[PULL 0/8] Audio 20200526 patches

[PULL 0/8] Audio 20200526 patches

[Gerd Hoffmann]

The following changes since commit fea8f3ed739536fca027cf56af7f5576f37ef9cd:

  Merge remote-tracking branch 'remotes/philmd-gitlab/tags/pflash-next-20200522' into staging (2020-05-22 18:54:47 +0100)

are available in the Git repository at:

  git://git.kraxel.org/qemu tags/audio-20200526-pull-request

for you to fetch changes up to b3b8a1fea6ed5004bbad2f70833caee70402bf02:

  hw/mips/mips_fulong2e: Remove unused 'audio/audio.h' include (2020-05-26 08:46:14 +0200)

----------------------------------------------------------------
audio: add JACK client audiodev.
audio: bugfixes and cleanups.

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

Bruce Rogers (1):
  audio: fix wavcapture segfault

Geoffrey McRae (1):
  audio/jack: add JACK client audiodev

Philippe Mathieu-Daudé (4):
  hw/audio/gus: Use AUDIO_HOST_ENDIANNESS definition from
    'audio/audio.h'
  audio: Let audio_sample_to_uint64() use const samples argument
  audio: Let capture_callback handler use const buffer argument
  hw/mips/mips_fulong2e: Remove unused 'audio/audio.h' include

Prasad J Pandit (1):
  es1370: check total frame count against current frame

Volker Rümelin (1):
  audio/mixeng: fix clang 10+ warning

 configure               |  17 +
 audio/audio.h           |   4 +-
 audio/audio_template.h  |   2 +
 audio/audio.c           |   5 +-
 audio/jackaudio.c       | 667 ++++++++++++++++++++++++++++++++++++++++
 audio/mixeng.c          |   9 +-
 audio/wavcapture.c      |   2 +-
 hw/audio/es1370.c       |   7 +-
 hw/audio/gus.c          |   8 +-
 hw/mips/mips_fulong2e.c |   1 -
 ui/vnc.c                |   2 +-
 audio/Makefile.objs     |   5 +
 qapi/audio.json         |  56 +++-
 13 files changed, 763 insertions(+), 22 deletions(-)
 create mode 100644 audio/jackaudio.c

-- 
2.18.4

[PULL 1/8] es1370: check total frame count against current frame

[Gerd Hoffmann]

From: Prasad J Pandit <pjp@fedoraproject.org>

A guest user may set channel frame count via es1370_write()
such that, in es1370_transfer_audio(), total frame count
'size' is lesser than the number of frames that are processed
'cnt'.

    int cnt = d->frame_cnt >> 16;
    int size = d->frame_cnt & 0xffff;

if (size < cnt), it results in incorrect calculations leading
to OOB access issue(s). Add check to avoid it.

Reported-by: Ren Ding <rding@gatech.edu>
Reported-by: Hanqing Zhao <hanqing@gatech.edu>
Signed-off-by: Prasad J Pandit <pjp@fedoraproject.org>
Message-id: 20200514200608.1744203-1-ppandit@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 hw/audio/es1370.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 89c4dabcd44f..5f8a83ff5624 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -643,6 +643,9 @@ static void es1370_transfer_audio (ES1370State *s, struct chan *d, int loop_sel,
     int csc_bytes = (csc + 1) << d->shift;
     int cnt = d->frame_cnt >> 16;
     int size = d->frame_cnt & 0xffff;
+    if (size < cnt) {
+        return;
+    }
     int left = ((size - cnt + 1) << 2) + d->leftover;
     int transferred = 0;
     int temp = MIN (max, MIN (left, csc_bytes));
@@ -651,7 +654,7 @@ static void es1370_transfer_audio (ES1370State *s, struct chan *d, int loop_sel,
     addr += (cnt << 2) + d->leftover;
 
     if (index == ADC_CHANNEL) {
-        while (temp) {
+        while (temp > 0) {
             int acquired, to_copy;
 
             to_copy = MIN ((size_t) temp, sizeof (tmpbuf));
@@ -669,7 +672,7 @@ static void es1370_transfer_audio (ES1370State *s, struct chan *d, int loop_sel,
     else {
         SWVoiceOut *voice = s->dac_voice[index];
 
-        while (temp) {
+        while (temp > 0) {
             int copied, to_copy;
 
             to_copy = MIN ((size_t) temp, sizeof (tmpbuf));
-- 
2.18.4

[PULL 2/8] hw/audio/gus: Use AUDIO_HOST_ENDIANNESS definition from 'audio/audio.h'

[Gerd Hoffmann]

From: Philippe Mathieu-Daudé <f4bug@amsat.org>

Use the generic AUDIO_HOST_ENDIANNESS definition instead
of a custom one.

Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-id: 20200505100750.27332-1-f4bug@amsat.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 hw/audio/gus.c | 8 +-------
 1 file changed, 1 insertion(+), 7 deletions(-)

diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index eb4a803fb53b..c8df2bde6b32 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -41,12 +41,6 @@
 #define ldebug(...)
 #endif
 
-#ifdef HOST_WORDS_BIGENDIAN
-#define GUS_ENDIANNESS 1
-#else
-#define GUS_ENDIANNESS 0
-#endif
-
 #define TYPE_GUS "gus"
 #define GUS(obj) OBJECT_CHECK (GUSState, (obj), TYPE_GUS)
 
@@ -256,7 +250,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp)
     as.freq = s->freq;
     as.nchannels = 2;
     as.fmt = AUDIO_FORMAT_S16;
-    as.endianness = GUS_ENDIANNESS;
+    as.endianness = AUDIO_HOST_ENDIANNESS;
 
     s->voice = AUD_open_out (
         &s->card,
-- 
2.18.4

[PULL 3/8] audio/jack: add JACK client audiodev

[Gerd Hoffmann]

From: Geoffrey McRae <geoff@hostfission.com>

This commit adds a new audiodev backend to allow QEMU to use JACK as
both an audio sink and source.

Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-Id: <20200512101603.E3DB73A038E@moya.office.hostfission.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 configure              |  17 ++
 audio/audio_template.h |   2 +
 audio/audio.c          |   1 +
 audio/jackaudio.c      | 667 +++++++++++++++++++++++++++++++++++++++++
 audio/Makefile.objs    |   5 +
 qapi/audio.json        |  56 +++-
 6 files changed, 746 insertions(+), 2 deletions(-)
 create mode 100644 audio/jackaudio.c

diff --git a/configure b/configure
index 2fc05c4465cb..b969dee675bb 100755
--- a/configure
+++ b/configure
@@ -3629,6 +3629,22 @@ for drv in $audio_drv_list; do
       oss_libs="$oss_lib"
     ;;
 
+    jack | try-jack)
+    if $pkg_config jack --exists; then
+        jack_libs=$($pkg_config jack --libs)
+        if test "$drv" = "try-jack"; then
+            audio_drv_list=$(echo "$audio_drv_list" | sed -e 's/try-jack/jack/')
+        fi
+    else
+        if test "$drv" = "try-jack"; then
+            audio_drv_list=$(echo "$audio_drv_list" | sed -e 's/try-jack//')
+        else
+            error_exit "$drv check failed" \
+                "Make sure to have the $drv libs and headers installed."
+        fi
+    fi
+    ;;
+
     *)
     echo "$audio_possible_drivers" | grep -q "\<$drv\>" || {
         error_exit "Unknown driver '$drv' selected" \
@@ -6904,6 +6920,7 @@ echo "PULSE_LIBS=$pulse_libs" >> $config_host_mak
 echo "COREAUDIO_LIBS=$coreaudio_libs" >> $config_host_mak
 echo "DSOUND_LIBS=$dsound_libs" >> $config_host_mak
 echo "OSS_LIBS=$oss_libs" >> $config_host_mak
+echo "JACK_LIBS=$jack_libs" >> $config_host_mak
 if test "$audio_win_int" = "yes" ; then
   echo "CONFIG_AUDIO_WIN_INT=y" >> $config_host_mak
 fi
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 7013d3041f91..8dd48ce14e9d 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -330,6 +330,8 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev)
             dev->u.coreaudio.TYPE);
     case AUDIODEV_DRIVER_DSOUND:
         return dev->u.dsound.TYPE;
+    case AUDIODEV_DRIVER_JACK:
+        return qapi_AudiodevJackPerDirectionOptions_base(dev->u.jack.TYPE);
     case AUDIODEV_DRIVER_OSS:
         return qapi_AudiodevOssPerDirectionOptions_base(dev->u.oss.TYPE);
     case AUDIODEV_DRIVER_PA:
diff --git a/audio/audio.c b/audio/audio.c
index 7a9e6803558b..95d9fb16caa5 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1969,6 +1969,7 @@ void audio_create_pdos(Audiodev *dev)
         CASE(ALSA, alsa, Alsa);
         CASE(COREAUDIO, coreaudio, Coreaudio);
         CASE(DSOUND, dsound, );
+        CASE(JACK, jack, Jack);
         CASE(OSS, oss, Oss);
         CASE(PA, pa, Pa);
         CASE(SDL, sdl, );
diff --git a/audio/jackaudio.c b/audio/jackaudio.c
new file mode 100644
index 000000000000..722ddb1dfe43
--- /dev/null
+++ b/audio/jackaudio.c
@@ -0,0 +1,667 @@
+/*
+ * QEMU JACK Audio Connection Kit Client
+ *
+ * Copyright (c) 2020 Geoffrey McRae (gnif)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include "qemu/osdep.h"
+#include "qemu/module.h"
+#include "qemu/atomic.h"
+#include "qemu-common.h"
+#include "audio.h"
+
+#define AUDIO_CAP "jack"
+#include "audio_int.h"
+
+#include <jack/jack.h>
+#include <jack/thread.h>
+
+struct QJack;
+
+typedef enum QJackState {
+    QJACK_STATE_DISCONNECTED,
+    QJACK_STATE_STOPPED,
+    QJACK_STATE_RUNNING,
+    QJACK_STATE_SHUTDOWN
+}
+QJackState;
+
+typedef struct QJackBuffer {
+    int          channels;
+    int          frames;
+    uint32_t     used;
+    int          rptr, wptr;
+    float      **data;
+}
+QJackBuffer;
+
+typedef struct QJackClient {
+    AudiodevJackPerDirectionOptions *opt;
+
+    bool out;
+    bool finished;
+    bool connect_ports;
+    int  packets;
+
+    QJackState      state;
+    jack_client_t  *client;
+    jack_nframes_t  freq;
+
+    struct QJack   *j;
+    int             nchannels;
+    int             buffersize;
+    jack_port_t   **port;
+    QJackBuffer     fifo;
+}
+QJackClient;
+
+typedef struct QJackOut {
+    HWVoiceOut  hw;
+    QJackClient c;
+}
+QJackOut;
+
+typedef struct QJackIn {
+    HWVoiceIn   hw;
+    QJackClient c;
+}
+QJackIn;
+
+static int qjack_client_init(QJackClient *c);
+static void qjack_client_connect_ports(QJackClient *c);
+static void qjack_client_fini(QJackClient *c);
+
+static void qjack_buffer_create(QJackBuffer *buffer, int channels, int frames)
+{
+    buffer->channels = channels;
+    buffer->frames   = frames;
+    buffer->used     = 0;
+    buffer->rptr     = 0;
+    buffer->wptr     = 0;
+    buffer->data     = g_malloc(channels * sizeof(float *));
+    for (int i = 0; i < channels; ++i) {
+        buffer->data[i] = g_malloc(frames * sizeof(float));
+    }
+}
+
+static void qjack_buffer_clear(QJackBuffer *buffer)
+{
+    assert(buffer->data);
+    atomic_store_release(&buffer->used, 0);
+    buffer->rptr = 0;
+    buffer->wptr = 0;
+}
+
+static void qjack_buffer_free(QJackBuffer *buffer)
+{
+    if (!buffer->data) {
+        return;
+    }
+
+    for (int i = 0; i < buffer->channels; ++i) {
+        g_free(buffer->data[i]);
+    }
+
+    g_free(buffer->data);
+    buffer->data = NULL;
+}
+
+/* write PCM interleaved */
+static int qjack_buffer_write(QJackBuffer *buffer, float *data, int size)
+{
+    assert(buffer->data);
+    const int samples = size / sizeof(float);
+    int frames        = samples / buffer->channels;
+    const int avail   = buffer->frames - atomic_load_acquire(&buffer->used);
+
+    if (frames > avail) {
+        frames = avail;
+    }
+
+    int copy = frames;
+    int wptr = buffer->wptr;
+
+    while (copy) {
+
+        for (int c = 0; c < buffer->channels; ++c) {
+            buffer->data[c][wptr] = *data++;
+        }
+
+        if (++wptr == buffer->frames) {
+            wptr = 0;
+        }
+
+        --copy;
+    }
+
+    buffer->wptr = wptr;
+
+    atomic_add(&buffer->used, frames);
+    return frames * buffer->channels * sizeof(float);
+};
+
+/* write PCM linear */
+static int qjack_buffer_write_l(QJackBuffer *buffer, float **dest, int frames)
+{
+    assert(buffer->data);
+    const int avail   = buffer->frames - atomic_load_acquire(&buffer->used);
+    int wptr = buffer->wptr;
+
+    if (frames > avail) {
+        frames = avail;
+    }
+
+    int right = buffer->frames - wptr;
+    if (right > frames) {
+        right = frames;
+    }
+
+    const int left = frames - right;
+    for (int c = 0; c < buffer->channels; ++c) {
