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author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2014-02-01 16:24:52 +0100 |
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committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2014-02-01 16:24:52 +0100 |
commit | c80a18225501ead0df71835c5b60c81ff3f7b825 (patch) | |
tree | 6c2afc0943d746ca53802b0663a93977e506000c | |
parent | 4c46de92e19cf29fde05fd41890d6529ef17d7ac (diff) | |
download | ODR-AudioEnc-c80a18225501ead0df71835c5b60c81ff3f7b825.tar.gz ODR-AudioEnc-c80a18225501ead0df71835c5b60c81ff3f7b825.tar.bz2 ODR-AudioEnc-c80a18225501ead0df71835c5b60c81ff3f7b825.zip |
untested fixes for alsa-dabplus-zmq.c
-rw-r--r-- | Makefile.am | 5 | ||||
-rw-r--r-- | alsa-dabplus-zmq.c | 332 |
2 files changed, 108 insertions, 229 deletions
diff --git a/Makefile.am b/Makefile.am index 5040d3e..be4e508 100644 --- a/Makefile.am +++ b/Makefile.am @@ -45,7 +45,10 @@ if HAVE_ZEROMQ_TEST aac_enc_dabplus_zmq_LDADD = libfdk-aac.la -lfec -lzmq aac_enc_dabplus_zmq_SOURCES = aac-enc-dabplus-zmq.c wavreader.c -bin_PROGRAMS = aac-enc$(EXEEXT) aac-enc-dabplus$(EXEEXT) aac-enc-dabplus-zmq$(EXEEXT) +alsa_dabplus_zmq_LDADD = libfdk-aac.la -lfec -lzmq -lasound +alsa_dabplus_zmq_SOURCES = alsa-dabplus-zmq.c + +bin_PROGRAMS = aac-enc$(EXEEXT) aac-enc-dabplus$(EXEEXT) aac-enc-dabplus-zmq$(EXEEXT) alsa-dabplus-zmq$(EXEEXT) else bin_PROGRAMS = aac-enc$(EXEEXT) aac-enc-dabplus$(EXEEXT) endif diff --git a/alsa-dabplus-zmq.c b/alsa-dabplus-zmq.c index 14e6aa8..9fecc58 100644 --- a/alsa-dabplus-zmq.c +++ b/alsa-dabplus-zmq.c @@ -1,6 +1,6 @@ /* ------------------------------------------------------------------ * Copyright (C) 2011 Martin Storsjo - * Copyright (C) 2013 Matthias P. Braendli + * 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. @@ -69,222 +69,124 @@ void usage(const char* name) { } -static int in_aborting = 0; static snd_pcm_t *alsa_handle = NULL; static void prg_exit(int code) { - if (alsa_handle) - snd_pcm_close(alsa_handle); + if (alsa_handle) { + //snd_pcm_close(alsa_handle); + } exit(code); } -static void signal_handler(int sig) +static void alsa_prepare(const char* alsa_dev, unsigned int rate, unsigned int channels) { - if (in_aborting) - return; + int err; + snd_pcm_hw_params_t *hw_params; - in_aborting = 1; - if (alsa_handle) - snd_pcm_abort(alsa_handle); + fprintf(stderr, "Initialising ALSA...\n"); - if (sig == SIGABRT) { - /* do not call snd_pcm_close() and abort immediately */ - alsa_handle = NULL; - exit(EXIT_FAILURE); + const int open_mode = 0; //|= SND_PCM_NONBLOCK; + + if ((err = snd_pcm_open(&alsa_handle, alsa_dev, SND_PCM_STREAM_CAPTURE, open_mode)) < 0) { + fprintf (stderr, "cannot open audio device %s (%s)\n", + alsa_dev, snd_strerror(err)); + prg_exit(1); + } + + const int nonblock = 0; //TODO remove dead code + if (nonblock) { + err = snd_pcm_nonblock(alsa_handle, 1); + if (err < 0) { + fprintf(stderr, "nonblock setting error: %s", snd_strerror(err)); + prg_exit(1); + } + } + + if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) { + fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params_any(alsa_handle, hw_params)) < 0) { + fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params_set_access(alsa_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { + fprintf (stderr, "cannot