Merge tag 'nvme-next-pull-request' of https://gitlab.com/birkelund/qemu into staging
hw/nvme updates
* basic support for directives
* simple support for endurance groups
* emulation of flexible data placement (tp4146)
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# gpg: Signature made Mon 06 Mar 2023 14:34:02 GMT
# gpg: using RSA key 522833AA75E2DCE6A24766C04DE1AF316D4F0DE9
# gpg: Good signature from "Klaus Jensen <its@irrelevant.dk>" [full]
# gpg: aka "Klaus Jensen <k.jensen@samsung.com>" [full]
# Primary key fingerprint: DDCA 4D9C 9EF9 31CC 3468 4272 63D5 6FC5 E55D A838
# Subkey fingerprint: 5228 33AA 75E2 DCE6 A247 66C0 4DE1 AF31 6D4F 0DE9
* tag 'nvme-next-pull-request' of https://gitlab.com/birkelund/qemu:
hw/nvme: flexible data placement emulation
hw/nvme: basic directives support
hw/nvme: add basic endurance group support
hw/nvme: store a pointer to the NvmeSubsystem in the NvmeNamespace
hw/nvme: move adjustment of data_units{read,written}
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
diff --git a/MAINTAINERS b/MAINTAINERS
index 011fd85..da29661 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -2490,6 +2490,7 @@
----------
Overall Audio backends
M: Gerd Hoffmann <kraxel@redhat.com>
+M: Marc-André Lureau <marcandre.lureau@redhat.com>
S: Odd Fixes
F: audio/
X: audio/alsaaudio.c
@@ -2785,6 +2786,7 @@
Graphics
M: Gerd Hoffmann <kraxel@redhat.com>
+M: Marc-André Lureau <marcandre.lureau@redhat.com>
S: Odd Fixes
F: ui/
F: include/ui/
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 714bfb6..057571d 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -222,11 +222,7 @@
return -1;
}
- pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
- if (!pfds) {
- dolog ("Could not initialize poll mode\n");
- return -1;
- }
+ pfds = g_new0(struct pollfd, count);
err = snd_pcm_poll_descriptors (handle, pfds, count);
if (err < 0) {
@@ -917,28 +913,23 @@
alsa_init_per_direction(aopts->in);
alsa_init_per_direction(aopts->out);
- /*
- * need to define them, as otherwise alsa produces no sound
- * doesn't set has_* so alsa_open can identify it wasn't set by the user
- */
+ /* don't set has_* so alsa_open can identify it wasn't set by the user */
if (!dev->u.alsa.out->has_period_length) {
- /* 1024 frames assuming 44100Hz */
- dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
+ /* 256 frames assuming 44100Hz */
+ dev->u.alsa.out->period_length = 5805;
}
if (!dev->u.alsa.out->has_buffer_length) {
/* 4096 frames assuming 44100Hz */
- dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
+ dev->u.alsa.out->buffer_length = 92880;
}
- /*
- * OptsVisitor sets unspecified optional fields to zero, but do not depend
- * on it...
- */
if (!dev->u.alsa.in->has_period_length) {
- dev->u.alsa.in->period_length = 0;
+ /* 256 frames assuming 44100Hz */
+ dev->u.alsa.in->period_length = 5805;
}
if (!dev->u.alsa.in->has_buffer_length) {
- dev->u.alsa.in->buffer_length = 0;
+ /* 4096 frames assuming 44100Hz */
+ dev->u.alsa.in->buffer_length = 92880;
}
return dev;
diff --git a/audio/audio.c b/audio/audio.c
index 4290309..70b0967 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -33,6 +33,7 @@
#include "qapi/qapi-visit-audio.h"
#include "qapi/qapi-commands-audio.h"
#include "qemu/cutils.h"
+#include "qemu/log.h"
#include "qemu/module.h"
#include "qemu/help_option.h"
#include "sysemu/sysemu.h"
@@ -148,26 +149,6 @@
}
}
-void *audio_calloc (const char *funcname, int nmemb, size_t size)
-{
- int cond;
- size_t len;
-
- len = nmemb * size;
- cond = !nmemb || !size;
- cond |= nmemb < 0;
- cond |= len < size;
-
- if (audio_bug ("audio_calloc", cond)) {
- AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
- funcname);
- AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
- return NULL;
- }
-
- return g_malloc0 (len);
-}
-
void AUD_vlog (const char *cap, const char *fmt, va_list ap)
{
if (cap) {
@@ -400,13 +381,6 @@
/*
* Capture
*/
-static void noop_conv (struct st_sample *dst, const void *src, int samples)
-{
- (void) src;
- (void) dst;
- (void) samples;
-}
-
static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
struct audsettings *as)
{
@@ -504,15 +478,8 @@
sw->info = hw->info;
sw->empty = 1;
sw->active = hw->enabled;
- sw->conv = noop_conv;
- sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
sw->vol = nominal_volume;
sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
- if (!sw->rate) {
- dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
- g_free (sw);
- return -1;
- }
QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
#ifdef DEBUG_CAPTURE
@@ -547,8 +514,8 @@
static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
{
size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
- if (audio_bug(__func__, live > hw->conv_buf->size)) {
- dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
+ if (audio_bug(__func__, live > hw->conv_buf.size)) {
+ dolog("live=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size);
return 0;
}
return live;
@@ -557,13 +524,13 @@
static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples)
{
size_t conv = 0;
- STSampleBuffer *conv_buf = hw->conv_buf;
+ STSampleBuffer *conv_buf = &hw->conv_buf;
