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) <kraxel@redhat.com>" [full]
# gpg:                 aka "Gerd Hoffmann <gerd@kraxel.org>" [full]
# gpg:                 aka "Gerd Hoffmann (private) <kraxel@gmail.com>" [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 <peter.maydell@linaro.org>

1 files changed, 7 insertions, 11 deletions

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index a8e62542f9..fcc2f62864 100644
--- a/audio/alsaaudio.c
+++ b/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,
 };