- import version 0.9.1 from upstream

1 files changed, 205 insertions, 0 deletions

diff --git a/src/lingot-audio-alsa.c b/src/lingot-audio-alsa.c
new file mode 100644
index 0000000..2c134f3
--- /dev/null
+++ b/src/lingot-audio-alsa.c
@@ -0,0 +1,205 @@
+/*
+ * lingot, a musical instrument tuner.
+ *
+ * Copyright (C) 2004-2011  Ibán Cereijo Graña, Jairo Chapela Martínez.
+ *
+ * This file is part of lingot.
+ *
+ * lingot is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * lingot is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with lingot; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include <stdlib.h>
+
+#include "lingot-defs.h"
+#include "lingot-audio-alsa.h"
+#include "lingot-i18n.h"
+#include "lingot-msg.h"
+
+LingotAudioHandler* lingot_audio_alsa_new(char* device, int sample_rate) {
+
+	LingotAudioHandler* audio = NULL;
+
+#	ifdef ALSA
+	const char* exception;
+	snd_pcm_hw_params_t* hw_params = NULL;
+	int err;
+	char error_message[1000];
+	unsigned int channels = 1;
+
+	audio = malloc(sizeof(LingotAudioHandler));
+	audio->read_buffer = NULL;
+	audio->audio_system = AUDIO_SYSTEM_ALSA;
+	audio->read_buffer_size = 128; // TODO: size up
+
+	// ALSA allocates some mem to load its config file when we call
+	// snd_card_next. Now that we're done getting the info, let's tell ALSA
+	// to unload the info and free up that mem
+	snd_config_update_free_global();
+
+	audio->capture_handle = NULL;
+
+	try {
+		if ((err = snd_pcm_open(&audio->capture_handle, device,
+				SND_PCM_STREAM_CAPTURE, 0)) < 0) {
+			sprintf(error_message, "Cannot open audio device %s.\n%s.", device,
+					snd_strerror(err));
+			throw(error_message);
+		}
+
+		strcpy(audio->device, device);
+
+		if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
+			sprintf(error_message,
+					"Cannot allocate hardware parameter structure.\n%s.",
+					snd_strerror(err));
+			throw(error_message);
+		}
+
+		if ((err = snd_pcm_hw_params_any(audio->capture_handle, hw_params)) < 0) {
+			sprintf(error_message,
+					"Cannot initialize hardware parameter structure.\n%s.",
+					snd_strerror(err));
+			throw(error_message);
+		}
+
+		if ((err = snd_pcm_hw_params_set_access(audio->capture_handle,
+				hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
+			sprintf(error_message, "Cannot set access type.\n%s", snd_strerror(
+					err));
+			throw(error_message);
+		}
+
+		if ((err = snd_pcm_hw_params_set_format(audio->capture_handle,
+				hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
+			sprintf(error_message, "Cannot set sample format.\n%s.",
+					snd_strerror(err));
+			throw(error_message);
+		}
+
+		unsigned int rate = sample_rate;
+
+		if ((err = snd_pcm_hw_params_set_rate_near(audio->capture_handle,
+				hw_params, &rate, 0)) < 0) {
+			sprintf(error_message, "Cannot set sample rate.\n%s.",
+					snd_strerror(err));
+			throw(error_message);
+		}
+
+		audio->real_sample_rate = rate;
+
+		if ((err = snd_pcm_hw_params_set_channels(audio->capture_handle,
+				hw_params, channels)) < 0) {
+			sprintf(error_message, "Cannot set channel count.\n%s.",
+					snd_strerror(err));
+			throw(error_message);
+		}
+
+		if ((err = snd_pcm_hw_params(audio->capture_handle, hw_params)) < 0) {
+			sprintf(error_message, "Cannot set parameters.\n%s.", snd_strerror(
+					err));
+			throw(error_message);
+		}
+
+		if ((err = snd_pcm_prepare(audio->capture_handle)) < 0) {
+			sprintf(error_message,
+					"Cannot prepare audio interface for use.\n%s.",
+					snd_strerror(err));
+			throw(error_message);
+		}
+
+		audio->read_buffer = malloc(channels * audio->read_buffer_size
+				* sizeof(SAMPLE_TYPE));
+		memset(audio->read_buffer, 0, audio->read_buffer_size
+				* sizeof(SAMPLE_TYPE));
+	} catch {
+		if (audio->capture_handle != NULL)
+			snd_pcm_close(audio->capture_handle);
+		free(audio);
+		audio = NULL;
+		lingot_msg_add_error(exception);
+	}
+
+	if (hw_params != NULL)
+		snd_pcm_hw_params_free(hw_params);
+
+#	else
+	lingot_msg_add_error(
+			_("The application has not been built with ALSA support"));
+#	endif
+
+	return audio;
+}
+
+void lingot_audio_alsa_destroy(LingotAudioHandler* audio) {
+#	ifdef ALSA
+	if (audio != NULL) {
+		snd_pcm_close(audio->capture_handle);
+		free(audio->read_buffer);
+	}
+#	endif
+}
+
+int lingot_audio_alsa_read(LingotAudioHandler* audio) {
+#	ifdef ALSA
+	int temp_sret;
+	int i;
+
+	temp_sret = snd_pcm_readi(audio->capture_handle, audio->read_buffer,
+			audio->read_buffer_size);
+
+	//	if (rand() < 0.001 * RAND_MAX)
+	//		temp_sret = 0;
+
+	if (temp_sret != audio->read_buffer_size) {
+		char buff[100];
+		sprintf(buff, _("Read from audio interface failed.\n%s."),
+				snd_strerror(temp_sret));
+		lingot_msg_add_error(buff);
+		return -1;
+	}
+
+	// float point conversion
+	for (i = 0; i < audio->read_buffer_size; i++) {
+		audio->flt_read_buffer[i] = audio->read_buffer[i];
+	}
+
+#	endif
+
+	return 0;
+}
+
+LingotAudioSystemProperties* lingot_audio_alsa_get_audio_system_properties(
+		audio_system_t audio_system) {
+
+	LingotAudioSystemProperties* result =
+			(LingotAudioSystemProperties*) malloc(1
+					* sizeof(LingotAudioSystemProperties));
+
+	// TODO
+	result->forced_sample_rate = 0;
+	result->n_devices = 0;
+	result->devices = NULL;
+
+	result->n_sample_rates = 5;
+	result->sample_rates = malloc(result->n_sample_rates * sizeof(int));
+	result->sample_rates[0] = 8000;
+	result->sample_rates[1] = 11025;
+	result->sample_rates[2] = 22050;
+	result->sample_rates[3] = 44100;
+	result->sample_rates[4] = 48000;
+
+	return result;
+}
+