[PULL 0/7] Audio 20210617 patches

[PULL 0/7] Audio 20210617 patches

[Gerd Hoffmann]

The following changes since commit 38848ce565849e5b867a5e08022b3c755039c11a:

  Merge remote-tracking branch 'remotes/pmaydell/tags/pull-target-arm-20210616' into staging (2021-06-16 17:02:30 +0100)

are available in the Git repository at:

  git://git.kraxel.org/qemu tags/audio-20210617-pull-request

for you to fetch changes up to 986bdbc6a29c4d7ef125299c5013783e30dc2cae:

  coreaudio: Fix output stream format settings (2021-06-17 12:00:26 +0200)

----------------------------------------------------------------
audio: bugfix collection.

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

Akihiko Odaki (2):
  audio: Fix format specifications of debug logs
  coreaudio: Fix output stream format settings

Philippe Mathieu-Daudé (1):
  hw/audio/sb16: Avoid assertion by restricting I/O sampling rate range

Volker Rümelin (4):
  alsaaudio: remove #ifdef DEBUG to avoid bit rot
  paaudio: remove unused stream flags
  audio: move code to audio/audio.c
  jackaudio: avoid that the client name contains the word (NULL)

 audio/audio_int.h            |  2 ++
 audio/alsaaudio.c            | 10 +++----
 audio/audio.c                | 15 ++++++++---
 audio/coreaudio.c            | 48 +++++++++------------------------
 audio/jackaudio.c            |  3 +--
 audio/paaudio.c              | 10 ++-----
 hw/audio/sb16.c              | 14 ++++++++++
 tests/qtest/fuzz-sb16-test.c | 52 ++++++++++++++++++++++++++++++++++++
 MAINTAINERS                  |  1 +
 tests/qtest/meson.build      |  1 +
 10 files changed, 102 insertions(+), 54 deletions(-)
 create mode 100644 tests/qtest/fuzz-sb16-test.c

-- 
2.31.1

[PULL 1/7] alsaaudio: remove #ifdef DEBUG to avoid bit rot

[Gerd Hoffmann]

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

Merge the #ifdef DEBUG code with the if statement a few lines
above to avoid bit rot.

Suggested-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/alsaaudio.c | 10 ++++------
 1 file changed, 4 insertions(+), 6 deletions(-)

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index fcc2f62864fb..2b9789e64771 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -34,6 +34,8 @@
 #define AUDIO_CAP "alsa"
 #include "audio_int.h"
 
+#define DEBUG_ALSA 0
+
 struct pollhlp {
     snd_pcm_t *handle;
     struct pollfd *pfds;
@@ -587,16 +589,12 @@ static int alsa_open(bool in, struct alsa_params_req *req,
 
     *handlep = handle;
 
-    if (obtfmt != req->fmt ||
-         obt->nchannels != req->nchannels ||
-         obt->freq != req->freq) {
+    if (DEBUG_ALSA || obtfmt != req->fmt ||
+        obt->nchannels != req->nchannels || obt->freq != req->freq) {
         dolog ("Audio parameters for %s\n", typ);
         alsa_dump_info(req, obt, obtfmt, apdo);
     }
 
-#ifdef DEBUG
-    alsa_dump_info(req, obt, obtfmt, apdo);
-#endif
     return 0;
 
  err:
-- 
2.31.1

[PULL 2/7] paaudio: remove unused stream flags

[Gerd Hoffmann]

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

In current code there are no calls to pa_stream_get_latency()
or pa_stream_get_time() to receive latency or time information.

Remove the flags PA_STREAM_INTERPOLATE_TIMING and
PA_STREAM_AUTO_TIMING_UPDATE which instruct PulseAudio to
calculate this information in regular intervals.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/paaudio.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index c97b22e970d8..14b4269c5500 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -463,10 +463,7 @@ static pa_stream *qpa_simple_new (
 
     pa_stream_set_state_callback(stream, stream_state_cb, c);
 
-    flags =
-        PA_STREAM_INTERPOLATE_TIMING
-        | PA_STREAM_AUTO_TIMING_UPDATE
-        | PA_STREAM_EARLY_REQUESTS;
+    flags = PA_STREAM_EARLY_REQUESTS;
 
     if (dev) {
         /* don't move the stream if the user specified a sink/source */
-- 
2.31.1

[PULL 3/7] audio: move code to audio/audio.c

[Gerd Hoffmann]

