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From: Gerd Hoffmann <kraxel@redhat.com>
To: qemu-devel@nongnu.org
Cc: Kővágó@redhat.com, Zoltán <DirtY.iCE.hu@gmail.com>,
	"Gerd Hoffmann" <kraxel@redhat.com>,
	"Markus Armbruster" <armbru@redhat.com>
Subject: [Qemu-devel] [PULL 01/26] audio: api for mixeng code free backends
Date: Thu, 19 Sep 2019 10:36:04 +0200	[thread overview]
Message-ID: <20190919083629.29998-2-kraxel@redhat.com> (raw)
In-Reply-To: <20190919083629.29998-1-kraxel@redhat.com>

From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>

This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only).  In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.

Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases.  Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required.  audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 6ba56ad4506c283aa903e8029f5fa53520b27544.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
 audio/audio_int.h      |  41 ++++++--
 audio/audio_template.h |   1 +
 audio/audio.c          | 216 ++++++++++++++++++++++++++++++++++++++++-
 3 files changed, 250 insertions(+), 8 deletions(-)

diff --git a/audio/audio_int.h b/audio/audio_int.h
index a674c5374a06..8fb1ca8a8d0f 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -65,6 +65,8 @@ typedef struct HWVoiceOut {
     uint64_t ts_helper;
 
     struct st_sample *mix_buf;
+    void *buf_emul;
+    size_t pos_emul, pending_emul, size_emul;
 
     size_t samples;
     QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
@@ -87,6 +89,8 @@ typedef struct HWVoiceIn {
     uint64_t ts_helper;
 
     struct st_sample *conv_buf;
+    void *buf_emul;
+    size_t pos_emul, pending_emul, size_emul;
 
     size_t samples;
     QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
@@ -147,17 +151,42 @@ struct audio_driver {
 };
 
 struct audio_pcm_ops {
-    int  (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
-    void (*fini_out)(HWVoiceOut *hw);
+    int    (*init_out)(HWVoiceOut *hw, audsettings *as, void *drv_opaque);
+    void   (*fini_out)(HWVoiceOut *hw);
     size_t (*run_out)(HWVoiceOut *hw, size_t live);
-    int  (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
+    size_t (*write)   (HWVoiceOut *hw, void *buf, size_t size);
+    /*
+     * get a buffer that after later can be passed to put_buffer_out; optional
+     * returns the buffer, and writes it's size to size (in bytes)
+     * this is unrelated to the above buffer_size_out function
+     */
+    void  *(*get_buffer_out)(HWVoiceOut *hw, size_t *size);
+    /*
+     * put back the buffer returned by get_buffer_out; optional
+     * buf must be equal the pointer returned by get_buffer_out,
+     * size may be smaller
+     */
+    size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size);
+    int    (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
 
-    int  (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
-    void (*fini_in) (HWVoiceIn *hw);
+    int    (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
+    void   (*fini_in) (HWVoiceIn *hw);
     size_t (*run_in)(HWVoiceIn *hw);
-    int  (*ctl_in)  (HWVoiceIn *hw, int cmd, ...);
+    size_t (*read)    (HWVoiceIn *hw, void *buf, size_t size);
+    void  *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
+    void   (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
+    int    (*ctl_in)  (HWVoiceIn *hw, int cmd, ...);
 };
 
+void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
+void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size);
+void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size);
+size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size);
+size_t audio_generic_put_buffer_out_nowrite(HWVoiceOut *hw, void *buf,
+                                            size_t size);
+size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size);
+size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size);
+
 struct capture_callback {
     struct audio_capture_ops ops;
     void *opaque;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 2562bf5f0062..ff4a173f1810 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -71,6 +71,7 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
 
 static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
 {
+    g_free(hw->buf_emul);
     g_free (HWBUF);
     HWBUF = NULL;
 }
diff --git a/audio/audio.c b/audio/audio.c
index e99fcd0694e9..e29a1e15eb30 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -573,6 +573,25 @@ size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
     return clipped;
 }
 
+static void audio_pcm_hw_clip_out2(HWVoiceOut *hw, void *pcm_buf, size_t len)
+{
+    size_t clipped = 0;
+    size_t pos = hw->rpos;
+
+    while (len) {
+        st_sample *src = hw->mix_buf + pos;
+        uint8_t *dst = advance(pcm_buf, clipped << hw->info.shift);
+        size_t samples_till_end_of_buf = hw->samples - 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->samples;
+        len -= samples_to_clip;
+        clipped += samples_to_clip;
+    }
+}
+
 /*
  * Soft voice (capture)
  */
@@ -1050,6 +1069,36 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
     mixeng_clear(hw->mix_buf, samples - n);
 }
 
