示例#1
0
static int auloop_reset(struct audio_loop *al)
{
	struct auplay_prm auplay_prm;
	struct ausrc_prm ausrc_prm;
	const struct config *cfg = conf_config();
	int err;

	if (!cfg)
		return ENOENT;

	/* Optional audio codec */
	if (str_isset(aucodec))
		start_codec(al, aucodec);

	al->auplay = mem_deref(al->auplay);
	al->ausrc = mem_deref(al->ausrc);
	al->ab = mem_deref(al->ab);

	al->srate = configv[al->index].srate;
	al->ch    = configv[al->index].ch;
	al->fs    = al->srate * al->ch * PTIME / 1000;

	(void)re_printf("Audio-loop: %uHz, %dch\n", al->srate, al->ch);

	err = aubuf_alloc(&al->ab, 320, 0);
	if (err)
		return err;

	auplay_prm.fmt        = AUFMT_S16LE;
	auplay_prm.srate      = al->srate;
	auplay_prm.ch         = al->ch;
	auplay_prm.frame_size = al->fs;
	err = auplay_alloc(&al->auplay, cfg->audio.play_mod, &auplay_prm,
			   cfg->audio.play_dev, write_handler, al);
	if (err) {
		DEBUG_WARNING("auplay %s,%s failed: %m\n",
			      cfg->audio.play_mod, cfg->audio.play_dev,
			      err);
		return err;
	}

	ausrc_prm.fmt        = AUFMT_S16LE;
	ausrc_prm.srate      = al->srate;
	ausrc_prm.ch         = al->ch;
	ausrc_prm.frame_size = al->fs;
	err = ausrc_alloc(&al->ausrc, NULL, cfg->audio.src_mod,
			  &ausrc_prm, cfg->audio.src_dev,
			  read_handler, error_handler, al);
	if (err) {
		DEBUG_WARNING("ausrc %s,%s failed: %m\n", cfg->audio.src_mod,
			      cfg->audio.src_dev, err);
		return err;
	}

	return err;
}
示例#2
0
文件: audio.c 项目: soramimi/qSIP
int audio_set_source(struct audio *au, const char *mod, const char *device)
{
	struct autx *tx;
	int err;

	if (!au) return EINVAL;

	tx = &au->tx;

	/* stop the audio device first */
	tx->ausrc = mem_deref(tx->ausrc);

	err = ausrc_alloc(&tx->ausrc, NULL, mod, &tx->ausrc_prm, device,
			  ausrc_read_handler, ausrc_error_handler, au, NULL);
	if (err) {
		DEBUG_WARNING("audio: set_source failed (%s.%s): %m\n", mod, device, err);
		return err;
	}
	(void)re_printf("%H", autx_print_pipeline, &au->tx);

	return 0;
}
示例#3
0
文件: audio.c 项目: AmesianX/baresip
int audio_set_source(struct audio *au, const char *mod, const char *device)
{
	struct autx *tx;
	int err;

	if (!au)
		return EINVAL;

	tx = &au->tx;

	/* stop the audio device first */
	tx->ausrc = mem_deref(tx->ausrc);

	err = ausrc_alloc(&tx->ausrc, NULL, mod, &tx->ausrc_prm, device,
			  ausrc_read_handler, ausrc_error_handler, au);
	if (err) {
		warning("audio: set_source failed (%s.%s): %m\n",
			mod, device, err);
		return err;
	}

	return 0;
}
示例#4
0
static int start_source(struct autx *tx, struct audio *a)
{
	const struct aucodec *ac = tx->ac;
	uint32_t srate_dsp = get_srate(tx->ac);
	int err;

	if (!ac)
		return 0;

	/* Optional resampler, if configured */
	if (a->cfg.srate_src && a->cfg.srate_src != srate_dsp &&
	    !tx->resamp) {

		srate_dsp = a->cfg.srate_src;

		(void)re_printf("enable ausrc resampler: %u --> %u Hz\n",
				get_srate(ac), srate_dsp);

		tx->sampv_rs = mem_zalloc(AUDIO_SAMPSZ * 2, NULL);
		if (!tx->sampv_rs)
			return ENOMEM;

		err = auresamp_alloc(&tx->resamp, AUDIO_SAMPSZ,
				     srate_dsp, ac->ch,
				     get_srate(ac), ac->ch);
		if (err)
			return err;
	}