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

Move the code to generate the pa_context_new() application name
argument to a function in audio/audio.c. The new function
audio_application_name() will also be used in the jackaudio
backend.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio_int.h | 2 ++
 audio/audio.c     | 9 +++++++++
 audio/paaudio.c   | 5 +----
 3 files changed, 12 insertions(+), 4 deletions(-)

diff --git a/audio/audio_int.h b/audio/audio_int.h
index 06f0913835b0..6d685e24a388 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -243,6 +243,8 @@ void *audio_calloc (const char *funcname, int nmemb, size_t size);
 
 void audio_run(AudioState *s, const char *msg);
 
+const char *audio_application_name(void);
+
 typedef struct RateCtl {
     int64_t start_ticks;
     int64_t bytes_sent;
diff --git a/audio/audio.c b/audio/audio.c
index 534278edfed2..052ca6cb789b 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -32,6 +32,7 @@
 #include "qapi/qapi-visit-audio.h"
 #include "qemu/cutils.h"
 #include "qemu/module.h"
+#include "qemu-common.h"
 #include "sysemu/replay.h"
 #include "sysemu/runstate.h"
 #include "ui/qemu-spice.h"
@@ -2172,6 +2173,14 @@ const char *audio_get_id(QEMUSoundCard *card)
     }
 }
 
+const char *audio_application_name(void)
+{
+    const char *vm_name;
+
+    vm_name = qemu_get_vm_name();
+    return vm_name ? vm_name : "qemu";
+}
+
 void audio_rate_start(RateCtl *rate)
 {
     memset(rate, 0, sizeof(RateCtl));
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 14b4269c5500..75401d53910a 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -2,7 +2,6 @@
 
 #include "qemu/osdep.h"
 #include "qemu/module.h"
-#include "qemu-common.h"
 #include "audio.h"
 #include "qapi/opts-visitor.h"
 
@@ -753,7 +752,6 @@ static int qpa_validate_per_direction_opts(Audiodev *dev,
 /* common */
 static void *qpa_conn_init(const char *server)
 {
-    const char *vm_name;
     PAConnection *c = g_malloc0(sizeof(PAConnection));
     QTAILQ_INSERT_TAIL(&pa_conns, c, list);
 
@@ -762,9 +760,8 @@ static void *qpa_conn_init(const char *server)
         goto fail;
     }
 
-    vm_name = qemu_get_vm_name();
     c->context = pa_context_new(pa_threaded_mainloop_get_api(c->mainloop),
-                                vm_name ? vm_name : "qemu");
+                                audio_application_name());
     if (!c->context) {
         goto fail;
     }
-- 
2.31.1

[PULL 4/7] jackaudio: avoid that the client name contains the word (NULL)

[Gerd Hoffmann]

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

Currently with jackaudio client name and qemu guest name unset,
the JACK client names are out-(NULL) and in-(NULL). These names
are user visible in the patch bay. Replace the function call to
qemu_get_vm_name() with a call to audio_application_name() which
replaces NULL with "qemu" to have more descriptive names.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/jackaudio.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

diff --git a/audio/jackaudio.c b/audio/jackaudio.c
index 3031c4e29bd4..e7de6d5433e9 100644
--- a/audio/jackaudio.c
+++ b/audio/jackaudio.c
@@ -26,7 +26,6 @@
 #include "qemu/module.h"
 #include "qemu/atomic.h"
 #include "qemu/main-loop.h"
-#include "qemu-common.h"
 #include "audio.h"
 
 #define AUDIO_CAP "jack"
@@ -412,7 +411,7 @@ static int qjack_client_init(QJackClient *c)
 
     snprintf(client_name, sizeof(client_name), "%s-%s",
         c->out ? "out" : "in",
-        c->opt->client_name ? c->opt->client_name : qemu_get_vm_name());
+        c->opt->client_name ? c->opt->client_name : audio_application_name());
 
     if (c->opt->exact_name) {
         options |= JackUseExactName;
-- 
2.31.1

[PULL 5/7] hw/audio/sb16: Avoid assertion by restricting I/O sampling rate range

[Gerd Hoffmann]

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

While the SB16 seems to work up to 48000 Hz, the "Sound Blaster Series
Hardware Programming Guide" limit the sampling range from 4000 Hz to
44100 Hz (Section 3-9, 3-10: Digitized Sound I/O Programming, tables
3-2 and 3-3).

Later, section 6-15 (DSP Commands) is more specific regarding the 41h /
42h registers (Set digitized sound output sampling rate):

  Valid sampling rates range from 5000 to 45000 Hz inclusive.

