Added audio backends, fifo support, see #78

README.md

@@ -68,6 +68,21 @@ GLava aims to be compatible with _most_ EWMH compliant window managers. Below is
 
 Note that some WMs listed without issues have specific overrides when using the `--desktop` flag. See `shaders/env_*.glsl` files for details.
 
+## Reading from MPD's FIFO output
+
+Add the following to your `~/.config/mpd.conf`:
+
+```
+audio_output {
+    type                    "fifo"
+    name                    "glava_fifo"
+    path                    "/tmp/mpd.fifo"
+    format                  "22050:16:2"
+}
+```
+
+Note the `22050` sample rate -- this is the reccommended setting for GLava. Restart MPD (if nessecary) and start GLava with `glava --audio=fifo`.
+
 ## Licensing
 
 GLava is licensed under the terms of the GPLv3, with the exemption of `khrplatform.h`, which is licensed under the terms in its header. GLava includes some (heavily modified) source code that originated from [cava](https://github.com/karlstav/cava), which was initially provided under the MIT license. The source files that originated from cava are the following:

fifo.c

@@ -7,65 +7,112 @@
 #include <fcntl.h>
 #include <math.h>
 #include <time.h>
+#include <string.h>
+#include <errno.h>
+#include <poll.h>
 
 #include "fifo.h"
 
-void* input_fifo(void* data) {
+/* Implementation struct storage */
-	struct audio_data* audio = (struct audio_data *) data;
-	int fd;
-	int n = 0;
-	signed char buf[1024];
-	int tempr, templ, lo;
-	int q, i;
-	int t = 0;
-	int size = 1024;
-	int bytes = 0;
-	int flags;
-	struct timespec req = { .tv_sec = 0, .tv_nsec = 10000000 };
 
-	fd = open(audio->source, O_RDONLY);
+typeof(*audio_impls) audio_impls[sizeof(audio_impls) / sizeof(struct audio_impl*)] = {};
-	flags = fcntl(fd, F_GETFL, 0);
+size_t audio_impls_idx = 0;
-	fcntl(fd, F_SETFL, flags | O_NONBLOCK);
 
-	while (1) {
+/* FIFO backend */
-		bytes = read(fd, buf, sizeof(buf));
 
-		if (bytes == -1) { /* if no bytes read, sleep 10ms and zero shared buffer */
+static void init(struct audio_data* audio) {
-			nanosleep (&req, NULL);
+    if (!audio->source) {
-			t++;
+        audio->source = strdup("/tmp/mpd.fifo");
-			if (t > 10) {
+    }
-				for (i = 0; i < 2048; i++)audio->audio_out_l[i] = 0;
-				for (i = 0; i < 2048; i++)audio->audio_out_r[i] = 0;
-				t = 0;
 }
-		} else { /* if bytes read, go ahead */
-			t = 0;
-			for (q = 0; q < (size / 4); q++) {
 
-				tempr = (buf[4 * q + 3] << 2);
+static void* entry(void* data) {
+    struct audio_data* audio = (struct audio_data *) data;
     
-				lo =  (buf[4 * q + 2] >> 6);
+    float* bl = (float*) audio->audio_out_l;
-				if (lo < 0) lo = abs(lo) + 1;
+    float* br = (float*) audio->audio_out_r;
-				if (tempr >= 0) tempr = tempr + lo;
+    size_t fsz = audio->audio_buf_sz;
-				else tempr = tempr - lo;
+    size_t ssz = audio->sample_sz;
     
-				templ = (buf[4 * q + 1] << 2);
+    int fd;
+    int16_t buf[ssz / 2];
+    size_t q;
+    int timeout = 50;
     
-				lo =  (buf[4 * q] >> 6);
+    struct timespec tv_last = {}, tv;
-				if (lo < 0) lo = abs(lo) + 1;
+    bool measured = false;
-				if (templ >= 0) templ = templ + lo;
-				else templ = templ - lo;
     
