mirror of
https://github.com/saitohirga/WSJT-X.git
synced 2024-08-24 23:26:36 -04:00
2. Tweaked code that measures sampling rates (in fivehz and fivehztx) so that it works well with both PortAudio and bare ALSA. 3. Added a "Using PortAudio" message to a2d.f90. git-svn-id: svn+ssh://svn.code.sf.net/p/wsjt/wsjt/trunk@84 ab8295b8-cf94-4d9e-aec4-7959e3be5d79
561 lines
18 KiB
C
561 lines
18 KiB
C
#include <stdio.h>
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#include <stdlib.h>
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#include <pthread.h>
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#include "conf.h"
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#ifdef USE_ALSA
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#include <alsa/asoundlib.h>
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#endif
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#include <inttypes.h>
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#include <time.h>
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#ifdef USE_PORTAUDIO
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extern void decode1_(int *iarg);
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extern void a2d_(int *iarg);
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#endif
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#ifdef USE_ALSA
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#if 0
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#define ALSA_LOG
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#define ALSA_LOG_BUFFERS
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#endif
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#if 0
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#define ALSA_PLAYBACK_LOG
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#define ALSA_CAPTURE_LOG
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#endif
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#define BUFFER_TIME 2000*1000
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typedef struct alsa_driver_s {
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snd_pcm_t *audio_fd;
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int capabilities;
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int open_mode;
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int has_pause_resume;
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int is_paused;
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int32_t output_sample_rate, input_sample_rate;
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double sample_rate_factor;
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uint32_t num_channels;
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uint32_t bits_per_sample;
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uint32_t bytes_per_frame;
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uint32_t bytes_in_buffer; /* number of bytes writen to audio hardware */
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int16_t *app_buffer_y1;
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int16_t *app_buffer_y2;
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int *app_buffer_offset;
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int app_buffer_length;
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double *Tsec;
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double *tbuf;
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int *ibuf;
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int *ndsec;
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int *tx_ok;
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int tx_starting;
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int tx_offset;
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int *tr_period;
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int *nwave;
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int *nmode;
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int *transmitting;
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snd_pcm_uframes_t buffer_size;
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snd_pcm_uframes_t period_size;
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int32_t mmap;
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} alsa_driver_t;
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alsa_driver_t alsa_driver_playback;
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alsa_driver_t alsa_driver_capture;
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void *alsa_capture_buffers[2];
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void *alsa_playback_buffers[2];
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static snd_output_t *jcd_out;
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/*
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* open the audio device for writing to
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*/
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static int ao_alsa_open(alsa_driver_t *this_gen, int32_t *input_rate, snd_pcm_stream_t direction ) {
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alsa_driver_t *this = (alsa_driver_t *) this_gen;
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char *pcm_device;
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snd_pcm_hw_params_t *params;
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snd_pcm_sw_params_t *swparams;
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snd_pcm_access_mask_t *mask;
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snd_pcm_uframes_t period_size_min;
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snd_pcm_uframes_t period_size_max;
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snd_pcm_uframes_t buffer_size_min;
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snd_pcm_uframes_t buffer_size_max;
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snd_pcm_format_t format;
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uint32_t buffer_time=BUFFER_TIME;
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snd_pcm_uframes_t buffer_time_to_size;
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int err, dir;
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int open_mode=1; /* NONBLOCK */
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/* int open_mode=0; BLOCK */
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int32_t rate=*input_rate;