There is no comment regarding error handling if the register is filled
with an out-of-range value.  (See also section 3-28 "8-bit or 16-bit
Auto-initialize Transfer"). Assume limits are enforced in hardware.

This fixes triggering an assertion in audio_calloc():

  #1 abort
  #2 audio_bug audio/audio.c:119:9
  #3 audio_calloc audio/audio.c:154:9
  #4 audio_pcm_sw_alloc_resources_out audio/audio_template.h:116:15
  #5 audio_pcm_sw_init_out audio/audio_template.h:175:11
  #6 audio_pcm_create_voice_pair_out audio/audio_template.h:410:9
  #7 AUD_open_out audio/audio_template.h:503:14
  #8 continue_dma8 hw/audio/sb16.c:216:20
  #9 dma_cmd8 hw/audio/sb16.c:276:5
  #10 command hw/audio/sb16.c:0
  #11 dsp_write hw/audio/sb16.c:949:13
  #12 portio_write softmmu/ioport.c:205:13
  #13 memory_region_write_accessor softmmu/memory.c:491:5
  #14 access_with_adjusted_size softmmu/memory.c:552:18
  #15 memory_region_dispatch_write softmmu/memory.c:0:13
  #16 flatview_write_continue softmmu/physmem.c:2759:23
  #17 flatview_write softmmu/physmem.c:2799:14
  #18 address_space_write softmmu/physmem.c:2891:18
  #19 cpu_outw softmmu/ioport.c:70:5

[*] http://www.baudline.com/solutions/full_duplex/sb16_pci/index.html

OSS-Fuzz Report: https://bugs.chromium.org/p/oss-fuzz/issues/detail?id=29174

Fixes: 85571bc7415 ("audio merge (malc)")
Buglink: https://bugs.launchpad.net/bugs/1910603
Tested-by: Qiang Liu <cyruscyliu@gmail.com>
Reviewed-by: Qiang Liu <cyruscyliu@gmail.com>
Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20210616104349.2398060-1-f4bug@amsat.org>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 hw/audio/sb16.c              | 14 ++++++++++
 tests/qtest/fuzz-sb16-test.c | 52 ++++++++++++++++++++++++++++++++++++
 MAINTAINERS                  |  1 +
 tests/qtest/meson.build      |  1 +
 4 files changed, 68 insertions(+)
 create mode 100644 tests/qtest/fuzz-sb16-test.c

diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index 8b2070041025..5cf121fe3639 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -115,6 +115,9 @@ struct SB16State {
     PortioList portio_list;
 };
 
+#define SAMPLE_RATE_MIN 5000
+#define SAMPLE_RATE_MAX 45000
+
 static void SB_audio_callback (void *opaque, int free);
 
 static int magic_of_irq (int irq)
@@ -241,6 +244,17 @@ static void dma_cmd8 (SB16State *s, int mask, int dma_len)
         int tmp = (256 - s->time_const);
         s->freq = (1000000 + (tmp / 2)) / tmp;
     }
+    if (s->freq < SAMPLE_RATE_MIN) {
+        qemu_log_mask(LOG_GUEST_ERROR,
+                      "sampling range too low: %d, increasing to %u\n",
+                      s->freq, SAMPLE_RATE_MIN);
+        s->freq = SAMPLE_RATE_MIN;
+    } else if (s->freq > SAMPLE_RATE_MAX) {
+        qemu_log_mask(LOG_GUEST_ERROR,
+                      "sampling range too high: %d, decreasing to %u\n",
+                      s->freq, SAMPLE_RATE_MAX);
+        s->freq = SAMPLE_RATE_MAX;
+    }
 