	/* Start Audio Source */
	if (!tx->ausrc && ausrc_find(NULL)) {

		struct ausrc_prm prm;

		prm.fmt        = AUFMT_S16LE;
		prm.srate      = srate_dsp;
		prm.ch         = ac->ch;
		prm.frame_size = calc_nsamp(prm.srate, prm.ch, tx->ptime);

		tx->psize = 2 * prm.frame_size;

		if (!tx->ab) {
			err = aubuf_alloc(&tx->ab, tx->psize * 2,
					  tx->psize * 30);
			if (err)
				return err;
		}

		err = ausrc_alloc(&tx->ausrc, NULL, a->cfg.src_mod,
				  &prm, a->cfg.src_dev,
				  ausrc_read_handler, ausrc_error_handler, a);
		if (err) {
			DEBUG_WARNING("start_source failed: %m\n", err);
			return err;
		}

		switch (a->cfg.txmode) {
#ifdef HAVE_PTHREAD
		case AUDIO_MODE_THREAD:
		case AUDIO_MODE_THREAD_REALTIME:
			if (!tx->u.thr.run) {
				tx->u.thr.run = true;
				err = pthread_create(&tx->u.thr.tid, NULL,
						     tx_thread, a);
				if (err) {
					tx->u.thr.tid = false;
					return err;
				}
			}
			break;
#endif

		case AUDIO_MODE_TMR:
			tmr_start(&tx->u.tmr, 1, timeout_tx, a);
			break;

		default:
			break;
		}
	}

	return 0;
}
示例#5
0
文件: auloop.c 项目: FOSSRIT/baresip
static int auloop_reset(struct audio_loop *al)
{
	struct auplay_prm auplay_prm;
	struct ausrc_prm ausrc_prm;
	const struct config *cfg = conf_config();
	int err;

	if (!cfg)
		return ENOENT;

	/* Optional audio codec */
	if (str_isset(aucodec))
		start_codec(al, aucodec);

	/* audio player/source must be stopped first */
	al->auplay = mem_deref(al->auplay);
	al->ausrc  = mem_deref(al->ausrc);

	al->sampv  = mem_deref(al->sampv);
	al->ab     = mem_deref(al->ab);

	al->srate = configv[al->index].srate;
	al->ch    = configv[al->index].ch;

	if (str_isset(aucodec)) {
		al->sampc = al->srate * al->ch * PTIME / 1000;
		al->sampv = mem_alloc(al->sampc * 2, NULL);
		if (!al->sampv)
			return ENOMEM;
	}

	info("Audio-loop: %uHz, %dch\n", al->srate, al->ch);

	err = aubuf_alloc(&al->ab, 320, 0);
	if (err)
		return err;

	auplay_prm.srate      = al->srate;
	auplay_prm.ch         = al->ch;
	auplay_prm.ptime      = PTIME;
	err = auplay_alloc(&al->auplay, cfg->audio.play_mod, &auplay_prm,
			   cfg->audio.play_dev, write_handler, al);
	if (err) {
		warning("auloop: auplay %s,%s failed: %m\n",
			cfg->audio.play_mod, cfg->audio.play_dev,
			err);
		return err;
	}

	ausrc_prm.srate      = al->srate;
	ausrc_prm.ch         = al->ch;
	ausrc_prm.ptime      = PTIME;
	err = ausrc_alloc(&al->ausrc, NULL, cfg->audio.src_mod,
			  &ausrc_prm, cfg->audio.src_dev,
			  read_handler, error_handler, al);
	if (err) {
		warning("auloop: ausrc %s,%s failed: %m\n", cfg->audio.src_mod,
			cfg->audio.src_dev, err);
		return err;
	}

	return err;
}
示例#6
0
static int start_source(struct autx *tx, struct audio *a)
{
	const struct aucodec *ac = tx->ac;
	uint32_t srate_dsp = get_srate(ac);
	uint32_t channels_dsp;
	bool resamp = false;
	int err;

	if (!ac)
		return 0;

	channels_dsp = ac->ch;

	if (a->cfg.srate_src && a->cfg.srate_src != srate_dsp) {
		resamp = true;
		srate_dsp = a->cfg.srate_src;
	}
	if (a->cfg.channels_src && a->cfg.channels_src != channels_dsp) {
		resamp = true;
		channels_dsp = a->cfg.channels_src;
	}

