Merge remote-tracking branch 'remotes/kraxel/tags/audio-20190821-pull-request' into staging

audio: second batch of -audiodev support, adding support for multiple backends.

# gpg: Signature made Wed 21 Aug 2019 09:40:37 BST
# gpg:                using RSA key 4CB6D8EED3E87138
# 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-20190821-pull-request:
  audio: fix memory leak reported by ASAN
  audio: use size_t where makes sense
  audio: remove read and write pcm_ops
  paaudio: fix playback glitches
  audio: do not run each backend in audio_run
  audio: remove audio_MIN, audio_MAX
  paaudio: properly disconnect streams in fini_*
  paaudio: do not move stream when sink/source name is specified
  audio: audiodev= parameters no longer optional when -audiodev present
  paaudio: prepare for multiple audiodev
  audio: add audiodev properties to frontends
  audio: add audiodev property to vnc and wav_capture
  audio: basic support for multi backend audio
  audio: reduce glob_audio_state usage
  audio: Add missing fall through comments

Signed-off-by: Peter Maydell <peter.maydell@linaro.org>

1 files changed, 9 insertions, 8 deletions

diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 4fee0673d9..f17e8d8dce 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -235,10 +235,10 @@ static void hda_audio_input_timer(void *opaque)
         goto out_timer;
     }
 
-    int64_t to_transfer = audio_MIN(wpos - rpos, wanted_rpos - rpos);
+    int64_t to_transfer = MIN(wpos - rpos, wanted_rpos - rpos);
     while (to_transfer) {
         uint32_t start = (rpos & B_MASK);
-        uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
+        uint32_t chunk = MIN(B_SIZE - start, to_transfer);
         int rc = hda_codec_xfer(
                 &st->state->hda, st->stream, false, st->buf + start, chunk);
         if (!rc) {
@@ -263,13 +263,13 @@ static void hda_audio_input_cb(void *opaque, int avail)
     int64_t wpos = st->wpos;
     int64_t rpos = st->rpos;
 
-    int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), avail);
+    int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), avail);
 
     hda_timer_sync_adjust(st, -((wpos - rpos) + to_transfer - (B_SIZE >> 1)));
 
     while (to_transfer) {
         uint32_t start = (uint32_t) (wpos & B_MASK);
-        uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
+        uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
         uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
         wpos += read;
         to_transfer -= read;
@@ -299,10 +299,10 @@ static void hda_audio_output_timer(void *opaque)
         goto out_timer;
     }
 
-    int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
+    int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
     while (to_transfer) {
         uint32_t start = (wpos & B_MASK);
-        uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
+        uint32_t chunk = MIN(B_SIZE - start, to_transfer);
         int rc = hda_codec_xfer(
                 &st->state->hda, st->stream, true, st->buf + start, chunk);
         if (!rc) {
@@ -327,7 +327,7 @@ static void hda_audio_output_cb(void *opaque, int avail)
     int64_t wpos = st->wpos;
     int64_t rpos = st->rpos;
 
-    int64_t to_transfer = audio_MIN(wpos - rpos, avail);
+    int64_t to_transfer = MIN(wpos - rpos, avail);
 
     if (wpos - rpos == B_SIZE) {
         /* drop buffer, reset timer adjust */
@@ -342,7 +342,7 @@ static void hda_audio_output_cb(void *opaque, int avail)
 
     while (to_transfer) {
         uint32_t start = (uint32_t) (rpos & B_MASK);
-        uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
+        uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
         uint32_t written = AUD_write(st->voice.out, st->buf + start, chunk);
         rpos += written;
         to_transfer -= written;
@@ -841,6 +841,7 @@ static const VMStateDescription vmstate_hda_audio = {
 };
 
 static Property hda_audio_properties[] = {
+    DEFINE_AUDIO_PROPERTIES(HDAAudioState, card),
     DEFINE_PROP_UINT32("debug", HDAAudioState, debug,   0),
     DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer,  true),
     DEFINE_PROP_BOOL("use-timer", HDAAudioState, use_timer,  true),