[PULL 00/30] Audio 20210115 patches

[PULL 00/30] Audio 20210115 patches

[Gerd Hoffmann]

The following changes since commit 45240eed4f064576d589ea60ebadf3c11d7ab891:

  Merge remote-tracking branch 'remotes/armbru/tags/pull-yank-2021-01-13' int=
o staging (2021-01-13 14:19:24 +0000)

are available in the Git repository at:

  git://git.kraxel.org/qemu tags/audio-20210115-pull-request

for you to fetch changes up to 8abf3feb4d464abadd5133d8810c8a3232cbbe6e:

  audio: space prohibited between function name and parenthesis'(' (2021-01-1=
5 11:49:26 +0100)

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

----------------------------------------------------------------

Volker R=C3=BCmelin (23):
  sdlaudio: remove leftover SDL1.2 code
  audio: fix bit-rotted code
  sdlaudio: add -audiodev sdl,out.buffer-count option
  sdlaudio: don't start playback in init routine
  sdlaudio: always clear the sample buffer
  sdlaudio: fill remaining sample buffer with silence
  sdlaudio: replace legacy functions with modern ones
  audio: split pcm_ops function get_buffer_in
  sdlaudio: add recording functions
  audio: break generic buffer dependency on mixing-engine
  sdlaudio: enable (in|out).mixing-engine=3Doff
  audio: remove remaining unused plive code
  paaudio: avoid to clip samples multiple times
  paaudio: wait for PA_STREAM_READY in qpa_write()
  paaudio: wait until the playback stream is ready
  paaudio: remove unneeded code
  paaudio: comment bugs in functions qpa_init_*
  paaudio: limit minreq to 75% of audio timer_rate
  paaudio: send recorded data in smaller chunks
  dsoundaudio: replace GetForegroundWindow()
  dsoundaudio: rename dsound_open()
  dsoundaudio: enable f32 audio sample format
  dsoundaudio: fix log message

Zhang Han (7):
  audio: Add braces for statements/fix braces' position
  audio: Add spaces around operator/delete redundant spaces
  audio: foo* bar" should be "foo *bar".
  audio: Fix lines over 90 characters
  audio: Don't use '%#' in format strings
  audio: Suspect code indent for conditional statements
  audio: space prohibited between function name and parenthesis'('

 audio/audio_int.h       |   2 +
 audio/audio_template.h  |  26 ++--
 audio/dsound_template.h |   2 +-
 audio/alsaaudio.c       |  18 +--
 audio/audio.c           |  74 +++++-----
 audio/audio_legacy.c    |   3 +-
 audio/audio_win_int.c   |  67 ++++++---
 audio/coreaudio.c       |  17 ++-
 audio/dsoundaudio.c     |  56 +++++---
 audio/jackaudio.c       |   3 +-
 audio/noaudio.c         |   1 +
 audio/ossaudio.c        |  13 +-
 audio/paaudio.c         |  73 ++++++++--
 audio/sdlaudio.c        | 309 ++++++++++++++++++++++++++++------------
 audio/spiceaudio.c      |   1 +
 qapi/audio.json         |  33 ++++-
 qemu-options.hx         |   8 +-
 17 files changed, 471 insertions(+), 235 deletions(-)

--=20
2.29.2

[PULL 01/30] sdlaudio: remove leftover SDL1.2 code

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/sdlaudio.c | 30 +++++++++++++-----------------
 1 file changed, 13 insertions(+), 17 deletions(-)

diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 21b7a0484bec..bf3cfb845616 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -240,28 +240,24 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
     }
 }
 
-#define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args, fail, unlock) \
-    static ret_type glue(sdl_, name)args_decl                           \
-    {                                                                   \
-        ret_type ret;                                                   \
-                                                                        \
-        SDL_LockAudio();                                                \
-                                                                        \
-        ret = glue(audio_generic_, name)args;                           \
-                                                                        \
-        SDL_UnlockAudio();                                              \
-        return ret;                                                     \
+#define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args)      \
+    static ret_type glue(sdl_, name)args_decl                  \
+    {                                                          \
+        ret_type ret;                                          \
+                                                               \
+        SDL_LockAudio();                                       \
+        ret = glue(audio_generic_, name)args;                  \
+        SDL_UnlockAudio();                                     \
+                                                               \
+        return ret;                                            \
     }
 
 SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
-                 (hw, size), *size = 0, sdl_unlock)
+                 (hw, size))
 SDL_WRAPPER_FUNC(put_buffer_out, size_t,
-                 (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
-                 /*nothing*/, sdl_unlock_and_post)
+                 (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size))
 SDL_WRAPPER_FUNC(write, size_t,
-                 (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
-                 /*nothing*/, sdl_unlock_and_post)
-
+                 (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size))
 #undef SDL_WRAPPER_FUNC
 
 static void sdl_fini_out (HWVoiceOut *hw)
-- 
2.29.2

[PULL 02/30] audio: fix bit-rotted code

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/alsaaudio.c | 2 +-
 audio/sdlaudio.c  | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index a8e62542f97e..6787e91bc198 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -599,7 +599,7 @@ 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;
 
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index bf3cfb845616..00cd12ba66a0 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -215,7 +215,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
         return;
     }
 
-    /* dolog ("in callback samples=%zu live=%zu\n", samples, sdl->live); */
+    /* dolog("callback: len=%d avail=%zu\n", len, hw->pending_emul); */
 
     while (hw->pending_emul && len) {
         size_t write_len;
-- 
2.29.2

[PULL 03/30] sdlaudio: add -audiodev sdl,out.buffer-count option

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Currently there is a crackling noise with SDL2 audio playback.
Commit bcf19777df: "audio/sdlaudio: Allow audio playback with
SDL2" already mentioned the crackling noise.

Add an out.buffer-count option to give users a chance to select
sane settings for glitch free audio playback. The idea was taken
from the coreaudio backend.

The in.buffer-count option will be used with one of the next
patches.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Acked-by: Markus Armbruster <armbru@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio_template.h |  2 +-
 audio/audio.c          |  2 +-
 audio/audio_legacy.c   |  3 ++-
 audio/sdlaudio.c       | 11 +++++++++--
 qapi/audio.json        | 33 ++++++++++++++++++++++++++++++++-
 qemu-options.hx        |  8 +++++++-
 6 files changed, 52 insertions(+), 7 deletions(-)

diff --git a/audio/audio_template.h b/audio/audio_template.h
index 8dd48ce14e9d..434df5d5e750 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -337,7 +337,7 @@ 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:
diff --git a/audio/audio.c b/audio/audio.c
index b48471bb3f64..d048d262835e 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -2003,7 +2003,7 @@ 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, );
 
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index ffdbd0bcce8c..0fe827b05735 100644
--- a/audio/audio_legacy.c
+++ b/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 */
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 00cd12ba66a0..431bfcfddd97 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -276,12 +276,18 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
     int endianness;
     int err;
     AudioFormat effective_fmt;
+    AudiodevSdlPerDirectionOptions *spdo = s->dev->u.sdl.out;
     struct audsettings obt_as;
 
     req.freq = as->freq;
     req.format = aud_to_sdlfmt (as->fmt);
     req.channels = as->nchannels;
-    req.samples = audio_buffer_samples(s->dev->u.sdl.out, as, 11610);
+    /*
+     * This is wrong. SDL samples are QEMU frames. The buffer size will be
+     * the requested buffer size multiplied by the number of channels.
+     */
+    req.samples = audio_buffer_samples(
+        qapi_AudiodevSdlPerDirectionOptions_base(spdo), as, 11610);
     req.callback = sdl_callback;
     req.userdata = sdl;
 
@@ -301,7 +307,8 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
     obt_as.endianness = endianness;
 
     audio_pcm_init_info (&hw->info, &obt_as);
-    hw->samples = obt.samples;
+    hw->samples = (spdo->has_buffer_count ? spdo->buffer_count : 4) *
+        obt.samples;
 
     s->initialized = 1;
     s->exit = 0;
diff --git a/qapi/audio.json b/qapi/audio.json
index 072ed79def50..9cba0df8a4e9 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -301,6 +301,37 @@
     '*out':    'AudiodevPaPerDirectionOptions',
     '*server': 'str' } }
 
+##
+# @AudiodevSdlPerDirectionOptions:
+#
+# Options of the SDL audio backend that are used for both playback and
+# recording.
+#
+# @buffer-count: number of buffers (default 4)
+#
+# Since: 6.0
+##
+{ 'struct': 'AudiodevSdlPerDirectionOptions',
+  'base': 'AudiodevPerDirectionOptions',
+  'data': {
+    '*buffer-count': 'uint32' } }
+
+##
+# @AudiodevSdlOptions:
+#
+# Options of the SDL audio backend.
+#
+# @in: options of the recording stream
+#
+# @out: options of the playback stream
+#
+# Since: 6.0
+##
+{ 'struct': 'AudiodevSdlOptions',
+  'data': {
+    '*in':  'AudiodevSdlPerDirectionOptions',
+    '*out': 'AudiodevSdlPerDirectionOptions' } }
+
 ##
 # @AudiodevWavOptions:
 #
@@ -385,6 +416,6 @@
     'jack':      'AudiodevJackOptions',
     'oss':       'AudiodevOssOptions',
     'pa':        'AudiodevPaOptions',
-    'sdl':       'AudiodevGenericOptions',
+    'sdl':       'AudiodevSdlOptions',
     'spice':     'AudiodevGenericOptions',
     'wav':       'AudiodevWavOptions' } }
diff --git a/qemu-options.hx b/qemu-options.hx
index 1698a0c751ff..4e02e9bd7604 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -588,6 +588,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
 #endif
 #ifdef CONFIG_AUDIO_SDL
     "-audiodev sdl,id=id[,prop[=value][,...]]\n"
+    "                in|out.buffer-count= number of buffers\n"
 #endif
 #ifdef CONFIG_SPICE
     "-audiodev spice,id=id[,prop[=value][,...]]\n"
@@ -745,7 +746,12 @@ SRST
 ``-audiodev sdl,id=id[,prop[=value][,...]]``
     Creates a backend using SDL. This backend is available on most
     systems, but you should use your platform's native backend if
-    possible. This backend has no backend specific properties.
+    possible.
+
+    SDL specific options are:
+
+    ``in|out.buffer-count=count``
+        Sets the count of the buffers.
 
 ``-audiodev spice,id=id[,prop[=value][,...]]``
     Creates a backend that sends audio through SPICE. This backend
-- 
2.29.2

[PULL 04/30] sdlaudio: don't start playback in init routine

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Every emulated audio device has a way to enable audio playback. Don't
start playback until the guest enables the audio device. This patch
keeps the SDL2 device pause state in sync with hw->enabled.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Tested-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/sdlaudio.c | 1 -
 1 file changed, 1 deletion(-)

diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 431bfcfddd97..68126a99ab32 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -312,7 +312,6 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
 
     s->initialized = 1;
     s->exit = 0;
-    SDL_PauseAudio (0);
     return 0;
 }
 
-- 
2.29.2

[PULL 05/30] sdlaudio: always clear the sample buffer

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Always fill the remaining audio callback buffer with silence.
SDL 2.0 doesn't initialize the audio callback buffer. This was
an incompatible change compared to SDL 1.2. For reference read
the SDL 1.2 to 2.0 migration guide.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-5-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/sdlaudio.c | 35 +++++++++++++++++------------------
 1 file changed, 17 insertions(+), 18 deletions(-)

diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 68126a99ab32..79eed23849ea 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -211,27 +211,26 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
     SDLAudioState *s = &glob_sdl;
     HWVoiceOut *hw = &sdl->hw;
 
-    if (s->exit) {
-        return;
-    }
+    if (!s->exit) {
 
-    /* dolog("callback: len=%d avail=%zu\n", len, hw->pending_emul); */
+        /* dolog("callback: len=%d avail=%zu\n", len, hw->pending_emul); */
 
-    while (hw->pending_emul && len) {
-        size_t write_len;
-        ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
-        if (start < 0) {
-            start += hw->size_emul;
+        while (hw->pending_emul && len) {
+            size_t write_len;
+            ssize_t start = (ssize_t)hw->pos_emul - hw->pending_emul;
+            if (start < 0) {
+                start += hw->size_emul;
+            }
+            assert(start >= 0 && start < hw->size_emul);
+
+            write_len = MIN(MIN(hw->pending_emul, len),
+                            hw->size_emul - start);
+
+            memcpy(buf, hw->buf_emul + start, write_len);
+            hw->pending_emul -= write_len;
+            len -= write_len;
+            buf += write_len;
         }
-        assert(start >= 0 && start < hw->size_emul);
-
-        write_len = MIN(MIN(hw->pending_emul, len),
-                        hw->size_emul - start);
-
-        memcpy(buf, hw->buf_emul + start, write_len);
-        hw->pending_emul -= write_len;
-        len -= write_len;
-        buf += write_len;
     }
 
     /* clear remaining buffer that we couldn't fill with data */
-- 
2.29.2

[PULL 06/30] sdlaudio: fill remaining sample buffer with silence

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Fill the remaining sample buffer with silence. To fill it with
zeroes is wrong for unsigned samples because this is silence
with a DC bias.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-6-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/sdlaudio.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 79eed23849ea..01ae4c600e80 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -235,7 +235,8 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
 
     /* clear remaining buffer that we couldn't fill with data */
     if (len) {
-        memset(buf, 0, len);
+        audio_pcm_info_clear_buf(&hw->info, buf,
+                                 len / hw->info.bytes_per_frame);
     }
 }
 
-- 
2.29.2

[PULL 07/30] sdlaudio: replace legacy functions with modern ones

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

With the modern audio functions it's possible to add new
features like audio recording.

As a side effect this patch fixes a bug where SDL2 can't be used
on Windows. This bug was reported on the qemu-devel mailing list at

https://lists.nongnu.org/archive/html/qemu-devel/2020-01/msg04043.html

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-7-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/sdlaudio.c | 107 ++++++++++++++++++++++-------------------------
 1 file changed, 50 insertions(+), 57 deletions(-)

diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 01ae4c600e80..47968c502027 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -41,15 +41,11 @@
 
 typedef struct SDLVoiceOut {
     HWVoiceOut hw;
-} SDLVoiceOut;
-
-static struct SDLAudioState {
     int exit;
     int initialized;
-    bool driver_created;
     Audiodev *dev;
-} glob_sdl;
-typedef struct SDLAudioState SDLAudioState;
+    SDL_AudioDeviceID devid;
+} SDLVoiceOut;
 
 static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
 {
@@ -155,9 +151,10 @@ static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness)
     return 0;
 }
 
-static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
+static SDL_AudioDeviceID sdl_open(SDL_AudioSpec *req, SDL_AudioSpec *obt,
+                                  int rec)
 {
-    int status;
+    SDL_AudioDeviceID devid;
 #ifndef _WIN32
     int err;
     sigset_t new, old;
@@ -166,18 +163,19 @@ static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
     err = sigfillset (&new);
     if (err) {
         dolog ("sdl_open: sigfillset failed: %s\n", strerror (errno));
-        return -1;
+        return 0;
     }
     err = pthread_sigmask (SIG_BLOCK, &new, &old);
     if (err) {
         dolog ("sdl_open: pthread_sigmask failed: %s\n", strerror (err));
-        return -1;
+        return 0;
     }
 #endif
 
-    status = SDL_OpenAudio (req, obt);
-    if (status) {
-        sdl_logerr ("SDL_OpenAudio failed\n");
+    devid = SDL_OpenAudioDevice(NULL, rec, req, obt, 0);
+    if (!devid) {
+        sdl_logerr("SDL_OpenAudioDevice for %s failed\n",
+                   rec ? "recording" : "playback");
     }
 
 #ifndef _WIN32
@@ -190,30 +188,32 @@ static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
         exit (EXIT_FAILURE);
     }
 #endif
-    return status;
+    return devid;
 }
 
-static void sdl_close (SDLAudioState *s)
+static void sdl_close_out(SDLVoiceOut *sdl)
 {
-    if (s->initialized) {
-        SDL_LockAudio();
-        s->exit = 1;
-        SDL_UnlockAudio();
-        SDL_PauseAudio (1);
-        SDL_CloseAudio ();
-        s->initialized = 0;
+    if (sdl->initialized) {
+        SDL_LockAudioDevice(sdl->devid);
+        sdl->exit = 1;
+        SDL_UnlockAudioDevice(sdl->devid);
+        SDL_PauseAudioDevice(sdl->devid, 1);
+        sdl->initialized = 0;
+    }
+    if (sdl->devid) {
+        SDL_CloseAudioDevice(sdl->devid);
+        sdl->devid = 0;
     }
 }
 
-static void sdl_callback (void *opaque, Uint8 *buf, int len)
+static void sdl_callback_out(void *opaque, Uint8 *buf, int len)
 {
     SDLVoiceOut *sdl = opaque;
-    SDLAudioState *s = &glob_sdl;
     HWVoiceOut *hw = &sdl->hw;
 
-    if (!s->exit) {
+    if (!sdl->exit) {
 
-        /* dolog("callback: len=%d avail=%zu\n", len, hw->pending_emul); */
+        /* dolog("callback_out: len=%d avail=%zu\n", len, hw->pending_emul); */
 
         while (hw->pending_emul && len) {
             size_t write_len;
@@ -240,43 +240,44 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
     }
 }
 
-#define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args)      \
+#define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args, dir) \
     static ret_type glue(sdl_, name)args_decl                  \
     {                                                          \
         ret_type ret;                                          \
+        glue(SDLVoice, dir) *sdl = (glue(SDLVoice, dir) *)hw;  \
                                                                \
-        SDL_LockAudio();                                       \
+        SDL_LockAudioDevice(sdl->devid);                       \
         ret = glue(audio_generic_, name)args;                  \
-        SDL_UnlockAudio();                                     \
+        SDL_UnlockAudioDevice(sdl->devid);                     \
                                                                \
         return ret;                                            \
     }
 
 SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
-                 (hw, size))
+                 (hw, size), Out)
 SDL_WRAPPER_FUNC(put_buffer_out, size_t,
-                 (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size))
+                 (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size), Out)
 SDL_WRAPPER_FUNC(write, size_t,
-                 (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size))
+                 (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size), Out)
 #undef SDL_WRAPPER_FUNC
 
-static void sdl_fini_out (HWVoiceOut *hw)
+static void sdl_fini_out(HWVoiceOut *hw)
 {
-    (void) hw;
+    SDLVoiceOut *sdl = (SDLVoiceOut *)hw;
 
-    sdl_close (&glob_sdl);
+    sdl_close_out(sdl);
 }
 
 static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
                         void *drv_opaque)
 {
-    SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
-    SDLAudioState *s = &glob_sdl;
+    SDLVoiceOut *sdl = (SDLVoiceOut *)hw;
     SDL_AudioSpec req, obt;
     int endianness;
     int err;
     AudioFormat effective_fmt;
-    AudiodevSdlPerDirectionOptions *spdo = s->dev->u.sdl.out;
+    Audiodev *dev = drv_opaque;
+    AudiodevSdlPerDirectionOptions *spdo = dev->u.sdl.out;
     struct audsettings obt_as;
 
     req.freq = as->freq;
@@ -288,16 +289,18 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
      */
     req.samples = audio_buffer_samples(
         qapi_AudiodevSdlPerDirectionOptions_base(spdo), as, 11610);
-    req.callback = sdl_callback;
+    req.callback = sdl_callback_out;
     req.userdata = sdl;
 
-    if (sdl_open (&req, &obt)) {
+    sdl->dev = dev;
+    sdl->devid = sdl_open(&req, &obt, 0);
+    if (!sdl->devid) {
         return -1;
     }
 
     err = sdl_to_audfmt(obt.format, &effective_fmt, &endianness);
     if (err) {
-        sdl_close (s);
+        sdl_close_out(sdl);
         return -1;
     }
 
@@ -310,41 +313,31 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
     hw->samples = (spdo->has_buffer_count ? spdo->buffer_count : 4) *
         obt.samples;
 
-    s->initialized = 1;
-    s->exit = 0;
+    sdl->initialized = 1;
+    sdl->exit = 0;
     return 0;
 }
 
 static void sdl_enable_out(HWVoiceOut *hw, bool enable)
 {
-    SDL_PauseAudio(!enable);
+    SDLVoiceOut *sdl = (SDLVoiceOut *)hw;
+
+    SDL_PauseAudioDevice(sdl->devid, !enable);
 }
 
 static void *sdl_audio_init(Audiodev *dev)
 {
-    SDLAudioState *s = &glob_sdl;
-    if (s->driver_created) {
-        sdl_logerr("Can't create multiple sdl backends\n");
-        return NULL;
-    }
-
     if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
         sdl_logerr ("SDL failed to initialize audio subsystem\n");
         return NULL;
     }
 
-    s->driver_created = true;
-    s->dev = dev;
-    return s;
+    return dev;
 }
 
 static void sdl_audio_fini (void *opaque)
 {
-    SDLAudioState *s = opaque;
-    sdl_close (s);
     SDL_QuitSubSystem (SDL_INIT_AUDIO);
-    s->driver_created = false;
-    s->dev = NULL;
 }
 
 static struct audio_pcm_ops sdl_pcm_ops = {
-- 
2.29.2

[PULL 08/30] audio: split pcm_ops function get_buffer_in

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Split off pcm_ops function run_buffer_in from get_buffer_in and
call run_buffer_in before get_buffer_in.

The next patch only needs the generic buffer management part
from audio_generic_get_buffer_in().

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-8-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio_int.h  |  2 ++
 audio/alsaaudio.c  |  1 +
 audio/audio.c      | 18 ++++++++++++++----
 audio/jackaudio.c  |  1 +
 audio/noaudio.c    |  1 +
 audio/ossaudio.c   |  1 +
 audio/spiceaudio.c |  1 +
 7 files changed, 21 insertions(+), 4 deletions(-)

diff --git a/audio/audio_int.h b/audio/audio_int.h
index 4775857bf245..06f0913835b0 100644
--- a/audio/audio_int.h
+++ b/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);
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 6787e91bc198..5a871aaf6b48 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -929,6 +929,7 @@ 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,
 };
 
diff --git a/audio/audio.c b/audio/audio.c
index d048d262835e..480b3cce1ffb 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1241,6 +1241,10 @@ 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;
@@ -1381,10 +1385,8 @@ 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);
@@ -1403,8 +1405,12 @@ 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;
     }
@@ -1505,6 +1511,10 @@ 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);
diff --git a/audio/jackaudio.c b/audio/jackaudio.c
index 3b7c18443dbe..f8afb5cc318c 100644
--- a/audio/jackaudio.c
+++ b/audio/jackaudio.c
@@ -657,6 +657,7 @@ static struct audio_pcm_ops jack_pcm_ops = {
     .init_in        = qjack_init_in,
     .fini_in        = qjack_fini_in,
     .read           = qjack_read,
+    .run_buffer_in  = audio_generic_run_buffer_in,
     .enable_in      = qjack_enable_in
 };
 
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 05798ea21032..aac87dbc93fb 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -124,6 +124,7 @@ static struct audio_pcm_ops no_pcm_ops = {
     .init_in  = no_init_in,
     .fini_in  = no_fini_in,
     .read     = no_read,
+    .run_buffer_in = audio_generic_run_buffer_in,
     .enable_in = no_enable_in
 };
 
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index a7dcaa31adeb..c1db89f23300 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -762,6 +762,7 @@ static struct audio_pcm_ops oss_pcm_ops = {
     .init_in  = oss_init_in,
     .fini_in  = oss_fini_in,
     .read     = oss_read,
+    .run_buffer_in = audio_generic_run_buffer_in,
     .enable_in = oss_enable_in
 };
 
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 8967cca12925..999bfbde47c5 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -293,6 +293,7 @@ static struct audio_pcm_ops audio_callbacks = {
     .init_in  = line_in_init,
     .fini_in  = line_in_fini,
     .read     = line_in_read,
+    .run_buffer_in = audio_generic_run_buffer_in,
     .enable_in = line_in_enable,
 #if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
     .volume_in = line_in_volume,
-- 
2.29.2

[PULL 09/30] sdlaudio: add recording functions

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Add audio recording functions. SDL 2.0.5 or later is required to
use the recording functions. Playback continues to work with
earlier SDL 2.0 versions.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-9-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/sdlaudio.c | 142 ++++++++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 139 insertions(+), 3 deletions(-)

diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 47968c502027..445cae8de578 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -47,6 +47,14 @@ typedef struct SDLVoiceOut {
     SDL_AudioDeviceID devid;
 } SDLVoiceOut;
 
+typedef struct SDLVoiceIn {
+    HWVoiceIn hw;
+    int exit;
+    int initialized;
+    Audiodev *dev;
+    SDL_AudioDeviceID devid;
+} SDLVoiceIn;
+
 static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
 {
     va_list ap;
@@ -240,6 +248,45 @@ static void sdl_callback_out(void *opaque, Uint8 *buf, int len)
     }
 }
 
+static void sdl_close_in(SDLVoiceIn *sdl)
+{
+    if (sdl->initialized) {
+        SDL_LockAudioDevice(sdl->devid);
+        sdl->exit = 1;
+        SDL_UnlockAudioDevice(sdl->devid);
+        SDL_PauseAudioDevice(sdl->devid, 1);
+        sdl->initialized = 0;
+    }
+    if (sdl->devid) {
+        SDL_CloseAudioDevice(sdl->devid);
+        sdl->devid = 0;
+    }
+}
+
+static void sdl_callback_in(void *opaque, Uint8 *buf, int len)
+{
+    SDLVoiceIn *sdl = opaque;
+    HWVoiceIn *hw = &sdl->hw;
+
+    if (sdl->exit) {
+        return;
+    }
+
+    /* dolog("callback_in: len=%d pending=%zu\n", len, hw->pending_emul); */
+
+    while (hw->pending_emul < hw->size_emul && len) {
+        size_t read_len = MIN(len, MIN(hw->size_emul - hw->pos_emul,
+                                       hw->size_emul - hw->pending_emul));
+
+        memcpy(hw->buf_emul + hw->pos_emul, buf, read_len);
+
+        hw->pending_emul += read_len;
+        hw->pos_emul = (hw->pos_emul + read_len) % hw->size_emul;
+        len -= read_len;
+        buf += read_len;
+    }
+}
+
 #define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args, dir) \
     static ret_type glue(sdl_, name)args_decl                  \
     {                                                          \
@@ -253,13 +300,30 @@ static void sdl_callback_out(void *opaque, Uint8 *buf, int len)
         return ret;                                            \
     }
 
+#define SDL_WRAPPER_VOID_FUNC(name, args_decl, args, dir)      \
+    static void glue(sdl_, name)args_decl                      \
+    {                                                          \
+        glue(SDLVoice, dir) *sdl = (glue(SDLVoice, dir) *)hw;  \
+                                                               \
+        SDL_LockAudioDevice(sdl->devid);                       \
+        glue(audio_generic_, name)args;                        \
+        SDL_UnlockAudioDevice(sdl->devid);                     \
+    }
+
 SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
                  (hw, size), Out)
 SDL_WRAPPER_FUNC(put_buffer_out, size_t,
                  (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size), Out)
 SDL_WRAPPER_FUNC(write, size_t,
                  (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size), Out)
+SDL_WRAPPER_FUNC(read, size_t, (HWVoiceIn *hw, void *buf, size_t size),
+                 (hw, buf, size), In)
+SDL_WRAPPER_FUNC(get_buffer_in, void *, (HWVoiceIn *hw, size_t *size),
+                 (hw, size), In)
+SDL_WRAPPER_VOID_FUNC(put_buffer_in, (HWVoiceIn *hw, void *buf, size_t size),
+                      (hw, buf, size), In)
 #undef SDL_WRAPPER_FUNC
+#undef SDL_WRAPPER_VOID_FUNC
 
 static void sdl_fini_out(HWVoiceOut *hw)
 {
@@ -325,6 +389,69 @@ static void sdl_enable_out(HWVoiceOut *hw, bool enable)
     SDL_PauseAudioDevice(sdl->devid, !enable);
 }
 