+        memcpy(buffer->data[c] + wptr, dest[c]        , right * sizeof(float));
+        memcpy(buffer->data[c]       , dest[c] + right, left  * sizeof(float));
+    }
+
+    wptr += frames;
+    if (wptr >= buffer->frames) {
+        wptr -= buffer->frames;
+    }
+    buffer->wptr = wptr;
+
+    atomic_add(&buffer->used, frames);
+    return frames;
+}
+
+/* read PCM interleaved */
+static int qjack_buffer_read(QJackBuffer *buffer, float *dest, int size)
+{
+    assert(buffer->data);
+    const int samples = size / sizeof(float);
+    int frames        = samples / buffer->channels;
+    const int avail   = atomic_load_acquire(&buffer->used);
+
+    if (frames > avail) {
+        frames = avail;
+    }
+
+    int copy = frames;
+    int rptr = buffer->rptr;
+
+    while (copy) {
+
+        for (int c = 0; c < buffer->channels; ++c) {
+            *dest++ = buffer->data[c][rptr];
+        }
+
+        if (++rptr == buffer->frames) {
+            rptr = 0;
+        }
+
+        --copy;
+    }
+
+    buffer->rptr = rptr;
+
+    atomic_sub(&buffer->used, frames);
+    return frames * buffer->channels * sizeof(float);
+}
+
+/* read PCM linear */
+static int qjack_buffer_read_l(QJackBuffer *buffer, float **dest, int frames)
+{
+    assert(buffer->data);
+    int copy       = frames;
+    const int used = atomic_load_acquire(&buffer->used);
+    int rptr       = buffer->rptr;
+
+    if (copy > used) {
+        copy = used;
+    }
+
+    int right = buffer->frames - rptr;
+    if (right > copy) {
+        right = copy;
+    }
+
+    const int left = copy - right;
+    for (int c = 0; c < buffer->channels; ++c) {
+        memcpy(dest[c]        , buffer->data[c] + rptr, right * sizeof(float));
+        memcpy(dest[c] + right, buffer->data[c]       , left  * sizeof(float));
+    }
+
+    rptr += copy;
+    if (rptr >= buffer->frames) {
+        rptr -= buffer->frames;
+    }
+    buffer->rptr = rptr;
+
+    atomic_sub(&buffer->used, copy);
+    return copy;
+}
+
+static int qjack_process(jack_nframes_t nframes, void *arg)
+{
+    QJackClient *c = (QJackClient *)arg;
+
+    if (c->state != QJACK_STATE_RUNNING) {
+        return 0;
+    }
+
+    /* get the buffers for the ports */
+    float *buffers[c->nchannels];
+    for (int i = 0; i < c->nchannels; ++i) {
+        buffers[i] = jack_port_get_buffer(c->port[i], nframes);
+    }
+
+    if (c->out) {
+        qjack_buffer_read_l(&c->fifo, buffers, nframes);
+    } else {
+        qjack_buffer_write_l(&c->fifo, buffers, nframes);
+    }
+
+    return 0;
+}
+
+static void qjack_port_registration(jack_port_id_t port, int reg, void *arg)
+{
+    if (reg) {
+        QJackClient *c = (QJackClient *)arg;
+        c->connect_ports = true;
+    }
+}
+
+static int qjack_xrun(void *arg)
+{
+    QJackClient *c = (QJackClient *)arg;
+    if (c->state != QJACK_STATE_RUNNING) {
+        return 0;
+    }
+
+    qjack_buffer_clear(&c->fifo);
+    return 0;
+}
+
+static void qjack_shutdown(void *arg)
+{
+    QJackClient *c = (QJackClient *)arg;
+    c->state = QJACK_STATE_SHUTDOWN;
+}
+
+static void qjack_client_recover(QJackClient *c)
+{
+    if (c->state == QJACK_STATE_SHUTDOWN) {
+        qjack_client_fini(c);
+    }
+
+    /* packets is used simply to throttle this */
+    if (c->state == QJACK_STATE_DISCONNECTED &&
+        c->packets % 100 == 0) {
+
+        /* if not finished then attempt to recover */
+        if (!c->finished) {
+            dolog("attempting to reconnect to server\n");
+            qjack_client_init(c);
+        }
+    }
+}
+
+static size_t qjack_write(HWVoiceOut *hw, void *buf, size_t len)
+{
+    QJackOut *jo = (QJackOut *)hw;
+    ++jo->c.packets;
+
+    if (jo->c.state != QJACK_STATE_RUNNING) {
+        qjack_client_recover(&jo->c);
+        return len;
+    }
+
+    qjack_client_connect_ports(&jo->c);
+    return qjack_buffer_write(&jo->c.fifo, buf, len);
+}
+
+static size_t qjack_read(HWVoiceIn *hw, void *buf, size_t len)
+{
+    QJackIn *ji = (QJackIn *)hw;
+    ++ji->c.packets;
+
+    if (ji->c.state != QJACK_STATE_RUNNING) {
+        qjack_client_recover(&ji->c);
+        return len;
+    }
+
+    qjack_client_connect_ports(&ji->c);
+    return qjack_buffer_read(&ji->c.fifo, buf, len);
+}
+
+static void qjack_client_connect_ports(QJackClient *c)
+{
+    if (!c->connect_ports || !c->opt->connect_ports) {
+        return;
+    }
+
+    c->connect_ports = false;
+    const char **ports;
+    ports = jack_get_ports(c->client, c->opt->connect_ports, NULL,
+        c->out ? JackPortIsInput : JackPortIsOutput);
+
+    if (!ports) {
+        return;
+    }
+
+    for (int i = 0; i < c->nchannels && ports[i]; ++i) {
+        const char *p = jack_port_name(c->port[i]);
+        if (jack_port_connected_to(c->port[i], ports[i])) {
+            continue;
+        }
+
+        if (c->out) {
+            dolog("connect %s -> %s\n", p, ports[i]);
+            jack_connect(c->client, p, ports[i]);
+        } else {
+            dolog("connect %s -> %s\n", ports[i], p);
+            jack_connect(c->client, ports[i], p);
+        }
+    }
+}
+
+static int qjack_client_init(QJackClient *c)
+{
+    jack_status_t status;
+    char client_name[jack_client_name_size()];
+    jack_options_t options = JackNullOption;
+
+    c->finished      = false;
+    c->connect_ports = true;
+
+    snprintf(client_name, sizeof(client_name), "%s-%s",
+        c->out ? "out" : "in",
+        c->opt->client_name ? c->opt->client_name : qemu_get_vm_name());
+
+    if (c->opt->exact_name) {
+        options |= JackUseExactName;
+    }
+
+    if (!c->opt->start_server) {
+        options |= JackNoStartServer;
+    }
+
+    if (c->opt->server_name) {
+        options |= JackServerName;
+    }
+
+    c->client = jack_client_open(client_name, options, &status,
+      c->opt->server_name);
+
+    if (c->client == NULL) {
+        dolog("jack_client_open failed: status = 0x%2.0x\n", status);
+        if (status & JackServerFailed) {
+            dolog("unable to connect to JACK server\n");
+        }
+        return -1;
+    }
+
+    c->freq = jack_get_sample_rate(c->client);
+
+    if (status & JackServerStarted) {
+        dolog("JACK server started\n");
+    }
+
+    if (status & JackNameNotUnique) {
+        dolog("JACK unique name assigned %s\n",
+          jack_get_client_name(c->client));
+    }
+
+    jack_set_process_callback(c->client, qjack_process , c);
+    jack_set_port_registration_callback(c->client, qjack_port_registration, c);
+    jack_set_xrun_callback(c->client, qjack_xrun, c);