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this->input_sample_rate=*input_rate;
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snd_pcm_hw_params_alloca(¶ms);
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snd_pcm_sw_params_alloca(&swparams);
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err = snd_output_stdio_attach(&jcd_out, stdout, 0);
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this->num_channels = 2;
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pcm_device="default";
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#ifdef ALSA_LOG
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printf("audio_alsa_out: Audio Device name = %s\n",pcm_device);
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printf("audio_alsa_out: Number of channels = %d\n",this->num_channels);
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#endif
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if (this->audio_fd) {
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printf("audio_alsa_out:Already open...WHY!");
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snd_pcm_close (this->audio_fd);
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this->audio_fd = NULL;
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}
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this->bytes_in_buffer = 0;
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/*
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* open audio device
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*/
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err=snd_pcm_open(&this->audio_fd, pcm_device, direction, open_mode);
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if(err <0 ) {
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printf ("audio_alsa_out: snd_pcm_open() of %s failed: %s\n", pcm_device, snd_strerror(err));
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printf ("audio_alsa_out: >>> check if another program already uses PCM <<<\n");
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return 0;
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}
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/* printf ("audio_alsa_out: snd_pcm_open() opened %s\n", pcm_device); */
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/* We wanted non blocking open but now put it back to normal */
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//snd_pcm_nonblock(this->audio_fd, 0);
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snd_pcm_nonblock(this->audio_fd, 1);
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/*
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* configure audio device
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*/
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err = snd_pcm_hw_params_any(this->audio_fd, params);
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if (err < 0) {
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printf ("audio_alsa_out: broken configuration for this PCM: no configurations available: %s\n"),
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snd_strerror(err);
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goto close;
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}
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/* set interleaved access */
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if (this->mmap != 0) {
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mask = alloca(snd_pcm_access_mask_sizeof());
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snd_pcm_access_mask_none(mask);
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snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_INTERLEAVED);
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snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
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snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_COMPLEX);
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err = snd_pcm_hw_params_set_access_mask(this->audio_fd, params, mask);
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if (err < 0) {
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printf ( "audio_alsa_out: mmap not availiable, falling back to compatiblity mode\n");
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this->mmap=0;
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err = snd_pcm_hw_params_set_access(this->audio_fd, params,
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SND_PCM_ACCESS_RW_NONINTERLEAVED);
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}
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} else {
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err = snd_pcm_hw_params_set_access(this->audio_fd, params,
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SND_PCM_ACCESS_RW_NONINTERLEAVED);
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}
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if (err < 0) {
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printf ( "audio_alsa_out: access type not available: %s\n", snd_strerror(err));
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goto close;
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}
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/* set the sample format S16 */
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/* ALSA automatically appends _LE or _BE depending on the CPU */
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format = SND_PCM_FORMAT_S16;
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err = snd_pcm_hw_params_set_format(this->audio_fd, params, format );
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if (err < 0) {
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printf ( "audio_alsa_out: sample format non available: %s\n", snd_strerror(err));
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goto close;
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}
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/* set the number of channels */
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err = snd_pcm_hw_params_set_channels(this->audio_fd, params, this->num_channels);
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if (err < 0) {
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printf ( "audio_alsa_out: Cannot set number of channels to %d (err=%d:%s)\n",
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this->num_channels, err, snd_strerror(err));