+static void sdl_fini_in(HWVoiceIn *hw)
+{
+    SDLVoiceIn *sdl = (SDLVoiceIn *)hw;
+
+    sdl_close_in(sdl);
+}
+
+static int sdl_init_in(HWVoiceIn *hw, audsettings *as, void *drv_opaque)
+{
+    SDLVoiceIn *sdl = (SDLVoiceIn *)hw;
+    SDL_AudioSpec req, obt;
+    int endianness;
+    int err;
+    AudioFormat effective_fmt;
+    Audiodev *dev = drv_opaque;
+    AudiodevSdlPerDirectionOptions *spdo = dev->u.sdl.in;
+    struct audsettings obt_as;
+
+    req.freq = as->freq;
+    req.format = aud_to_sdlfmt(as->fmt);
+    req.channels = as->nchannels;
+    /* SDL samples are QEMU frames */
+    req.samples = audio_buffer_frames(
+        qapi_AudiodevSdlPerDirectionOptions_base(spdo), as, 11610);
+    req.callback = sdl_callback_in;
+    req.userdata = sdl;
+
+    sdl->dev = dev;
+    sdl->devid = sdl_open(&req, &obt, 1);
+    if (!sdl->devid) {
+        return -1;
+    }
+
+    err = sdl_to_audfmt(obt.format, &effective_fmt, &endianness);
+    if (err) {
+        sdl_close_in(sdl);
+        return -1;
+    }
+
+    obt_as.freq = obt.freq;
+    obt_as.nchannels = obt.channels;
+    obt_as.fmt = effective_fmt;
+    obt_as.endianness = endianness;
+
+    audio_pcm_init_info(&hw->info, &obt_as);
+    hw->samples = (spdo->has_buffer_count ? spdo->buffer_count : 4) *
+        obt.samples;
+    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;
+
+    sdl->initialized = 1;
+    sdl->exit = 0;
+    return 0;
+}
+
+static void sdl_enable_in(HWVoiceIn *hw, bool enable)
+{
+    SDLVoiceIn *sdl = (SDLVoiceIn *)hw;
+
+    SDL_PauseAudioDevice(sdl->devid, !enable);
+}
+
 static void *sdl_audio_init(Audiodev *dev)
 {
     if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
@@ -350,6 +477,15 @@ static struct audio_pcm_ops sdl_pcm_ops = {
   /* wrapper for audio_generic_put_buffer_out */
     .put_buffer_out = sdl_put_buffer_out,
     .enable_out = sdl_enable_out,
+    .init_in = sdl_init_in,
+    .fini_in = sdl_fini_in,
+  /* wrapper for audio_generic_read */
+    .read = sdl_read,
+  /* wrapper for audio_generic_get_buffer_in */
+    .get_buffer_in = sdl_get_buffer_in,
+  /* wrapper for audio_generic_put_buffer_in */
+    .put_buffer_in = sdl_put_buffer_in,
+    .enable_in = sdl_enable_in,
 };
 
 static struct audio_driver sdl_audio_driver = {
@@ -360,9 +496,9 @@ static struct audio_driver sdl_audio_driver = {
     .pcm_ops        = &sdl_pcm_ops,
     .can_be_default = 1,
     .max_voices_out = 1,
-    .max_voices_in  = 0,
-    .voice_size_out = sizeof (SDLVoiceOut),
-    .voice_size_in  = 0
+    .max_voices_in  = 1,
+    .voice_size_out = sizeof(SDLVoiceOut),
+    .voice_size_in  = sizeof(SDLVoiceIn),
 };
 
 static void register_audio_sdl(void)
-- 
2.29.2

[PULL 10/30] audio: break generic buffer dependency on mixing-engine

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Break the unnecessary dependency of the generic buffer management
code on mixing-engine. This is required for the next patch.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio.c | 11 ++++-------
 1 file changed, 4 insertions(+), 7 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 480b3cce1ffb..22d769db0c99 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1388,9 +1388,8 @@ void audio_run(AudioState *s, const char *msg)
 void audio_generic_run_buffer_in(HWVoiceIn *hw)
 {
     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;
     }
 
@@ -1452,10 +1451,8 @@ 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;
     }
 
-- 
2.29.2

[PULL 11/30] sdlaudio: enable (in|out).mixing-engine=off

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Enable the SDL2 backend options -audiodev sdl,out.mixing-
engine=off,in.mixing-engine=off.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-11-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/sdlaudio.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 445cae8de578..c68c62a3e4f1 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -495,8 +495,8 @@ static struct audio_driver sdl_audio_driver = {
     .fini           = sdl_audio_fini,
     .pcm_ops        = &sdl_pcm_ops,
     .can_be_default = 1,
-    .max_voices_out = 1,
-    .max_voices_in  = 1,
+    .max_voices_out = INT_MAX,
+    .max_voices_in  = INT_MAX,
     .voice_size_out = sizeof(SDLVoiceOut),
     .voice_size_in  = sizeof(SDLVoiceIn),
 };
-- 
2.29.2

[PULL 12/30] audio: remove remaining unused plive code

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Commit 73ad33ef7b "audio: remove plive" forgot to remove this code.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-12-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio.c | 17 +----------------
 1 file changed, 1 insertion(+), 16 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 22d769db0c99..34c9cb918279 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -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;
-            }
-        }
     }
 }
 
-- 
2.29.2

[PULL 13/30] paaudio: avoid to clip samples multiple times

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

The pulseaudio backend currently converts, clips and copies audio
playback samples in the mixing-engine sample buffer multiple
times.

In qpa_get_buffer_out() the function pa_stream_begin_write()
returns a rather large buffer and this allows audio_pcm_hw_run_out()
in audio/audio.c to copy all samples in the mixing-engine buffer
to the pulse audio buffer. Immediately after copying, qpa_write()
notices with a call to pa_stream_writable_size() that pulse audio
only needs a smaller part of the copied samples and ignores the
rest. This copy and ignore process happens several times for each
audio sample.

To fix this behaviour, call pa_stream_writable_size() in
qpa_get_buffer_out() to limit the number of samples
audio_pcm_hw_run_out() will convert. With this change the
pulseaudio pcm_ops functions put_buffer_out and write are no
longer identical and a separate qpa_put_buffer_out is needed.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-13-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/paaudio.c | 32 +++++++++++++++++++++++++++++++-
 1 file changed, 31 insertions(+), 1 deletion(-)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index b05208469831..229bcfcae838 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -207,6 +207,7 @@ static void *qpa_get_buffer_out(HWVoiceOut *hw, size_t *size)
     PAVoiceOut *p = (PAVoiceOut *) hw;
     PAConnection *c = p->g->conn;
     void *ret;
+    size_t l;
     int r;
 
     pa_threaded_mainloop_lock(c->mainloop);
@@ -214,12 +215,19 @@ static void *qpa_get_buffer_out(HWVoiceOut *hw, size_t *size)
     CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
                     "pa_threaded_mainloop_lock failed\n");
 
+    l = pa_stream_writable_size(p->stream);
+    CHECK_SUCCESS_GOTO(c, l != (size_t) -1, unlock_and_fail,
+                       "pa_stream_writable_size failed\n");
+
     *size = -1;
     r = pa_stream_begin_write(p->stream, &ret, size);
     CHECK_SUCCESS_GOTO(c, r >= 0, unlock_and_fail,
                        "pa_stream_begin_write failed\n");
 
     pa_threaded_mainloop_unlock(c->mainloop);
+    if (*size > l) {
+        *size = l;
+    }
     return ret;
 
 unlock_and_fail:
@@ -228,6 +236,28 @@ unlock_and_fail:
     return NULL;
 }
 
+static size_t qpa_put_buffer_out(HWVoiceOut *hw, void *data, size_t length)
+{
+    PAVoiceOut *p = (PAVoiceOut *)hw;
+    PAConnection *c = p->g->conn;
+    int r;
+
+    pa_threaded_mainloop_lock(c->mainloop);
+
+    CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+                    "pa_threaded_mainloop_lock failed\n");
+
+    r = pa_stream_write(p->stream, data, length, NULL, 0LL, PA_SEEK_RELATIVE);
+    CHECK_SUCCESS_GOTO(c, r >= 0, unlock_and_fail, "pa_stream_write failed\n");
+
+    pa_threaded_mainloop_unlock(c->mainloop);
+    return length;
+
+unlock_and_fail:
+    pa_threaded_mainloop_unlock(c->mainloop);
+    return 0;
+}
+
 static size_t qpa_write(HWVoiceOut *hw, void *data, size_t length)
 {
     PAVoiceOut *p = (PAVoiceOut *) hw;
@@ -861,7 +891,7 @@ static struct audio_pcm_ops qpa_pcm_ops = {
     .fini_out = qpa_fini_out,
     .write    = qpa_write,
     .get_buffer_out = qpa_get_buffer_out,
-    .put_buffer_out = qpa_write, /* pa handles it */
+    .put_buffer_out = qpa_put_buffer_out,
     .volume_out = qpa_volume_out,
 
     .init_in  = qpa_init_in,
-- 
2.29.2

[PULL 14/30] paaudio: wait for PA_STREAM_READY in qpa_write()

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Don't call pa_stream_writable_size() in qpa_write() before the
playback stream is ready. This prevents a lot of the following
pulseaudio error messages.

pulseaudio: pa_stream_writable_size failed
pulseaudio: Reason: Bad state

To reproduce start qemu with
-parallel none -device gus,audiodev=audio0
-audiodev pa,id=audio0,out.mixing-engine=off

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-14-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/paaudio.c | 6 ++++++
 1 file changed, 6 insertions(+)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index 229bcfcae838..1a7252b16d6d 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -269,6 +269,11 @@ static size_t qpa_write(HWVoiceOut *hw, void *data, size_t length)
 
     CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
                     "pa_threaded_mainloop_lock failed\n");
+    if (pa_stream_get_state(p->stream) != PA_STREAM_READY) {
+        /* wait for stream to become ready */
+        l = 0;
+        goto unlock;
+    }
 
     l = pa_stream_writable_size(p->stream);
 
@@ -282,6 +287,7 @@ static size_t qpa_write(HWVoiceOut *hw, void *data, size_t length)
     r = pa_stream_write(p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
     CHECK_SUCCESS_GOTO(c, r >= 0, unlock_and_fail, "pa_stream_write failed\n");
 
+unlock:
     pa_threaded_mainloop_unlock(c->mainloop);
     return l;
 
-- 
2.29.2

[PULL 15/30] paaudio: wait until the playback stream is ready

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Don't call pa_stream_writable_size() in qpa_get_buffer_out()
before the playback stream is ready. This prevents a lot of the
following pulseaudio error messages.

pulseaudio: pa_stream_writable_size failed
pulseaudio: Reason: Bad state

To reproduce start qemu with
-parallel none -device gus,audiodev=audio0 -audiodev pa,id=audio0

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-15-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/paaudio.c | 7 +++++++
 1 file changed, 7 insertions(+)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index 1a7252b16d6d..4a1ffda753d5 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -214,6 +214,12 @@ static void *qpa_get_buffer_out(HWVoiceOut *hw, size_t *size)
 
     CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
                     "pa_threaded_mainloop_lock failed\n");
+    if (pa_stream_get_state(p->stream) != PA_STREAM_READY) {
+        /* wait for stream to become ready */
+        l = 0;
+        ret = NULL;
+        goto unlock;
+    }
 
     l = pa_stream_writable_size(p->stream);
     CHECK_SUCCESS_GOTO(c, l != (size_t) -1, unlock_and_fail,
@@ -224,6 +230,7 @@ static void *qpa_get_buffer_out(HWVoiceOut *hw, size_t *size)
     CHECK_SUCCESS_GOTO(c, r >= 0, unlock_and_fail,
                        "pa_stream_begin_write failed\n");
 
+unlock:
     pa_threaded_mainloop_unlock(c->mainloop);
     if (*size > l) {
         *size = l;
-- 
2.29.2

[PULL 16/30] paaudio: remove unneeded code

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Commit baea032ec7 "audio/paaudio: fix ignored buffer_length setting"
added code to handle buffer_length defaults. This was unnecessary
because the audio_buffer_* functions in audio/audio.c already handle
this. Remove the unneeded code.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-16-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/paaudio.c | 10 ++--------
 1 file changed, 2 insertions(+), 8 deletions(-)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index 4a1ffda753d5..86038f3e1392 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -539,8 +539,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
 
     audio_pcm_init_info (&hw->info, &obt_as);
     hw->samples = audio_buffer_samples(
-        qapi_AudiodevPaPerDirectionOptions_base(ppdo),
-        &obt_as, ppdo->buffer_length);
+        qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440);
 
     return 0;
 
@@ -587,8 +586,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 
     audio_pcm_init_info (&hw->info, &obt_as);
     hw->samples = audio_buffer_samples(
-        qapi_AudiodevPaPerDirectionOptions_base(ppdo),
-        &obt_as, ppdo->buffer_length);
+        qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440);
 
     return 0;
 
@@ -738,10 +736,6 @@ static void qpa_volume_in(HWVoiceIn *hw, Volume *vol)
 static int qpa_validate_per_direction_opts(Audiodev *dev,
                                            AudiodevPaPerDirectionOptions *pdo)
 {
-    if (!pdo->has_buffer_length) {
-        pdo->has_buffer_length = true;
-        pdo->buffer_length = 46440;
-    }
     if (!pdo->has_latency) {
         pdo->has_latency = true;
         pdo->latency = 15000;
-- 
2.29.2

[PULL 17/30] paaudio: comment bugs in functions qpa_init_*

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

The audio buffer size in audio/paaudio.c is typically larger
than expected. Just comment the bugs in qpa_init_in() and
qpa_init_out() for now. Fixing these bugs may break glitch free
audio playback with fine tuned user audio settings.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-17-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/paaudio.c | 8 ++++++++
 1 file changed, 8 insertions(+)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index 86038f3e1392..ff3dd01c96dd 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -538,6 +538,10 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
     }
 
     audio_pcm_init_info (&hw->info, &obt_as);
+    /*
+     * This is wrong. hw->samples counts in frames. hw->samples will be
+     * number of channels times larger than expected.
+     */
     hw->samples = audio_buffer_samples(
         qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440);
 
@@ -585,6 +589,10 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     }
 
     audio_pcm_init_info (&hw->info, &obt_as);
+    /*
+     * This is wrong. hw->samples counts in frames. hw->samples will be
+     * number of channels times larger than expected.
+     */
     hw->samples = audio_buffer_samples(
         qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440);
 
-- 
2.29.2

[PULL 18/30] paaudio: limit minreq to 75% of audio timer_rate

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Currently with the playback buffer attribute minreq = -1 and flag
PA_STREAM_EARLY_REQUESTS PulseAudio uses minreq = tlength / 4.
To improve audio playback with larger PulseAudio server side
buffers, limit minreq to a maximum of 75% of audio timer_rate.
That way there is a good chance qemu receives a stream buffer
size update before it tries to write data to the playback stream.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-18-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/paaudio.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index ff3dd01c96dd..318686829428 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -517,7 +517,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
     ss.rate = as->freq;
 
     ba.tlength = pa_usec_to_bytes(ppdo->latency, &ss);
-    ba.minreq = -1;
+    ba.minreq = pa_usec_to_bytes(MIN(ppdo->latency >> 2,
+                                     (g->dev->timer_period >> 2) * 3), &ss);
     ba.maxlength = -1;
     ba.prebuf = -1;
 
-- 
2.29.2

[PULL 19/30] paaudio: send recorded data in smaller chunks

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Tell PulseAudio to send recorded audio data in smaller chunks
than timer_period, so there's a good chance that qemu can read
recorded audio data every time it looks for new data.

PulseAudio tries to send buffer updates at a fragsize / 2 rate.
With fragsize = timer_period / 2 * 3 the update rate is 75% of
timer_period. The lower limit for the recording buffer size
maxlength is fragsize * 2.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-19-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/paaudio.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index 318686829428..1e6f4448ce37 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -568,8 +568,9 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     ss.channels = as->nchannels;
     ss.rate = as->freq;
 
-    ba.fragsize = pa_usec_to_bytes(ppdo->latency, &ss);
-    ba.maxlength = pa_usec_to_bytes(ppdo->latency * 2, &ss);
+    ba.fragsize = pa_usec_to_bytes((g->dev->timer_period >> 1) * 3, &ss);
+    ba.maxlength = pa_usec_to_bytes(
+        MAX(ppdo->latency, g->dev->timer_period * 3), &ss);
     ba.minreq = -1;
     ba.prebuf = -1;
 
-- 
2.29.2

[PULL 20/30] dsoundaudio: replace GetForegroundWindow()

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

GetForegroundWindow() doesn't necessarily return the own window
handle. It just returns a handle to the currently active window
and can even return NULL. At the time dsound_open() gets called
the active window is most likely the shell window and not the
QEMU window.

Replace GetForegroundWindow() with GetDesktopWindow() which
always returns a valid window handle, and at the same time
replace the DirectSound buffer flag DSBCAPS_STICKYFOCUS with
DSBCAPS_GLOBALFOCUS where Windows only expects a valid window
handle for DirectSound function SetCooperativeLevel(). The
Microsoft online docs for IDirectSound::SetCooperativeLevel
recommend this in the remarks.

This fixes a bug where you can't hear sound from the guest.

To reproduce start qemu with -machine pcspk-audiodev=audio0
-device intel-hda -device hda-duplex,audiodev=audio0
-audiodev dsound,id=audio0,out.mixing-engine=off
from a shell and start audio playback with the hda device in the
guest. The guest will be silent. To hear guest audio you have to
activate the shell window once.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-20-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/dsound_template.h | 2 +-
 audio/dsoundaudio.c     | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 9c5ce625ab91..0678f2de38be 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -205,7 +205,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
         NULL
         );
 #else
-    bd.dwFlags = DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2;
+    bd.dwFlags = DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2;
     hr = IDirectSound_CreateSoundBuffer (
         s->dsound,
         &bd,
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 4cdf19ab6799..0fbdf770ac75 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -347,7 +347,7 @@ static int dsound_open (dsound *s)
     HRESULT hr;
     HWND hwnd;
 
-    hwnd = GetForegroundWindow ();
+    hwnd = GetDesktopWindow();
     hr = IDirectSound_SetCooperativeLevel (
         s->dsound,
         hwnd,
-- 
2.29.2

[PULL 21/30] dsoundaudio: rename dsound_open()

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Rename dsound_open() to dsound_set_cooperative_level(). The
only task of that function is to set the cooperative level for
DirectSound.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-21-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/dsoundaudio.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 0fbdf770ac75..d3695f3af653 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -342,7 +342,7 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
     dsound_unlock_out (dsb, p1, p2, blen1, blen2);
 }
 
-static int dsound_open (dsound *s)
+static int dsound_set_cooperative_level(dsound *s)
 {
     HRESULT hr;
     HWND hwnd;
@@ -673,7 +673,7 @@ static void *dsound_audio_init(Audiodev *dev)
         }
     }
 
-    err = dsound_open (s);
+    err = dsound_set_cooperative_level(s);
     if (err) {
         dsound_audio_fini (s);
         return NULL;
-- 
2.29.2

[PULL 22/30] dsoundaudio: enable f32 audio sample format

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

Enable the f32 audio sample format for the DirectSound backend.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-22-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio_win_int.c | 65 +++++++++++++++++++++++++++++--------------
 1 file changed, 44 insertions(+), 21 deletions(-)

diff --git a/audio/audio_win_int.c b/audio/audio_win_int.c
index b938fd667bae..b7db34900c7c 100644
--- a/audio/audio_win_int.c
+++ b/audio/audio_win_int.c
@@ -5,6 +5,7 @@
 
 #define AUDIO_CAP "win-int"
 #include <windows.h>
+#include <mmreg.h>
 #include <mmsystem.h>
 
 #include "audio.h"
@@ -16,7 +17,6 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
 {
     memset (wfx, 0, sizeof (*wfx));
 
-    wfx->wFormatTag = WAVE_FORMAT_PCM;
     wfx->nChannels = as->nchannels;
     wfx->nSamplesPerSec = as->freq;
     wfx->nAvgBytesPerSec = as->freq << (as->nchannels == 2);
@@ -26,11 +26,13 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
     switch (as->fmt) {
     case AUDIO_FORMAT_S8:
     case AUDIO_FORMAT_U8:
+        wfx->wFormatTag = WAVE_FORMAT_PCM;
         wfx->wBitsPerSample = 8;
         break;
 
     case AUDIO_FORMAT_S16:
     case AUDIO_FORMAT_U16:
+        wfx->wFormatTag = WAVE_FORMAT_PCM;
         wfx->wBitsPerSample = 16;
         wfx->nAvgBytesPerSec <<= 1;
         wfx->nBlockAlign <<= 1;
@@ -38,6 +40,14 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
 
     case AUDIO_FORMAT_S32:
     case AUDIO_FORMAT_U32:
+        wfx->wFormatTag = WAVE_FORMAT_PCM;
+        wfx->wBitsPerSample = 32;
+        wfx->nAvgBytesPerSec <<= 2;
+        wfx->nBlockAlign <<= 2;
+        break;
+
+    case AUDIO_FORMAT_F32:
+        wfx->wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
         wfx->wBitsPerSample = 32;
         wfx->nAvgBytesPerSec <<= 2;
         wfx->nBlockAlign <<= 2;
@@ -54,12 +64,6 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
 int waveformat_to_audio_settings (WAVEFORMATEX *wfx,
                                   struct audsettings *as)
 {
-    if (wfx->wFormatTag != WAVE_FORMAT_PCM) {
-        dolog ("Invalid wave format, tag is not PCM, but %d\n",
-               wfx->wFormatTag);
-        return -1;
-    }
-
     if (!wfx->nSamplesPerSec) {
         dolog ("Invalid wave format, frequency is zero\n");
         return -1;
@@ -83,23 +87,42 @@ int waveformat_to_audio_settings (WAVEFORMATEX *wfx,
         return -1;
     }
 
-    switch (wfx->wBitsPerSample) {
-    case 8:
-        as->fmt = AUDIO_FORMAT_U8;
-        break;
+    if (wfx->wFormatTag == WAVE_FORMAT_PCM) {
+        switch (wfx->wBitsPerSample) {
+        case 8:
+            as->fmt = AUDIO_FORMAT_U8;
+            break;
 
-    case 16:
-        as->fmt = AUDIO_FORMAT_S16;
-        break;
+        case 16:
+            as->fmt = AUDIO_FORMAT_S16;
+            break;
 
-    case 32:
-        as->fmt = AUDIO_FORMAT_S32;
-        break;
+        case 32:
+            as->fmt = AUDIO_FORMAT_S32;
+            break;
 
-    default:
-        dolog ("Invalid wave format, bits per sample is not "
-               "8, 16 or 32, but %d\n",
-               wfx->wBitsPerSample);
+        default:
+            dolog("Invalid PCM wave format, bits per sample is not "
+                  "8, 16 or 32, but %d\n",
+                  wfx->wBitsPerSample);
+            return -1;
+        }
+    } else if (wfx->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) {
+        switch (wfx->wBitsPerSample) {
+        case 32:
+            as->fmt = AUDIO_FORMAT_F32;
+            break;
+
+        default:
+            dolog("Invalid IEEE_FLOAT wave format, bits per sample is not "
+                  "32, but %d\n",
+                  wfx->wBitsPerSample);
+            return -1;
+        }
+    } else {
+        dolog("Invalid wave format, tag is not PCM and not IEEE_FLOAT, "
+              "but %d\n",
+              wfx->wFormatTag);
         return -1;
     }
 
-- 
2.29.2

[PULL 23/30] dsoundaudio: fix log message

[Gerd Hoffmann]

From: Volker Rümelin <vr_qemu@t-online.de>

There is a mismatch between message and used argument. Change
the argument from frequency to format.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-23-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio_win_int.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/audio/audio_win_int.c b/audio/audio_win_int.c
index b7db34900c7c..5ea8157dfcc8 100644
--- a/audio/audio_win_int.c
+++ b/audio/audio_win_int.c
@@ -54,7 +54,7 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
         break;
 
     default:
-        dolog ("Internal logic error: Bad audio format %d\n", as->freq);
+        dolog("Internal logic error: Bad audio format %d\n", as->fmt);
         return -1;
     }
 
-- 
2.29.2

[PULL 24/30] audio: Add braces for statements/fix braces' position

[Gerd Hoffmann]

From: Zhang Han <zhanghan64@huawei.com>

Fix problems about braces:
-braces are necessary for all arms of if/for/while statements
-else should follow close brace '}'

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-2-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio_template.h | 12 ++++--------
 audio/alsaaudio.c      | 15 +++++----------
 audio/audio.c          | 26 ++++++++++++--------------
 audio/coreaudio.c      |  3 +--
 audio/dsoundaudio.c    |  9 +++------
 audio/ossaudio.c       | 12 ++++--------
 6 files changed, 29 insertions(+), 48 deletions(-)

diff --git a/audio/audio_template.h b/audio/audio_template.h
index 434df5d5e750..f77210e40435 100644
--- a/audio/audio_template.h
+++ b/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),
@@ -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;
     }
 
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 5a871aaf6b48..fcc2f62864fb 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;
         }
 
@@ -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);
diff --git a/audio/audio.c b/audio/audio.c
index 34c9cb918279..6734c8af708c 100644
--- a/audio/audio.c
+++ b/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;
 
@@ -1532,8 +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);
         }
@@ -1848,8 +1847,7 @@ CaptureVoiceOut *AUD_add_capture(
     if (cap) {
         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
         return cap;
-    }
-    else {
+    } else {
         HWVoiceOut *hw;
         CaptureVoiceOut *cap;
 
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 79a9d40bf8ef..408b5871261c 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -524,8 +524,7 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
     } else if (frameRange.mMaximum < frames) {
         core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum;
         dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum);
-    }
-    else {
+    } else {
         core->audioDevicePropertyBufferFrameSize = frames;
     }
 
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index d3695f3af653..e59bd83dc717 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -404,8 +404,7 @@ static void dsound_enable_out(HWVoiceOut *hw, bool enable)
                 dsound_logerr (hr, "Could not stop playing buffer\n");
                 return;
             }
-        }
-        else {
+        } else {
             dolog ("warning: Voice is not playing\n");
         }
     }
@@ -509,8 +508,7 @@ static void dsound_enable_in(HWVoiceIn *hw, bool enable)
                 dsound_logerr (hr, "Could not stop capturing\n");
                 return;
             }
-        }
-        else {
+        } else {
             dolog ("warning: Voice is not capturing\n");
         }
     }
@@ -659,8 +657,7 @@ static void *dsound_audio_init(Audiodev *dev)
         );
     if (FAILED (hr)) {
         dsound_logerr (hr, "Could not create DirectSoundCapture instance\n");
-    }
-    else {
+    } else {
         hr = IDirectSoundCapture_Initialize (s->dsound_capture, NULL);
         if (FAILED (hr)) {
             dsound_logerr (hr, "Could not initialize DirectSoundCapture\n");
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index c1db89f23300..60eff66424b9 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -142,16 +142,14 @@ static int aud_to_ossfmt (AudioFormat fmt, int endianness)
     case AUDIO_FORMAT_S16:
         if (endianness) {
             return AFMT_S16_BE;
-        }
-        else {
+        } else {
             return AFMT_S16_LE;
         }
 
     case AUDIO_FORMAT_U16:
         if (endianness) {
             return AFMT_U16_BE;
-        }
-        else {
+        } else {
             return AFMT_U16_LE;
         }
 
@@ -542,16 +540,14 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
             int trig = 0;
             if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
                 oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
-            }
-            else {
+            } else {
                 trig = PCM_ENABLE_OUTPUT;
                 if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
                     oss_logerr (
                         errno,
                         "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
                         );
-                }
-                else {
+                } else {
                     oss->mmapped = 1;
                 }
             }
-- 
2.29.2

[PULL 25/30] audio: Add spaces around operator/delete redundant spaces

[Gerd Hoffmann]

From: Zhang Han <zhanghan64@huawei.com>

Fix problems about spaces:
-operator needs spaces around it, add them.
-somespaces are redundant, remove them.

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-3-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio_template.h | 2 +-
 audio/coreaudio.c      | 2 +-
 audio/dsoundaudio.c    | 2 +-
 audio/jackaudio.c      | 2 +-
 4 files changed, 4 insertions(+), 4 deletions(-)

diff --git a/audio/audio_template.h b/audio/audio_template.h
index f77210e40435..f82593bb8db2 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -208,7 +208,7 @@ static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp)
         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);
+        glue (audio_pcm_hw_free_resources_ , TYPE) (hw);
         g_free (hw);
         *hwp = NULL;
     }
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 408b5871261c..6ca0d79c1f5e 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -270,7 +270,7 @@ static void coreaudio_logstatus (OSStatus status)
 {
     const char *str = "BUG";
 
-    switch(status) {
+    switch (status) {
     case kAudioHardwareNoError:
         str = "kAudioHardwareNoError";
         break;
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index e59bd83dc717..21c2891772de 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -133,7 +133,7 @@ static void dsound_log_hresult (HRESULT hr)
         break;
 #endif
 #ifdef DSERR_GENERIC
-    case DSERR_GENERIC :
+    case DSERR_GENERIC:
         str = "An undetermined error occurred inside the DirectSound subsystem";
         break;
 #endif
diff --git a/audio/jackaudio.c b/audio/jackaudio.c
index f8afb5cc318c..3031c4e29bd4 100644
--- a/audio/jackaudio.c
+++ b/audio/jackaudio.c
@@ -277,7 +277,7 @@ static int qjack_process(jack_nframes_t nframes, void *arg)
         if (likely(c->enabled)) {
             qjack_buffer_read_l(&c->fifo, buffers, nframes);
         } else {
-            for(int i = 0; i < c->nchannels; ++i) {
+            for (int i = 0; i < c->nchannels; ++i) {
                 memset(buffers[i], 0, nframes * sizeof(float));
             }
         }
-- 
2.29.2

[PULL 26/30] audio: foo* bar" should be "foo *bar".

[Gerd Hoffmann]

From: Zhang Han <zhanghan64@huawei.com>

transfer "foo* " to "foo *"

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-4-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/coreaudio.c | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 6ca0d79c1f5e..b7c02e0e516d 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -421,12 +421,12 @@ COREAUDIO_WRAPPER_FUNC(write, size_t, (HWVoiceOut *hw, void *buf, size_t size),
 /* callback to feed audiooutput buffer */
 static OSStatus audioDeviceIOProc(
     AudioDeviceID inDevice,
-    const AudioTimeStamp* inNow,
-    const AudioBufferList* inInputData,
-    const AudioTimeStamp* inInputTime,
-    AudioBufferList* outOutputData,
-    const AudioTimeStamp* inOutputTime,
-    void* hwptr)
+    const AudioTimeStamp *inNow,
+    const AudioBufferList *inInputData,
+    const AudioTimeStamp *inInputTime,
+    AudioBufferList *outOutputData,
+    const AudioTimeStamp *inOutputTime,
+    void *hwptr)
 {
     UInt32 frameCount, pending_frames;
     void *out = outOutputData->mBuffers[0].mData;
-- 
2.29.2

[PULL 27/30] audio: Fix lines over 90 characters

[Gerd Hoffmann]

From: Zhang Han <zhanghan64@huawei.com>

Fix the line width of code.

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-5-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/dsoundaudio.c | 37 +++++++++++++++++++++++++++----------
 1 file changed, 27 insertions(+), 10 deletions(-)

diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 21c2891772de..3b4afb949631 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -89,7 +89,9 @@ static void dsound_log_hresult (HRESULT hr)
 #endif
 #ifdef DSERR_ALLOCATED
     case DSERR_ALLOCATED:
-        str = "The request failed because resources, such as a priority level, were already in use by another caller";
+        str = "The request failed because resources, "
+              "such as a priority level, were already in use "
+              "by another caller";
         break;
 #endif
 #ifdef DSERR_ALREADYINITIALIZED
@@ -104,7 +106,8 @@ static void dsound_log_hresult (HRESULT hr)
 #endif
 #ifdef DSERR_BADSENDBUFFERGUID
     case DSERR_BADSENDBUFFERGUID:
-        str = "The GUID specified in an audiopath file does not match a valid mix-in buffer";
+        str = "The GUID specified in an audiopath file "
+              "does not match a valid mix-in buffer";
         break;
 #endif
 #ifdef DSERR_BUFFERLOST
@@ -114,22 +117,31 @@ static void dsound_log_hresult (HRESULT hr)
 #endif
 #ifdef DSERR_BUFFERTOOSMALL
     case DSERR_BUFFERTOOSMALL:
-        str = "The buffer size is not great enough to enable effects processing";
+        str = "The buffer size is not great enough to "
+              "enable effects processing";
         break;
 #endif
 #ifdef DSERR_CONTROLUNAVAIL
     case DSERR_CONTROLUNAVAIL:
-        str = "The buffer control (volume, pan, and so on) requested by the caller is not available. Controls must be specified when the buffer is created, using the dwFlags member of DSBUFFERDESC";
+        str = "The buffer control (volume, pan, and so on) "
+              "requested by the caller is not available. "
+              "Controls must be specified when the buffer is created, "
+              "using the dwFlags member of DSBUFFERDESC";
         break;
 #endif
 #ifdef DSERR_DS8_REQUIRED
     case DSERR_DS8_REQUIRED:
-        str = "A DirectSound object of class CLSID_DirectSound8 or later is required for the requested functionality. For more information, see IDirectSound8 Interface";
+        str = "A DirectSound object of class CLSID_DirectSound8 or later "
+              "is required for the requested functionality. "
+              "For more information, see IDirectSound8 Interface";
         break;
 #endif
 #ifdef DSERR_FXUNAVAILABLE
     case DSERR_FXUNAVAILABLE:
-        str = "The effects requested could not be found on the system, or they are in the wrong order or in the wrong location; for example, an effect expected in hardware was found in software";
+        str = "The effects requested could not be found on the system, "
+              "or they are in the wrong order or in the wrong location; "
+              "for example, an effect expected in hardware "
+              "was found in software";
         break;
 #endif
 #ifdef DSERR_GENERIC
@@ -154,7 +166,8 @@ static void dsound_log_hresult (HRESULT hr)
 #endif
 #ifdef DSERR_NODRIVER
     case DSERR_NODRIVER:
-        str = "No sound driver is available for use, or the given GUID is not a valid DirectSound device ID";
+        str = "No sound driver is available for use, "
+              "or the given GUID is not a valid DirectSound device ID";
         break;
 #endif
 #ifdef DSERR_NOINTERFACE
@@ -169,12 +182,14 @@ static void dsound_log_hresult (HRESULT hr)
 #endif
 #ifdef DSERR_OTHERAPPHASPRIO
     case DSERR_OTHERAPPHASPRIO:
-        str = "Another application has a higher priority level, preventing this call from succeeding";
+        str = "Another application has a higher priority level, "
+              "preventing this call from succeeding";
         break;
 #endif
 #ifdef DSERR_OUTOFMEMORY
     case DSERR_OUTOFMEMORY:
-        str = "The DirectSound subsystem could not allocate sufficient memory to complete the caller's request";
+        str = "The DirectSound subsystem could not allocate "
+               "sufficient memory to complete the caller's request";
         break;
 #endif
 #ifdef DSERR_PRIOLEVELNEEDED
@@ -189,7 +204,9 @@ static void dsound_log_hresult (HRESULT hr)
 #endif
 #ifdef DSERR_UNINITIALIZED
     case DSERR_UNINITIALIZED:
-        str = "The Initialize method has not been called or has not been called successfully before other methods were called";
+        str = "The Initialize method has not been called "
+              "or has not been called successfully "
+              "before other methods were called";
         break;
 #endif
 #ifdef DSERR_UNSUPPORTED
-- 
2.29.2

[PULL 28/30] audio: Don't use '%#' in format strings

[Gerd Hoffmann]

From: Zhang Han <zhanghan64@huawei.com>

Use '0x' prefix instead of '%#'

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-6-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/dsoundaudio.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 3b4afb949631..cfc79c129eee 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -215,7 +215,7 @@ static void dsound_log_hresult (HRESULT hr)
         break;
 #endif
     default:
-        AUD_log (AUDIO_CAP, "Reason: Unknown (HRESULT %#lx)\n", hr);
+        AUD_log (AUDIO_CAP, "Reason: Unknown (HRESULT 0x%lx)\n", hr);
         return;
     }
 
-- 
2.29.2

[PULL 29/30] audio: Suspect code indent for conditional statements

[Gerd Hoffmann]

From: Zhang Han <zhanghan64@huawei.com>

Fix code indent.