while (samples) {
uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame);
size_t proc = MIN(samples, conv_buf->size - conv_buf->pos);
- hw->conv(conv_buf->samples + conv_buf->pos, src, proc);
+ hw->conv(conv_buf->buffer + conv_buf->pos, src, proc);
conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
samples -= proc;
conv += proc;
@@ -575,56 +542,65 @@
/*
* Soft voice (capture)
*/
-static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
+static void audio_pcm_sw_resample_in(SWVoiceIn *sw,
+ size_t frames_in_max, size_t frames_out_max,
+ size_t *total_in, size_t *total_out)
{
HWVoiceIn *hw = sw->hw;
- size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
- struct st_sample *src, *dst = sw->buf;
+ struct st_sample *src, *dst;
+ size_t live, rpos, frames_in, frames_out;
+
+ live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+ rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size);
+
+ /* resample conv_buf from rpos to end of buffer */
+ src = hw->conv_buf.buffer + rpos;
+ frames_in = MIN(frames_in_max, hw->conv_buf.size - rpos);
+ dst = sw->resample_buf.buffer;
+ frames_out = frames_out_max;
+ st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
+ rpos += frames_in;
+ *total_in = frames_in;
+ *total_out = frames_out;
+
+ /* resample conv_buf from start of buffer if there are input frames left */
+ if (frames_in_max - frames_in && rpos == hw->conv_buf.size) {
+ src = hw->conv_buf.buffer;
+ frames_in = frames_in_max - frames_in;
+ dst += frames_out;
+ frames_out = frames_out_max - frames_out;
+ st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
+ *total_in += frames_in;
+ *total_out += frames_out;
+ }
+}
+
+static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len)
+{
+ HWVoiceIn *hw = sw->hw;
+ size_t live, frames_out_max, total_in, total_out;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
if (!live) {
return 0;
}
- if (audio_bug(__func__, live > hw->conv_buf->size)) {
- dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
+ if (audio_bug(__func__, live > hw->conv_buf.size)) {
+ dolog("live_in=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size);
return 0;
}
- rpos = audio_ring_posb(hw->conv_buf->pos, live, hw->conv_buf->size);
+ frames_out_max = MIN(buf_len / sw->info.bytes_per_frame,
+ sw->resample_buf.size);
- samples = size / sw->info.bytes_per_frame;
-
- swlim = (live * sw->ratio) >> 32;
- swlim = MIN (swlim, samples);
-
- while (swlim) {
- src = hw->conv_buf->samples + rpos;
- if (hw->conv_buf->pos > rpos) {
- isamp = hw->conv_buf->pos - rpos;
- } else {
- isamp = hw->conv_buf->size - rpos;
- }
-
- if (!isamp) {
- break;
- }
- osamp = swlim;
-
- st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
- swlim -= osamp;
- rpos = (rpos + isamp) % hw->conv_buf->size;
- dst += osamp;
- ret += osamp;
- total += isamp;
- }
+ audio_pcm_sw_resample_in(sw, live, frames_out_max, &total_in, &total_out);
if (!hw->pcm_ops->volume_in) {
- mixeng_volume (sw->buf, ret, &sw->vol);
+ mixeng_volume(sw->resample_buf.buffer, total_out, &sw->vol);
}
+ sw->clip(buf, sw->resample_buf.buffer, total_out);
- sw->clip (buf, sw->buf, ret);
- sw->total_hw_samples_acquired += total;
- return ret * sw->info.bytes_per_frame;
+ sw->total_hw_samples_acquired += total_in;
+ return total_out * sw->info.bytes_per_frame;
}
/*
@@ -660,8 +636,8 @@
if (nb_live1) {
size_t live = smin;
- if (audio_bug(__func__, live > hw->mix_buf->size)) {
- dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
+ if (audio_bug(__func__, live > hw->mix_buf.size)) {
+ dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size);
return 0;
}
return live;
@@ -678,17 +654,17 @@
static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
{
size_t clipped = 0;
- size_t pos = hw->mix_buf->pos;
+ size_t pos = hw->mix_buf.pos;
while (len) {
- st_sample *src = hw->mix_buf->samples + pos;
+ st_sample *src = hw->mix_buf.buffer + pos;
uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
- size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
+ size_t samples_till_end_of_buf = hw->mix_buf.size - pos;
size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
hw->clip(dst, src, samples_to_clip);
- pos = (pos + samples_to_clip) % hw->mix_buf->size;
+ pos = (pos + samples_to_clip) % hw->mix_buf.size;
len -= samples_to_clip;
clipped += samples_to_clip;
}
@@ -697,84 +673,113 @@
/*
* Soft voice (playback)
*/
-static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
+static void audio_pcm_sw_resample_out(SWVoiceOut *sw,
+ size_t frames_in_max, size_t frames_out_max,
+ size_t *total_in, size_t *total_out)
{
- size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, blck;
- size_t hw_free;
- size_t ret = 0, pos = 0, total = 0;
-
- if (!sw) {
- return size;
- }
-
- hwsamples = sw->hw->mix_buf->size;
+ HWVoiceOut *hw = sw->hw;
+ struct st_sample *src, *dst;
+ size_t live, wpos, frames_in, frames_out;
live = sw->total_hw_samples_mixed;
- if (audio_bug(__func__, live > hwsamples)) {
- dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
+ wpos = (hw->mix_buf.pos + live) % hw->mix_buf.size;
+
+ /* write to mix_buf from wpos to end of buffer */
+ src = sw->resample_buf.buffer;