+    jack_on_shutdown(c->client, qjack_shutdown, c);
+
+    /*
+     * ensure the buffersize is no smaller then 512 samples, some (all?) qemu
+     * virtual devices do not work correctly otherwise
+     */
+    if (c->buffersize < 512) {
+        c->buffersize = 512;
+    }
+
+    /* create a 2 period buffer */
+    qjack_buffer_create(&c->fifo, c->nchannels, c->buffersize * 2);
+
+    /* allocate and register the ports */
+    c->port = g_malloc(sizeof(jack_port_t *) * c->nchannels);
+    for (int i = 0; i < c->nchannels; ++i) {
+
+        char port_name[16];
+        snprintf(
+            port_name,
+            sizeof(port_name),
+            c->out ? "output %d" : "input %d",
+            i);
+
+        c->port[i] = jack_port_register(
+            c->client,
+            port_name,
+            JACK_DEFAULT_AUDIO_TYPE,
+            c->out ? JackPortIsOutput : JackPortIsInput,
+            0);
+    }
+
+    /* activate the session */
+    jack_activate(c->client);
+    c->buffersize = jack_get_buffer_size(c->client);
+
+    qjack_client_connect_ports(c);
+    c->state = QJACK_STATE_RUNNING;
+    return 0;
+}
+
+static int qjack_init_out(HWVoiceOut *hw, struct audsettings *as,
+    void *drv_opaque)
+{
+    QJackOut *jo  = (QJackOut *)hw;
+    Audiodev *dev = (Audiodev *)drv_opaque;
+
+    if (jo->c.state != QJACK_STATE_DISCONNECTED) {
+        return 0;
+    }
+
+    jo->c.out       = true;
+    jo->c.nchannels = as->nchannels;
+    jo->c.opt       = dev->u.jack.out;
+    int ret = qjack_client_init(&jo->c);
+    if (ret != 0) {
+        return ret;
+    }
+
+    /* report the buffer size to qemu */
+    hw->samples = jo->c.buffersize;
+
+    /* report the audio format we support */
+    struct audsettings os = {
+        .freq       = jo->c.freq,
+        .nchannels  = jo->c.nchannels,
+        .fmt        = AUDIO_FORMAT_F32,
+        .endianness = 0
+    };
+    audio_pcm_init_info(&hw->info, &os);
+
+    dolog("JACK output configured for %dHz (%d samples)\n",
+        jo->c.freq, jo->c.buffersize);
+
+    return 0;
+}
+
+static int qjack_init_in(HWVoiceIn *hw, struct audsettings *as,
+    void *drv_opaque)
+{
+    QJackIn  *ji  = (QJackIn *)hw;
+    Audiodev *dev = (Audiodev *)drv_opaque;
+
+    if (ji->c.state != QJACK_STATE_DISCONNECTED) {
+        return 0;
+    }
+
+    ji->c.out       = false;
+    ji->c.nchannels = as->nchannels;
+    ji->c.opt       = dev->u.jack.in;
+    int ret = qjack_client_init(&ji->c);
+    if (ret != 0) {
+        return ret;
+    }
+
+    /* report the buffer size to qemu */
+    hw->samples = ji->c.buffersize;
+
+    /* report the audio format we support */
+    struct audsettings is = {
+        .freq       = ji->c.freq,
+        .nchannels  = ji->c.nchannels,
+        .fmt        = AUDIO_FORMAT_F32,
+        .endianness = 0
+    };
+    audio_pcm_init_info(&hw->info, &is);
+
+    dolog("JACK input configured for %dHz (%d samples)\n",
+        ji->c.freq, ji->c.buffersize);
+
+    return 0;
+}
+
+static void qjack_client_fini(QJackClient *c)
+{
+    switch (c->state) {
+    case QJACK_STATE_RUNNING:
+        /* fallthrough */
+
+    case QJACK_STATE_STOPPED:
+        for (int i = 0; i < c->nchannels; ++i) {
+            jack_port_unregister(c->client, c->port[i]);
+        }
+        jack_deactivate(c->client);
+        /* fallthrough */
+
+    case QJACK_STATE_SHUTDOWN:
+        jack_client_close(c->client);
+        /* fallthrough */
+
+    case QJACK_STATE_DISCONNECTED:
+        break;
+    }
+
+    qjack_buffer_free(&c->fifo);
+    g_free(c->port);
+
+    c->state = QJACK_STATE_DISCONNECTED;
+}
+
+static void qjack_fini_out(HWVoiceOut *hw)
+{
+    QJackOut *jo = (QJackOut *)hw;
+    jo->c.finished = true;
+    qjack_client_fini(&jo->c);
+}
+
+static void qjack_fini_in(HWVoiceIn *hw)
+{
+    QJackIn *ji = (QJackIn *)hw;
+    ji->c.finished = true;
+    qjack_client_fini(&ji->c);
+}
+
+static void qjack_enable_out(HWVoiceOut *hw, bool enable)
+{
+}
+
+static void qjack_enable_in(HWVoiceIn *hw, bool enable)
+{
+}
+
+static int qjack_thread_creator(jack_native_thread_t *thread,
+    const pthread_attr_t *attr, void *(*function)(void *), void *arg)
+{
+    int ret = pthread_create(thread, attr, function, arg);
+    if (ret != 0) {
+        return ret;
+    }
+
+    /* set the name of the thread */
+    pthread_setname_np(*thread, "jack-client");
+
+    return ret;
+}
+
+static void *qjack_init(Audiodev *dev)
+{
+    assert(dev->driver == AUDIODEV_DRIVER_JACK);
+
+    dev->u.jack.has_in = false;
+
+    return dev;
+}
+
+static void qjack_fini(void *opaque)
+{
+}
+
+static struct audio_pcm_ops jack_pcm_ops = {
+    .init_out       = qjack_init_out,
+    .fini_out       = qjack_fini_out,
+    .write          = qjack_write,
+    .run_buffer_out = audio_generic_run_buffer_out,
+    .enable_out     = qjack_enable_out,
+
+    .init_in        = qjack_init_in,
+    .fini_in        = qjack_fini_in,
+    .read           = qjack_read,
+    .enable_in      = qjack_enable_in
+};
+
+static struct audio_driver jack_driver = {
+    .name           = "jack",
+    .descr          = "JACK Audio Connection Kit Client",
+    .init           = qjack_init,
+    .fini           = qjack_fini,
+    .pcm_ops        = &jack_pcm_ops,
+    .can_be_default = 1,
+    .max_voices_out = INT_MAX,
+    .max_voices_in  = INT_MAX,
+    .voice_size_out = sizeof(QJackOut),
+    .voice_size_in  = sizeof(QJackIn)
+};
+
+static void qjack_error(const char *msg)
+{
+    dolog("E: %s\n", msg);
+}
+
+static void qjack_info(const char *msg)
+{
+    dolog("I: %s\n", msg);
+}
+
+static void register_audio_jack(void)
+{
+    audio_driver_register(&jack_driver);
+    jack_set_thread_creator(qjack_thread_creator);
+    jack_set_error_function(qjack_error);
+    jack_set_info_function(qjack_info);
+}
+type_init(register_audio_jack);
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index d7490a379f3e..b4a4c11f3122 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -28,3 +28,8 @@ common-obj-$(CONFIG_AUDIO_SDL) += sdl.mo
 sdl.mo-objs = sdlaudio.o
 sdl.mo-cflags := $(SDL_CFLAGS)
 sdl.mo-libs := $(SDL_LIBS)
+
+# jack module
+common-obj-$(CONFIG_AUDIO_JACK) += jack.mo
+jack.mo-objs = jackaudio.o
+jack.mo-libs := $(JACK_LIBS)
diff --git a/qapi/audio.json b/qapi/audio.json
index c31251f45b57..f62bd0d7f6ec 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -152,6 +152,55 @@
     '*out':     'AudiodevPerDirectionOptions',
     '*latency': 'uint32' } }
 