     if (dma_len != -1) {
         s->block_size = dma_len << s->fmt_stereo;
diff --git a/tests/qtest/fuzz-sb16-test.c b/tests/qtest/fuzz-sb16-test.c
new file mode 100644
index 000000000000..51030cd7dc40
--- /dev/null
+++ b/tests/qtest/fuzz-sb16-test.c
@@ -0,0 +1,52 @@
+/*
+ * QTest fuzzer-generated testcase for sb16 audio device
+ *
+ * Copyright (c) 2021 Philippe Mathieu-Daudé <f4bug@amsat.org>
+ *
+ * SPDX-License-Identifier: GPL-2.0-or-later
+ */
+
+#include "qemu/osdep.h"
+#include "libqos/libqtest.h"
+
+/*
+ * This used to trigger the assert in audio_calloc
+ * https://bugs.launchpad.net/qemu/+bug/1910603
+ */
+static void test_fuzz_sb16_0x1c(void)
+{
+    QTestState *s = qtest_init("-M q35 -display none "
+                               "-device sb16,audiodev=snd0 "
+                               "-audiodev none,id=snd0");
+    qtest_outw(s, 0x22c, 0x41);
+    qtest_outb(s, 0x22c, 0x00);
+    qtest_outw(s, 0x22c, 0x1004);
+    qtest_outw(s, 0x22c, 0x001c);
+    qtest_quit(s);
+}
+
+static void test_fuzz_sb16_0x91(void)
+{
+    QTestState *s = qtest_init("-M pc -display none "
+                               "-device sb16,audiodev=none "
+                               "-audiodev id=none,driver=none");
+    qtest_outw(s, 0x22c, 0xf141);
+    qtest_outb(s, 0x22c, 0x00);
+    qtest_outb(s, 0x22c, 0x24);
+    qtest_outb(s, 0x22c, 0x91);
+    qtest_quit(s);
+}
+
+int main(int argc, char **argv)
+{
+    const char *arch = qtest_get_arch();
+
+    g_test_init(&argc, &argv, NULL);
+
+   if (strcmp(arch, "i386") == 0) {
+        qtest_add_func("fuzz/test_fuzz_sb16/1c", test_fuzz_sb16_0x1c);
+        qtest_add_func("fuzz/test_fuzz_sb16/91", test_fuzz_sb16_0x91);
+   }
+
+   return g_test_run();
+}
diff --git a/MAINTAINERS b/MAINTAINERS
index 636bf2f53655..4842cc26e5ce 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -2221,6 +2221,7 @@ F: qapi/audio.json
 F: tests/qtest/ac97-test.c
 F: tests/qtest/es1370-test.c
 F: tests/qtest/intel-hda-test.c
+F: tests/qtest/fuzz-sb16-test.c
 
 Block layer core
 M: Kevin Wolf <kwolf@redhat.com>
diff --git a/tests/qtest/meson.build b/tests/qtest/meson.build
index c3a223a83d6a..b03e85417001 100644
--- a/tests/qtest/meson.build
+++ b/tests/qtest/meson.build
@@ -20,6 +20,7 @@ slow_qtests = {
 qtests_generic = \
   (config_all_devices.has_key('CONFIG_MEGASAS_SCSI_PCI') ? ['fuzz-megasas-test'] : []) + \
   (config_all_devices.has_key('CONFIG_VIRTIO_SCSI') ? ['fuzz-virtio-scsi-test'] : []) + \
+  (config_all_devices.has_key('CONFIG_SB16') ? ['fuzz-sb16-test'] : []) + \
   [
   'cdrom-test',
   'device-introspect-test',
-- 
2.31.1

[PULL 6/7] audio: Fix format specifications of debug logs

[Gerd Hoffmann]

From: Akihiko Odaki <akihiko.odaki@gmail.com>

Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-id: 20210616141411.53892-1-akihiko.odaki@gmail.com
Message-Id: <20210616141411.53892-1-akihiko.odaki@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 052ca6cb789b..59453ef85673 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -705,7 +705,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
 
     if (live == hwsamples) {
 #ifdef DEBUG_OUT
-        dolog ("%s is full %d\n", sw->name, live);
+        dolog ("%s is full %zu\n", sw->name, live);
 #endif
         return 0;
     }
@@ -995,7 +995,7 @@ static size_t audio_get_avail (SWVoiceIn *sw)
     }
 
     ldebug (
-        "%s: get_avail live %d ret %" PRId64 "\n",
+        "%s: get_avail live %zu ret %" PRId64 "\n",
         SW_NAME (sw),
         live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
         );
@@ -1022,7 +1022,7 @@ static size_t audio_get_free(SWVoiceOut *sw)
     dead = sw->hw->mix_buf->size - live;
 
 #ifdef DEBUG_OUT
-    dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
+    dolog ("%s: get_free live %zu dead %zu ret %" PRId64 "\n",
            SW_NAME (sw),
            live, dead, (((int64_t) dead << 32) / sw->ratio) *
            sw->info.bytes_per_frame);
-- 
2.31.1

[PULL 7/7] coreaudio: Fix output stream format settings

[Gerd Hoffmann]

From: Akihiko Odaki <akihiko.odaki@gmail.com>

Before commit 7d6948cd98cf5ad8a3458a4ce7fdbcb79bcd1212, it was coded to
retrieve the initial output stream format settings, modify the frame
rate, and set again. However, I removed a frame rate modification code by
mistake in the commit. It also assumes the initial output stream format
is consistent with what QEMU expects, but that expectation is not in the
code, which makes it harder to understand and will lead to breakage if
the initial settings change.

