adjusting volume and getting current volume.

ao-play-async/ao_playasync.c

@@ -109,6 +109,11 @@ typedef struct {
     int             paused;
 
     ao_device      *ao_device;
+    int             dev_bits_per_sample;
+    int             dev_endianess;
+    int             dev_channels;
+    int             dev_rate;
+
 #ifdef USE_WINDOWS_THREADS
     HANDLE          mutex;
     HANDLE          pause_mutex;
@@ -128,6 +133,7 @@ typedef struct {
     double          music_duration;
     int             at_music_id;
     int             buf_size;
+    int             volume_in_10000;    // volume in 100000 steps, i.e. 100000 equals 100%
 } AO_Handle;
 
 
@@ -222,6 +228,55 @@ static void clear(AO_Handle *h)
     MUTEX_UNLOCK(h->clear_mutex);
 }
 
+static int inline littleEndian()
+{
+  int n = 1;
+  return (*(char *)&n) == 1;
+}
+
+static void inline adjustVolume(AO_Handle *handle, char *_buf, int buf_size, int volume_in_10000)
+{
+  int bytes_per_sample = (handle->dev_bits_per_sample >> 3);
+  register int endianess = handle->dev_endianess;
+  register unsigned char *buf = (unsigned char *) _buf;
+
+  register int i;
+  register long long sample;
+  register int k;
+
+  int little_endian = (endianess == AO_FMT_LITTLE);
+  if (!little_endian && endianess == AO_FMT_NATIVE) little_endian = littleEndian();
+
+  for(i = 0; i < buf_size; i += bytes_per_sample) {
+    if (little_endian) {
+      sample = buf[bytes_per_sample + i - 1] > 127 ? -1 : 0;
+      for(k = bytes_per_sample + i - 1; k >= i; k--) {
+        sample <<= 8;
+        sample |= buf[k];
+      }
+    } else {
+      sample = (buf[i] > 127) ? -1 : 0;
+      for(k = i; k < (i + bytes_per_sample); k++) {
+        sample <<= 8;
+        sample += buf[k];
+      }
+    }
+    sample *= volume_in_10000;
+    sample /= 10000;
+    if (little_endian) {
+      for(k = i; k < (i + bytes_per_sample); k++) {
+        buf[k] = sample&0xff;
+        sample >>= 8;
+      }
+    } else {
+      for(k = i + bytes_per_sample - 1; k >= i; k--) {
+        buf[k] = sample&0xff;
+        sample >>= 8;
+      }
+    }
+  }
+}
+
 #ifdef USE_PTHREADS
 static void *run(void *arg)
 #endif
@@ -243,6 +298,11 @@ static DWORD run(LPVOID arg)
           handle->music_duration = q->music_duration;
           handle->at_music_id = q->music_id;
 
+          if (handle->volume_in_10000 != 10000) {
+              // adjust volume
+              adjustVolume(handle, q->buf, q->buflen, handle->volume_in_10000);
+          }
+
           if (q->command == STOP) {
               go_on = (1 == 0);
           } else {
@@ -311,6 +371,12 @@ void *ao_create_async(int bits, int rate, int channels, int byte_format)
     }
 
     handle->ao_device = dev;
+    handle->dev_bits_per_sample = bits;
+    handle->dev_channels = channels;
+    handle->dev_rate = rate;
+    handle->dev_endianess = byte_format;
+    handle->volume_in_10000 = 10000;
+
     handle->play_queue = NULL;
     handle->last_frame = NULL;
     handle->at_second = -1;
@@ -428,14 +494,15 @@ void ao_stop_async(void *ao_handle)
 #define AO_FMT_BIG    2
 #define AO_FMT_NATIVE 4
 
-static inline void make_sample_bytes(int32_t sample, int bytes_per_sample, int endianess, unsigned char b[4])
+static inline void make_sample_bytes(int32_t sample, int bytes_per_sample, int big_endian, unsigned char b[4])
 {
   int i;
+
   for (i = 0; i < bytes_per_sample; i++) {
      b[i] = sample&0xff;
-     sample = sample >> 8;
+     sample >>=  8;
   }
-  if (endianess == AO_FMT_BIG) {
+  if (big_endian) {
      unsigned char b1[4] = { 0, 0, 0, 0 };
      for(i = 0; i < bytes_per_sample; i++) {
         b1[bytes_per_sample - i - 1] = b[i];
@@ -451,7 +518,12 @@ void *convertFlac(void *mem, int buf_len, BufferInfo_t *info, int *audio_size)
   // buf_size equals number of samples of 32bit for all channels. So buf_size for flac = 4 * buf_len * channels
 
   int bytes = info->sample_bits / 8;
-  int endianness = info->endiannes;
+  int endianess = info->endiannes;
+
+  int little_endian = (endianess == AO_FMT_LITTLE);
+  if (!little_endian && endianess == AO_FMT_NATIVE) little_endian = littleEndian();
+  int big_endian = !little_endian;
+
   int store_size = info->channels * bytes * buf_len;
   unsigned char *new_mem = (unsigned char *) malloc(store_size);
   *audio_size = store_size;
@@ -464,7 +536,7 @@ void *convertFlac(void *mem, int buf_len, BufferInfo_t *info, int *audio_size)
       int32_t *chan = buffer[channel];
       int32_t sample = chan[k];
       unsigned char b[4];
-      make_sample_bytes(sample, bytes, endianness, b);
+      make_sample_bytes(sample, bytes, big_endian, b);
       for(int j = 0; j < bytes; j++) {
         new_mem[i++] = b[j];
       }
@@ -557,6 +629,28 @@ int ao_bufsize_async(void *ao_handle)
     return s;
 }
 
+void ao_set_volume_async(void *ao_handle, double percentage)
+{
+    AO_Handle *h = (AO_Handle *) ao_handle;
+    MUTEX_LOCK(h->mutex);
+    int volume_10000 = (int) (percentage * 100.0);
+    if (volume_10000 >= 9990 && volume_10000 <= 10010) {
+      volume_10000 = 10000;
+    }
+    h->volume_in_10000 = volume_10000;
+    MUTEX_UNLOCK(h->mutex);
+}
+
+
+double ao_volume_async(void *ao_handle)
+{
+    AO_Handle *h = (AO_Handle *) ao_handle;
+    MUTEX_LOCK(h->mutex);
+    double volume = h->volume_in_10000 / 100.0;
+    MUTEX_UNLOCK(h->mutex);
+    return volume;
+}
+
 void ao_pause_async(void *ao_handle, int paused)
 {
     AO_Handle *h = (AO_Handle *) ao_handle;
@@ -578,3 +672,4 @@ void ao_pause_async(void *ao_handle, int paused)
 }
 
 
+

ao-play-async/ao_playasync.h

@@ -40,6 +40,9 @@ AOPLAYASYNC_EXPORT double ao_music_duration_async(void *handle);
 
 AOPLAYASYNC_EXPORT void ao_pause_async(void *ao_handle, int paused);
 
+AOPLAYASYNC_EXPORT void ao_set_volume_async(void *ao_handle, double percentage);
+AOPLAYASYNC_EXPORT double ao_volume_async(void *ao_handle);
+
 AOPLAYASYNC_EXPORT int ao_bufsize_async(void *handle);
 
 #endif // AO_PLAYASYNC_H