+ frames_in = frames_in_max;
+ dst = hw->mix_buf.buffer + wpos;
+ frames_out = MIN(frames_out_max, hw->mix_buf.size - wpos);
+ st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
+ wpos += frames_out;
+ *total_in = frames_in;
+ *total_out = frames_out;
+
+ /* write to mix_buf from start of buffer if there are input frames left */
+ if (frames_in_max - frames_in > 0 && wpos == hw->mix_buf.size) {
+ src += frames_in;
+ frames_in = frames_in_max - frames_in;
+ dst = hw->mix_buf.buffer;
+ frames_out = frames_out_max - frames_out;
+ st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
+ *total_in += frames_in;
+ *total_out += frames_out;
+ }
+}
+
+static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
+{
+ HWVoiceOut *hw = sw->hw;
+ size_t live, dead, hw_free, sw_max, fe_max;
+ size_t frames_in_max, frames_out_max, total_in, total_out;
+
+ live = sw->total_hw_samples_mixed;
+ if (audio_bug(__func__, live > hw->mix_buf.size)) {
+ dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size);
return 0;
}
- if (live == hwsamples) {
+ if (live == hw->mix_buf.size) {
#ifdef DEBUG_OUT
dolog ("%s is full %zu\n", sw->name, live);
#endif
return 0;
}
- wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
-
- dead = hwsamples - live;
- hw_free = audio_pcm_hw_get_free(sw->hw);
+ dead = hw->mix_buf.size - live;
+ hw_free = audio_pcm_hw_get_free(hw);
hw_free = hw_free > live ? hw_free - live : 0;
- samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
- samples = MIN(samples, size / sw->info.bytes_per_frame);
- if (samples) {
- sw->conv(sw->buf, buf, samples);
+ frames_out_max = MIN(dead, hw_free);
+ sw_max = st_rate_frames_in(sw->rate, frames_out_max);
+ fe_max = MIN(buf_len / sw->info.bytes_per_frame + sw->resample_buf.pos,
+ sw->resample_buf.size);
+ frames_in_max = MIN(sw_max, fe_max);
+ if (!frames_in_max) {
+ return 0;
+ }
+
+ if (frames_in_max > sw->resample_buf.pos) {
+ sw->conv(sw->resample_buf.buffer + sw->resample_buf.pos,
+ buf, frames_in_max - sw->resample_buf.pos);
if (!sw->hw->pcm_ops->volume_out) {
- mixeng_volume(sw->buf, samples, &sw->vol);
+ mixeng_volume(sw->resample_buf.buffer + sw->resample_buf.pos,
+ frames_in_max - sw->resample_buf.pos, &sw->vol);
}
}
- while (samples) {
- dead = hwsamples - live;
- left = hwsamples - wpos;
- blck = MIN (dead, left);
- if (!blck) {
- break;
- }
- isamp = samples;
- osamp = blck;
- st_rate_flow_mix (
- sw->rate,
- sw->buf + pos,
- sw->hw->mix_buf->samples + wpos,
- &isamp,
- &osamp
- );
- ret += isamp;
- samples -= isamp;
- pos += isamp;
- live += osamp;
- wpos = (wpos + osamp) % hwsamples;
- total += osamp;
- }
+ audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max,
+ &total_in, &total_out);
- sw->total_hw_samples_mixed += total;
+ sw->total_hw_samples_mixed += total_out;
sw->empty = sw->total_hw_samples_mixed == 0;
+ /*
+ * Upsampling may leave one audio frame in the resample buffer. Decrement
+ * total_in by one if there was a leftover frame from the previous resample
+ * pass in the resample buffer. Increment total_in by one if the current
+ * resample pass left one frame in the resample buffer.
+ */
+ if (frames_in_max - total_in == 1) {
+ /* copy one leftover audio frame to the beginning of the buffer */
+ *sw->resample_buf.buffer = *(sw->resample_buf.buffer + total_in);
+ total_in += 1 - sw->resample_buf.pos;
+ sw->resample_buf.pos = 1;
+ } else if (total_in >= sw->resample_buf.pos) {
+ total_in -= sw->resample_buf.pos;
+ sw->resample_buf.pos = 0;
+ }
+
#ifdef DEBUG_OUT
dolog (
- "%s: write size %zu ret %zu total sw %zu\n",
- SW_NAME (sw),
- size / sw->info.bytes_per_frame,
- ret,
+ "%s: write size %zu written %zu total mixed %zu\n",
+ SW_NAME(sw),
+ buf_len / sw->info.bytes_per_frame,
+ total_in,
sw->total_hw_samples_mixed
);
#endif
- return ret * sw->info.bytes_per_frame;
+ return total_in * sw->info.bytes_per_frame;
}
#ifdef DEBUG_AUDIO
@@ -992,18 +997,6 @@
}
}
-/**
- * audio_frontend_frames_in() - returns the number of frames the resampling
- * code generates from frames_in frames
- *
- * @sw: audio recording frontend
- * @frames_in: number of frames
- */
-static size_t audio_frontend_frames_in(SWVoiceIn *sw, size_t frames_in)
-{
- return (int64_t)frames_in * sw->ratio >> 32;
-}
-
static size_t audio_get_avail (SWVoiceIn *sw)
{
size_t live;
@@ -1013,33 +1006,21 @@
}
live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
- dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
- sw->hw->conv_buf->size);
+ if (audio_bug(__func__, live > sw->hw->conv_buf.size)) {
+ dolog("live=%zu sw->hw->conv_buf.size=%zu\n", live,
+ sw->hw->conv_buf.size);
return 0;
}
ldebug (
- "%s: get_avail live %zu frontend frames %zu\n",
+ "%s: get_avail live %zu frontend frames %u\n",
SW_NAME (sw),
- live, audio_frontend_frames_in(sw, live)
+ live, st_rate_frames_out(sw->rate, live)
);
return live;
}
-/**
- * audio_frontend_frames_out() - returns the number of frames needed to
- * get frames_out frames after resampling
- *
- * @sw: audio playback frontend
- * @frames_out: number of frames
- */
-static size_t audio_frontend_frames_out(SWVoiceOut *sw, size_t frames_out)
-{
