Restructure: Move sound from common code to 'libsound'
This commit is contained in:
parent
b4eb2d63a2
commit
f25b46ffef
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@ -33,7 +33,7 @@
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#include "mncc_console.h"
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#include "cause.h"
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#include "../common/call.h"
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#include "../common/sound.h"
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#include "../libsound/sound.h"
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static int new_callref = 0; /* toward mobile */
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@ -0,0 +1,6 @@
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AM_CPPFLAGS = -Wall -Wextra -g $(all_includes)
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noinst_LIBRARIES = libsound.a
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libsound_a_SOURCES = \
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sound_alsa.c
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@ -0,0 +1,10 @@
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enum paging_signal;
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void *sound_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int channels, double paging_frequency, int samplerate, int latspl, double bandwidth, double sample_deviation);
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int sound_start(void *inst);
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void sound_close(void *inst);
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int sound_write(void *inst, sample_t **samples, uint8_t **power, int num, enum paging_signal *paging_signal, int *on, int channels);
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int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf_level_db);
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int sound_get_tosend(void *inst, int latspl);
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@ -0,0 +1,485 @@
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/* Sound device access
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*
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* (C) 2016 by Andreas Eversberg <jolly@eversberg.eu>
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* All Rights Reserved
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*
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* This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 3 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdlib.h>
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#include <stdint.h>
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#include <math.h>
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#include <alsa/asoundlib.h>
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#include "../common/sample.h"
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#include "../common/debug.h"
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#include "../common/sender.h"
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typedef struct sound {
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snd_pcm_t *phandle, *chandle;
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int pchannels, cchannels;
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double spl_deviation; /* how much deviation is one sample step */
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double paging_phaseshift; /* phase to shift every sample */
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double paging_phase; /* current phase */
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double rx_frequency[2]; /* rx frequency of radio connected to channel */
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dispmeasparam_t *dmp[2];
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} sound_t;
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static int set_hw_params(snd_pcm_t *handle, int samplerate, int *channels)
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{
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snd_pcm_hw_params_t *hw_params = NULL;
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int rc;
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unsigned int rrate;
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rc = snd_pcm_hw_params_malloc(&hw_params);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "Failed to allocate hw_params! (%s)\n", snd_strerror(rc));
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goto error;
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}
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rc = snd_pcm_hw_params_any(handle, hw_params);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "cannot initialize hardware parameter structure (%s)\n", snd_strerror(rc));
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goto error;
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}
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rc = snd_pcm_hw_params_set_rate_resample(handle, hw_params, 0);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "cannot set real hardware rate (%s)\n", snd_strerror(rc));
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goto error;
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}
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rc = snd_pcm_hw_params_set_access (handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "cannot set access to interleaved (%s)\n", snd_strerror(rc));
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goto error;
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}
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rc = snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "cannot set sample format (%s)\n", snd_strerror(rc));
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goto error;
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}
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rrate = samplerate;
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rc = snd_pcm_hw_params_set_rate_near(handle, hw_params, &rrate, 0);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "cannot set sample rate (%s)\n", snd_strerror(rc));
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goto error;
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}
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if ((int)rrate != samplerate) {
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PDEBUG(DSOUND, DEBUG_ERROR, "Rate doesn't match (requested %dHz, get %dHz)\n", samplerate, rrate);
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rc = -EIO;
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goto error;
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}
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*channels = 1;
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rc = snd_pcm_hw_params_set_channels(handle, hw_params, *channels);
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if (rc < 0) {
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*channels = 2;
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rc = snd_pcm_hw_params_set_channels(handle, hw_params, *channels);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "cannot set channel count to 1 nor 2 (%s)\n", snd_strerror(rc));
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goto error;
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}
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}
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rc = snd_pcm_hw_params(handle, hw_params);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "cannot set parameters (%s)\n", snd_strerror(rc));
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goto error;
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}
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snd_pcm_hw_params_free(hw_params);