+##
+# @AudiodevJackPerDirectionOptions:
+#
+# Options of the JACK backend that are used for both playback and
+# recording.
+#
+# @server-name: select from among several possible concurrent server instances
+# (default: environment variable $JACK_DEFAULT_SERVER if set, else "default")
+#
+# @client-name: the client name to use. The server will modify this name to
+# create a unique variant, if needed unless @exact-name is true (default: the
+# guest's name)
+#
+# @connect-ports: if set, a regular expression of JACK client port name(s) to
+# monitor for and automatically connect to
+#
+# @start-server: start a jack server process if one is not already present
+# (default: false)
+#
+# @exact-name: use the exact name requested otherwise JACK automatically
+# generates a unique one, if needed (default: false)
+#
+# Since: 5.1
+##
+{ 'struct': 'AudiodevJackPerDirectionOptions',
+  'base': 'AudiodevPerDirectionOptions',
+  'data': {
+    '*server-name':   'str',
+    '*client-name':   'str',
+    '*connect-ports': 'str',
+    '*start-server':  'bool',
+    '*exact-name':    'bool' } }
+
+##
+# @AudiodevJackOptions:
+#
+# Options of the JACK audio backend.
+#
+# @in: options of the capture stream
+#
+# @out: options of the playback stream
+#
+# Since: 5.1
+##
+{ 'struct': 'AudiodevJackOptions',
+  'data': {
+    '*in':  'AudiodevJackPerDirectionOptions',
+    '*out': 'AudiodevJackPerDirectionOptions' } }
+
 ##
 # @AudiodevOssPerDirectionOptions:
 #
@@ -297,11 +346,13 @@
 #
 # An enumeration of possible audio backend drivers.
 #
+# @jack: JACK audio backend (since 5.1)
+#
 # Since: 4.0
 ##
 { 'enum': 'AudiodevDriver',
-  'data': [ 'none', 'alsa', 'coreaudio', 'dsound', 'oss', 'pa', 'sdl',
-            'spice', 'wav' ] }
+  'data': [ 'none', 'alsa', 'coreaudio', 'dsound', 'jack', 'oss', 'pa',
+            'sdl', 'spice', 'wav' ] }
 