- return ((int64_t)frames_out << 32) / sw->ratio;
-}
-
static size_t audio_get_free(SWVoiceOut *sw)
{
size_t live, dead;
@@ -1050,17 +1031,17 @@
live = sw->total_hw_samples_mixed;
- if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
- dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
- sw->hw->mix_buf->size);
+ if (audio_bug(__func__, live > sw->hw->mix_buf.size)) {
+ dolog("live=%zu sw->hw->mix_buf.size=%zu\n", live,
+ sw->hw->mix_buf.size);
return 0;
}
- dead = sw->hw->mix_buf->size - live;
+ dead = sw->hw->mix_buf.size - live;
#ifdef DEBUG_OUT
- dolog("%s: get_free live %zu dead %zu frontend frames %zu\n",
- SW_NAME(sw), live, dead, audio_frontend_frames_out(sw, dead));
+ dolog("%s: get_free live %zu dead %zu frontend frames %u\n",
+ SW_NAME(sw), live, dead, st_rate_frames_in(sw->rate, dead));
#endif
return dead;
@@ -1076,32 +1057,40 @@
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
SWVoiceOut *sw = &sc->sw;
- int rpos2 = rpos;
+ size_t rpos2 = rpos;
n = samples;
while (n) {
- size_t till_end_of_hw = hw->mix_buf->size - rpos2;
- size_t to_write = MIN(till_end_of_hw, n);
- size_t bytes = to_write * hw->info.bytes_per_frame;
- size_t written;
+ size_t till_end_of_hw = hw->mix_buf.size - rpos2;
+ size_t to_read = MIN(till_end_of_hw, n);
+ size_t live, frames_in, frames_out;
- sw->buf = hw->mix_buf->samples + rpos2;
- written = audio_pcm_sw_write (sw, NULL, bytes);
- if (written - bytes) {
- dolog("Could not mix %zu bytes into a capture "
+ sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;
+ sw->resample_buf.size = to_read;
+ live = sw->total_hw_samples_mixed;
+
+ audio_pcm_sw_resample_out(sw,
+ to_read, sw->hw->mix_buf.size - live,
+ &frames_in, &frames_out);
+
+ sw->total_hw_samples_mixed += frames_out;
+ sw->empty = sw->total_hw_samples_mixed == 0;
+
+ if (to_read - frames_in) {
+ dolog("Could not mix %zu frames into a capture "
"buffer, mixed %zu\n",
- bytes, written);
+ to_read, frames_in);
break;
}
- n -= to_write;
- rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
+ n -= to_read;
+ rpos2 = (rpos2 + to_read) % hw->mix_buf.size;
}
}
}
- n = MIN(samples, hw->mix_buf->size - rpos);
- mixeng_clear(hw->mix_buf->samples + rpos, n);
- mixeng_clear(hw->mix_buf->samples, samples - n);
+ n = MIN(samples, hw->mix_buf.size - rpos);
+ mixeng_clear(hw->mix_buf.buffer + rpos, n);
+ mixeng_clear(hw->mix_buf.buffer, samples - n);
}
static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
@@ -1127,7 +1116,7 @@
live -= proc;
clipped += proc;
- hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
+ hw->mix_buf.pos = (hw->mix_buf.pos + proc) % hw->mix_buf.size;
if (proc == 0 || proc < decr) {
break;
@@ -1181,12 +1170,14 @@
size_t free;
if (hw_free > sw->total_hw_samples_mixed) {
- free = audio_frontend_frames_out(sw,
+ free = st_rate_frames_in(sw->rate,
MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
} else {
free = 0;
}
- if (free > 0) {
+ if (free > sw->resample_buf.pos) {
+ free = MIN(free, sw->resample_buf.size)
+ - sw->resample_buf.pos;
sw->callback.fn(sw->callback.opaque,
free * sw->info.bytes_per_frame);
}
@@ -1198,8 +1189,8 @@
live = 0;
}
- if (audio_bug(__func__, live > hw->mix_buf->size)) {
- dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
+ if (audio_bug(__func__, live > hw->mix_buf.size)) {
+ dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size);
continue;
}
@@ -1227,13 +1218,13 @@
continue;
}
- prev_rpos = hw->mix_buf->pos;
+ prev_rpos = hw->mix_buf.pos;
played = audio_pcm_hw_run_out(hw, live);
replay_audio_out(&played);
- if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
- dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
- hw->mix_buf->pos, hw->mix_buf->size, played);
- hw->mix_buf->pos = 0;
+ if (audio_bug(__func__, hw->mix_buf.pos >= hw->mix_buf.size)) {
+ dolog("hw->mix_buf.pos=%zu hw->mix_buf.size=%zu played=%zu\n",
+ hw->mix_buf.pos, hw->mix_buf.size, played);
+ hw->mix_buf.pos = 0;
}
#ifdef DEBUG_OUT
@@ -1314,10 +1305,10 @@
if (replay_mode != REPLAY_MODE_PLAY) {
captured = audio_pcm_hw_run_in(
- hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
+ hw, hw->conv_buf.size - audio_pcm_hw_get_live_in(hw));
}
- replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
- hw->conv_buf->size);
+ replay_audio_in(&captured, hw->conv_buf.buffer, &hw->conv_buf.pos,
+ hw->conv_buf.size);
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
@@ -1330,8 +1321,9 @@
size_t sw_avail = audio_get_avail(sw);
size_t avail;
- avail = audio_frontend_frames_in(sw, sw_avail);
+ avail = st_rate_frames_out(sw->rate, sw_avail);
if (avail > 0) {
+ avail = MIN(avail, sw->resample_buf.size);
sw->callback.fn(sw->callback.opaque,
avail * sw->info.bytes_per_frame);
}
@@ -1350,14 +1342,14 @@
SWVoiceOut *sw;
captured = live = audio_pcm_hw_get_live_out (hw, NULL);
- rpos = hw->mix_buf->pos;
+ rpos = hw->mix_buf.pos;
while (live) {
- size_t left = hw->mix_buf->size - rpos;
+ size_t left = hw->mix_buf.size - rpos;
size_t to_capture = MIN(live, left);
struct st_sample *src;
struct capture_callback *cb;
- src = hw->mix_buf->samples + rpos;
+ src = hw->mix_buf.buffer + rpos;
hw->clip (cap->buf, src, to_capture);
mixeng_clear (src, to_capture);