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return 0;
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error:
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if (hw_params) {
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snd_pcm_hw_params_free(hw_params);
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}
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return rc;
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}
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static int sound_prepare(sound_t *sound)
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{
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int rc;
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rc = snd_pcm_prepare(sound->phandle);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
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return rc;
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}
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rc = snd_pcm_prepare(sound->chandle);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
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return rc;
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}
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return 0;
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}
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void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int channels, double __attribute__((unused)) paging_frequency, int samplerate, int __attribute((unused)) latspl, double max_deviation, double __attribute__((unused)) max_modulation)
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{
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sound_t *sound;
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int rc;
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if (channels < 1 || channels > 2) {
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PDEBUG(DSOUND, DEBUG_ERROR, "Cannot use more than two channels with the same sound card!\n");
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return NULL;
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}
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sound = calloc(1, sizeof(sound_t));
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if (!sound) {
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PDEBUG(DSOUND, DEBUG_ERROR, "Failed to alloc memory!\n");
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return NULL;
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}
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sound->spl_deviation = max_deviation / 32767.0;
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sound->paging_phaseshift = 1.0 / ((double)samplerate / 1000.0);
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rc = snd_pcm_open(&sound->phandle, audiodev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "Failed to open '%s' for playback! (%s)\n", audiodev, snd_strerror(rc));
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goto error;
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}
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rc = snd_pcm_open(&sound->chandle, audiodev, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "Failed to open '%s' for capture! (%s)\n", audiodev, snd_strerror(rc));
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goto error;
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}
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rc = set_hw_params(sound->phandle, samplerate, &sound->pchannels);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "Failed to set playback hw params\n");
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goto error;
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}
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if (sound->pchannels < channels) {
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PDEBUG(DSOUND, DEBUG_ERROR, "Sound card only supports %d channel for playback.\n", sound->pchannels);
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goto error;
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}
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PDEBUG(DSOUND, DEBUG_DEBUG, "Playback with %d channels.\n", sound->pchannels);
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rc = set_hw_params(sound->chandle, samplerate, &sound->cchannels);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "Failed to set capture hw params\n");
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goto error;
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}
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if (sound->cchannels < channels) {
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PDEBUG(DSOUND, DEBUG_ERROR, "Sound card only supports %d channel for capture.\n", sound->cchannels);
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goto error;
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}
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PDEBUG(DSOUND, DEBUG_DEBUG, "Capture with %d channels.\n", sound->cchannels);
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rc = sound_prepare(sound);
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if (rc < 0)
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goto error;
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if (rx_frequency) {
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sender_t *sender;
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int i;
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for (i = 0; i < channels; i++) {
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sound->rx_frequency[i] = rx_frequency[i];
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sender = get_sender_by_empfangsfrequenz(sound->rx_frequency[i]);
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if (!sender)
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continue;
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sound->dmp[i] = display_measurements_add(sender, "RX Level", "%.1f dB", DISPLAY_MEAS_PEAK, DISPLAY_MEAS_LEFT, -96.0, 0.0, -INFINITY);
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}
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}
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return sound;
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error:
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sound_close(sound);
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return NULL;
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}
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/* start streaming */
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int sound_start(void *inst)
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{
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sound_t *sound = (sound_t *)inst;
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int16_t buff[2];
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/* trigger capturing */
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snd_pcm_readi(sound->chandle, buff, 1);
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return 0;
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}
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void sound_close(void *inst)
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{
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sound_t *sound = (sound_t *)inst;
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if (sound->phandle != NULL)
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snd_pcm_close(sound->phandle);
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if (sound->chandle != NULL)
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snd_pcm_close(sound->chandle);
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free(sound);
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}
