From 7c8c97b1a8f844d7e0e682abe0dcba01c2973a66 Mon Sep 17 00:00:00 2001
From: dec05eba <dec05eba@protonmail.com>
Date: Sun, 29 Mar 2020 09:10:24 +0200
Subject: add alsa

---
 src/sound.cpp | 84 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 84 insertions(+)
 create mode 100644 src/sound.cpp

(limited to 'src/sound.cpp')

diff --git a/src/sound.cpp b/src/sound.cpp
new file mode 100644
index 0000000..fcfa7ae
--- /dev/null
+++ b/src/sound.cpp
@@ -0,0 +1,84 @@
+#include "../include/sound.hpp"
+
+#include <stdlib.h>
+#include <stdio.h>
+
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#include <alsa/asoundlib.h>
+
+int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int num_channels, unsigned int period_frame_size) {
+    int rc;
+    snd_pcm_t *handle;
+
+    rc = snd_pcm_open(&handle, name, SND_PCM_STREAM_CAPTURE, 0);
+    if(rc < 0) {
+        fprintf(stderr, "unable to open pcm device 'default', reason: %s\n", snd_strerror(rc));
+        return rc;
+    }
+
+    snd_pcm_hw_params_t *params;
+    snd_pcm_hw_params_alloca(&params);
+    // Fill the params with default values
+    snd_pcm_hw_params_any(handle, params);
+    // Interleaved mode
+    snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
+    // Signed 16--bit little-endian format
+    snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
+    snd_pcm_hw_params_set_channels(handle, params, num_channels);
+
+    // 44100 bits/second samling rate (CD quality)
+    unsigned int val = 44100;
+    int dir;
+    snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);
+
+    snd_pcm_uframes_t frames = period_frame_size;
+    snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir);
+
+    // Write the parmeters to the driver
+    rc = snd_pcm_hw_params(handle, params);
+    if(rc < 0) {
+        fprintf(stderr, "unable to set hw parameters, reason: %s\n", snd_strerror(rc));
+        snd_pcm_close(handle);
+        return rc;
+    }
+
+    // Use a buffer large enough to hold one period
+    snd_pcm_hw_params_get_period_size(params, &frames, &dir);
+    int buffer_size = frames * 2 * num_channels; // 2 bytes/sample, @num_channels channels
+    char *buffer = (char*)malloc(buffer_size);
+    if(!buffer) {
+        fprintf(stderr, "failed to allocate buffer for audio\n");
+        snd_pcm_close(handle);
+        return -1;
+    }
+
+    device->handle = (void*)handle;
+    device->buffer = buffer;
+    device->buffer_size = buffer_size;
+    device->frames = frames;
+    return 0;
+}
+
+int sound_device_close(SoundDevice *device) {
+    /* TODO: Is this also needed in @sound_device_get_by_name on failure? */
+    snd_pcm_drain((snd_pcm_t*)device->handle);
+    snd_pcm_close((snd_pcm_t*)device->handle);
+    free(device->buffer);
+}
+
+int sound_device_read_next_chunk(SoundDevice *device, char **buffer) {
+    int rc = snd_pcm_readi((snd_pcm_t*)device->handle, device->buffer, device->frames);
+    if (rc == -EPIPE) {
+        /* overrun */
+        fprintf(stderr, "overrun occured\n");
+        snd_pcm_prepare((snd_pcm_t*)device->handle);
+        return 0;
+    } else if(rc < 0) {
+        fprintf(stderr, "failed to read from sound device, reason: %s\n", snd_strerror(rc));
+        return rc;
+    } else if (rc != (int)device->frames) {
+        fprintf(stderr, "short read, read %d frames\n", rc);
+    }
+    *buffer = device->buffer;
+    return rc;
+}
-- 
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