 ##
 # @Audiodev:
@@ -327,6 +378,7 @@
     'alsa':      'AudiodevAlsaOptions',
     'coreaudio': 'AudiodevCoreaudioOptions',
     'dsound':    'AudiodevDsoundOptions',
+    'jack':      'AudiodevJackOptions',
     'oss':       'AudiodevOssOptions',
     'pa':        'AudiodevPaOptions',
     'sdl':       'AudiodevGenericOptions',
-- 
2.18.4

[PULL 4/8] audio/mixeng: fix clang 10+ warning

[Gerd Hoffmann]

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

The code in CONV_NATURAL_FLOAT() and CLIP_NATURAL_FLOAT()
seems to use the constant 2^31-0.5 to convert float to integer
and back. But the float type lacks the required precision and
the constant used for the conversion is 2^31. This is equiva-
lent to a [-1.f, 1.f] <-> [INT32_MIN, INT32_MAX + 1] mapping.

This patch explicitly writes down the used constant. The
compiler generated code doesn't change.

The constant 2^31 has an exact float representation and the
clang 10 compiler stops complaining about an implicit int to
float conversion with a changed value.

A few notes:
- The conversion of 1.f to INT32_MAX + 1 doesn't overflow. The
  type of the destination variable is int64_t.
- At a later stage one of the clip_* functions in
  audio/mixeng_template.h limits INT32_MAX + 1 to the integer
  range.
- The clip_natural_float_* functions in audio/mixeng.c convert
  INT32_MAX and INT32_MAX + 1 to 1.f.