-				if (audio->channels == 1) audio->audio_out_l[n] = (tempr + templ) / 2;
+    if ((fd = open(audio->source, O_RDONLY)) == -1) {
+        fprintf(stderr, "failed to open FIFO audio source \"%s\": %s\n", audio->source, strerror(errno));
+        exit(EXIT_FAILURE);
+    }
+
+    struct pollfd pfd = {
+        .fd     = fd,
+        .events = POLLIN
+    };
+
+    size_t buffer_offset = (fsz - (ssz / 4));
+    
+    while (true) {
+
+        /* The poll timeout is set to accommodate an approximate UPS, but has little purpose except
+           for effectively setting the rate of empty samples in the event of the FIFO descriptor
+           blocking for long periods of time. */
+        
+        switch (poll(&pfd, 1, timeout)) {
+            case -1:
+                fprintf(stderr, "FIFO backend: poll() failed (%s)\n", strerror(errno));
+                exit(EXIT_FAILURE);
+            case 0:
+                pthread_mutex_lock(&audio->mutex);
+                
+                memmove(bl, &bl[ssz / 4], buffer_offset * sizeof(float));
+                memmove(br, &br[ssz / 4], buffer_offset * sizeof(float));
+                
+                for (q = 0; q < (ssz / 4); ++q) bl[buffer_offset + q] = 0;
+                for (q = 0; q < (ssz / 4); ++q) br[buffer_offset + q] = 0;
+            
+                audio->modified = true;
+                
+                pthread_mutex_unlock(&audio->mutex);
+                break;
+            default: {
+                read(fd, buf, sizeof(buf));
+                clock_gettime(CLOCK_REALTIME, measured ? &tv : &tv_last);
+                if (measured) {
+                    /* Set the timeout slightly higher than the delay between samples to prevent empty writes */
+                    timeout = (((tv.tv_sec - tv_last.tv_sec) * 1000) + ((tv.tv_nsec - tv_last.tv_nsec) / 1000000)) + 1;
+                    tv_last = tv;
+                } else measured = true;
+            
+                pthread_mutex_lock(&audio->mutex);
+                
+                memmove(bl, &bl[ssz / 4], buffer_offset * sizeof(float));
+                memmove(br, &br[ssz / 4], buffer_offset * sizeof(float));
+            
+                for (size_t n = 0, q = 0; q < (ssz / 2); q += 2) {
+
+                    size_t idx = (fsz - (ssz / 4)) + n;
+                
+                    if (audio->channels == 1) {
+                        float sample = ((buf[q] + buf[q + 1]) / 2) / (float) 65535;
+                        bl[idx] = sample;
+                        br[idx] = sample;
+                    }
                 
-				/* stereo storing channels in buffer */
                     if (audio->channels == 2) {
-					audio->audio_out_l[n] = templ;
+                        bl[idx] = buf[q] / (float) 65535;
-					audio->audio_out_r[n] = tempr;
+                        br[idx] = buf[q + 1] / (float) 65535;
                     }
                 
                     n++;
-				if (n == 2048 - 1)n = 0;
+                }
+            
+                audio->modified = true;
+            
+                pthread_mutex_unlock(&audio->mutex);
+                break;
             }
         }
         
@@ -78,3 +125,5 @@ void* input_fifo(void* data) {
     
     return 0;
 }
+
+AUDIO_ATTACH(fifo);

fifo.h

@@ -14,4 +14,26 @@ struct audio_data {
     pthread_mutex_t mutex;
 };
 
-void* input_fifo(void* data);
+struct audio_impl {
+    const char* name;
+    void  (*init)(struct audio_data* data);
+    void* (*entry)(void* data);
+};
+
+#define AUDIO_FUNC(F)                                   \
+    .F = (typeof(((struct audio_impl*) NULL)->F)) &F
+
+extern struct audio_impl* audio_impls[4];
+extern size_t audio_impls_idx;
+
+static inline void register_audio_impl(struct audio_impl* impl) { audio_impls[audio_impls_idx++] = impl; }
+
+#define AUDIO_ATTACH(N)                                         \
+    static struct audio_impl N##_var = {                        \
+        .name = #N,                                             \
+        AUDIO_FUNC(init),                                       \
+        AUDIO_FUNC(entry),                                      \
+    };                                                          \
+    void __attribute__((constructor)) _##N##_construct(void) { \
+        register_audio_impl(&N##_var);                          \
+    }

glava.c

@@ -177,6 +177,7 @@ static const char* help_str =
     "-b, --backend           specifies a window creation backend to use. By default, the most\n"
     "                          appropriate backend will be used for the underlying windowing\n"
     "                          system.\n"
+    "-a, --audio=BACKEND     specifies an audio input backend to use.\n"
     "-V, --version           print application version and exit\n"
     "\n"
     "The REQUEST argument is evaluated identically to the \'#request\' preprocessor directive\n"
@@ -188,13 +189,17 @@ static const char* help_str =
     "The FILE argument may be any file path. All specified file paths are relative to the\n"
     "active configuration root (usually ~/.config/glava if present).\n"
     "\n"
+    "The BACKEND argument may be any of the following strings (for this particular build):\n"
+    "%s"
+    "\n"
     GLAVA_VERSION_STRING "\n";
 