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goto close;
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}
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#if SND_LIB_VERSION >= 0x010009
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/* Restrict a configuration space to contain only real hardware rates */
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err = snd_pcm_hw_params_set_rate_resample(this->audio_fd, params, 0);
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#endif
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/* set the stream rate [Hz] */
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dir=0;
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err = snd_pcm_hw_params_set_rate_near(this->audio_fd, params, &rate, &dir);
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if (err < 0) {
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printf ( "audio_alsa_out: rate not available: %s\n", snd_strerror(err));
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goto close;
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}
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this->output_sample_rate = (uint32_t)rate;
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if (this->input_sample_rate != this->output_sample_rate) {
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printf ( "audio_alsa_out: audio rate : %d requested, %d provided by device/sec\n",
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this->input_sample_rate, this->output_sample_rate);
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}
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buffer_time_to_size = ( (uint64_t)buffer_time * rate) / 1000000;
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err = snd_pcm_hw_params_get_buffer_size_min(params, &buffer_size_min);
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err = snd_pcm_hw_params_get_buffer_size_max(params, &buffer_size_max);
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dir=0;
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err = snd_pcm_hw_params_get_period_size_min(params, &period_size_min,&dir);
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dir=0;
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err = snd_pcm_hw_params_get_period_size_max(params, &period_size_max,&dir);
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#ifdef ALSA_LOG_BUFFERS
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printf("Buffer size range from %lu to %lu\n",buffer_size_min, buffer_size_max);
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printf("Period size range from %lu to %lu\n",period_size_min, period_size_max);
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printf("Buffer time size %lu\n",buffer_time_to_size);
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#endif
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this->buffer_size = buffer_time_to_size;
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if (buffer_size_max < this->buffer_size)
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this->buffer_size = buffer_size_max;
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if (buffer_size_min > this->buffer_size)
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this->buffer_size = buffer_size_min;
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this->period_size = this->buffer_size/8;
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if (this->period_size > 2048)
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this->period_size = 2048;
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this->buffer_size = this->period_size*8;
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#ifdef ALSA_LOG_BUFFERS
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printf("To choose buffer_size = %ld\n",this->buffer_size);
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printf("To choose period_size = %ld\n",this->period_size);
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#endif
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#if 0
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/* Set period to buffer size ratios at 8 periods to 1 buffer */
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dir=-1;
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periods=8;
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err = snd_pcm_hw_params_set_periods_near(this->audio_fd, params, &periods ,&dir);
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if (err < 0) {
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xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
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"audio_alsa_out: unable to set any periods: %s\n", snd_strerror(err));
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goto close;
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}
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/* set the ring-buffer time [us] (large enough for x us|y samples ...) */
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dir=0;
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err = snd_pcm_hw_params_set_buffer_time_near(this->audio_fd, params, &buffer_time, &dir);
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if (err < 0) {
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xprintf (this->class->xine, XINE_VERBOSITY_DEBUG,
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"audio_alsa_out: buffer time not available: %s\n", snd_strerror(err));
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goto close;
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}
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#endif
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#if 1
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/* set the period time [us] (interrupt every x us|y samples ...) */
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dir=0;
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err = snd_pcm_hw_params_set_period_size_near(this->audio_fd, params, &(this->period_size), &dir);
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if (err < 0) {
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printf ( "audio_alsa_out: period time not available: %s\n", snd_strerror(err));
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goto close;
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}
