* [PULL 0/8] Audio 20200526 patches
@ 2020-05-26 7:56 Gerd Hoffmann
2020-05-26 7:56 ` [PULL 1/8] es1370: check total frame count against current frame Gerd Hoffmann
` (9 more replies)
0 siblings, 10 replies; 12+ messages in thread
From: Gerd Hoffmann @ 2020-05-26 7:56 UTC (permalink / raw)
To: qemu-devel
Cc: Philippe Mathieu-Daudé,
Markus Armbruster, Aleksandar Markovic, Gerd Hoffmann,
Aleksandar Rikalo, Aurelien Jarno
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
^ permalink raw reply [flat|nested] 12+ messages in thread
* [PULL 1/8] es1370: check total frame count against current frame
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
@ 2020-05-26 7:56 ` Gerd Hoffmann
2020-05-26 7:56 ` [PULL 2/8] hw/audio/gus: Use AUDIO_HOST_ENDIANNESS definition from 'audio/audio.h' Gerd Hoffmann
` (8 subsequent siblings)
9 siblings, 0 replies; 12+ messages in thread
From: Gerd Hoffmann @ 2020-05-26 7:56 UTC (permalink / raw)
To: qemu-devel
Cc: Prasad J Pandit, Philippe Mathieu-Daudé,
Markus Armbruster, Aleksandar Markovic, Gerd Hoffmann,
Aleksandar Rikalo, Aurelien Jarno
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
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [PULL 2/8] hw/audio/gus: Use AUDIO_HOST_ENDIANNESS definition from 'audio/audio.h'
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
2020-05-26 7:56 ` [PULL 1/8] es1370: check total frame count against current frame Gerd Hoffmann
@ 2020-05-26 7:56 ` Gerd Hoffmann
2020-05-26 7:56 ` [PULL 3/8] audio/jack: add JACK client audiodev Gerd Hoffmann
` (7 subsequent siblings)
9 siblings, 0 replies; 12+ messages in thread
From: Gerd Hoffmann @ 2020-05-26 7:56 UTC (permalink / raw)
To: qemu-devel
Cc: Philippe Mathieu-Daudé,
Markus Armbruster, Philippe Mathieu-Daudé,
Aleksandar Markovic, Gerd Hoffmann, Aleksandar Rikalo,
Aurelien Jarno
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
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [PULL 3/8] audio/jack: add JACK client audiodev
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
2020-05-26 7:56 ` [PULL 1/8] es1370: check total frame count against current frame Gerd Hoffmann
2020-05-26 7:56 ` [PULL 2/8] hw/audio/gus: Use AUDIO_HOST_ENDIANNESS definition from 'audio/audio.h' Gerd Hoffmann
@ 2020-05-26 7:56 ` Gerd Hoffmann
2020-05-26 7:56 ` [PULL 4/8] audio/mixeng: fix clang 10+ warning Gerd Hoffmann
` (6 subsequent siblings)
9 siblings, 0 replies; 12+ messages in thread
From: Gerd Hoffmann @ 2020-05-26 7:56 UTC (permalink / raw)
To: qemu-devel
Cc: Geoffrey McRae, Philippe Mathieu-Daudé,
Markus Armbruster, Aleksandar Markovic, Gerd Hoffmann,
Aleksandar Rikalo, Aurelien Jarno
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
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [PULL 4/8] audio/mixeng: fix clang 10+ warning
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
` (2 preceding siblings ...)
2020-05-26 7:56 ` [PULL 3/8] audio/jack: add JACK client audiodev Gerd Hoffmann
@ 2020-05-26 7:56 ` Gerd Hoffmann
2020-05-26 7:56 ` [PULL 5/8] audio: fix wavcapture segfault Gerd Hoffmann
` (5 subsequent siblings)
9 siblings, 0 replies; 12+ messages in thread
From: Gerd Hoffmann @ 2020-05-26 7:56 UTC (permalink / raw)
To: qemu-devel
Cc: Philippe Mathieu-Daudé,
Volker Rümelin, Markus Armbruster, Aleksandar Markovic,
Gerd Hoffmann, Aleksandar Rikalo, Aurelien Jarno
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
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [PULL 5/8] audio: fix wavcapture segfault
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
` (3 preceding siblings ...)
2020-05-26 7:56 ` [PULL 4/8] audio/mixeng: fix clang 10+ warning Gerd Hoffmann
@ 2020-05-26 7:56 ` Gerd Hoffmann
2020-05-26 7:56 ` [PULL 6/8] audio: Let audio_sample_to_uint64() use const samples argument Gerd Hoffmann
` (4 subsequent siblings)
9 siblings, 0 replies; 12+ messages in thread
From: Gerd Hoffmann @ 2020-05-26 7:56 UTC (permalink / raw)
To: qemu-devel
Cc: Philippe Mathieu-Daudé,
Markus Armbruster, Bruce Rogers, Aleksandar Markovic,
Gerd Hoffmann, Aleksandar Rikalo, Aurelien Jarno
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
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [PULL 6/8] audio: Let audio_sample_to_uint64() use const samples argument
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
` (4 preceding siblings ...)
2020-05-26 7:56 ` [PULL 5/8] audio: fix wavcapture segfault Gerd Hoffmann
@ 2020-05-26 7:56 ` Gerd Hoffmann
2020-05-26 7:56 ` [PULL 7/8] audio: Let capture_callback handler use const buffer argument Gerd Hoffmann
` (3 subsequent siblings)
9 siblings, 0 replies; 12+ messages in thread
From: Gerd Hoffmann @ 2020-05-26 7:56 UTC (permalink / raw)
To: qemu-devel
Cc: Philippe Mathieu-Daudé,
Markus Armbruster, Philippe Mathieu-Daudé,
Aleksandar Markovic, Gerd Hoffmann, Aleksandar Rikalo,
Aurelien Jarno
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
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [PULL 7/8] audio: Let capture_callback handler use const buffer argument
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
` (5 preceding siblings ...)
