Merge remote-tracking branch 'remotes/kraxel/tags/audio-20210115-pull-request' into staging

audio: improvements for sdl, pulse, fsound.
audio: cleanups & codestyle fixes.

# gpg: Signature made Fri 15 Jan 2021 13:20:56 GMT
# gpg:                using RSA key A0328CFFB93A17A79901FE7D4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <[email protected]>" [full]
# gpg:                 aka "Gerd Hoffmann <[email protected]>" [full]
# gpg:                 aka "Gerd Hoffmann (private) <[email protected]>" [full]
# Primary key fingerprint: A032 8CFF B93A 17A7 9901  FE7D 4CB6 D8EE D3E8 7138

* remotes/kraxel/tags/audio-20210115-pull-request: (30 commits)
  audio: space prohibited between function name and parenthesis'('
  audio: Suspect code indent for conditional statements
  audio: Don't use '%#' in format strings
  audio: Fix lines over 90 characters
  audio: foo* bar" should be "foo *bar".
  audio: Add spaces around operator/delete redundant spaces
  audio: Add braces for statements/fix braces' position
  dsoundaudio: fix log message
  dsoundaudio: enable f32 audio sample format
  dsoundaudio: rename dsound_open()
  dsoundaudio: replace GetForegroundWindow()
  paaudio: send recorded data in smaller chunks
  paaudio: limit minreq to 75% of audio timer_rate
  paaudio: comment bugs in functions qpa_init_*
  paaudio: remove unneeded code
  paaudio: wait until the playback stream is ready
  paaudio: wait for PA_STREAM_READY in qpa_write()
  paaudio: avoid to clip samples multiple times
  audio: remove remaining unused plive code
  sdlaudio: enable (in|out).mixing-engine=off
  ...

Signed-off-by: Peter Maydell <[email protected]>

Author: pm215

audio/alsaaudio.c

@@ -278,32 +278,28 @@ static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
     case AUDIO_FORMAT_S16:
         if (endianness) {
             return SND_PCM_FORMAT_S16_BE;
-        }
-        else {
+        } else {
             return SND_PCM_FORMAT_S16_LE;
         }
 
     case AUDIO_FORMAT_U16:
         if (endianness) {
             return SND_PCM_FORMAT_U16_BE;
-        }
-        else {
+        } else {
             return SND_PCM_FORMAT_U16_LE;
         }
 
     case AUDIO_FORMAT_S32:
         if (endianness) {
             return SND_PCM_FORMAT_S32_BE;
-        }
-        else {
+        } else {
             return SND_PCM_FORMAT_S32_LE;
         }
 
     case AUDIO_FORMAT_U32:
         if (endianness) {
             return SND_PCM_FORMAT_U32_BE;
-        }
-        else {
+        } else {
             return SND_PCM_FORMAT_U32_LE;
         }
 
@@ -599,7 +595,7 @@ static int alsa_open(bool in, struct alsa_params_req *req,
     }
 
 #ifdef DEBUG
-    alsa_dump_info(req, obt, obtfmt, pdo);
+    alsa_dump_info(req, obt, obtfmt, apdo);
 #endif
     return 0;
 
@@ -722,8 +718,7 @@ static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
             alsa_logerr (err, "Could not stop %s\n", typ);
             return -1;
         }
-    }
-    else {
+    } else {
         err = snd_pcm_prepare (handle);
         if (err < 0) {
             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
@@ -929,6 +924,7 @@ static struct audio_pcm_ops alsa_pcm_ops = {
     .init_in  = alsa_init_in,
     .fini_in  = alsa_fini_in,
     .read     = alsa_read,
+    .run_buffer_in = audio_generic_run_buffer_in,
     .enable_in = alsa_enable_in,
 };
 

audio/audio.c

@@ -344,8 +344,7 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
 
     if (info->is_signed || info->is_float) {
         memset(buf, 0x00, len * info->bytes_per_frame);
-    }
-    else {
+    } else {
         switch (info->bits) {
         case 8:
             memset(buf, 0x80, len * info->bytes_per_frame);
@@ -584,8 +583,7 @@ static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
     rpos = hw->conv_buf->pos - live;
     if (rpos >= 0) {
         return rpos;
-    }
-    else {
+    } else {
         return hw->conv_buf->size + rpos;
     }
 }
@@ -788,10 +786,14 @@ static int audio_is_timer_needed(AudioState *s)
     HWVoiceOut *hwo = NULL;
 
     while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
-        if (!hwo->poll_mode) return 1;
+        if (!hwo->poll_mode) {
+            return 1;
+        }
     }
     while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
-        if (!hwi->poll_mode) return 1;
+        if (!hwi->poll_mode) {
+            return 1;
+        }
     }
     return 0;
 }
@@ -908,8 +910,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
                     audio_reset_timer (s);
                 }
             }
-        }
-        else {
+        } else {
             if (hw->enabled) {
                 int nb_active = 0;
 