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#endif
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dir=0;
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err = snd_pcm_hw_params_get_period_size(params, &(this->period_size), &dir);
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dir=0;
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err = snd_pcm_hw_params_set_buffer_size_near(this->audio_fd, params, &(this->buffer_size));
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if (err < 0) {
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printf ( "audio_alsa_out: buffer time not available: %s\n", snd_strerror(err));
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goto close;
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}
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err = snd_pcm_hw_params_get_buffer_size(params, &(this->buffer_size));
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#ifdef ALSA_LOG_BUFFERS
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printf("was set period_size = %ld\n",this->period_size);
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printf("was set buffer_size = %ld\n",this->buffer_size);
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#endif
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if (2*this->period_size > this->buffer_size) {
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printf ( "audio_alsa_out: buffer to small, could not use\n");
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goto close;
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}
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/* write the parameters to device */
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err = snd_pcm_hw_params(this->audio_fd, params);
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if (err < 0) {
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printf ( "audio_alsa_out: pcm hw_params failed: %s\n", snd_strerror(err));
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goto close;
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}
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/* Check for pause/resume support */
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this->has_pause_resume = ( snd_pcm_hw_params_can_pause (params)
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&& snd_pcm_hw_params_can_resume (params) );
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// printf( "audio_alsa_out:open pause_resume=%d\n", this->has_pause_resume);
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this->sample_rate_factor = (double) this->output_sample_rate / (double) this->input_sample_rate;
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this->bytes_per_frame = snd_pcm_frames_to_bytes (this->audio_fd, 1);
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/*
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* audio buffer size handling
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*/
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/* Copy current parameters into swparams */
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err = snd_pcm_sw_params_current(this->audio_fd, swparams);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to determine current swparams: %s\n", snd_strerror(err));
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goto close;
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}
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/* align all transfers to 1 sample */
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err = snd_pcm_sw_params_set_xfer_align(this->audio_fd, swparams, 1);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set transfer alignment: %s\n", snd_strerror(err));
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goto close;
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}
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/* allow the transfer when at least period_size samples can be processed */
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err = snd_pcm_sw_params_set_avail_min(this->audio_fd, swparams, this->period_size);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set available min: %s\n", snd_strerror(err));
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goto close;
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}
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if (direction == SND_PCM_STREAM_PLAYBACK) {
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/* start the transfer when the buffer contains at least period_size samples */
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err = snd_pcm_sw_params_set_start_threshold(this->audio_fd, swparams, this->buffer_size);
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} else {
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err = snd_pcm_sw_params_set_start_threshold(this->audio_fd, swparams, -1);
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}
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set start threshold: %s\n", snd_strerror(err));
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goto close;
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}
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if (direction == SND_PCM_STREAM_PLAYBACK) {
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/* never stop the transfer, even on xruns */
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err = snd_pcm_sw_params_set_stop_threshold(this->audio_fd, swparams, this->buffer_size);
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} else {
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err = snd_pcm_sw_params_set_stop_threshold(this->audio_fd, swparams, this->buffer_size);
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}
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set stop threshold: %s\n", snd_strerror(err));
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goto close;
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}
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/* Install swparams into current parameters */
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err = snd_pcm_sw_params(this->audio_fd, swparams);
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if (err < 0) {
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printf ( "audio_alsa_out: Unable to set swparams: %s\n", snd_strerror(err));
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goto close;