@@ -1365,10 +1357,10 @@
cb->ops.capture (cb->opaque, cap->buf,
to_capture * hw->info.bytes_per_frame);
}
- rpos = (rpos + to_capture) % hw->mix_buf->size;
+ rpos = (rpos + to_capture) % hw->mix_buf.size;
live -= to_capture;
}
- hw->mix_buf->pos = rpos;
+ hw->mix_buf.pos = rpos;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (!sw->active && sw->empty) {
@@ -1927,7 +1919,7 @@
audio_pcm_init_info (&hw->info, as);
- cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
+ cap->buf = g_malloc0_n(hw->mix_buf.size, hw->info.bytes_per_frame);
if (hw->info.is_float) {
hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
@@ -1979,7 +1971,7 @@
sw = sw1;
}
QLIST_REMOVE (cap, entries);
- g_free (cap->hw.mix_buf);
+ g_free(cap->hw.mix_buf.buffer);
g_free (cap->buf);
g_free (cap);
}
diff --git a/audio/audio_int.h b/audio/audio_int.h
index e87ce01..d51d63f 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -58,7 +58,7 @@
typedef struct STSampleBuffer {
size_t pos, size;
- st_sample samples[];
+ st_sample *buffer;
} STSampleBuffer;
typedef struct HWVoiceOut {
@@ -71,7 +71,7 @@
f_sample *clip;
uint64_t ts_helper;
- STSampleBuffer *mix_buf;
+ STSampleBuffer mix_buf;
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
@@ -93,7 +93,7 @@
size_t total_samples_captured;
uint64_t ts_helper;
- STSampleBuffer *conv_buf;
+ STSampleBuffer conv_buf;
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
@@ -108,8 +108,7 @@
AudioState *s;
struct audio_pcm_info info;
t_sample *conv;
- int64_t ratio;
- struct st_sample *buf;
+ STSampleBuffer resample_buf;
void *rate;
size_t total_hw_samples_mixed;
int active;
@@ -126,10 +125,9 @@
AudioState *s;
int active;
struct audio_pcm_info info;
- int64_t ratio;
void *rate;
size_t total_hw_samples_acquired;
- struct st_sample *buf;
+ STSampleBuffer resample_buf;
f_sample *clip;
HWVoiceIn *hw;
char *name;
@@ -151,8 +149,8 @@
int can_be_default;
int max_voices_out;
int max_voices_in;
- int voice_size_out;
- int voice_size_in;
+ size_t voice_size_out;
+ size_t voice_size_in;
QLIST_ENTRY(audio_driver) next;
};
@@ -251,7 +249,6 @@
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
int audio_bug (const char *funcname, int cond);
-void *audio_calloc (const char *funcname, int nmemb, size_t size);
void audio_run(AudioState *s, const char *msg);
@@ -294,9 +291,6 @@
#define ldebug(fmt, ...) (void)0
#endif
-#define AUDIO_STRINGIFY_(n) #n
-#define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n)
-
typedef struct AudiodevListEntry {
Audiodev *dev;
QSIMPLEQ_ENTRY(AudiodevListEntry) next;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 42b4712..e42326c 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -40,7 +40,7 @@
struct audio_driver *drv)
{
int max_voices = glue (drv->max_voices_, TYPE);
- int voice_size = glue (drv->voice_size_, TYPE);
+ size_t voice_size = glue(drv->voice_size_, TYPE);
if (glue (s->nb_hw_voices_, TYPE) > max_voices) {
if (!max_voices) {
@@ -63,16 +63,17 @@
}
if (audio_bug(__func__, voice_size && !max_voices)) {
- dolog ("drv=`%s' voice_size=%d max_voices=0\n",
- drv->name, voice_size);
+ dolog("drv=`%s' voice_size=%zu max_voices=0\n",
+ drv->name, voice_size);
}
}
static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
{
g_free(hw->buf_emul);
- g_free (HWBUF);
- HWBUF = NULL;
+ g_free(HWBUF.buffer);
+ HWBUF.buffer = NULL;
+ HWBUF.size = 0;
}
static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
@@ -83,56 +84,67 @@
dolog("Attempted to allocate empty buffer\n");
}
- HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
- HWBUF->size = samples;
+ HWBUF.buffer = g_new0(st_sample, samples);
+ HWBUF.size = samples;
+ HWBUF.pos = 0;
} else {
- HWBUF = NULL;
+ HWBUF.buffer = NULL;
+ HWBUF.size = 0;
}
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
{
- g_free (sw->buf);
+ g_free(sw->resample_buf.buffer);
+ sw->resample_buf.buffer = NULL;
+ sw->resample_buf.size = 0;
if (sw->rate) {
st_rate_stop (sw->rate);
}
-
- sw->buf = NULL;
sw->rate = NULL;
}
static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
{
- int samples;
+ HW *hw = sw->hw;
+ uint64_t samples;
if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
return 0;
}
-#ifdef DAC
- samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
-#else
- samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32;
-#endif
+ samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq);
+ if (samples == 0) {
+ uint64_t f_fe_min;
+ uint64_t f_be = (uint32_t)hw->info.freq;
- sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample));
- if (!sw->buf) {
- dolog ("Could not allocate buffer for `%s' (%d samples)\n",
- SW_NAME (sw), samples);
+ /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
+ f_fe_min = (f_be + HWBUF.size - 1) / HWBUF.size;
+ qemu_log_mask(LOG_UNIMP,
+ AUDIO_CAP ": The guest selected a " NAME " sample rate"
+ " of %d Hz for %s. Only sample rates >= %" PRIu64 " Hz"
+ " are supported.\n",
+ sw->info.freq, sw->name, f_fe_min);
return -1;
}
+ /*
+ * Allocate one additional audio frame that is needed for upsampling
+ * if the resample buffer size is small. For large buffer sizes take
+ * care of overflows and truncation.
