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

---
 include/sound.hpp | 27 +++++++++++++++
 project.conf      |  5 ++-
 src/sound.cpp     | 84 +++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 115 insertions(+), 1 deletion(-)
 create mode 100644 include/sound.hpp
 create mode 100644 src/sound.cpp

diff --git a/include/sound.hpp b/include/sound.hpp
new file mode 100644
index 0000000..41dd043
--- /dev/null
+++ b/include/sound.hpp
@@ -0,0 +1,27 @@
+#ifndef GPU_SCREEN_RECORDER_H
+#define GPU_SCREEN_RECORDER_H
+
+typedef struct {
+    void *handle;
+    char *buffer;
+    int buffer_size;
+    unsigned int frames;
+} SoundDevice;
+
+/*
+    Get a sound device by name, returning the device into the @device parameter.
+    The device should be closed with @sound_device_close after it has been used
+    to clean up internal resources.
+    Returns 0 on success, or a negative value on failure.
+*/
+int sound_device_get_by_name(SoundDevice *device, const char *name = "default", unsigned int num_channels = 1, unsigned int period_frame_size = 32);
+
+int sound_device_close(SoundDevice *device);
+
+/*
+    Returns the next chunk of audio into @buffer.
+    Returns the size of the buffer, or a negative value on failure.
+*/
+int sound_device_read_next_chunk(SoundDevice *device, char **buffer);
+
+#endif /* GPU_SCREEN_RECORDER_H */
diff --git a/project.conf b/project.conf
index bc30fab..f7a8e64 100644
--- a/project.conf
+++ b/project.conf
@@ -6,6 +6,7 @@ platforms = ["posix"]
 
 [config]
 include_dirs = ["/opt/cuda/targets/x86_64-linux/include"]
+libs = ["/usr/lib/libcuda.so"]
 
 [dependencies]
 ffnvcodec = ">=9"
@@ -20,4 +21,6 @@ xcomposite = ">=0.2"
 xdamage = "1"
 
 # TODO: Remove this dependency, this is needed right now for glfwMakeContextCurrent
-glfw3 = "3"
\ No newline at end of file
+glfw3 = "3"
+
+alsa = "1"
\ No newline at end of file
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;
+}