Merge tag 'for-upstream' of https://gitlab.com/bonzini/qemu into staging

[mirror_qemu.git] / audio / audio_template.h

diff --git a/audio/audio_template.h b/audio/audio_template.h
index dd87170cbdf98d8ebed37053abe73d49833f21a9..dc0c74aa74645f549e198c8e2d4b47c856602955 100644 (file)
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -95,55 +95,54 @@ static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
 
 static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
 {
-    g_free (sw->buf);
+    g_free(sw->resample_buf.buffer);
+    sw->resample_buf.buffer = NULL;
+    sw->resample_buf.size = 0;
 
     if (sw->rate) {
         st_rate_stop (sw->rate);
     }
-
-    sw->buf = NULL;
     sw->rate = NULL;
 }
 
 static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
 {
-    int samples;
+    HW *hw = sw->hw;
+    uint64_t samples;
 
     if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
         return 0;
     }
 
-#ifdef DAC
-    samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio;
-#else
-    samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32;
-#endif
-    if (audio_bug(__func__, samples < 0)) {
-        dolog("Can not allocate buffer for `%s' (%d samples)\n",
-              SW_NAME(sw), samples);
-        return -1;
-    }
-
+    samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq);
     if (samples == 0) {
-        HW *hw = sw->hw;
-        size_t f_fe_min;
+        uint64_t f_fe_min;
+        uint64_t f_be = (uint32_t)hw->info.freq;
 
         /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
-        f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size;
+        f_fe_min = (f_be + HWBUF.size - 1) / HWBUF.size;
         qemu_log_mask(LOG_UNIMP,
                       AUDIO_CAP ": The guest selected a " NAME " sample rate"
-                      " of %d Hz for %s. Only sample rates >= %zu Hz are"
-                      " supported.\n",
+                      " of %d Hz for %s. Only sample rates >= %" PRIu64 " Hz"
+                      " are supported.\n",
                       sw->info.freq, sw->name, f_fe_min);
         return -1;
     }
 
-    sw->buf = g_new0(st_sample, samples);
+    /*
+     * Allocate one additional audio frame that is needed for upsampling
+     * if the resample buffer size is small. For large buffer sizes take
+     * care of overflows and truncation.
+     */
+    samples = samples < SIZE_MAX ? samples + 1 : SIZE_MAX;
+    sw->resample_buf.buffer = g_new0(st_sample, samples);
+    sw->resample_buf.size = samples;
+    sw->resample_buf.pos = 0;
 
 #ifdef DAC
-    sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);
+    sw->rate = st_rate_start(sw->info.freq, hw->info.freq);
 #else
-    sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq);
+    sw->rate = st_rate_start(hw->info.freq, sw->info.freq);
 #endif
 
     return 0;
@@ -162,11 +161,8 @@ static int glue (audio_pcm_sw_init_, TYPE) (
     sw->hw = hw;
     sw->active = 0;
 #ifdef DAC
-    sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq;
     sw->total_hw_samples_mixed = 0;
     sw->empty = 1;
-#else
-    sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
 #endif
 
     if (sw->info.is_float) {
@@ -366,6 +362,10 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev)
     case AUDIODEV_DRIVER_PA:
         return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE);
 #endif
+#ifdef CONFIG_AUDIO_PIPEWIRE
+    case AUDIODEV_DRIVER_PIPEWIRE:
+        return qapi_AudiodevPipewirePerDirectionOptions_base(dev->u.pipewire.TYPE);
+#endif
 #ifdef CONFIG_AUDIO_SDL
     case AUDIODEV_DRIVER_SDL:
         return qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE);