[Qemu-devel] [PULL 0/3] audio patch queue

[Qemu-devel] [PULL 0/3] audio patch queue

[Gerd Hoffmann]

  Hi,

Three little audio patches, adding replay support
and fixing audio with SDL2.

please pull,
  Gerd

The following changes since commit 3b1d8169844fafee184366b0e0d7080534758b4d:

  tests-aio-multithread: use atomic_read properly (2017-02-27 12:54:08 +0000)

are available in the git repository at:

  git://git.kraxel.org/qemu tags/pull-audio-20170227-1

for you to fetch changes up to 516452fe6accef3f94b4bcf060b2518038f50c08:

  audio/sdlaudio: Allow audio playback with SDL2 (2017-02-27 14:37:59 +0100)

----------------------------------------------------------------
audio: replay support, sdl2 fix.

----------------------------------------------------------------
Pavel Dovgalyuk (2):
      replay: add record/replay for audio passthrough
      audio: make audio poll timer deterministic

Thomas Huth (1):
      audio/sdlaudio: Allow audio playback with SDL2

 audio/audio.c            | 11 +++++--
 audio/audio.h            |  5 +++
 audio/mixeng.c           | 31 +++++++++++++++++++
 audio/sdlaudio.c         | 48 +++++++++++++++++++++++++++++
 docs/replay.txt          |  7 +++++
 include/sysemu/replay.h  |  7 +++++
 replay/Makefile.objs     |  1 +
 replay/replay-audio.c    | 79 ++++++++++++++++++++++++++++++++++++++++++++++++
 replay/replay-internal.h |  4 +++
 9 files changed, 190 insertions(+), 3 deletions(-)
 create mode 100644 replay/replay-audio.c

[Qemu-devel] [PULL 1/3] replay: add record/replay for audio passthrough

[Gerd Hoffmann]

From: Pavel Dovgalyuk <pavel.dovgaluk@ispras.ru>

This patch adds recording and replaying audio data. Is saves synchronization
information for audio out and inputs from the microphone.

v2: removed unneeded whitespace change

Signed-off-by: Pavel Dovgalyuk <pavel.dovgaluk@ispras.ru>
Message-id: 20170202055054.4848.94901.stgit@PASHA-ISP.lan02.inno
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio.c            |  9 ++++--
 audio/audio.h            |  5 +++
 audio/mixeng.c           | 31 +++++++++++++++++++
 docs/replay.txt          |  7 +++++
 include/sysemu/replay.h  |  7 +++++
 replay/Makefile.objs     |  1 +
 replay/replay-audio.c    | 79 ++++++++++++++++++++++++++++++++++++++++++++++++
 replay/replay-internal.h |  4 +++
 8 files changed, 141 insertions(+), 2 deletions(-)
 create mode 100644 replay/replay-audio.c

diff --git a/audio/audio.c b/audio/audio.c
index c845a44..21f7b0b 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -28,6 +28,7 @@
 #include "qemu/timer.h"
 #include "sysemu/sysemu.h"
 #include "qemu/cutils.h"
+#include "sysemu/replay.h"
 
 #define AUDIO_CAP "audio"
 #include "audio_int.h"
@@ -1387,6 +1388,7 @@ static void audio_run_out (AudioState *s)
 
         prev_rpos = hw->rpos;
         played = hw->pcm_ops->run_out (hw, live);
+        replay_audio_out(&played);
         if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
             dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
                    hw->rpos, hw->samples, played);
@@ -1450,9 +1452,12 @@ static void audio_run_in (AudioState *s)
 
     while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) {
         SWVoiceIn *sw;
-        int captured, min;
+        int captured = 0, min;
 
-        captured = hw->pcm_ops->run_in (hw);
+        if (replay_mode != REPLAY_MODE_PLAY) {
+            captured = hw->pcm_ops->run_in(hw);
+        }
+        replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples);
 
         min = audio_pcm_hw_find_min_in (hw);
         hw->total_samples_captured += captured - min;
diff --git a/audio/audio.h b/audio/audio.h
index c3c5198..f4339a1 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -166,4 +166,9 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
 bool audio_is_cleaning_up(void);
 void audio_cleanup(void);
 
+void audio_sample_to_uint64(void *samples, int pos,
+                            uint64_t *left, uint64_t *right);
+void audio_sample_from_uint64(void *samples, int pos,
+                            uint64_t left, uint64_t right);
+
 #endif /* QEMU_AUDIO_H */
diff --git a/audio/mixeng.c b/audio/mixeng.c
index 66c0328..c23508e 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -267,6 +267,37 @@ f_sample *mixeng_clip[2][2][2][3] = {
     }
 };
 
+
+void audio_sample_to_uint64(void *samples, int pos,
+                            uint64_t *left, uint64_t *right)
+{
+    struct st_sample *sample = samples;
+    sample += pos;
+#ifdef FLOAT_MIXENG
+    error_report(
+        "Coreaudio and floating point samples are not supported by replay yet");
+    abort();
+#else
+    *left = sample->l;
+    *right = sample->r;
+#endif
+}
+
+void audio_sample_from_uint64(void *samples, int pos,
+                            uint64_t left, uint64_t right)
+{
+    struct st_sample *sample = samples;
+    sample += pos;
+#ifdef FLOAT_MIXENG
+    error_report(
+        "Coreaudio and floating point samples are not supported by replay yet");
+    abort();
+#else
+    sample->l = left;
+    sample->r = right;
+#endif
+}
+
 /*
  * August 21, 1998
  * Copyright 1998 Fabrice Bellard.
diff --git a/docs/replay.txt b/docs/replay.txt
index 03e1931..486c1e0 100644
--- a/docs/replay.txt
+++ b/docs/replay.txt
@@ -225,3 +225,10 @@ recording the virtual machine this filter puts all packets coming from
 the outer world into the log. In replay mode packets from the log are
 injected into the network device. All interactions with network backend
 in replay mode are disabled.
+
+Audio devices
+-------------
+
+Audio data is recorded and replay automatically. The command line for recording
+and replaying must contain identical specifications of audio hardware, e.g.:
+ -soundhw ac97
diff --git a/include/sysemu/replay.h b/include/sysemu/replay.h
index 7aad20b..f1c0712 100644
--- a/include/sysemu/replay.h
+++ b/include/sysemu/replay.h
@@ -152,6 +152,13 @@ void replay_unregister_net(ReplayNetState *rns);
 void replay_net_packet_event(ReplayNetState *rns, unsigned flags,
                              const struct iovec *iov, int iovcnt);
 
+/* Audio */
+
+/*! Saves/restores number of played samples of audio out operation. */
+void replay_audio_out(int *played);
+/*! Saves/restores recorded samples of audio in operation. */
+void replay_audio_in(int *recorded, void *samples, int *wpos, int size);
+
 /* VM state operations */
 