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}
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#ifdef ALSA_LOG
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snd_pcm_dump_setup(this->audio_fd, jcd_out);
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snd_pcm_sw_params_dump(swparams, jcd_out);
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#endif
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return this->output_sample_rate;
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close:
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snd_pcm_close (this->audio_fd);
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this->audio_fd=NULL;
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return 0;
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}
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int16_t zero_buffer[65536];
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int playback_callback(alsa_driver_t *alsa_driver_playback) {
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alsa_driver_t *this = alsa_driver_playback;
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int result;
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struct timeval tv;
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double stime;
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int nsec;
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int i,n;
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static int ic;
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int16_t b0[2048];
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// printf("playback callback\n");
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gettimeofday(&tv, NULL);
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stime = (double) tv.tv_sec + ((double)tv.tv_usec / 1000000.0) +
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*(this->ndsec) * 0.1;
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*(this->Tsec) = stime;
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if(!(this->tx_starting) && (*(this->tx_ok)) ) {
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nsec = (int)stime;
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n = nsec / *(this->tr_period);
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ic = (int)(stime - *(this->tr_period) * n) * this->output_sample_rate;
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ic = ic % *(this->nwave);
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this->tx_offset = ic;
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}
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this->tx_starting = *(this->tx_ok);
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*(this->transmitting) = *(this->tx_ok);
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if(*(this->tx_ok)) {
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/*
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alsa_playback_buffers[0] = this->app_buffer_y1 + this->tx_offset;
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alsa_playback_buffers[1] = this->app_buffer_y1 + this->tx_offset;
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*/
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alsa_playback_buffers[0] = b0;
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alsa_playback_buffers[1] = b0;
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for(i=0; i<this->period_size; i++) {
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b0[i]=this->app_buffer_y1[ic];
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ic++;
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if(ic >= *this->nwave) {
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ic=ic % *this->nwave;
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if(*this->nmode == 2)
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*this->tx_ok=0;
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}
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}
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} else {
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alsa_playback_buffers[0] = zero_buffer;
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alsa_playback_buffers[1] = zero_buffer;
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}
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result = snd_pcm_writen(this->audio_fd, alsa_playback_buffers, this->period_size);
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this->tx_offset += this->period_size;
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if (result != this->period_size) {
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printf("playback writei failed. Expected %d samples, sent only %d\n", this->period_size, result);
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#ifdef ALSA_PLAYBACK_LOG
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snd_pcm_status_t *pcm_stat;
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snd_pcm_status_alloca(&pcm_stat);
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snd_pcm_status(this->audio_fd, pcm_stat);
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snd_pcm_status_dump(pcm_stat, jcd_out);
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#endif
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}
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fivehztx_(); //Call fortran routine
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}
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int capture_callback(alsa_driver_t *alsa_driver_capture) {
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alsa_driver_t *this = alsa_driver_capture;
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int result;
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struct timeval tv;
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double stime;
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int ib;
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#ifdef ALSA_CAPTURE_LOG
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printf("capture callback %d samples\n", this->period_size);
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#endif
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#ifdef ALSA_CAPTURE_LOG
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snd_pcm_status_t *pcm_stat;
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snd_pcm_status_alloca(&pcm_stat);