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-7-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/paaudio.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index 1e6f4448ce37..c97b22e970d8 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -480,7 +480,7 @@ static pa_stream *qpa_simple_new (
     }
 
     if (r < 0) {
-      goto fail;
+        goto fail;
     }
 
     pa_threaded_mainloop_unlock(c->mainloop);
-- 
2.29.2

[PULL 30/30] audio: space prohibited between function name and parenthesis'('

[Gerd Hoffmann]

From: Zhang Han <zhanghan64@huawei.com>

Delete spaces between function name and open parenthesis'('

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-8-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio_template.h | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

diff --git a/audio/audio_template.h b/audio/audio_template.h
index f82593bb8db2..c6714946aaed 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -203,13 +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;
     }
 }
-- 
2.29.2

Re: [PULL 00/30] Audio 20210115 patches

[no-reply]

Patchew URL: https://patchew.org/QEMU/20210115132146.1443592-1-kraxel@redhat.com/



Hi,

This series seems to have some coding style problems. See output below for
more information:

Type: series
Message-id: 20210115132146.1443592-1-kraxel@redhat.com
Subject: [PULL 00/30] Audio 20210115 patches

=== TEST SCRIPT BEGIN ===
#!/bin/bash
git rev-parse base > /dev/null || exit 0
git config --local diff.renamelimit 0
git config --local diff.renames True
git config --local diff.algorithm histogram
./scripts/checkpatch.pl --mailback base..
=== TEST SCRIPT END ===

Updating 3c8cf5a9c21ff8782164d1def7f44bd888713384
From https://github.com/patchew-project/qemu
 * [new tag]         patchew/20210115132146.1443592-1-kraxel@redhat.com -> patchew/20210115132146.1443592-1-kraxel@redhat.com
 - [tag update]      patchew/cover.1610638428.git.jag.raman@oracle.com -> patchew/cover.1610638428.git.jag.raman@oracle.com
Switched to a new branch 'test'
1494b2f audio: space prohibited between function name and parenthesis'('
f6e4a58 audio: Suspect code indent for conditional statements
53a6160 audio: Don't use '%#' in format strings
fad4fe1 audio: Fix lines over 90 characters
f66b15a audio: foo* bar" should be "foo *bar".
a90865a audio: Add spaces around operator/delete redundant spaces
01dd516 audio: Add braces for statements/fix braces' position
7c87251 dsoundaudio: fix log message
4df2f7a dsoundaudio: enable f32 audio sample format
0ab413d dsoundaudio: rename dsound_open()
62c5cae dsoundaudio: replace GetForegroundWindow()
5c6d84c paaudio: send recorded data in smaller chunks
4799266 paaudio: limit minreq to 75% of audio timer_rate
7ad5624 paaudio: comment bugs in functions qpa_init_*
fc82ae0 paaudio: remove unneeded code
3b4756e paaudio: wait until the playback stream is ready
9efa6c0 paaudio: wait for PA_STREAM_READY in qpa_write()
8408153 paaudio: avoid to clip samples multiple times
b12f285 audio: remove remaining unused plive code
df4b9c7 sdlaudio: enable (in|out).mixing-engine=off
0c9ff0d audio: break generic buffer dependency on mixing-engine
c25b215 sdlaudio: add recording functions
34d824d audio: split pcm_ops function get_buffer_in
278b4db sdlaudio: replace legacy functions with modern ones
b4c73f3 sdlaudio: fill remaining sample buffer with silence
d52cce6 sdlaudio: always clear the sample buffer
9acac2a sdlaudio: don't start playback in init routine
402f1bb sdlaudio: add -audiodev sdl,out.buffer-count option
d3a03cb audio: fix bit-rotted code
eb9b30a sdlaudio: remove leftover SDL1.2 code

=== OUTPUT BEGIN ===
1/30 Checking commit eb9b30a750bc (sdlaudio: remove leftover SDL1.2 code)
2/30 Checking commit d3a03cb7f82b (audio: fix bit-rotted code)
3/30 Checking commit 402f1bb02a33 (sdlaudio: add -audiodev sdl,out.buffer-count option)
4/30 Checking commit 9acac2ac79a3 (sdlaudio: don't start playback in init routine)
5/30 Checking commit d52cce6a394a (sdlaudio: always clear the sample buffer)
6/30 Checking commit b4c73f3196f8 (sdlaudio: fill remaining sample buffer with silence)
7/30 Checking commit 278b4db01c96 (sdlaudio: replace legacy functions with modern ones)
ERROR: spaces required around that '*' (ctx:WxV)
#137: FILE: audio/sdlaudio.c:247:
+        glue(SDLVoice, dir) *sdl = (glue(SDLVoice, dir) *)hw;  \
                             ^

ERROR: spaces required around that '*' (ctx:WxB)
#137: FILE: audio/sdlaudio.c:247:
+        glue(SDLVoice, dir) *sdl = (glue(SDLVoice, dir) *)hw;  \
                                                         ^

total: 2 errors, 0 warnings, 222 lines checked

Patch 7/30 has style problems, please review.  If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.

8/30 Checking commit 34d824d723c9 (audio: split pcm_ops function get_buffer_in)
9/30 Checking commit c25b21570d2d (sdlaudio: add recording functions)
ERROR: spaces required around that '*' (ctx:WxV)
#89: FILE: audio/sdlaudio.c:306:
+        glue(SDLVoice, dir) *sdl = (glue(SDLVoice, dir) *)hw;  \
                             ^

ERROR: spaces required around that '*' (ctx:WxB)
#89: FILE: audio/sdlaudio.c:306:
+        glue(SDLVoice, dir) *sdl = (glue(SDLVoice, dir) *)hw;  \
                                                         ^

total: 2 errors, 0 warnings, 185 lines checked

Patch 9/30 has style problems, please review.  If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.

10/30 Checking commit 0c9ff0d05558 (audio: break generic buffer dependency on mixing-engine)
11/30 Checking commit df4b9c7bc97a (sdlaudio: enable (in|out).mixing-engine=off)
12/30 Checking commit b12f2859ee02 (audio: remove remaining unused plive code)
13/30 Checking commit 84081537b4bf (paaudio: avoid to clip samples multiple times)
14/30 Checking commit 9efa6c036626 (paaudio: wait for PA_STREAM_READY in qpa_write())
15/30 Checking commit 3b4756e48f9d (paaudio: wait until the playback stream is ready)
16/30 Checking commit fc82ae0f7214 (paaudio: remove unneeded code)
17/30 Checking commit 7ad56244f8a0 (paaudio: comment bugs in functions qpa_init_*)
18/30 Checking commit 4799266d2ce7 (paaudio: limit minreq to 75% of audio timer_rate)
19/30 Checking commit 5c6d84ce19a3 (paaudio: send recorded data in smaller chunks)
20/30 Checking commit 62c5cae8b7ab (dsoundaudio: replace GetForegroundWindow())
21/30 Checking commit 0ab413dc7a17 (dsoundaudio: rename dsound_open())
22/30 Checking commit 4df2f7a26559 (dsoundaudio: enable f32 audio sample format)
23/30 Checking commit 7c87251cd059 (dsoundaudio: fix log message)
24/30 Checking commit 01dd5165938e (audio: Add braces for statements/fix braces' position)
25/30 Checking commit a90865a46c6c (audio: Add spaces around operator/delete redundant spaces)
ERROR: space prohibited between function name and open parenthesis '('
#27: FILE: audio/audio_template.h:211:
+        glue (audio_pcm_hw_free_resources_ , TYPE) (hw);

total: 1 errors, 0 warnings, 32 lines checked

Patch 25/30 has style problems, please review.  If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.

26/30 Checking commit f66b15a8e97a (audio: foo* bar" should be "foo *bar".)
27/30 Checking commit fad4fe1d4ed3 (audio: Fix lines over 90 characters)
28/30 Checking commit 53a6160b07e9 (audio: Don't use '%#' in format strings)
29/30 Checking commit f6e4a582154a (audio: Suspect code indent for conditional statements)
30/30 Checking commit 1494b2f2386b (audio: space prohibited between function name and parenthesis'(')
=== OUTPUT END ===

Test command exited with code: 1


The full log is available at
http://patchew.org/logs/20210115132146.1443592-1-kraxel@redhat.com/testing.checkpatch/?type=message.
---
Email generated automatically by Patchew [https://patchew.org/].
Please send your feedback to patchew-devel@redhat.com

Re: [PULL 00/30] Audio 20210115 patches

[Peter Maydell]

On Fri, 15 Jan 2021 at 13:50, Gerd Hoffmann <kraxel@redhat.com> wrote:
>
> The following changes since commit 45240eed4f064576d589ea60ebadf3c11d7ab891:
>
>   Merge remote-tracking branch 'remotes/armbru/tags/pull-yank-2021-01-13' int=
> o staging (2021-01-13 14:19:24 +0000)
>
> are available in the Git repository at:
>
>   git://git.kraxel.org/qemu tags/audio-20210115-pull-request
>
> for you to fetch changes up to 8abf3feb4d464abadd5133d8810c8a3232cbbe6e:
>
>   audio: space prohibited between function name and parenthesis'(' (2021-01-1=
> 5 11:49:26 +0100)
>
> ----------------------------------------------------------------
> audio: improvements for sdl, pulse, fsound.
> audio: cleanups & codestyle fixes.
>
> ----------------------------------------------------------------
>

Applied, thanks.

Please update the changelog at https://wiki.qemu.org/ChangeLog/6.0
for any user-visible changes.

-- PMM