 /*! Called at the start of execution.
diff --git a/replay/Makefile.objs b/replay/Makefile.objs
index b2afd40..cee6539 100644
--- a/replay/Makefile.objs
+++ b/replay/Makefile.objs
@@ -6,3 +6,4 @@ common-obj-y += replay-input.o
 common-obj-y += replay-char.o
 common-obj-y += replay-snapshot.o
 common-obj-y += replay-net.o
+common-obj-y += replay-audio.o
\ No newline at end of file
diff --git a/replay/replay-audio.c b/replay/replay-audio.c
new file mode 100644
index 0000000..3d83743
--- /dev/null
+++ b/replay/replay-audio.c
@@ -0,0 +1,79 @@
+/*
+ * replay-audio.c
+ *
+ * Copyright (c) 2010-2017 Institute for System Programming
+ *                         of the Russian Academy of Sciences.
+ *
+ * This work is licensed under the terms of the GNU GPL, version 2 or later.
+ * See the COPYING file in the top-level directory.
+ *
+ */
+
+#include "qemu/osdep.h"
+#include "qemu/error-report.h"
+#include "sysemu/replay.h"
+#include "replay-internal.h"
+#include "sysemu/sysemu.h"
+#include "audio/audio.h"
+
+void replay_audio_out(int *played)
+{
+    if (replay_mode == REPLAY_MODE_RECORD) {
+        replay_save_instructions();
+        replay_mutex_lock();
+        replay_put_event(EVENT_AUDIO_OUT);
+        replay_put_dword(*played);
+        replay_mutex_unlock();
+    } else if (replay_mode == REPLAY_MODE_PLAY) {
+        replay_account_executed_instructions();
+        replay_mutex_lock();
+        if (replay_next_event_is(EVENT_AUDIO_OUT)) {
+            *played = replay_get_dword();
+            replay_finish_event();
+            replay_mutex_unlock();
+        } else {
+            replay_mutex_unlock();
+            error_report("Missing audio out event in the replay log");
+            abort();
+        }
+    }
+}
+
+void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
+{
+    int pos;
+    uint64_t left, right;
+    if (replay_mode == REPLAY_MODE_RECORD) {
+        replay_save_instructions();
+        replay_mutex_lock();
+        replay_put_event(EVENT_AUDIO_IN);
+        replay_put_dword(*recorded);
+        replay_put_dword(*wpos);
+        for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
+             ; pos = (pos + 1) % size) {
+            audio_sample_to_uint64(samples, pos, &left, &right);
+            replay_put_qword(left);
+            replay_put_qword(right);
+        }
+        replay_mutex_unlock();
+    } else if (replay_mode == REPLAY_MODE_PLAY) {
+        replay_account_executed_instructions();
+        replay_mutex_lock();
+        if (replay_next_event_is(EVENT_AUDIO_IN)) {
+            *recorded = replay_get_dword();
+            *wpos = replay_get_dword();
+            for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
+                 ; pos = (pos + 1) % size) {
+                left = replay_get_qword();
+                right = replay_get_qword();
+                audio_sample_from_uint64(samples, pos, left, right);
+            }
+            replay_finish_event();
+            replay_mutex_unlock();
+        } else {
+            replay_mutex_unlock();
+            error_report("Missing audio in event in the replay log");
+            abort();
+        }
+    }
+}
diff --git a/replay/replay-internal.h b/replay/replay-internal.h
index c26d079..ed66ed8 100644
--- a/replay/replay-internal.h
+++ b/replay/replay-internal.h
@@ -29,6 +29,10 @@ enum ReplayEvents {
     /* for character device read all event */
     EVENT_CHAR_READ_ALL,
     EVENT_CHAR_READ_ALL_ERROR,
+    /* for audio out event */
+    EVENT_AUDIO_OUT,
+    /* for audio in event */
+    EVENT_AUDIO_IN,
     /* for clock read/writes */
     /* some of greater codes are reserved for clocks */
     EVENT_CLOCK,
-- 
1.8.3.1

[Qemu-devel] [PULL 2/3] audio: make audio poll timer deterministic

[Gerd Hoffmann]

From: Pavel Dovgalyuk <pavel.dovgaluk@ispras.ru>

This patch changes resetting strategy of the audio polling timer.
It does not change expiration time if the timer is already set.
This patch is needed to make this timer deterministic and to use execution
record/replay for audio devices.

audio_reset_timer is used in the function audio_vm_change_state_handler.
Therefore every time VM is stopped or restarted the timer will be reset
to new timeout. Virtual clock does not proceed while VM is stopped.
Therefore there is no need in resetting the timeout when VM restarts.

v2: updated commit message
v3: now using timer_mod_anticipate function (as suggested by Yurii Zubrytskyi)

Signed-off-by: Pavel Dovgalyuk <pavel.dovgaluk@ispras.ru>
Message-id: 20170214071510.6112.76764.stgit@PASHA-ISP
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/audio/audio.c b/audio/audio.c
index 21f7b0b..c8898d8 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1113,7 +1113,7 @@ static int audio_is_timer_needed (void)
 static void audio_reset_timer (AudioState *s)
 {
     if (audio_is_timer_needed ()) {
-        timer_mod (s->ts,
+        timer_mod_anticipate_ns(s->ts,
             qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks);
     }
     else {
-- 
1.8.3.1

[Qemu-devel] [PULL 3/3] audio/sdlaudio: Allow audio playback with SDL2

[Gerd Hoffmann]

From: Thomas Huth <thuth@redhat.com>

When compiling with SDL2, the semaphore trick used in sdlaudio.c
does not work - QEMU locks up completely in this case. To avoid
the hang and get at least some audio playback up and running (it's
a little bit crackling, but better than nothing), we can use the
SDL locking functions SDL_LockAudio() and SDL_UnlockAudio() to sync
with the sound playback thread instead.