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snd_pcm_status(this->audio_fd, pcm_stat);
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snd_pcm_status_dump(pcm_stat, jcd_out);
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#endif
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gettimeofday(&tv, NULL);
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stime = (double) tv.tv_sec + ((double)tv.tv_usec / 1000000.0) +
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*(this->ndsec) * 0.1;
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*(this->Tsec) = stime;
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ib=*(this->ibuf);
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this->tbuf[ib++] = stime;
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if(ib>=1024)
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ib = 0;
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*(this->ibuf) = ib;
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alsa_capture_buffers[0]=this->app_buffer_y1 + *(this->app_buffer_offset);
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alsa_capture_buffers[1]=this->app_buffer_y2 + *(this->app_buffer_offset);
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result = snd_pcm_readn(this->audio_fd, alsa_capture_buffers, this->period_size);
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*(this->app_buffer_offset) += this->period_size;
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if ( *(this->app_buffer_offset) >= this->app_buffer_length )
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*(this->app_buffer_offset)=0; /* FIXME: implement proper wrapping */
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#ifdef ALSA_CAPTURE_LOG
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printf("result=%d\n",result);
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snd_pcm_status(this->audio_fd, pcm_stat);
|
|
snd_pcm_status_dump(pcm_stat, jcd_out);
|
|
#endif
|
|
fivehz_(); //Call fortran routine
|
|
}
|
|
|
|
int playback_xrun(alsa_driver_t *alsa_driver_playback) {
|
|
alsa_driver_t *this = alsa_driver_playback;
|
|
snd_pcm_status_t *pcm_stat;
|
|
snd_pcm_status_alloca(&pcm_stat);
|
|
printf("playback xrun\n");
|
|
snd_pcm_status(this->audio_fd, pcm_stat);
|
|
snd_pcm_status_dump(pcm_stat, jcd_out);
|
|
snd_pcm_prepare(this->audio_fd);
|
|
}
|
|
|
|
int capture_xrun(alsa_driver_t *alsa_driver_capture) {
|
|
alsa_driver_t *this = alsa_driver_capture;
|
|
snd_pcm_status_t *pcm_stat;
|
|
snd_pcm_status_alloca(&pcm_stat);
|
|
printf("capture xrun\n");
|
|
snd_pcm_status(this->audio_fd, pcm_stat);
|
|
snd_pcm_status_dump(pcm_stat, jcd_out);
|
|
}
|
|
|
|
void ao_alsa_loop(void *iarg) {
|
|
int playback_nfds;
|
|
int capture_nfds;
|
|
struct pollfd *pfd;
|
|
int nfds;
|
|
int capture_index;
|
|
unsigned short playback_revents;
|
|
unsigned short capture_revents;
|
|
playback_nfds = snd_pcm_poll_descriptors_count (
|
|
alsa_driver_playback.audio_fd);
|
|
capture_nfds = snd_pcm_poll_descriptors_count (
|
|
alsa_driver_capture.audio_fd);
|
|
pfd = (struct pollfd *) malloc (sizeof (struct pollfd) *
|
|
(playback_nfds + capture_nfds));
|
|
|
|
nfds=0;
|
|
snd_pcm_poll_descriptors (alsa_driver_playback.audio_fd,
|
|
&pfd[0],
|
|
playback_nfds);
|
|
nfds += playback_nfds;
|
|
snd_pcm_poll_descriptors (alsa_driver_capture.audio_fd,
|
|
&pfd[nfds],
|
|
capture_nfds);
|
|
capture_index = nfds;
|
|
nfds += capture_nfds;
|
|
while(1) {
|
|
if (poll (pfd, nfds, 100000) < 0) {
|
|
printf("poll failed\n");
|
|
return;
|
|
}
|
|
snd_pcm_poll_descriptors_revents(alsa_driver_playback.audio_fd, &pfd[0], playback_nfds, &playback_revents);
|
|
snd_pcm_poll_descriptors_revents(alsa_driver_capture.audio_fd, &pfd[capture_index], capture_nfds, &capture_revents);
|
|
//if ((playback_revents & POLLERR) || ((capture_revents) & POLLERR)) {
|
|
if (((capture_revents) & POLLERR)) {
|
|
printf("pollerr\n");
|
|
capture_xrun(&alsa_driver_capture);
|
|
return;
|
|
}
|
|
if (((playback_revents) & POLLERR)) {
|
|
printf("pollerr\n");
|
|
playback_xrun(&alsa_driver_capture);
|
|
return;
|
|
}
|
|
if (playback_revents & POLLOUT) {
|
|
playback_callback(&alsa_driver_playback);
|
|
}
|
|
if (capture_revents & POLLIN) {
|
|
capture_callback(&alsa_driver_capture);
|
|
}
|
|
}
|
|
|
|
return;
|
|
}
|
|
#endif
|
|
|
|
extern void decode1_(int *iarg);
|
|
int start_threads_(int *ndevin, int *ndevout, short y1[], short y2[],
|
|
int *nbuflen, int *iwrite, short iwave[],
|
|
int *nwave, int *nfsample, int *nsamperbuf,
|
|
int *TRPeriod, int *TxOK, int *ndebug,
|
|
int *Transmitting, double *Tsec, int *ngo, int *nmode,
|
|
double tbuf[], int *ibuf, int *ndsec)
|
|
{
|
|
pthread_t thread1,thread2;
|
|
int iret1,iret2;
|
|
int iarg1 = 1,iarg2 = 2;
|
|
#ifdef USE_ALSA
|
|
//int32_t rate=11025;
|
|
int32_t rate=*nfsample;
|
|
alsa_driver_capture.app_buffer_y1 = y1;
|
|
alsa_driver_capture.app_buffer_y2 = y2;
|
|
alsa_driver_capture.app_buffer_offset = iwrite;
|
|
alsa_driver_capture.app_buffer_length = *nbuflen;
|
|
alsa_driver_capture.Tsec = Tsec;
|
|
alsa_driver_capture.tbuf = tbuf;
|
|
alsa_driver_capture.ibuf = ibuf;
|
|
alsa_driver_capture.ndsec = ndsec;
|
|
alsa_driver_playback.Tsec = Tsec;
|
|
alsa_driver_playback.app_buffer_y1 = iwave;
|
|
alsa_driver_playback.tx_ok = TxOK;
|
|
alsa_driver_playback.tr_period = TRPeriod;
|
|
alsa_driver_playback.nwave = nwave;
|
|
alsa_driver_playback.nmode = nmode;
|
|
alsa_driver_playback.transmitting = Transmitting;
|
|
alsa_driver_playback.ndsec = ndsec;
|
|
// printf("start_threads: creating thread for decode1\n");
|
|
iret1 = pthread_create(&thread1,NULL,decode1_,&iarg1);
|
|
/* Open audio card. */
|
|
printf("Starting alsa routines.\n");
|
|
ao_alsa_open(&alsa_driver_playback, &rate, SND_PCM_STREAM_PLAYBACK);
|
|
ao_alsa_open(&alsa_driver_capture, &rate, SND_PCM_STREAM_CAPTURE);
|
|
|
|
/*
|
|
* Start audio io thread
|
|
*/
|
|
iret2 = pthread_create(&thread2, NULL, ao_alsa_loop, NULL);
|
|
snd_pcm_prepare(alsa_driver_capture.audio_fd);
|
|
snd_pcm_start(alsa_driver_capture.audio_fd);
|
|
snd_pcm_prepare(alsa_driver_playback.audio_fd);
|
|
//snd_pcm_start(alsa_driver_playback.audio_fd);
|
|
#endif
|
|
#ifdef USE_PORTAUDIO
|
|
/* snd_pcm_start */
|
|
// printf("start_threads: creating thread for a2d\n");
|
|
iret1 = pthread_create(&thread1,NULL,a2d_,&iarg1);
|
|
// printf("start_threads: creating thread for decode1_\n");
|
|
iret2 = pthread_create(&thread2,NULL,decode1_,&iarg2);
|
|
#endif
|
|
}
|