@@ -956,8 +957,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
                 }
             }
             sw->total_hw_samples_acquired = hw->total_samples_captured;
-        }
-        else {
+        } else {
             if (hw->enabled) {
                 int nb_active = 0;
 
@@ -1132,7 +1132,7 @@ static void audio_run_out (AudioState *s)
 
     while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
         size_t played, live, prev_rpos, free;
-        int nb_live, cleanup_required;
+        int nb_live;
 
         live = audio_pcm_hw_get_live_out (hw, &nb_live);
         if (!nb_live) {
@@ -1194,7 +1194,6 @@ static void audio_run_out (AudioState *s)
             audio_capture_mix_and_clear (hw, prev_rpos, played);
         }
 
-        cleanup_required = 0;
         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
             if (!sw->active && sw->empty) {
                 continue;
@@ -1210,7 +1209,6 @@ static void audio_run_out (AudioState *s)
 
             if (!sw->total_hw_samples_mixed) {
                 sw->empty = 1;
-                cleanup_required |= !sw->active && !sw->callback.fn;
             }
 
             if (sw->active) {
@@ -1220,19 +1218,6 @@ static void audio_run_out (AudioState *s)
                 }
             }
         }
-
-        if (cleanup_required) {
-            SWVoiceOut *sw1;
-
-            sw = hw->sw_head.lh_first;
-            while (sw) {
-                sw1 = sw->entries.le_next;
-                if (!sw->active && !sw->callback.fn) {
-                    audio_close_out (sw);
-                }
-                sw = sw1;
-            }
-        }
     }
 }
 
@@ -1241,6 +1226,10 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
     size_t conv = 0;
     STSampleBuffer *conv_buf = hw->conv_buf;
 
+    if (hw->pcm_ops->run_buffer_in) {
+        hw->pcm_ops->run_buffer_in(hw);
+    }
+
     while (samples) {
         size_t proc;
         size_t size = samples * hw->info.bytes_per_frame;
@@ -1381,14 +1370,11 @@ void audio_run(AudioState *s, const char *msg)
 #endif
 }
 
-void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
+void audio_generic_run_buffer_in(HWVoiceIn *hw)
 {
-    ssize_t start;
-
     if (unlikely(!hw->buf_emul)) {
-        size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame;
-        hw->buf_emul = g_malloc(calc_size);
-        hw->size_emul = calc_size;
+        hw->size_emul = hw->samples * hw->info.bytes_per_frame;
+        hw->buf_emul = g_malloc(hw->size_emul);
         hw->pos_emul = hw->pending_emul = 0;
     }
 
@@ -1403,8 +1389,12 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
             break;
         }
     }
+}
+
+void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
+{
+    ssize_t start = (ssize_t)hw->pos_emul - hw->pending_emul;
 
-    start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
     if (start < 0) {
         start += hw->size_emul;
     }
@@ -1446,10 +1436,8 @@ void audio_generic_run_buffer_out(HWVoiceOut *hw)
 void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
 {
     if (unlikely(!hw->buf_emul)) {
-        size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame;
-
-        hw->buf_emul = g_malloc(calc_size);
-        hw->size_emul = calc_size;
+        hw->size_emul = hw->samples * hw->info.bytes_per_frame;
+        hw->buf_emul = g_malloc(hw->size_emul);
         hw->pos_emul = hw->pending_emul = 0;
     }
 
@@ -1505,6 +1493,10 @@ size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
 {
     size_t total = 0;
 
+    if (hw->pcm_ops->run_buffer_in) {
+        hw->pcm_ops->run_buffer_in(hw);
+    }
+
     while (total < size) {
         size_t src_size = size - total;
         void *src = hw->pcm_ops->get_buffer_in(hw, &src_size);
@@ -1540,8 +1532,7 @@ static int audio_driver_init(AudioState *s, struct audio_driver *drv,
         audio_init_nb_voices_in(s, drv);
         s->drv = drv;
         return 0;
-    }
-    else {
+    } else {
         if (msg) {
             dolog("Could not init `%s' audio driver\n", drv->name);
         }
@@ -1856,8 +1847,7 @@ CaptureVoiceOut *AUD_add_capture(
     if (cap) {
         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
         return cap;
-    }
-    else {
+    } else {
         HWVoiceOut *hw;
         CaptureVoiceOut *cap;
 