-static const char* opt_str = "dhvVe:Cm:b:r:";
+static const char* opt_str = "dhvVe:Cm:b:r:a:";
 static struct option p_opts[] = {
     {"help",        no_argument,       0, 'h'},
     {"verbose",     no_argument,       0, 'v'},
     {"desktop",     no_argument,       0, 'd'},
+    {"audio",       required_argument, 0, 'a'},
     {"request",     required_argument, 0, 'r'},
     {"entry",       required_argument, 0, 'e'},
     {"force-mod",   required_argument, 0, 'm'},
@@ -226,7 +231,8 @@ int main(int argc, char** argv) {
         * user_path     = SHADER_USER_PATH,
         * entry         = "rc.glsl",
         * force         = NULL,
-        * backend      = NULL;
+        * backend       = NULL,
+        * audio_impl_name = "pulseaudio";
     const char* system_shader_paths[] = { user_path, install_path, NULL };
     
     char** requests    = malloc(1);
@@ -244,18 +250,25 @@ int main(int argc, char** argv) {
             case 'e': entry           = optarg; break;
             case 'm': force           = optarg; break;
             case 'b': backend         = optarg; break;
+            case 'a': audio_impl_name = optarg; break;
             case '?': exit(EXIT_FAILURE); break;
             case 'V':
                 puts(GLAVA_VERSION_STRING);
                 exit(EXIT_SUCCESS);
                 break;
             default:
-            case 'h':
+            case 'h': {
-                printf(help_str, argc > 0 ? argv[0] : "glava");
+                char buf[2048];
+                size_t bsz = 0;
+                for (size_t t = 0; t < audio_impls_idx; ++t)
+                    bsz += snprintf(buf + bsz, sizeof(buf) - bsz, "\t\"%s\"%s\n", audio_impls[t]->name,
+                                    !strcmp(audio_impls[t]->name, audio_impl_name) ? " (default)" : "");
+                printf(help_str, argc > 0 ? argv[0] : "glava", buf);
                 exit(EXIT_SUCCESS);
                 break;
             }
         }
+    }
 
     if (copy_mode) {
         copy_cfg(install_path, user_path, verbose);
@@ -306,13 +319,27 @@ int main(int argc, char** argv) {
         .sample_sz    = rd->samplesize_request,
         .modified     = false
     };
-    if (!audio.source) {
+
-        get_pulse_default_sink(&audio);
+    struct audio_impl* impl = NULL;
-        if (verbose) printf("Using default PulseAudio sink: %s\n", audio.source);
+    
+    for (size_t t = 0; t < audio_impls_idx; ++t) {
+        if (!strcmp(audio_impls[t]->name, audio_impl_name)) {
+            impl = audio_impls[t];
+            break;
+        }
     }
 
+    if (!impl) {
+        fprintf(stderr, "The specified audio backend (\"%s\") is not available.\n", audio_impl_name);
+        exit(EXIT_FAILURE);
+    }
+    
+    impl->init(&audio);
+    
+    if (verbose) printf("Using audio source: %s\n", audio.source);
+    
     pthread_t thread;
-    pthread_create(&thread, NULL, input_pulse, (void*) &audio);
+    pthread_create(&thread, NULL, impl->entry, (void*) &audio);
     
     float lb[rd->bufsize_request], rb[rd->bufsize_request];
     while (rd->alive) {

pulse_input.c

@@ -54,7 +54,9 @@ static void pulseaudio_context_state_callback(pa_context* pulseaudio_context, vo
 }
 
 
-void get_pulse_default_sink(struct audio_data* audio) {
+static void init(struct audio_data* audio) {
+
+    if (audio->source) return;
     
 	pa_mainloop_api* mainloop_api;
 	pa_context* pulseaudio_context;
@@ -106,7 +108,7 @@ void get_pulse_default_sink(struct audio_data* audio) {
 #error "Unsupported float format (requires 32 bit IEEE (little or big endian) floating point support)"
 #endif
 
-void* input_pulse(void* data) {
+static void* entry(void* data) {
     struct audio_data* audio = (struct audio_data*) data;
     int i, n;
     size_t ssz = audio->sample_sz;
@@ -188,3 +190,5 @@ void* input_pulse(void* data) {
     
 	return 0;
 }
+
+AUDIO_ATTACH(pulseaudio);

shaders/rc.glsl

@@ -98,9 +98,14 @@
    default when GLava itself is a desktop window. */
 #request setclickthrough false
 
-/* PulseAudio source. Can be a number or a name of an audio
+/* Audio source
-   sink or device to record from. Set to "auto" to use the
+
-   default output device. */
+   When the "pulseaudio" backend is set, this can be a number or
+   a name of an audio sink or device to record from. Set to "auto"
+   to use the default output device.
+   
+   When the "fifo" backend is set, "auto" is interpreted as
+   "/tmp/mpd.fifo". Otherwise, a valid path should be provided. */
 #request setsource "auto"
 
 /* Buffer swap interval (vsync), set to '0' to prevent
@@ -189,7 +194,11 @@
    
    Lower sample rates also can make output more choppy, when
    not using interpolation. It's generally OK to leave this
-   value unless you have a strange PulseAudio configuration. */
+   value unless you have a strange PulseAudio configuration.
+
+   This option does nothing when using the "fifo" audio
+   backend. Instead, an ideal rate should be be configured
+   in the application generating the output. */
 #request setsamplerate 22050
 
 /*                    ** DEPRECATED **