Signed-off-by: Thomas Huth <thuth@redhat.com>
Message-id: 1485852398-2327-1-git-send-email-thuth@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/sdlaudio.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 48 insertions(+)

diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index db69fe1..e8d91d2 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -38,10 +38,14 @@
 #define AUDIO_CAP "sdl"
 #include "audio_int.h"
 
+#define USE_SEMAPHORE (SDL_MAJOR_VERSION < 2)
+
 typedef struct SDLVoiceOut {
     HWVoiceOut hw;
     int live;
+#if USE_SEMAPHORE
     int rpos;
+#endif
     int decr;
 } SDLVoiceOut;
 
@@ -53,8 +57,10 @@ static struct {
 
 static struct SDLAudioState {
     int exit;
+#if USE_SEMAPHORE
     SDL_mutex *mutex;
     SDL_sem *sem;
+#endif
     int initialized;
     bool driver_created;
 } glob_sdl;
@@ -73,31 +79,45 @@ static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
 
 static int sdl_lock (SDLAudioState *s, const char *forfn)
 {
+#if USE_SEMAPHORE
     if (SDL_LockMutex (s->mutex)) {
         sdl_logerr ("SDL_LockMutex for %s failed\n", forfn);
         return -1;
     }
+#else
+    SDL_LockAudio();
+#endif
+
     return 0;
 }
 
 static int sdl_unlock (SDLAudioState *s, const char *forfn)
 {
+#if USE_SEMAPHORE
     if (SDL_UnlockMutex (s->mutex)) {
         sdl_logerr ("SDL_UnlockMutex for %s failed\n", forfn);
         return -1;
     }
+#else
+    SDL_UnlockAudio();
+#endif
+
     return 0;
 }
 
 static int sdl_post (SDLAudioState *s, const char *forfn)
 {
+#if USE_SEMAPHORE
     if (SDL_SemPost (s->sem)) {
         sdl_logerr ("SDL_SemPost for %s failed\n", forfn);
         return -1;
     }
+#endif
+
     return 0;
 }
 
+#if USE_SEMAPHORE
 static int sdl_wait (SDLAudioState *s, const char *forfn)
 {
     if (SDL_SemWait (s->sem)) {
@@ -106,6 +126,7 @@ static int sdl_wait (SDLAudioState *s, const char *forfn)
     }
     return 0;
 }
+#endif
 
 static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
 {
@@ -246,6 +267,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
         int to_mix, decr;
 
         /* dolog ("in callback samples=%d\n", samples); */
+#if USE_SEMAPHORE
         sdl_wait (s, "sdl_callback");
         if (s->exit) {
             return;
@@ -264,6 +286,11 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
         if (!sdl->live) {
             goto again;
         }
+#else
+        if (s->exit || !sdl->live) {
+            break;
+        }
+#endif
 
         /* dolog ("in callback live=%d\n", live); */
         to_mix = audio_MIN (samples, sdl->live);
@@ -274,7 +301,11 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
 
             /* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
             hw->clip (buf, src, chunk);
+#if USE_SEMAPHORE
             sdl->rpos = (sdl->rpos + chunk) % hw->samples;
+#else
+            hw->rpos = (hw->rpos + chunk) % hw->samples;
+#endif
             to_mix -= chunk;
             buf += chunk << hw->info.shift;
         }
@@ -282,12 +313,21 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
         sdl->live -= decr;
         sdl->decr += decr;
 
+#if USE_SEMAPHORE
     again:
         if (sdl_unlock (s, "sdl_callback")) {
             return;
         }
+#endif
     }
     /* dolog ("done len=%d\n", len); */
+
+#if (SDL_MAJOR_VERSION >= 2)
+    /* SDL2 does not clear the remaining buffer for us, so do it on our own */
+    if (samples) {
+        memset(buf, 0, samples << hw->info.shift);
+    }
+#endif
 }
 
 static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
@@ -315,8 +355,12 @@ static int sdl_run_out (HWVoiceOut *hw, int live)
     decr = audio_MIN (sdl->decr, live);
     sdl->decr -= decr;
 
