Merge remote-tracking branch 'remotes/kraxel/tags/audio-20210617-pull-request' into staging

audio: bugfix collection.

# gpg: Signature made Thu 17 Jun 2021 13:40:56 BST
# gpg:                using RSA key A0328CFFB93A17A79901FE7D4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <kraxel@redhat.com>" [full]
# gpg:                 aka "Gerd Hoffmann <gerd@kraxel.org>" [full]
# gpg:                 aka "Gerd Hoffmann (private) <kraxel@gmail.com>" [full]
# Primary key fingerprint: A032 8CFF B93A 17A7 9901  FE7D 4CB6 D8EE D3E8 7138

* remotes/kraxel/tags/audio-20210617-pull-request:
  coreaudio: Fix output stream format settings
  audio: Fix format specifications of debug logs
  hw/audio/sb16: Avoid assertion by restricting I/O sampling rate range
  jackaudio: avoid that the client name contains the word (NULL)
  audio: move code to audio/audio.c
  paaudio: remove unused stream flags
  alsaaudio: remove #ifdef DEBUG to avoid bit rot

Signed-off-by: Peter Maydell <peter.maydell@linaro.org>

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>

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:

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"
@@ -704,7 +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;
     }
@@ -994,7 +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
         );
@@ -1021,7 +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);
@@ -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));

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;

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,

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;

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"
 
@@ -463,10 +462,7 @@ 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 */
@@ -756,7 +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);
 
@@ -765,9 +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;
     }

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;

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();
+}

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',