2020-05-26 7:56 ` [PULL 6/8] audio: Let audio_sample_to_uint64() use const samples argument Gerd Hoffmann
@ 2020-05-26 7:56 ` Gerd Hoffmann
2020-05-26 7:56 ` [PULL 8/8] hw/mips/mips_fulong2e: Remove unused 'audio/audio.h' include Gerd Hoffmann
` (2 subsequent siblings)
9 siblings, 0 replies; 12+ messages in thread
From: Gerd Hoffmann @ 2020-05-26 7:56 UTC (permalink / raw)
To: qemu-devel
Cc: Philippe Mathieu-Daudé,
Markus Armbruster, Philippe Mathieu-Daudé,
Aleksandar Markovic, Gerd Hoffmann, Aleksandar Rikalo,
Aurelien Jarno
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
^ permalink raw reply related [flat|nested] 12+ messages in thread
* [PULL 8/8] hw/mips/mips_fulong2e: Remove unused 'audio/audio.h' include
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
` (6 preceding siblings ...)
2020-05-26 7:56 ` [PULL 7/8] audio: Let capture_callback handler use const buffer argument Gerd Hoffmann
@ 2020-05-26 7:56 ` Gerd Hoffmann
2020-05-26 13:05 ` [PULL 0/8] Audio 20200526 patches Peter Maydell
2020-05-27 6:30 ` Markus Armbruster
9 siblings, 0 replies; 12+ messages in thread
From: Gerd Hoffmann @ 2020-05-26 7:56 UTC (permalink / raw)
To: qemu-devel
Cc: Philippe Mathieu-Daudé,
Markus Armbruster, Philippe Mathieu-Daudé,
Aleksandar Markovic, Gerd Hoffmann, Aleksandar Rikalo,
Aurelien Jarno
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
^ permalink raw reply related [flat|nested] 12+ messages in thread
* Re: [PULL 0/8] Audio 20200526 patches
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
` (7 preceding siblings ...)
2020-05-26 7:56 ` [PULL 8/8] hw/mips/mips_fulong2e: Remove unused 'audio/audio.h' include Gerd Hoffmann
@ 2020-05-26 13:05 ` Peter Maydell
2020-05-27 6:30 ` Markus Armbruster
9 siblings, 0 replies; 12+ messages in thread
From: Peter Maydell @ 2020-05-26 13:05 UTC (permalink / raw)
To: Gerd Hoffmann
Cc: QEMU Developers, Markus Armbruster, Aleksandar Markovic,
Aleksandar Rikalo, Philippe Mathieu-Daudé,
Aurelien Jarno
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
^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: [PULL 0/8] Audio 20200526 patches
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
` (8 preceding siblings ...)
2020-05-26 13:05 ` [PULL 0/8] Audio 20200526 patches Peter Maydell
@ 2020-05-27 6:30 ` Markus Armbruster
2020-05-27 9:56 ` Gerd Hoffmann
9 siblings, 1 reply; 12+ messages in thread
From: Markus Armbruster @ 2020-05-27 6:30 UTC (permalink / raw)
To: Gerd Hoffmann
Cc: qemu-devel, Markus Armbruster, Aleksandar Markovic,
Aleksandar Rikalo, Philippe Mathieu-Daudé,
Aurelien Jarno
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".
^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: [PULL 0/8] Audio 20200526 patches
2020-05-27 6:30 ` Markus Armbruster
@ 2020-05-27 9:56 ` Gerd Hoffmann
0 siblings, 0 replies; 12+ messages in thread
From: Gerd Hoffmann @ 2020-05-27 9:56 UTC (permalink / raw)
To: Markus Armbruster
Cc: Aleksandar Markovic, Aleksandar Rikalo,
Philippe Mathieu-Daudé,
qemu-devel, Aurelien Jarno
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
^ permalink raw reply [flat|nested] 12+ messages in thread
end of thread, other threads:[~2020-05-27 9:57 UTC | newest]
Thread overview: 12+ messages (download: mbox.gz / follow: Atom feed)
-- links below jump to the message on this page --
2020-05-26 7:56 [PULL 0/8] Audio 20200526 patches Gerd Hoffmann
2020-05-26 7:56 ` [PULL 1/8] es1370: check total frame count against current frame Gerd Hoffmann
2020-05-26 7:56 ` [PULL 2/8] hw/audio/gus: Use AUDIO_HOST_ENDIANNESS definition from 'audio/audio.h' Gerd Hoffmann
2020-05-26 7:56 ` [PULL 3/8] audio/jack: add JACK client audiodev Gerd Hoffmann
2020-05-26 7:56 ` [PULL 4/8] audio/mixeng: fix clang 10+ warning Gerd Hoffmann
2020-05-26 7:56 ` [PULL 5/8] audio: fix wavcapture segfault Gerd Hoffmann
2020-05-26 7:56 ` [PULL 6/8] audio: Let audio_sample_to_uint64() use const samples argument Gerd Hoffmann
2020-05-26 7:56 ` [PULL 7/8] audio: Let capture_callback handler use const buffer argument Gerd Hoffmann
2020-05-26 7:56 ` [PULL 8/8] hw/mips/mips_fulong2e: Remove unused 'audio/audio.h' include Gerd Hoffmann
2020-05-26 13:05 ` [PULL 0/8] Audio 20200526 patches Peter Maydell
2020-05-27 6:30 ` Markus Armbruster
2020-05-27 9:56 ` Gerd Hoffmann
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