+#if USE_SEMAPHORE
     sdl->live = live - decr;
     hw->rpos = sdl->rpos;
+#else
+    sdl->live = live;
+#endif
 
     if (sdl->live > 0) {
         sdl_unlock_and_post (s, "sdl_run_out");
@@ -405,6 +449,7 @@ static void *sdl_audio_init (void)
         return NULL;
     }
 
+#if USE_SEMAPHORE
     s->mutex = SDL_CreateMutex ();
     if (!s->mutex) {
         sdl_logerr ("Failed to create SDL mutex\n");
@@ -419,6 +464,7 @@ static void *sdl_audio_init (void)
         SDL_QuitSubSystem (SDL_INIT_AUDIO);
         return NULL;
     }
+#endif
 
     s->driver_created = true;
     return s;
@@ -428,8 +474,10 @@ static void sdl_audio_fini (void *opaque)
 {
     SDLAudioState *s = opaque;
     sdl_close (s);
+#if USE_SEMAPHORE
     SDL_DestroySemaphore (s->sem);
     SDL_DestroyMutex (s->mutex);
+#endif
     SDL_QuitSubSystem (SDL_INIT_AUDIO);
     s->driver_created = false;
 }
-- 
1.8.3.1

Re: [Qemu-devel] [PULL 0/3] audio patch queue

[Peter Maydell]

On 27 February 2017 at 15:11, Gerd Hoffmann <kraxel@redhat.com> wrote:
>   Hi,
>
> Three little audio patches, adding replay support
> and fixing audio with SDL2.
>
> please pull,
>   Gerd
>
> The following changes since commit 3b1d8169844fafee184366b0e0d7080534758b4d:
>
>   tests-aio-multithread: use atomic_read properly (2017-02-27 12:54:08 +0000)
>
> are available in the git repository at:
>
>   git://git.kraxel.org/qemu tags/pull-audio-20170227-1
>
> for you to fetch changes up to 516452fe6accef3f94b4bcf060b2518038f50c08:
>
>   audio/sdlaudio: Allow audio playback with SDL2 (2017-02-27 14:37:59 +0100)
>
> ----------------------------------------------------------------
> audio: replay support, sdl2 fix.

I'm afraid this fails to build on OSX:

/Users/pm215/src/qemu-for-merges/audio/mixeng.c:277:5: error: implicit
declaration of function 'error_report' is invalid in C99
[-Werror,-Wimplicit-function-declaration]
    error_report(
    ^

Forgotten include of "qapi/error.h" (it's probably being pulled in
implicitly by some other header on other platforms).

thanks
-- PMM

[Qemu-devel] [PULL 2/3] audio: make audio poll timer deterministic

[Gerd Hoffmann]

From: Pavel Dovgalyuk <pavel.dovgaluk@ispras.ru>

This patch changes resetting strategy of the audio polling timer.
It does not change expiration time if the timer is already set.
This patch is needed to make this timer deterministic and to use execution
record/replay for audio devices.

audio_reset_timer is used in the function audio_vm_change_state_handler.
Therefore every time VM is stopped or restarted the timer will be reset
to new timeout. Virtual clock does not proceed while VM is stopped.
Therefore there is no need in resetting the timeout when VM restarts.

v2: updated commit message
v3: now using timer_mod_anticipate function (as suggested by Yurii Zubrytskyi)

Signed-off-by: Pavel Dovgalyuk <pavel.dovgaluk@ispras.ru>
Message-id: 20170214071510.6112.76764.stgit@PASHA-ISP
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/audio/audio.c b/audio/audio.c
index 21f7b0b..c8898d8 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1113,7 +1113,7 @@ static int audio_is_timer_needed (void)
 static void audio_reset_timer (AudioState *s)
 {
     if (audio_is_timer_needed ()) {
-        timer_mod (s->ts,
+        timer_mod_anticipate_ns(s->ts,
             qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks);
     }
     else {
-- 
1.8.3.1