+ */
+ samples = samples < SIZE_MAX ? samples + 1 : SIZE_MAX;
+ sw->resample_buf.buffer = g_new0(st_sample, samples);
+ sw->resample_buf.size = samples;
+ sw->resample_buf.pos = 0;
+
#ifdef DAC
- sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);
+ sw->rate = st_rate_start(sw->info.freq, hw->info.freq);
#else
- sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq);
+ sw->rate = st_rate_start(hw->info.freq, sw->info.freq);
#endif
- if (!sw->rate) {
- g_free (sw->buf);
- sw->buf = NULL;
- return -1;
- }
+
return 0;
}
@@ -149,11 +161,8 @@
sw->hw = hw;
sw->active = 0;
#ifdef DAC
- sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq;
sw->total_hw_samples_mixed = 0;
sw->empty = 1;
-#else
- sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
#endif
if (sw->info.is_float) {
@@ -264,13 +273,11 @@
return NULL;
}
- hw = audio_calloc(__func__, 1, glue(drv->voice_size_, TYPE));
- if (!hw) {
- dolog ("Can not allocate voice `%s' size %d\n",
- drv->name, glue (drv->voice_size_, TYPE));
- return NULL;
- }
-
+ /*
+ * Since glue(s->nb_hw_voices_, TYPE) is != 0, glue(drv->voice_size_, TYPE)
+ * is guaranteed to be != 0. See the audio_init_nb_voices_* functions.
+ */
+ hw = g_malloc0(glue(drv->voice_size_, TYPE));
hw->s = s;
hw->pcm_ops = drv->pcm_ops;
@@ -418,33 +425,28 @@
hw_as = *as;
}
- sw = audio_calloc(__func__, 1, sizeof(*sw));
- if (!sw) {
- dolog ("Could not allocate soft voice `%s' (%zu bytes)\n",
- sw_name ? sw_name : "unknown", sizeof (*sw));
- goto err1;
- }
+ sw = g_new0(SW, 1);
sw->s = s;
hw = glue(audio_pcm_hw_add_, TYPE)(s, &hw_as);
if (!hw) {
- goto err2;
+ dolog("Could not create a backend for voice `%s'\n", sw_name);
+ goto err1;
}
glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as)) {
- goto err3;
+ goto err2;
}
return sw;
-err3:
+err2:
glue (audio_pcm_hw_del_sw_, TYPE) (sw);
glue (audio_pcm_hw_gc_, TYPE) (&hw);
-err2:
- g_free (sw);
err1:
+ g_free(sw);
return NULL;
}
@@ -515,8 +517,8 @@
HW *hw = sw->hw;
if (!hw) {
- dolog ("Internal logic error voice `%s' has no hardware store\n",
- SW_NAME (sw));
+ dolog("Internal logic error: voice `%s' has no backend\n",
+ SW_NAME(sw));
goto fail;
}
@@ -527,7 +529,6 @@
} else {
sw = glue(audio_pcm_create_voice_pair_, TYPE)(s, name, as);
if (!sw) {
- dolog ("Failed to create voice `%s'\n", name);
return NULL;
}
}
diff --git a/audio/mixeng.c b/audio/mixeng.c
index 100a306..69f6549 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -414,12 +414,7 @@
*/
void *st_rate_start (int inrate, int outrate)
{
- struct rate *rate = audio_calloc(__func__, 1, sizeof(*rate));
-
- if (!rate) {
- dolog ("Could not allocate resampler (%zu bytes)\n", sizeof (*rate));
- return NULL;
- }
+ struct rate *rate = g_new0(struct rate, 1);
rate->opos = 0;
@@ -445,6 +440,86 @@
g_free (opaque);
}
+/**
+ * st_rate_frames_out() - returns the number of frames the resampling code
+ * generates from frames_in frames
+ *
+ * @opaque: pointer to struct rate
+ * @frames_in: number of frames
+ *
+ * When upsampling, there may be more than one correct result. In this case,
+ * the function returns the maximum number of output frames the resampling
+ * code can generate.
+ */
+uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in)
+{
+ struct rate *rate = opaque;
+ uint64_t opos_end, opos_delta;
+ uint32_t ipos_end;
+ uint32_t frames_out;
+
+ if (rate->opos_inc == 1ULL << 32) {
+ return frames_in;
+ }
+
+ /* no output frame without at least one input frame */
+ if (!frames_in) {
+ return 0;
+ }
+
+ /* last frame read was at rate->ipos - 1 */
+ ipos_end = rate->ipos - 1 + frames_in;
+ opos_end = (uint64_t)ipos_end << 32;
+
+ /* last frame written was at rate->opos - rate->opos_inc */
+ if (opos_end + rate->opos_inc <= rate->opos) {
+ return 0;
+ }
+ opos_delta = opos_end - rate->opos + rate->opos_inc;
+ frames_out = opos_delta / rate->opos_inc;
+
+ return opos_delta % rate->opos_inc ? frames_out : frames_out - 1;
+}
+
+/**
+ * st_rate_frames_in() - returns the number of frames needed to
+ * get frames_out frames after resampling
+ *
+ * @opaque: pointer to struct rate
+ * @frames_out: number of frames
+ *
+ * When downsampling, there may be more than one correct result. In this
+ * case, the function returns the maximum number of input frames needed.