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static void gen_paging_tone(sound_t *sound, int16_t *samples, int length, enum paging_signal paging_signal, int on)
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{
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double phaseshift, phase;
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int i;
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switch (paging_signal) {
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case PAGING_SIGNAL_NOTONE:
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/* no tone if paging signal is on */
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on = !on;
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// fall through
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case PAGING_SIGNAL_TONE:
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/* tone if paging signal is on */
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if (on) {
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phaseshift = sound->paging_phaseshift;
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phase = sound->paging_phase;
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for (i = 0; i < length; i++) {
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if (phase < 0.5)
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*samples++ = 30000;
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else
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*samples++ = -30000;
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phase += phaseshift;
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if (phase >= 1.0)
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phase -= 1.0;
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}
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sound->paging_phase = phase;
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} else
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memset(samples, 0, length << 1);
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break;
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case PAGING_SIGNAL_NEGATIVE:
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/* negative signal if paging signal is on */
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on = !on;
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// fall through
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case PAGING_SIGNAL_POSITIVE:
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/* positive signal if paging signal is on */
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if (on)
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memset(samples, 127, length << 1);
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else
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memset(samples, 128, length << 1);
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break;
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case PAGING_SIGNAL_NONE:
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break;
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}
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}
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int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused)) **power, int num, enum paging_signal *paging_signal, int *on, int channels)
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{
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sound_t *sound = (sound_t *)inst;
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double spl_deviation = sound->spl_deviation;
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int32_t value;
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int16_t buff[num << 1];
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int rc;
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int i, ii;
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if (sound->pchannels == 2) {
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/* two channels */
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if (paging_signal && on && paging_signal[0] != PAGING_SIGNAL_NONE) {
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int16_t paging[num << 1];
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gen_paging_tone(sound, paging, num, paging_signal[0], on[0]);
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for (i = 0, ii = 0; i < num; i++) {
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value = samples[0][i] / spl_deviation;
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if (value > 32767)
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value = 32767;
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else if (value < -32767)
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value = -32767;
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buff[ii++] = value;
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buff[ii++] = paging[i];
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}
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} else if (channels == 2) {
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for (i = 0, ii = 0; i < num; i++) {
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value = samples[0][i] / spl_deviation;
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if (value > 32767)
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value = 32767;
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else if (value < -32767)
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value = -32767;
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buff[ii++] = value;
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value = samples[1][i] / spl_deviation;
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if (value > 32767)
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value = 32767;
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else if (value < -32767)
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value = -32767;
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buff[ii++] = value;
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}
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} else {
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for (i = 0, ii = 0; i < num; i++) {
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value = samples[0][i] / spl_deviation;
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if (value > 32767)
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value = 32767;
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else if (value < -32767)
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value = -32767;
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buff[ii++] = value;
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buff[ii++] = value;
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}
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}
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} else {
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/* one channel */
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for (i = 0, ii = 0; i < num; i++) {
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value = samples[0][i] / spl_deviation;
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if (value > 32767)
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value = 32767;
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else if (value < -32767)
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value = -32767;
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buff[ii++] = value;
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}
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}
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rc = snd_pcm_writei(sound->phandle, buff, num);
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if (rc < 0) {
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PDEBUG(DSOUND, DEBUG_ERROR, "failed to write audio to interface (%s)\n", snd_strerror(rc));
