From df016a1998147ec328d7e800a7f7582e053720d1 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 4 Jan 2020 10:11:18 +0100 Subject: [PATCH 1/6] hda-codec: fix playback rate control MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since commit 1930616b98 "audio: make mixeng optional" the function hda_audio_output_cb can no longer assume the function parameter avail contains the free buffer size. With the playback mixing-engine turned off this leads to a broken playback rate control and playback buffer drops in regular intervals. This patch moves down the rate calculation, so the correct buffer fill level is used for the calculation. Signed-off-by: Volker Rümelin Message-id: 20200104091122.13971-1-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann --- hw/audio/hda-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c index f17e8d8dce..768ba31e79 100644 --- a/hw/audio/hda-codec.c +++ b/hw/audio/hda-codec.c @@ -338,8 +338,6 @@ static void hda_audio_output_cb(void *opaque, int avail) return; } - hda_timer_sync_adjust(st, (wpos - rpos) - to_transfer - (B_SIZE >> 1)); - while (to_transfer) { uint32_t start = (uint32_t) (rpos & B_MASK); uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer); @@ -351,6 +349,8 @@ static void hda_audio_output_cb(void *opaque, int avail) break; } } + + hda_timer_sync_adjust(st, (wpos - rpos) - (B_SIZE >> 1)); } static void hda_audio_compat_input_cb(void *opaque, int avail) From c435fea72bbae2c37e4ae375cbecfee0d4a5470c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 4 Jan 2020 10:11:19 +0100 Subject: [PATCH 2/6] hda-codec: fix recording rate control MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Apply previous commit to hda_audio_input_cb for the same reasons. Signed-off-by: Volker Rümelin Message-id: 20200104091122.13971-2-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann --- hw/audio/hda-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c index 768ba31e79..e711a99a41 100644 --- a/hw/audio/hda-codec.c +++ b/hw/audio/hda-codec.c @@ -265,8 +265,6 @@ static void hda_audio_input_cb(void *opaque, int avail) int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), avail); - hda_timer_sync_adjust(st, -((wpos - rpos) + to_transfer - (B_SIZE >> 1))); - while (to_transfer) { uint32_t start = (uint32_t) (wpos & B_MASK); uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer); @@ -278,6 +276,8 @@ static void hda_audio_input_cb(void *opaque, int avail) break; } } + + hda_timer_sync_adjust(st, -((wpos - rpos) - (B_SIZE >> 1))); } static void hda_audio_output_timer(void *opaque) From 4db3e634c77a671fadbba6d4d39e0d21232e5609 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 4 Jan 2020 10:11:20 +0100 Subject: [PATCH 3/6] paaudio: drop recording stream in qpa_fini_in MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Every call to pa_stream_peek which returns a data length > 0 should have a corresponding pa_stream_drop. A call to qpa_read does not necessarily call pa_stream_drop immediately after a call to pa_stream_peek. Test in qpa_fini_in if a last pa_stream_drop is needed. This prevents following messages in the libvirt log file after a recording stream gets closed and a new one opened. pulseaudio: pa_stream_drop failed pulseaudio: Reason: Bad state pulseaudio: pa_stream_drop failed pulseaudio: Reason: Bad state To reproduce start qemu with -audiodev pa,id=audio0,in.mixing-engine=off and in the guest start and stop Audacity several times. Signed-off-by: Volker Rümelin Message-id: 20200104091122.13971-3-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann --- audio/paaudio.c | 22 ++++++++++++++++++---- 1 file changed, 18 insertions(+), 4 deletions(-) diff --git a/audio/paaudio.c b/audio/paaudio.c index 55a91f8980..7db1dc15f0 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -536,7 +536,6 @@ static void qpa_simple_disconnect(PAConnection *c, pa_stream *stream) { int err; - pa_threaded_mainloop_lock(c->mainloop); /* * wait until actually connects. workaround pa bug #247 * https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/247 @@ -550,7 +549,6 @@ static void qpa_simple_disconnect(PAConnection *c, pa_stream *stream) dolog("Failed to disconnect! err=%d\n", err); } pa_stream_unref(stream); - pa_threaded_mainloop_unlock(c->mainloop); } static void qpa_fini_out (HWVoiceOut *hw) @@ -558,8 +556,12 @@ static void qpa_fini_out (HWVoiceOut *hw) PAVoiceOut *pa = (PAVoiceOut *) hw; if (pa->stream) { - qpa_simple_disconnect(pa->g->conn, pa->stream); + PAConnection *c = pa->g->conn; + + pa_threaded_mainloop_lock(c->mainloop); + qpa_simple_disconnect(c, pa->stream); pa->stream = NULL; + pa_threaded_mainloop_unlock(c->mainloop); } } @@ -568,8 +570,20 @@ static void qpa_fini_in (HWVoiceIn *hw) PAVoiceIn *pa = (PAVoiceIn *) hw; if (pa->stream) { - qpa_simple_disconnect(pa->g->conn, pa->stream); + PAConnection *c = pa->g->conn; + + pa_threaded_mainloop_lock(c->mainloop); + if (pa->read_length) { + int r = pa_stream_drop(pa->stream); + if (r) { + qpa_logerr(pa_context_errno(c->context), + "pa_stream_drop failed\n"); + } + pa->read_length = 0; + } + qpa_simple_disconnect(c, pa->stream); pa->stream = NULL; + pa_threaded_mainloop_unlock(c->mainloop); } } From acc3b63e1bdf806de1a520522dd43e494461d3bb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 4 Jan 2020 10:11:21 +0100 Subject: [PATCH 4/6] paaudio: try to drain the recording stream MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There is no guarantee a single call to pa_stream_peek every timer_period microseconds can read a recording stream faster than the data gets produced at the source. Let qpa_read try to drain the recording stream. To reproduce the problem: Start qemu with -audiodev pa,id=audio0,in.mixing-engine=off On the host connect the qemu recording stream to the monitor of a hardware output device. While the problem can also be seen with a hardware input device, it's obvious with the monitor of a hardware output device. In the guest start audio recording with audacity and notice the slow recording data rate. Signed-off-by: Volker Rümelin Message-id: 20200104091122.13971-4-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann --- audio/paaudio.c | 41 +++++++++++++++++++++++++---------------- 1 file changed, 25 insertions(+), 16 deletions(-) diff --git a/audio/paaudio.c b/audio/paaudio.c index 7db1dc15f0..b23274550e 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -156,34 +156,43 @@ static size_t qpa_read(HWVoiceIn *hw, void *data, size_t length) { PAVoiceIn *p = (PAVoiceIn *) hw; PAConnection *c = p->g->conn; - size_t l; - int r; + size_t total = 0; pa_threaded_mainloop_lock(c->mainloop); CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail, "pa_threaded_mainloop_lock failed\n"); - if (!p->read_length) { - r = pa_stream_peek(p->stream, &p->read_data, &p->read_length); - CHECK_SUCCESS_GOTO(c, r == 0, unlock_and_fail, - "pa_stream_peek failed\n"); - } + while (total < length) { + size_t l; + int r; - l = MIN(p->read_length, length); - memcpy(data, p->read_data, l); + if (!p->read_length) { + r = pa_stream_peek(p->stream, &p->read_data, &p->read_length); + CHECK_SUCCESS_GOTO(c, r == 0, unlock_and_fail, + "pa_stream_peek failed\n"); + if (!p->read_length) { + /* buffer is empty */ + break; + } + } - p->read_data += l; - p->read_length -= l; + l = MIN(p->read_length, length - total); + memcpy((char *)data + total, p->read_data, l); - if (!p->read_length) { - r = pa_stream_drop(p->stream); - CHECK_SUCCESS_GOTO(c, r == 0, unlock_and_fail, - "pa_stream_drop failed\n"); + p->read_data += l; + p->read_length -= l; + total += l; + + if (!p->read_length) { + r = pa_stream_drop(p->stream); + CHECK_SUCCESS_GOTO(c, r == 0, unlock_and_fail, + "pa_stream_drop failed\n"); + } } pa_threaded_mainloop_unlock(c->mainloop); - return l; + return total; unlock_and_fail: pa_threaded_mainloop_unlock(c->mainloop); From 7c9eb86e679b3b6992f97bd60440dbd1a9a75929 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 4 Jan 2020 10:11:22 +0100 Subject: [PATCH 5/6] paaudio: wait until the recording stream is ready MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Don't call pa_stream_peek before the recording stream is ready. Information to reproduce the problem. Start and stop Audacity in the guest several times because the problem is racy. libvirt log file: -audiodev pa,id=audio0,server=localhost,out.latency=30000, out.mixing-engine=off,in.mixing-engine=off \ -sandbox on,obsolete=deny,elevateprivileges=deny,spawn=deny, resourcecontrol=deny \ -msg timestamp=on : Domain id=4 is tainted: custom-argv char device redirected to /dev/pts/1 (label charserial0) audio: Device pcspk: audiodev default parameter is deprecated, please specify audiodev=audio0 audio: Device hda: audiodev default parameter is deprecated, please specify audiodev=audio0 pulseaudio: pa_stream_peek failed pulseaudio: Reason: Bad state pulseaudio: pa_stream_peek failed pulseaudio: Reason: Bad state Signed-off-by: Volker Rümelin Message-id: 20200104091122.13971-5-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann --- audio/paaudio.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/audio/paaudio.c b/audio/paaudio.c index b23274550e..dbfe48c03a 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -162,6 +162,10 @@ static size_t qpa_read(HWVoiceIn *hw, void *data, size_t length) CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail, "pa_threaded_mainloop_lock failed\n"); + if (pa_stream_get_state(p->stream) != PA_STREAM_READY) { + /* wait for stream to become ready */ + goto unlock; + } while (total < length) { size_t l; @@ -191,6 +195,7 @@ static size_t qpa_read(HWVoiceIn *hw, void *data, size_t length) } } +unlock: pa_threaded_mainloop_unlock(c->mainloop); return total; From 40ad46d3cc463fab5a23db466f77e37aff23f927 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Thu, 19 Dec 2019 21:34:05 +0100 Subject: [PATCH 6/6] audio: fix integer overflow MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Tell the compiler to do a 32bit * 32bit -> 64bit multiplication because period_ticks is a 64bit variable. The overflow occurs for audio timer periods larger than 4294967us. Fixes: be1092afa0 "audio: fix audio timer rate conversion bug" Signed-off-by: Volker Rümelin Message-id: 8893a235-66a8-8fbe-7d95-862e29da90b1@t-online.de Signed-off-by: Gerd Hoffmann --- audio/audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/audio/audio.c b/audio/audio.c index 56fae55047..abea027fdf 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -1738,7 +1738,7 @@ static AudioState *audio_init(Audiodev *dev, const char *name) if (dev->timer_period <= 0) { s->period_ticks = 1; } else { - s->period_ticks = dev->timer_period * SCALE_US; + s->period_ticks = dev->timer_period * (int64_t)SCALE_US; } e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);