+ */
+uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out)
+{
+ struct rate *rate = opaque;
+ uint64_t opos_start, opos_end;
+ uint32_t ipos_start, ipos_end;
+
+ if (rate->opos_inc == 1ULL << 32) {
+ return frames_out;
+ }
+
+ if (frames_out) {
+ opos_start = rate->opos;
+ ipos_start = rate->ipos;
+ } else {
+ uint64_t offset;
+
+ /* add offset = ceil(opos_inc) to opos and ipos to avoid an underflow */
+ offset = (rate->opos_inc + (1ULL << 32) - 1) & ~((1ULL << 32) - 1);
+ opos_start = rate->opos + offset;
+ ipos_start = rate->ipos + (offset >> 32);
+ }
+ /* last frame written was at opos_start - rate->opos_inc */
+ opos_end = opos_start - rate->opos_inc + rate->opos_inc * frames_out;
+ ipos_end = (opos_end >> 32) + 1;
+
+ /* last frame read was at ipos_start - 1 */
+ return ipos_end + 1 > ipos_start ? ipos_end + 1 - ipos_start : 0;
+}
+
void mixeng_clear (struct st_sample *buf, int len)
{
memset (buf, 0, len * sizeof (struct st_sample));
diff --git a/audio/mixeng.h b/audio/mixeng.h
index 2dcd6df..f9de7cf 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -52,6 +52,8 @@
void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf,
size_t *isamp, size_t *osamp);
void st_rate_stop (void *opaque);
+uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in);
+uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out);
void mixeng_clear (struct st_sample *buf, int len);
void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol);
diff --git a/audio/rate_template.h b/audio/rate_template.h
index b432719..6648f0d 100644
--- a/audio/rate_template.h
+++ b/audio/rate_template.h
@@ -40,8 +40,6 @@
int64_t t;
#endif
- ilast = rate->ilast;
-
istart = ibuf;
iend = ibuf + *isamp;
@@ -59,15 +57,17 @@
return;
}
- while (obuf < oend) {
+ /* without input samples, there's nothing to do */
+ if (ibuf >= iend) {
+ *osamp = 0;
+ return;
+ }
- /* Safety catch to make sure we have input samples. */
- if (ibuf >= iend) {
- break;
- }
+ ilast = rate->ilast;
+
+ while (true) {
/* read as many input samples so that ipos > opos */
-
while (rate->ipos <= (rate->opos >> 32)) {
ilast = *ibuf++;
rate->ipos++;
@@ -78,6 +78,11 @@
}
}
+ /* make sure that the next output sample can be written */
+ if (obuf >= oend) {
+ break;
+ }
+
icur = *ibuf;
/* wrap ipos and opos around long before they overflow */
diff --git a/hw/i386/pc_piix.c b/hw/i386/pc_piix.c
index 2f16011..4bf15f9 100644
--- a/hw/i386/pc_piix.c
+++ b/hw/i386/pc_piix.c
@@ -422,6 +422,7 @@
}
pc_xen_hvm_init_pci(machine);
+ xen_igd_reserve_slot(pcms->bus);
pci_create_simple(pcms->bus, -1, "xen-platform");
}
#endif
diff --git a/hw/xen/xen_pt.c b/hw/xen/xen_pt.c
index 8db0532..85c93cf 100644
--- a/hw/xen/xen_pt.c
+++ b/hw/xen/xen_pt.c
@@ -57,6 +57,7 @@
#include <sys/ioctl.h>
#include "hw/pci/pci.h"
+#include "hw/pci/pci_bus.h"
#include "hw/qdev-properties.h"
#include "hw/qdev-properties-system.h"
#include "hw/xen/xen.h"
@@ -780,15 +781,6 @@
s->hostaddr.bus, s->hostaddr.slot, s->hostaddr.function,
s->dev.devfn);
- xen_host_pci_device_get(&s->real_device,
- s->hostaddr.domain, s->hostaddr.bus,
- s->hostaddr.slot, s->hostaddr.function,
- errp);
- if (*errp) {
- error_append_hint(errp, "Failed to \"open\" the real pci device");
- return;
- }
-
s->is_virtfn = s->real_device.is_virtfn;
if (s->is_virtfn) {
XEN_PT_LOG(d, "%04x:%02x:%02x.%d is a SR-IOV Virtual Function\n",
@@ -803,8 +795,10 @@
s->io_listener = xen_pt_io_listener;
/* Setup VGA bios for passthrough GFX */
- if ((s->real_device.domain == 0) && (s->real_device.bus == 0) &&
- (s->real_device.dev == 2) && (s->real_device.func == 0)) {
+ if ((s->real_device.domain == XEN_PCI_IGD_DOMAIN) &&
+ (s->real_device.bus == XEN_PCI_IGD_BUS) &&
+ (s->real_device.dev == XEN_PCI_IGD_DEV) &&
+ (s->real_device.func == XEN_PCI_IGD_FN)) {
if (!is_igd_vga_passthrough(&s->real_device)) {
error_setg(errp, "Need to enable igd-passthru if you're trying"
" to passthrough IGD GFX");
@@ -950,11 +944,58 @@
PCI_DEVICE(obj)->cap_present |= QEMU_PCI_CAP_EXPRESS;