+static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
+{
+    size_t clipped = 0;
+
+    while (live) {
+        size_t size, decr, proc;
+        void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
+        if (!buf) {
+            /* retrying will likely won't help, drop everything. */
+            hw->rpos = (hw->rpos + live) % hw->samples;
+            return clipped + live;
+        }
+
+        decr = MIN(size >> hw->info.shift, live);
+        audio_pcm_hw_clip_out2(hw, buf, decr);
+        proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
+            hw->info.shift;
+
+        live -= proc;
+        clipped += proc;
+        hw->rpos = (hw->rpos + proc) % hw->samples;
+
+        if (proc == 0 || proc < decr) {
+            break;
+        }
+    }
+
+    return clipped;
+}
+
 static void audio_run_out (AudioState *s)
 {
     HWVoiceOut *hw = NULL;
@@ -1097,7 +1146,11 @@ static void audio_run_out (AudioState *s)
         }
 
         prev_rpos = hw->rpos;
-        played = hw->pcm_ops->run_out (hw, live);
+        if (hw->pcm_ops->run_out) {
+            played = hw->pcm_ops->run_out(hw, live);
+        } else {
+            played = audio_pcm_hw_run_out(hw, live);
+        }
         replay_audio_out(&played);
         if (audio_bug(__func__, hw->rpos >= hw->samples)) {
             dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
@@ -1156,6 +1209,35 @@ static void audio_run_out (AudioState *s)
     }
 }
 
+static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
+{
+    size_t conv = 0;
+
+    while (samples) {
+        size_t proc;
+        size_t size = samples << hw->info.shift;
+        void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
+
+        assert((size & hw->info.align) == 0);
+        if (size == 0) {
+            hw->pcm_ops->put_buffer_in(hw, buf, size);
+            break;
+        }
+
+        proc = MIN(size >> hw->info.shift,
+                   hw->samples - hw->wpos);
+
+        hw->conv(hw->conv_buf + hw->wpos, buf, proc);
+        hw->wpos = (hw->wpos + proc) % hw->samples;
+
+        samples -= proc;
+        conv += proc;
+        hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift);
+    }
+
+    return conv;
+}
+
 static void audio_run_in (AudioState *s)
 {
     HWVoiceIn *hw = NULL;
@@ -1165,7 +1247,12 @@ static void audio_run_in (AudioState *s)
         size_t captured = 0, min;
 
         if (replay_mode != REPLAY_MODE_PLAY) {
-            captured = hw->pcm_ops->run_in(hw);
+            if (hw->pcm_ops->run_in) {
+                captured = hw->pcm_ops->run_in(hw);
+            } else {
+                captured = audio_pcm_hw_run_in(
+                    hw, hw->samples - audio_pcm_hw_get_live_in(hw));
+            }
         }
         replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples);
 
@@ -1259,12 +1346,137 @@ void audio_run(AudioState *s, const char *msg)
 #endif
 }
 