	/* Optional resampler, if configured */
	if (resamp && !tx->sampv_rs) {

		info("audio: enable ausrc resampler:"
		     " %uHz/%uch <-- %uHz/%uch\n",
		     get_srate(ac), ac->ch, srate_dsp, channels_dsp);

		tx->sampv_rs = mem_zalloc(AUDIO_SAMPSZ * 2, NULL);
		if (!tx->sampv_rs)
			return ENOMEM;

		err = auresamp_setup(&tx->resamp,
				     srate_dsp, channels_dsp,
				     get_srate(ac), ac->ch);
		if (err) {
			warning("audio: could not setup ausrc resampler"
				" (%m)\n", err);
			return err;
		}
	}

	/* Start Audio Source */
	if (!tx->ausrc && ausrc_find(NULL)) {

		struct ausrc_prm prm;

		prm.srate      = srate_dsp;
		prm.ch         = channels_dsp;
		prm.ptime      = tx->ptime;

		tx->psize = 2 * calc_nsamp(prm.srate, prm.ch, prm.ptime);

		if (!tx->aubuf) {
			err = aubuf_alloc(&tx->aubuf, tx->psize * 2,
					  tx->psize * 30);
			if (err)
				return err;
		}

		err = ausrc_alloc(&tx->ausrc, NULL, a->cfg.src_mod,
				  &prm, tx->device,
				  ausrc_read_handler, ausrc_error_handler, a);
		if (err) {
			warning("audio: start_source failed (%s.%s): %m\n",
				a->cfg.src_mod, tx->device, err);
			return err;
		}

		switch (a->cfg.txmode) {
#ifdef HAVE_PTHREAD
		case AUDIO_MODE_THREAD:
		case AUDIO_MODE_THREAD_REALTIME:
			if (!tx->u.thr.run) {
				tx->u.thr.run = true;
				err = pthread_create(&tx->u.thr.tid, NULL,
						     tx_thread, a);
				if (err) {
					tx->u.thr.tid = false;
					return err;
				}
			}
			break;
#endif

		case AUDIO_MODE_TMR:
			tmr_start(&tx->u.tmr, 1, timeout_tx, a);
			break;

		default:
			break;
		}
	}

	return 0;
}
示例#7
0
文件: auloop.c 项目: alfredh/baresip
static int auloop_reset(struct audio_loop *al, uint32_t srate, uint32_t ch)
{
	struct auplay_prm auplay_prm;
	struct ausrc_prm ausrc_prm;
	const struct config *cfg = conf_config();
	int err;

	if (!cfg)
		return ENOENT;

	if (cfg->audio.src_fmt != cfg->audio.play_fmt) {
		warning("auloop: ausrc_format and auplay_format"
			" must be the same\n");
		return EINVAL;
	}

	al->fmt = cfg->audio.src_fmt;

	/* audio player/source must be stopped first */
	al->auplay = mem_deref(al->auplay);
	al->ausrc  = mem_deref(al->ausrc);

	al->aubuf  = mem_deref(al->aubuf);

	al->srate = srate;
	al->ch    = ch;

	info("Audio-loop: %uHz, %dch, %s\n", al->srate, al->ch,
	     aufmt_name(al->fmt));

	err = aubuf_alloc(&al->aubuf, 320, 0);
	if (err)
		return err;

	auplay_prm.srate      = al->srate;
	auplay_prm.ch         = al->ch;
	auplay_prm.ptime      = PTIME;
	auplay_prm.fmt        = al->fmt;
	err = auplay_alloc(&al->auplay, baresip_auplayl(),
			   cfg->audio.play_mod, &auplay_prm,
			   cfg->audio.play_dev, write_handler, al);
	if (err) {
		warning("auloop: auplay %s,%s failed: %m\n",
			cfg->audio.play_mod, cfg->audio.play_dev,
			err);
		return err;
	}

	ausrc_prm.srate      = al->srate;
	ausrc_prm.ch         = al->ch;
	ausrc_prm.ptime      = PTIME;
	ausrc_prm.fmt        = al->fmt;
	err = ausrc_alloc(&al->ausrc, baresip_ausrcl(),
			  NULL, cfg->audio.src_mod,
			  &ausrc_prm, cfg->audio.src_dev,
			  read_handler, error_handler, al);
	if (err) {
		warning("auloop: ausrc %s,%s failed: %m\n", cfg->audio.src_mod,
			cfg->audio.src_dev, err);
		return err;
	}

	return err;
}