}
+void xen_igd_reserve_slot(PCIBus *pci_bus)
+{
+ if (!xen_igd_gfx_pt_enabled()) {
+ return;
+ }
+
+ XEN_PT_LOG(0, "Reserving PCI slot 2 for IGD\n");
+ pci_bus->slot_reserved_mask |= XEN_PCI_IGD_SLOT_MASK;
+}
+
+static void xen_igd_clear_slot(DeviceState *qdev, Error **errp)
+{
+ ERRP_GUARD();
+ PCIDevice *pci_dev = (PCIDevice *)qdev;
+ XenPCIPassthroughState *s = XEN_PT_DEVICE(pci_dev);
+ XenPTDeviceClass *xpdc = XEN_PT_DEVICE_GET_CLASS(s);
+ PCIBus *pci_bus = pci_get_bus(pci_dev);
+
+ xen_host_pci_device_get(&s->real_device,
+ s->hostaddr.domain, s->hostaddr.bus,
+ s->hostaddr.slot, s->hostaddr.function,
+ errp);
+ if (*errp) {
+ error_append_hint(errp, "Failed to \"open\" the real pci device");
+ return;
+ }
+
+ if (!(pci_bus->slot_reserved_mask & XEN_PCI_IGD_SLOT_MASK)) {
+ xpdc->pci_qdev_realize(qdev, errp);
+ return;
+ }
+
+ if (is_igd_vga_passthrough(&s->real_device) &&
+ s->real_device.domain == XEN_PCI_IGD_DOMAIN &&
+ s->real_device.bus == XEN_PCI_IGD_BUS &&
+ s->real_device.dev == XEN_PCI_IGD_DEV &&
+ s->real_device.func == XEN_PCI_IGD_FN &&
+ s->real_device.vendor_id == PCI_VENDOR_ID_INTEL) {
+ pci_bus->slot_reserved_mask &= ~XEN_PCI_IGD_SLOT_MASK;
+ XEN_PT_LOG(pci_dev, "Intel IGD found, using slot 2\n");
+ }
+ xpdc->pci_qdev_realize(qdev, errp);
+}
+
static void xen_pci_passthrough_class_init(ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS(klass);
PCIDeviceClass *k = PCI_DEVICE_CLASS(klass);
+ XenPTDeviceClass *xpdc = XEN_PT_DEVICE_CLASS(klass);
+ xpdc->pci_qdev_realize = dc->realize;
+ dc->realize = xen_igd_clear_slot;
k->realize = xen_pt_realize;
k->exit = xen_pt_unregister_device;
k->config_read = xen_pt_pci_read_config;
@@ -977,6 +1018,7 @@
.instance_size = sizeof(XenPCIPassthroughState),
.instance_finalize = xen_pci_passthrough_finalize,
.class_init = xen_pci_passthrough_class_init,
+ .class_size = sizeof(XenPTDeviceClass),
.instance_init = xen_pci_passthrough_instance_init,
.interfaces = (InterfaceInfo[]) {
{ INTERFACE_CONVENTIONAL_PCI_DEVICE },
diff --git a/hw/xen/xen_pt.h b/hw/xen/xen_pt.h
index cf10fc7..e184699 100644
--- a/hw/xen/xen_pt.h
+++ b/hw/xen/xen_pt.h
@@ -40,7 +40,20 @@
#define TYPE_XEN_PT_DEVICE "xen-pci-passthrough"
OBJECT_DECLARE_SIMPLE_TYPE(XenPCIPassthroughState, XEN_PT_DEVICE)
+#define XEN_PT_DEVICE_CLASS(klass) \
+ OBJECT_CLASS_CHECK(XenPTDeviceClass, klass, TYPE_XEN_PT_DEVICE)
+#define XEN_PT_DEVICE_GET_CLASS(obj) \
+ OBJECT_GET_CLASS(XenPTDeviceClass, obj, TYPE_XEN_PT_DEVICE)
+
+typedef void (*XenPTQdevRealize)(DeviceState *qdev, Error **errp);
+
+typedef struct XenPTDeviceClass {
+ PCIDeviceClass parent_class;
+ XenPTQdevRealize pci_qdev_realize;
+} XenPTDeviceClass;
+
uint32_t igd_read_opregion(XenPCIPassthroughState *s);
+void xen_igd_reserve_slot(PCIBus *pci_bus);
void igd_write_opregion(XenPCIPassthroughState *s, uint32_t val);
void xen_igd_passthrough_isa_bridge_create(XenPCIPassthroughState *s,
XenHostPCIDevice *dev);
@@ -75,6 +88,13 @@
#define XEN_PCI_INTEL_OPREGION 0xfc
+#define XEN_PCI_IGD_DOMAIN 0
+#define XEN_PCI_IGD_BUS 0
+#define XEN_PCI_IGD_DEV 2
+#define XEN_PCI_IGD_FN 0
+#define XEN_PCI_IGD_SLOT_MASK \
+ (1UL << PCI_SLOT(PCI_DEVFN(XEN_PCI_IGD_DEV, XEN_PCI_IGD_FN)))
+
typedef enum {
XEN_PT_GRP_TYPE_HARDWIRED = 0, /* 0 Hardwired reg group */
XEN_PT_GRP_TYPE_EMU, /* emul reg group */
diff --git a/hw/xen/xen_pt_config_init.c b/hw/xen/xen_pt_config_init.c
index cde898b..8b9b554 100644
--- a/hw/xen/xen_pt_config_init.c
+++ b/hw/xen/xen_pt_config_init.c
@@ -1924,7 +1924,7 @@
if (reg->init) {
uint32_t host_mask, size_mask;
unsigned int offset;
- uint32_t val;
+ uint32_t val = 0;
/* initialize emulate register */
rc = reg->init(s, reg_entry->reg,
diff --git a/hw/xen/xen_pt_stub.c b/hw/xen/xen_pt_stub.c
index 2d8cac8..5c10844 100644
--- a/hw/xen/xen_pt_stub.c
+++ b/hw/xen/xen_pt_stub.c
@@ -20,3 +20,7 @@
error_setg(errp, "Xen PCI passthrough support not built in");
}
}
+
+void xen_igd_reserve_slot(PCIBus *pci_bus)
+{
+}