+void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
+{
+    ssize_t start;
+
+    if (unlikely(!hw->buf_emul)) {
+        size_t calc_size = hw->samples << hw->info.shift;
+        hw->buf_emul = g_malloc(calc_size);
+        hw->size_emul = calc_size;
+        hw->pos_emul = hw->pending_emul = 0;
+    }
+
+    while (hw->pending_emul < hw->size_emul) {
+        size_t read_len = MIN(hw->size_emul - hw->pos_emul,
+                              hw->size_emul - hw->pending_emul);
+        size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul,
+                                        read_len);
+        hw->pending_emul += read;
+        if (read < read_len) {
+            break;
+        }
+    }
+
+    start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+    if (start < 0) {
+        start += hw->size_emul;
+    }
+    assert(start >= 0 && start < hw->size_emul);
+
+    *size = MIN(hw->pending_emul, hw->size_emul - start);
+    return hw->buf_emul + start;
+}
+
+void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
+{
+    assert(size <= hw->pending_emul);
+    hw->pending_emul -= size;
+}
+
+void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
+{
+    if (unlikely(!hw->buf_emul)) {
+        size_t calc_size = hw->samples << hw->info.shift;
+
+        hw->buf_emul = g_malloc(calc_size);
+        hw->size_emul = calc_size;
+        hw->pos_emul = hw->pending_emul = 0;
+    }
+
+    *size = MIN(hw->size_emul - hw->pending_emul,
+                hw->size_emul - hw->pos_emul);
+    return hw->buf_emul + hw->pos_emul;
+}
+
+size_t audio_generic_put_buffer_out_nowrite(HWVoiceOut *hw, void *buf,
+                                            size_t size)
+{
+    assert(buf == hw->buf_emul + hw->pos_emul &&
+           size + hw->pending_emul <= hw->size_emul);
+
+    hw->pending_emul += size;
+    hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
+
+    return size;
+}
+
+size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
+{
+    audio_generic_put_buffer_out_nowrite(hw, buf, size);
+
+    while (hw->pending_emul) {
+        size_t write_len, written;
+        ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+        if (start < 0) {
+            start += hw->size_emul;
+        }
+        assert(start >= 0 && start < hw->size_emul);
+
+        write_len = MIN(hw->pending_emul, hw->size_emul - start);
+
+        written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
+        hw->pending_emul -= written;
+
+        if (written < write_len) {
+            break;
+        }
+    }
+
+    /*
+     * fake we have written everything. non-written data remain in pending_emul,
+     * so we do not have to clip them multiple times
+     */
+    return size;
+}
+
+size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
+{
+    size_t dst_size, copy_size;
+    void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
+    copy_size = MIN(size, dst_size);
+
+    memcpy(dst, buf, copy_size);
+    return hw->pcm_ops->put_buffer_out(hw, buf, copy_size);
+}
+
+size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
+{
+    size_t dst_size, copy_size;
+    void *dst = hw->pcm_ops->get_buffer_in(hw, &dst_size);
+    copy_size = MIN(size, dst_size);
+
+    memcpy(dst, buf, copy_size);
+    hw->pcm_ops->put_buffer_in(hw, buf, copy_size);
+    return copy_size;
+}
+
+
 static int audio_driver_init(AudioState *s, struct audio_driver *drv,
                              bool msg, Audiodev *dev)
 {
     s->drv_opaque = drv->init(dev);
 
     if (s->drv_opaque) {
+        if (!drv->pcm_ops->get_buffer_in) {
+            drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in;
+            drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in;
+        }
+        if (!drv->pcm_ops->get_buffer_out) {
+            drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out;
+            drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out;
+        }
+
         audio_init_nb_voices_out(s, drv);
         audio_init_nb_voices_in(s, drv);
         s->drv = drv;
-- 
2.18.1



  reply	other threads:[~2019-09-19  8:46 UTC|newest]

Thread overview: 31+ messages / expand[flat|nested]  mbox.gz  Atom feed  top
2019-09-19  8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
2019-09-19  8:36 ` Gerd Hoffmann [this message]
2019-09-19  8:36 ` [Qemu-devel] [PULL 02/26] alsaaudio: port to the new audio backend api Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 03/26] coreaudio: " Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 04/26] dsoundaudio: " Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 05/26] noaudio: " Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 06/26] ossaudio: " Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 07/26] paaudio: " Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 08/26] sdlaudio: " Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 09/26] spiceaudio: " Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 10/26] wavaudio: " Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 11/26] audio: remove remains of the old " Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 12/26] audio: unify input and output mixeng buffer management Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 13/26] audio: common rate control code for timer based outputs Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 14/26] audio: split ctl_* functions into enable_* and volume_* Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 15/26] audio: add mixeng option (documentation) Gerd Hoffmann
2019-09-19 15:27   ` Eric Blake
2019-09-19  8:36 ` [Qemu-devel] [PULL 16/26] audio: make mixeng optional Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 17/26] paaudio: get/put_buffer functions Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 18/26] audio: support more than two channels in volume setting Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 19/26] audio: replace shift in audio_pcm_info with bytes_per_frame Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 20/26] audio: basic support for multichannel audio Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 21/26] paaudio: channel-map option Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 22/26] usb-audio: do not count on avail bytes actually available Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 23/26] usb-audio: support more than two channels of audio Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 24/26] usbaudio: change playback counters to 64 bit Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 25/26] audio: fix buffer-length typo in documentation Gerd Hoffmann
2019-09-19  8:36 ` [Qemu-devel] [PULL 26/26] audio: fix ALSA period-length " Gerd Hoffmann
2019-09-19 15:15 ` [Qemu-devel] [PULL 00/26] Audio 20190919 patches Peter Maydell
2019-09-19 15:28   ` Eric Blake
2019-09-19 21:26     ` Zoltán Kővágó

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