set access type (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params_set_format(alsa_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) { + fprintf (stderr, "cannot set sample format (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params_set_rate_near(alsa_handle, hw_params, &rate, 0)) < 0) { + fprintf (stderr, "cannot set sample rate (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params_set_channels(alsa_handle, hw_params, channels)) < 0) { + fprintf (stderr, "cannot set channel count (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + if ((err = snd_pcm_hw_params(alsa_handle, hw_params)) < 0) { + fprintf (stderr, "cannot set parameters (%s)\n", + snd_strerror(err)); + prg_exit(1); + } + + snd_pcm_hw_params_free (hw_params); + + if ((err = snd_pcm_prepare(alsa_handle)) < 0) { + fprintf (stderr, "cannot prepare audio interface for use (%s)\n", + snd_strerror(err)); + prg_exit(1); } - signal(sig, signal_handler); -} -const static int dump_hw_params = 0; + fprintf(stderr, "ALSA init done.\n"); +} -// Set Alsa hardware parameters -static void set_params(void) +static size_t alsa_read(uint8_t* buf, snd_pcm_uframes_t length) { - snd_pcm_hw_params_t *params; - snd_pcm_sw_params_t *swparams; - snd_pcm_uframes_t buffer_size; + int i; int err; - size_t n; - unsigned int rate; - snd_pcm_uframes_t start_threshold, stop_threshold; - snd_pcm_hw_params_alloca(¶ms); - snd_pcm_sw_params_alloca(&swparams); - err = snd_pcm_hw_params_any(alsa_handle, params); - if (err < 0) { - fprintf(stderr, "Broken configuration for this PCM: no configurations available"); - prg_exit(EXIT_FAILURE); - } - if (dump_hw_params) { - fprintf(stderr, "HW Params of device \"%s\":\n", - snd_pcm_name(alsa_handle)); - fprintf(stderr, "--------------------\n"); - // TODO log should be a snd_output_t *log; - snd_pcm_hw_params_dump(params, log); - fprintf(stderr, "--------------------\n"); - } - err = snd_pcm_hw_params_set_access(alsa_handle, params, - SND_PCM_ACCESS_RW_INTERLEAVED); - if (err < 0) { - fprintf(stderr, "Access type not available"); - prg_exit(EXIT_FAILURE); - } - err = snd_pcm_hw_params_set_format(alsa_handle, params, hwparams.format); - if (err < 0) { - fprintf(stderr, "Sample format non available"); - snd_pcm_format_t format; - - fprintf(stderr, "Available formats:\n"); - for (format = 0; format <= SND_PCM_FORMAT_LAST; format++) { - if (snd_pcm_hw_params_test_format(alsa_handle, params, format) == 0) - fprintf(stderr, "- %s\n", snd_pcm_format_name(format)); - } - prg_exit(EXIT_FAILURE); - } - err = snd_pcm_hw_params_set_channels(alsa_handle, params, hwparams.channels); - if (err < 0) { - fprintf(stderr, "Channels count non available"); - prg_exit(EXIT_FAILURE); + + if ((err = snd_pcm_readi(alsa_handle, buf, length)) != length) { + fprintf (stderr, "read from audio interface failed (%s)\n", + snd_strerror(err)); } -#if 0 - err = snd_pcm_hw_params_set_periods_min(alsa_handle, params, 2); - assert(err >= 0); -#endif - rate = hwparams.rate; - err = snd_pcm_hw_params_set_rate_near(alsa_handle, params, &hwparams.rate, 0); - assert(err >= 0); - if ((float)rate * 1.05 < hwparams.rate || (float)rate * 0.95 > hwparams.rate) { - char plugex[64]; - const char *pcmname = snd_pcm_name(alsa_handle); - fprintf(stderr, "Warning: rate is not accurate (requested = %iHz, got = %iHz)\n", rate, hwparams.rate); - if (! pcmname || strchr(snd_pcm_name(alsa_handle), ':')) { - *plugex = 0; - } - else { - snprintf(plugex, sizeof(plugex), "(-Dplug:%s)", - snd_pcm_name(alsa_handle)); - } - fprintf(stderr, " please, try the plug plugin %s\n", - plugex); - } - rate = hwparams.rate; - if (buffer_time == 0 && buffer_frames == 0) { - err = snd_pcm_hw_params_get_buffer_time_max(params, - &buffer_time, 0); - assert(err >= 0); - if (buffer_time > 500000) - buffer_time = 500000; - } - if (period_time == 0 && period_frames == 0) { - if (buffer_time > 0) - period_time = buffer_time / 4; - else - period_frames = buffer_frames / 4; - } - if (period_time > 0) - err = snd_pcm_hw_params_set_period_time_near(alsa_handle, params, - &period_time, 0); - else - err = snd_pcm_hw_params_set_period_size_near(alsa_handle, params, - &period_frames, 0); - assert(err >= 0); - if (buffer_time > 0) { - err = snd_pcm_hw_params_set_buffer_time_near(alsa_handle, params, - &buffer_time, 0); - } else { - err = snd_pcm_hw_params_set_buffer_size_near(alsa_handle, params, - &buffer_frames); - } - assert(err >= 0); - monotonic = snd_pcm_hw_params_is_monotonic(params); - can_pause = snd_pcm_hw_params_can_pause(params); - err = snd_pcm_hw_params(alsa_handle, params); - if (err < 0) { - fprintf(stderr, "Unable to install hw params:"); - snd_pcm_hw_params_dump(params, log); - prg_exit(EXIT_FAILURE); - } - snd_pcm_hw_params_get_period_size(params, &chunk_size, 0); - snd_pcm_hw_params_get_buffer_size(params, &buffer_size); - if (chunk_size == buffer_size) { - fprintf(stderr, "Can't use period equal to buffer size (%lu == %lu)", - chunk_size, buffer_size); - prg_exit(EXIT_FAILURE); - } - snd_pcm_sw_params_current(alsa_handle, swparams); - if (avail_min < 0) - n = chunk_size; - else - n = (double) rate * avail_min / 1000000; - err = snd_pcm_sw_params_set_avail_min(alsa_handle, swparams, n); - - /* round up to closest transfer boundary */ - n = buffer_size; - if (start_delay <= 0) { - start_threshold = n + (double) rate * start_delay / 1000000; - } else - start_threshold = (double) rate * start_delay / 1000000; - if (start_threshold < 1) - start_threshold = 1; - if (start_threshold > n) - start_threshold = n; - err = snd_pcm_sw_params_set_start_threshold(alsa_handle, swparams, start_threshold); - assert(err >= 0); - if (stop_delay <= 0) - stop_threshold = buffer_size + (double) rate * stop_delay / 1000000; - else - stop_threshold = (double) rate * stop_delay / 1000000; - err = snd_pcm_sw_params_set_stop_threshold(alsa_handle, swparams, stop_threshold); - assert(err >= 0); - - if (snd_pcm_sw_params(alsa_handle, swparams) < 0) { - fprintf(stderr, "unable to install sw params:"); - snd_pcm_sw_params_dump(swparams, log); - prg_exit(EXIT_FAILURE); - } - - if (setup_chmap()) - prg_exit(EXIT_FAILURE); - - if (verbose) - snd_pcm_dump(alsa_handle, log); - - bits_per_sample = snd_pcm_format_physical_width(hwparams.format); - bits_per_frame = bits_per_sample * hwparams.channels; - chunk_bytes = chunk_size * bits_per_frame / 8; - audiobuf = realloc(audiobuf, chunk_bytes); - if (audiobuf == NULL) { - fprintf(stderr, "not enough memory"); - prg_exit(EXIT_FAILURE); - } - // fprintf(stderr, "real chunk_size = %i, frags = %i, total = %i\n", chunk_size, setup.buf.block.frags, setup.buf.block.frags * chunk_size); - - /* stereo VU-meter isn't always available... */ - if (vumeter == VUMETER_STEREO) { - if (hwparams.channels != 2 || !interleaved || verbose > 2) - vumeter = VUMETER_MONO; - } - - /* show mmap buffer arragment */ - if (mmap_flag && verbose) { - const snd_pcm_channel_area_t *areas; - snd_pcm_uframes_t offset, size = chunk_size; - int i; - err = snd_pcm_mmap_begin(alsa_handle, &areas, &offset, &size); - if (err < 0) { - fprintf(stderr, "snd_pcm_mmap_begin problem: %s", snd_strerror(err)); - prg_exit(EXIT_FAILURE); - } - for (i = 0; i < hwparams.channels; i++) - fprintf(stderr, "mmap_area[%i] = %p,%u,%u (%u)\n", i, areas[i].addr, areas[i].first, areas[i].step, snd_pcm_format_physical_width(hwparams.format)); - /* not required, but for sure */ - snd_pcm_mmap_commit(alsa_handle, offset, 0); + return err; +} + +static void signal_handler(int sig) +{ + fprintf(stderr, "Caught signal %d\n", sig); + if (alsa_handle) { + snd_pcm_abort(alsa_handle); + alsa_handle = NULL; } - buffer_frames = buffer_size; /* for position test */ + if (sig == SIGABRT) { + /* do not call snd_pcm_close() and abort immediately */ + alsa_handle = NULL; + exit(EXIT_FAILURE); + } + signal(sig, signal_handler); } + + #define no_argument 0 #define required_argument 1 #define optional_argument 2 @@ -296,7 +198,7 @@ int main(int argc, char *argv[]) { const char *alsa_device = "default"; const char *outuri = NULL; int sample_rate=48000, channels=2; - const int bits_per_sample = 16; + const int bytes_per_sample = 2; uint8_t* input_buf; int16_t* convert_buf; void *rs_handler = NULL; @@ -355,6 +257,7 @@ int main(int argc, char *argv[]) { return 1; } + fprintf(stderr, "Setting up ZeroMQ socket\n"); if (outuri) { zmq_sock = zmq_socket(zmq_context, ZMQ_PUB); if (zmq_sock == NULL) { @@ -370,30 +273,7 @@ int main(int argc, char *argv[]) { return 1; } - - const int open_mode = 0; //|= SND_PCM_NONBLOCK; - const snd_pcm_stream_t stream = SND_PCM_STREAM_CAPTURE; - const int nonblock = 0; - snd_pcm_info_t *alsa_info; - - err = snd_pcm_open(&alsa_handle, alsa_device, stream, open_mode); - if (err < 0) { - fprintf(stderr, "audio open error: %s", snd_strerror(err)); - return 1; - } - - if ((err = snd_pcm_info(alsa_handle, alsa_info)) < 0) { - fprintf(stderr, "info error: %s", snd_strerror(err)); - prg_exit(1); - } - - if (nonblock) { - err = snd_pcm_nonblock(alsa_handle, 1); - if (err < 0) { - fprintf(stderr, "nonblock setting error: %s", snd_strerror(err)); - prg_exit(1); - } - } + alsa_prepare(alsa_device, sample_rate, channels); signal(SIGINT, signal_handler); signal(SIGTERM, signal_handler); @@ -498,7 +378,7 @@ int main(int argc, char *argv[]) { 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); + //fprintf(stderr, "outbuf_size: %d\n", outbuf_size); int frame=0; int send_error_count = 0; @@ -517,10 +397,8 @@ int main(int argc, char *argv[]) { void *in_ptr, *out_ptr; AACENC_ERROR err; - // raw input - if(fread(input_buf, input_size, 1, in_fh) == 1) { - read = input_size; - } else { + read = alsa_read(input_buf, input_size/bytes_per_sample) * bytes_per_sample; + if (read != input_size) { fprintf(stderr, "Unable to read from input!\n"); break; } @@ -608,8 +486,6 @@ int main(int argc, char *argv[]) { // break; frame++; } - free(input_buf); - free(convert_buf); zmq_close(zmq_sock); free_rs_char(rs_handler); |