Buglink: https://bugs.launchpad.net/bugs/1878627
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200523201712.23908-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/mixeng.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/audio/mixeng.c b/audio/mixeng.c
index 739a500449ce..924dcb858af5 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -271,11 +271,12 @@ f_sample *mixeng_clip[2][2][2][3] = {
 #define CONV_NATURAL_FLOAT(x) (x)
 #define CLIP_NATURAL_FLOAT(x) (x)
 #else
-static const float float_scale = UINT_MAX / 2.f;
+/* macros to map [-1.f, 1.f] <-> [INT32_MIN, INT32_MAX + 1] */
+static const float float_scale = (int64_t)INT32_MAX + 1;
 #define CONV_NATURAL_FLOAT(x) ((x) * float_scale)
 
 #ifdef RECIPROCAL
-static const float float_scale_reciprocal = 2.f / UINT_MAX;
+static const float float_scale_reciprocal = 1.f / ((int64_t)INT32_MAX + 1);
 #define CLIP_NATURAL_FLOAT(x) ((x) * float_scale_reciprocal)
 #else
 #define CLIP_NATURAL_FLOAT(x) ((x) / float_scale)
-- 
2.18.4

[PULL 5/8] audio: fix wavcapture segfault

[Gerd Hoffmann]

From: Bruce Rogers <brogers@suse.com>

Commit 571a8c522e caused the HMP wavcapture command to segfault when
processing audio data in audio_pcm_sw_write(), where a NULL
sw->hw->pcm_ops is dereferenced. This fix checks that the pointer is
valid before dereferincing it. A similar fix is also made in the
parallel function audio_pcm_sw_read().

Fixes: 571a8c522e (audio: split ctl_* functions into enable_* and
volume_*)
Signed-off-by: Bruce Rogers <brogers@suse.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20200521172931.121903-1-brogers@suse.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 95d9fb16caa5..ce8c6dec5f47 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -649,7 +649,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
         total += isamp;
     }
 
-    if (!hw->pcm_ops->volume_in) {
+    if (hw->pcm_ops && !hw->pcm_ops->volume_in) {
         mixeng_volume (sw->buf, ret, &sw->vol);
     }
 
@@ -736,7 +736,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
     if (swlim) {
         sw->conv (sw->buf, buf, swlim);
 
-        if (!sw->hw->pcm_ops->volume_out) {
+        if (sw->hw->pcm_ops && !sw->hw->pcm_ops->volume_out) {
             mixeng_volume (sw->buf, swlim, &sw->vol);
         }
     }
-- 
2.18.4

[PULL 6/8] audio: Let audio_sample_to_uint64() use const samples argument

[Gerd Hoffmann]

From: Philippe Mathieu-Daudé <f4bug@amsat.org>

The samples are the input to convert to u64. As we should
not modify them, mark the argument const.

Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20200505132603.8575-2-f4bug@amsat.org>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio.h  | 2 +-
 audio/mixeng.c | 4 ++--
 2 files changed, 3 insertions(+), 3 deletions(-)

diff --git a/audio/audio.h b/audio/audio.h
index 0db3c7dd5e06..f27a12298fe5 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -163,7 +163,7 @@ int wav_start_capture(AudioState *state, CaptureState *s, const char *path,
 bool audio_is_cleaning_up(void);
 void audio_cleanup(void);
 
-void audio_sample_to_uint64(void *samples, int pos,
+void audio_sample_to_uint64(const 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);
diff --git a/audio/mixeng.c b/audio/mixeng.c
index 924dcb858af5..f27deb165b67 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -339,10 +339,10 @@ f_sample *mixeng_clip_float[2] = {
     clip_natural_float_from_stereo,
 };
 
-void audio_sample_to_uint64(void *samples, int pos,
+void audio_sample_to_uint64(const void *samples, int pos,
                             uint64_t *left, uint64_t *right)
 {
-    struct st_sample *sample = samples;
+    const struct st_sample *sample = samples;
     sample += pos;
 #ifdef FLOAT_MIXENG
     error_report(
-- 
2.18.4

[PULL 7/8] audio: Let capture_callback handler use const buffer argument

[Gerd Hoffmann]

From: Philippe Mathieu-Daudé <f4bug@amsat.org>

The buffer is the captured input to pass to backends.
As we should not modify it, mark the argument const.

Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20200505132603.8575-3-f4bug@amsat.org>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio.h      | 2 +-
 audio/wavcapture.c | 2 +-
 ui/vnc.c           | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

diff --git a/audio/audio.h b/audio/audio.h
index f27a12298fe5..b883ebfb1f8e 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -60,7 +60,7 @@ typedef enum {
 
 struct audio_capture_ops {
     void (*notify) (void *opaque, audcnotification_e cmd);
-    void (*capture) (void *opaque, void *buf, int size);
+    void (*capture) (void *opaque, const void *buf, int size);
     void (*destroy) (void *opaque);
 };
 
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index 8d7ce2eda145..17e87ed6f45e 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -71,7 +71,7 @@ static void wav_destroy (void *opaque)
     g_free (wav->path);
 }
 
-static void wav_capture (void *opaque, void *buf, int size)
+static void wav_capture(void *opaque, const void *buf, int size)
 {
     WAVState *wav = opaque;
 
diff --git a/ui/vnc.c b/ui/vnc.c
index 1d7138a3a073..12a12714e129 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -1177,7 +1177,7 @@ static void audio_capture_destroy(void *opaque)
 {
 }
 
-static void audio_capture(void *opaque, void *buf, int size)
+static void audio_capture(void *opaque, const void *buf, int size)
 {
     VncState *vs = opaque;
 
-- 
2.18.4

[PULL 8/8] hw/mips/mips_fulong2e: Remove unused 'audio/audio.h' include

[Gerd Hoffmann]

From: Philippe Mathieu-Daudé <f4bug@amsat.org>

The Fuloong machine never had to use "audio/audio.h", remove it.

Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Reviewed-by: Huacai Chen <chenhc@lemote.com>
Message-id: 20200515084209.9419-1-f4bug@amsat.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 hw/mips/mips_fulong2e.c | 1 -
 1 file changed, 1 deletion(-)

diff --git a/hw/mips/mips_fulong2e.c b/hw/mips/mips_fulong2e.c
index ef02d54b335c..05b9efa516cf 100644
--- a/hw/mips/mips_fulong2e.c
+++ b/hw/mips/mips_fulong2e.c
@@ -33,7 +33,6 @@
 #include "hw/mips/mips.h"
 #include "hw/mips/cpudevs.h"
 #include "hw/pci/pci.h"
-#include "audio/audio.h"
 #include "qemu/log.h"
 #include "hw/loader.h"
 #include "hw/ide/pci.h"
-- 
2.18.4

Re: [PULL 0/8] Audio 20200526 patches

[Peter Maydell]

On Tue, 26 May 2020 at 08:59, Gerd Hoffmann <kraxel@redhat.com> wrote:
>
> The following changes since commit fea8f3ed739536fca027cf56af7f5576f37ef9cd:
>
>   Merge remote-tracking branch 'remotes/philmd-gitlab/tags/pflash-next-20200522' into staging (2020-05-22 18:54:47 +0100)
>
> are available in the Git repository at:
>
>   git://git.kraxel.org/qemu tags/audio-20200526-pull-request
>
> for you to fetch changes up to b3b8a1fea6ed5004bbad2f70833caee70402bf02:
>
>   hw/mips/mips_fulong2e: Remove unused 'audio/audio.h' include (2020-05-26 08:46:14 +0200)
>
> ----------------------------------------------------------------
> audio: add JACK client audiodev.
> audio: bugfixes and cleanups.
>
> ----------------------------------------------------------------

Applied, thanks.

Please update the changelog at https://wiki.qemu.org/ChangeLog/5.1
for any user-visible changes.

-- PMM

Re: [PULL 0/8] Audio 20200526 patches

[Markus Armbruster]

Gerd Hoffmann <kraxel@redhat.com> writes:

> The following changes since commit fea8f3ed739536fca027cf56af7f5576f37ef9cd:
>
>   Merge remote-tracking branch 'remotes/philmd-gitlab/tags/pflash-next-20200522' into staging (2020-05-22 18:54:47 +0100)
>
> are available in the Git repository at:
>
>   git://git.kraxel.org/qemu tags/audio-20200526-pull-request
>
> for you to fetch changes up to b3b8a1fea6ed5004bbad2f70833caee70402bf02:
>
>   hw/mips/mips_fulong2e: Remove unused 'audio/audio.h' include (2020-05-26 08:46:14 +0200)
>
> ----------------------------------------------------------------
> audio: add JACK client audiodev.
> audio: bugfixes and cleanups.
>
> ----------------------------------------------------------------

$ qemu-system-aarch64 -S -accel qtest -display none -M akita -monitor stdio
(qemu) qemu-system-aarch64: /work/armbru/qemu/audio/paaudio.c:779: qpa_audio_init: Assertion `dev->driver == AUDIODEV_DRIVER_PA' failed.

Same for 40p, akita, borzoi, cheetah, integratorcp, milkymist, musicpal,
n800, n810, realview-eb, realview-eb-mpcore, realview-pb-a8,
realview-pbx-a9, spitz, terrier, versatileab, versatilepb, vexpress-a15,
vexpress-a9, xlnx-zcu102, z2.

git-bisect blames commit 2e44570321 "audio/jack: add JACK client
audiodev".

Re: [PULL 0/8] Audio 20200526 patches

[Gerd Hoffmann]

On Wed, May 27, 2020 at 08:30:18AM +0200, Markus Armbruster wrote:
> Gerd Hoffmann <kraxel@redhat.com> writes:
> 
> > The following changes since commit fea8f3ed739536fca027cf56af7f5576f37ef9cd:
> >
> >   Merge remote-tracking branch 'remotes/philmd-gitlab/tags/pflash-next-20200522' into staging (2020-05-22 18:54:47 +0100)
> >
> > are available in the Git repository at:
> >
> >   git://git.kraxel.org/qemu tags/audio-20200526-pull-request
> >
> > for you to fetch changes up to b3b8a1fea6ed5004bbad2f70833caee70402bf02:
> >
> >   hw/mips/mips_fulong2e: Remove unused 'audio/audio.h' include (2020-05-26 08:46:14 +0200)
> >
> > ----------------------------------------------------------------
> > audio: add JACK client audiodev.
> > audio: bugfixes and cleanups.
> >
> > ----------------------------------------------------------------
> 
> $ qemu-system-aarch64 -S -accel qtest -display none -M akita -monitor stdio
> (qemu) qemu-system-aarch64: /work/armbru/qemu/audio/paaudio.c:779: qpa_audio_init: Assertion `dev->driver == AUDIODEV_DRIVER_PA' failed.

Hmm, seems to be a dependency issue in our build system.
rm -rf $builddir, rebuild, try again, assert gone.

take care,
  Gerd