This change explicitly sets all of the output stream settings to solve
these problems.

Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20210616141721.54091-1-akihiko.odaki@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/coreaudio.c | 48 +++++++++++++----------------------------------
 1 file changed, 13 insertions(+), 35 deletions(-)

diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 578ec9b8b2e6..f570e1ee60ef 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -39,7 +39,6 @@ typedef struct coreaudioVoiceOut {
     int frameSizeSetting;
     uint32_t bufferCount;
     UInt32 audioDevicePropertyBufferFrameSize;
-    AudioStreamBasicDescription outputStreamBasicDescription;
     AudioDeviceIOProcID ioprocid;
     bool enabled;
 } coreaudioVoiceOut;
@@ -114,24 +113,6 @@ static OSStatus coreaudio_set_framesize(AudioDeviceID id, UInt32 *framesize)
                                       framesize);
 }
 
-static OSStatus coreaudio_get_streamformat(AudioDeviceID id,
-                                           AudioStreamBasicDescription *d)
-{
-    UInt32 size = sizeof(*d);
-    AudioObjectPropertyAddress addr = {
-        kAudioDevicePropertyStreamFormat,
-        kAudioDevicePropertyScopeOutput,
-        kAudioObjectPropertyElementMaster
-    };
-
-    return AudioObjectGetPropertyData(id,
-                                      &addr,
-                                      0,
-                                      NULL,
-                                      &size,
-                                      d);
-}
-
 static OSStatus coreaudio_set_streamformat(AudioDeviceID id,
                                            AudioStreamBasicDescription *d)
 {
@@ -373,6 +354,17 @@ static OSStatus init_out_device(coreaudioVoiceOut *core)
     OSStatus status;
     AudioValueRange frameRange;
 
+    AudioStreamBasicDescription streamBasicDescription = {
+        .mBitsPerChannel = core->hw.info.bits,
+        .mBytesPerFrame = core->hw.info.bytes_per_frame,
+        .mBytesPerPacket = core->hw.info.bytes_per_frame,
+        .mChannelsPerFrame = core->hw.info.nchannels,
+        .mFormatFlags = kLinearPCMFormatFlagIsFloat,
+        .mFormatID = kAudioFormatLinearPCM,
+        .mFramesPerPacket = 1,
+        .mSampleRate = core->hw.info.freq
+    };
+
     status = coreaudio_get_voice(&core->outputDeviceID);
     if (status != kAudioHardwareNoError) {
         coreaudio_playback_logerr (status,
@@ -432,29 +424,16 @@ static OSStatus init_out_device(coreaudioVoiceOut *core)
     }
     core->hw.samples = core->bufferCount * core->audioDevicePropertyBufferFrameSize;
 
-    /* get StreamFormat */
-    status = coreaudio_get_streamformat(core->outputDeviceID,
-                                        &core->outputStreamBasicDescription);
-    if (status == kAudioHardwareBadObjectError) {
-        return 0;
-    }
-    if (status != kAudioHardwareNoError) {
-        coreaudio_playback_logerr (status,
-                                    "Could not get Device Stream properties\n");
-        core->outputDeviceID = kAudioDeviceUnknown;
-        return status;
-    }
-
     /* set Samplerate */
     status = coreaudio_set_streamformat(core->outputDeviceID,
-                                        &core->outputStreamBasicDescription);
+                                        &streamBasicDescription);
     if (status == kAudioHardwareBadObjectError) {
         return 0;
     }
     if (status != kAudioHardwareNoError) {
         coreaudio_playback_logerr (status,
                                    "Could not set samplerate %lf\n",
-                                   core->outputStreamBasicDescription.mSampleRate);
+                                   streamBasicDescription.mSampleRate);
         core->outputDeviceID = kAudioDeviceUnknown;
         return status;
     }
@@ -598,7 +577,6 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
         qapi_AudiodevCoreaudioPerDirectionOptions_base(cpdo), as, 11610);
 
     core->bufferCount = cpdo->has_buffer_count ? cpdo->buffer_count : 4;
-    core->outputStreamBasicDescription.mSampleRate = (Float64) as->freq;
 
     status = AudioObjectAddPropertyListener(kAudioObjectSystemObject,
                                             &voice_addr, handle_voice_change,
-- 
2.31.1