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if (rc == -EPIPE) {
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sound_prepare(sound);
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sound_start(sound);
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}
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return rc;
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}
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if (rc != num)
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PDEBUG(DSOUND, DEBUG_ERROR, "short write to audio interface, written %d bytes, got %d bytes\n", num, rc);
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return rc;
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}
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#define KEEP_FRAMES 8 /* minimum frames not to read, due to bug in ALSA */
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int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf_level_db)
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{
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sound_t *sound = (sound_t *)inst;
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double spl_deviation = sound->spl_deviation;
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int16_t buff[num << 1];
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int32_t spl;
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int32_t max[2], a;
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int in, rc;
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int i, ii;
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/* get samples in rx buffer */
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in = snd_pcm_avail(sound->chandle);
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/* if not more than KEEP_FRAMES frames available, try next time */
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if (in <= KEEP_FRAMES)
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return 0;
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/* read some frames less than in buffer, because snd_pcm_readi() seems
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* to corrupt last frames */
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in -= KEEP_FRAMES;
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if (in > num)
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in = num;
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rc = snd_pcm_readi(sound->chandle, buff, in);
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if (rc < 0) {
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if (errno == EAGAIN)
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return 0;
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PDEBUG(DSOUND, DEBUG_ERROR, "failed to read audio from interface (%s)\n", snd_strerror(rc));
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/* recover read */
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if (rc == -EPIPE) {
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sound_prepare(sound);
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sound_start(sound);
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}
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return rc;
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}
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if (rc == 0)
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return rc;
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if (sound->cchannels == 2) {
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if (channels < 2) {
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for (i = 0, ii = 0; i < rc; i++) {
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spl = buff[ii++];
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spl += buff[ii++];
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a = (spl >= 0) ? spl : -spl;
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if (i == 0 || a > max[0])
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max[0] = a;
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samples[0][i] = (double)spl * spl_deviation;
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}
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} else {
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for (i = 0, ii = 0; i < rc; i++) {
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spl = buff[ii++];
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a = (spl >= 0) ? spl : -spl;
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if (i == 0 || a > max[0])
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max[0] = a;
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samples[0][i] = (double)spl * spl_deviation;
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spl = buff[ii++];
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a = (spl >= 0) ? spl : -spl;
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if (i == 0 || a > max[1])
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max[1] = a;
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samples[1][i] = (double)spl * spl_deviation;
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}
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}
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} else {
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for (i = 0, ii = 0; i < rc; i++) {
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spl = buff[ii++];
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a = (spl >= 0) ? spl : -spl;
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if (i == 0 || a > max[0])
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max[0] = a;
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samples[0][i] = (double)spl * spl_deviation;
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}
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}
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sender_t *sender;
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for (i = 0; i < channels; i++) {
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sender = get_sender_by_empfangsfrequenz(sound->rx_frequency[i]);
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if (!sender)
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continue;
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display_measurements_update(sound->dmp[i], log10((double)max[i] / 32768.0) * 20, 0.0);
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rf_level_db[i] = 0.0;
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}
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|
||||
return rc;
|
||||
}
|
||||
|
||||
/*
|
||||
* get playback buffer space
|
||||
*
|
||||
* return number of samples to be sent */
|
||||
int sound_get_tosend(void *inst, int latspl)
|
||||
{
|
||||
sound_t *sound = (sound_t *)inst;
|
||||
int rc;
|
||||
snd_pcm_sframes_t delay;
|
||||
int tosend;
|
||||
|
||||
rc = snd_pcm_delay(sound->phandle, &delay);
|
||||
if (rc < 0) {
|
||||
if (rc == -32)
|
||||
PDEBUG(DSOUND, DEBUG_ERROR, "Buffer underrun: Please use higher latency and enable real time scheduling\n");
|
||||
else
|
||||
PDEBUG(DSOUND, DEBUG_ERROR, "failed to get delay from interface (%s)\n", snd_strerror(rc));
|
||||
if (rc == -EPIPE) {
|
||||
sound_prepare(sound);
|
||||
sound_start(sound);
|
||||
}
|
||||
return rc;
|
||||
}
|
||||
|
||||
tosend = latspl - delay;
|
||||
return tosend;
|
||||
}
|
||||
|
||||
int sound_is_stereo_capture(void *inst)
|
||||
{
|
||||
sound_t *sound = (sound_t *)inst;
|
||||
|
||||
if (sound->cchannels == 2)
|
||||
return 1;
|
||||
return 0;
|
||||
}
|
||||
|
||||
int sound_is_stereo_playback(void *inst)
|
||||
{
|
||||
sound_t *sound = (sound_t *)inst;
|
||||
|
||||
if (sound->pchannels == 2)
|
||||
return 1;
|
||||
return 0;
|
||||
}
|
||||
|
Loading…
Reference in New Issue