1s %1s %1s", + level(0, &peak_left), + level(1, &peak_right), + status & STATUS_PAD_INSERTED ? "P" : " ", + status & STATUS_UNDERRUN ? "U" : " ", + status & STATUS_OVERRUN ? "O" : " "); + } + + status = 0; + } + + fflush(stdout); + } + fprintf(stderr, "\n"); + + zmq_sock.close(); + free_rs_char(rs_handler); + + aacEncClose(&encoder); +} + -- cgit v1.2.3