@@ -2003,7 +1993,7 @@ void audio_create_pdos(Audiodev *dev)
         CASE(JACK, jack, Jack);
         CASE(OSS, oss, Oss);
         CASE(PA, pa, Pa);
-        CASE(SDL, sdl, );
+        CASE(SDL, sdl, Sdl);
         CASE(SPICE, spice, );
         CASE(WAV, wav, );
 

audio/audio_int.h

@@ -172,12 +172,14 @@ struct audio_pcm_ops {
     int    (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
     void   (*fini_in) (HWVoiceIn *hw);
     size_t (*read)    (HWVoiceIn *hw, void *buf, size_t size);
+    void   (*run_buffer_in)(HWVoiceIn *hw);
     void  *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
     void   (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
     void   (*enable_in)(HWVoiceIn *hw, bool enable);
     void   (*volume_in)(HWVoiceIn *hw, Volume *vol);
 };
 
+void audio_generic_run_buffer_in(HWVoiceIn *hw);
 void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
 void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size);
 void audio_generic_run_buffer_out(HWVoiceOut *hw);

audio/audio_legacy.c

@@ -286,7 +286,8 @@ static void handle_sdl(Audiodev *dev)
 {
     /* SDL is output only */
     get_samples_to_usecs("QEMU_SDL_SAMPLES", &dev->u.sdl.out->buffer_length,
-                         &dev->u.sdl.out->has_buffer_length, dev->u.sdl.out);
+        &dev->u.sdl.out->has_buffer_length,
+        qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.out));
 }
 
 /* wav */

audio/audio_template.h

@@ -47,8 +47,7 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
 #ifdef DAC
             dolog ("Driver `%s' does not support " NAME "\n", drv->name);
 #endif
-        }
-        else {
+        } else {
             dolog ("Driver `%s' does not support %d " NAME " voices, max %d\n",
                    drv->name,
                    glue (s->nb_hw_voices_, TYPE),
@@ -204,13 +203,13 @@ static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp)
 
     if (!hw->sw_head.lh_first) {
 #ifdef DAC
-        audio_detach_capture (hw);
+        audio_detach_capture(hw);
 #endif
-        QLIST_REMOVE (hw, entries);
-        glue (hw->pcm_ops->fini_, TYPE) (hw);
-        glue (s->nb_hw_voices_, TYPE) += 1;
-        glue (audio_pcm_hw_free_resources_ ,TYPE) (hw);
-        g_free (hw);
+        QLIST_REMOVE(hw, entries);
+        glue(hw->pcm_ops->fini_, TYPE) (hw);
+        glue(s->nb_hw_voices_, TYPE) += 1;
+        glue(audio_pcm_hw_free_resources_ , TYPE) (hw);
+        g_free(hw);
         *hwp = NULL;
     }
 }
@@ -337,7 +336,7 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev)
     case AUDIODEV_DRIVER_PA:
         return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE);
     case AUDIODEV_DRIVER_SDL:
-        return dev->u.sdl.TYPE;
+        return qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE);
     case AUDIODEV_DRIVER_SPICE:
         return dev->u.spice.TYPE;
     case AUDIODEV_DRIVER_WAV:
@@ -387,8 +386,7 @@ static SW *glue(audio_pcm_create_voice_pair_, TYPE)(
 
     if (pdo->fixed_settings) {
         hw_as = audiodev_to_audsettings(pdo);
-    }
-    else {
+    } else {
         hw_as = *as;
     }
 
@@ -498,8 +496,7 @@ SW *glue (AUD_open_, TYPE) (
         if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, as)) {
             goto fail;
         }
-    }
-    else {
+    } else {
         sw = glue(audio_pcm_create_voice_pair_, TYPE)(s, name, as);
         if (!sw) {
             dolog ("Failed to create voice `%s'\n", name);
@@ -553,8 +550,7 @@ uint64_t glue (AUD_get_elapsed_usec_, TYPE) (SW *sw, QEMUAudioTimeStamp *ts)
 
     if (cur_ts >= old_ts) {
         delta = cur_ts - old_ts;
-    }
-    else {
+    } else {
         delta = UINT64_MAX - old_ts + cur_ts;
     }