Re: [PULL 0/7] Audio 20210617 patches

[no-reply]

Patchew URL: https://patchew.org/QEMU/20210617124107.2386073-1-kraxel@redhat.com/



Hi,

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

Type: series
Message-id: 20210617124107.2386073-1-kraxel@redhat.com
Subject: [PULL 0/7] Audio 20210617 patches

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

Updating 3c8cf5a9c21ff8782164d1def7f44bd888713384
From https://github.com/patchew-project/qemu
 - [tag update]      patchew/20210617121628.20116-1-peter.maydell@linaro.org -> patchew/20210617121628.20116-1-peter.maydell@linaro.org
 * [new tag]         patchew/20210617124107.2386073-1-kraxel@redhat.com -> patchew/20210617124107.2386073-1-kraxel@redhat.com
Switched to a new branch 'test'
a6c9846 coreaudio: Fix output stream format settings
32f3f1e audio: Fix format specifications of debug logs
b51279d hw/audio/sb16: Avoid assertion by restricting I/O sampling rate range
e864a29 jackaudio: avoid that the client name contains the word (NULL)
c4a86a7 audio: move code to audio/audio.c
2800db0 paaudio: remove unused stream flags
0b3fd3c alsaaudio: remove #ifdef DEBUG to avoid bit rot

=== OUTPUT BEGIN ===
1/7 Checking commit 0b3fd3ca5f98 (alsaaudio: remove #ifdef DEBUG to avoid bit rot)
2/7 Checking commit 2800db079dd3 (paaudio: remove unused stream flags)
3/7 Checking commit c4a86a755701 (audio: move code to audio/audio.c)
4/7 Checking commit e864a29cb0c2 (jackaudio: avoid that the client name contains the word (NULL))
5/7 Checking commit b51279d277e6 (hw/audio/sb16: Avoid assertion by restricting I/O sampling rate range)
6/7 Checking commit 32f3f1e7b9c4 (audio: Fix format specifications of debug logs)
ERROR: space prohibited between function name and open parenthesis '('
#25: FILE: audio/audio.c:708:
+        dolog ("%s is full %zu\n", sw->name, live);

ERROR: space prohibited between function name and open parenthesis '('
#43: FILE: audio/audio.c:1025:
+    dolog ("%s: get_free live %zu dead %zu ret %" PRId64 "\n",

total: 2 errors, 0 warnings, 24 lines checked

Patch 6/7 has style problems, please review.  If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.

7/7 Checking commit a6c98463e79b (coreaudio: Fix output stream format settings)
=== OUTPUT END ===

Test command exited with code: 1


The full log is available at
http://patchew.org/logs/20210617124107.2386073-1-kraxel@redhat.com/testing.checkpatch/?type=message.
---
Email generated automatically by Patchew [https://patchew.org/].
Please send your feedback to patchew-devel@redhat.com

Re: [PULL 0/7] Audio 20210617 patches

[Peter Maydell]

On Thu, 17 Jun 2021 at 13:55, Gerd Hoffmann <kraxel@redhat.com> wrote:
>
> The following changes since commit 38848ce565849e5b867a5e08022b3c755039c11a:
>
>   Merge remote-tracking branch 'remotes/pmaydell/tags/pull-target-arm-20210616' into staging (2021-06-16 17:02:30 +0100)
>
> are available in the Git repository at:
>
>   git://git.kraxel.org/qemu tags/audio-20210617-pull-request
>
> for you to fetch changes up to 986bdbc6a29c4d7ef125299c5013783e30dc2cae:
>
>   coreaudio: Fix output stream format settings (2021-06-17 12:00:26 +0200)
>
> ----------------------------------------------------------------
> audio: bugfix collection.
>
> ----------------------------------------------------------------
>
> Akihiko Odaki (2):
>   audio: Fix format specifications of debug logs
>   coreaudio: Fix output stream format settings
>
> Philippe Mathieu-Daudé (1):
>   hw/audio/sb16: Avoid assertion by restricting I/O sampling rate range
>
> Volker Rümelin (4):
>   alsaaudio: remove #ifdef DEBUG to avoid bit rot
>   paaudio: remove unused stream flags
>   audio: move code to audio/audio.c
>   jackaudio: avoid that the client name contains the word (NULL)
>


Applied, thanks.

Please update the changelog at https://wiki.qemu.org/ChangeLog/6.1
for any user-visible changes.

-- PMM