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From: "Kővágó, Zoltán" <dirty.ice.hu@gmail.com>
To: qemu-devel@nongnu.org
Cc: Gerd Hoffmann <kraxel@redhat.com>
Subject: [Qemu-devel] [PATCH 20/25] audio: replace shift in audio_pcm_info with bytes_per_frame
Date: Sun, 25 Aug 2019 20:46:22 +0200
Message-ID: <a5d9a317768e9021671f28e360d88cc4a043a588.1566755452.git.DirtY.iCE.hu@gmail.com> (raw)
In-Reply-To: <cover.1566755452.git.DirtY.iCE.hu@gmail.com>

The bit shifting trick worked because the number of bytes per frame was
always a power-of-two (since QEMU only supports mono, stereo and 8, 16
and 32 bit samples).  But if we want to add support for surround sound,
this no longer holds true.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
 audio/audio_int.h       |  3 +-
 audio/dsound_template.h | 10 +++---
 audio/alsaaudio.c       | 11 +++---
 audio/audio.c           | 74 ++++++++++++++++++++---------------------
 audio/coreaudio.c       |  4 +--
 audio/dsoundaudio.c     |  4 +--
 audio/noaudio.c         |  2 +-
 audio/ossaudio.c        | 14 ++++----
 audio/spiceaudio.c      |  5 +--
 audio/wavaudio.c        |  6 ++--
 10 files changed, 67 insertions(+), 66 deletions(-)

diff --git a/audio/audio_int.h b/audio/audio_int.h
index e826734f8c..29fb0ededd 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -43,8 +43,7 @@ struct audio_pcm_info {
     int sign;
     int freq;
     int nchannels;
-    int align;
-    int shift;
+    int bytes_per_frame;
     int bytes_per_second;
     int swap_endianness;
 };
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 8cc5d69f36..ebc6ba04c1 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -98,8 +98,8 @@ static int glue (dsound_lock_, TYPE) (
         goto fail;
     }
 
-    if ((p1p && *p1p && (*blen1p & info->align)) ||
-        (p2p && *p2p && (*blen2p & info->align))) {
+    if ((p1p && *p1p && (*blen1p % info->bytes_per_frame)) ||
+        (p2p && *p2p && (*blen2p % info->bytes_per_frame))) {
         dolog("DirectSound returned misaligned buffer %ld %ld\n",
               *blen1p, *blen2p);
         glue(dsound_unlock_, TYPE)(buf, *p1p, p2p ? *p2p : NULL, *blen1p,
@@ -247,14 +247,14 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
     obt_as.endianness = 0;
     audio_pcm_init_info (&hw->info, &obt_as);
 
-    if (bc.dwBufferBytes & hw->info.align) {
+    if (bc.dwBufferBytes % hw->info.bytes_per_frame) {
         dolog (
             "GetCaps returned misaligned buffer size %ld, alignment %d\n",
-            bc.dwBufferBytes, hw->info.align + 1
+            bc.dwBufferBytes, hw->info.bytes_per_frame
             );
     }
     hw->size_emul = bc.dwBufferBytes;
-    ds->samples = bc.dwBufferBytes >> hw->info.shift;
+    ds->samples = bc.dwBufferBytes / hw->info.bytes_per_frame;
     ds->s = s;
 
 #ifdef DEBUG_DSOUND
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 95468bcf6d..b3b21e07a2 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -604,7 +604,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
 {
     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
     size_t pos = 0;
-    size_t len_frames = len >> hw->info.shift;
+    size_t len_frames = len / hw->info.bytes_per_frame;
 
     while (len_frames) {
         char *src = advance(buf, pos);
@@ -650,7 +650,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
             }
         }
 
-        pos += written << hw->info.shift;
+        pos += written * hw->info.bytes_per_frame;
         if (written < len_frames) {
             break;
         }
@@ -816,7 +816,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
         void *dst = advance(buf, pos);
         snd_pcm_sframes_t nread;
 
-        nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
+        nread = snd_pcm_readi(
+            alsa->handle, dst, len / hw->info.bytes_per_frame);
 
         if (nread <= 0) {
             switch (nread) {
@@ -842,8 +843,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
             }
         }
 
-        pos += nread << hw->info.shift;
-        len -= nread << hw->info.shift;
+        pos += nread * hw->info.bytes_per_frame;
+        len -= nread * hw->info.bytes_per_frame;
     }
 
     return pos;
diff --git a/audio/audio.c b/audio/audio.c
index 60a1d16dea..f46bd5dc3d 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -299,12 +299,13 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
 
 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
 {
-    int bits = 8, sign = 0, shift = 0;
+    int bits = 8, sign = 0, mul;
 
     switch (as->fmt) {
     case AUDIO_FORMAT_S8:
         sign = 1;
     case AUDIO_FORMAT_U8:
+        mul = 1;
         break;
 
     case AUDIO_FORMAT_S16:
@@ -312,7 +313,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
         /* fall through */
     case AUDIO_FORMAT_U16:
         bits = 16;
-        shift = 1;
+        mul = 2;
         break;
 
     case AUDIO_FORMAT_S32:
@@ -320,7 +321,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
         /* fall through */
     case AUDIO_FORMAT_U32:
         bits = 32;
-        shift = 2;
+        mul = 4;
         break;
 
     default:
@@ -331,9 +332,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
     info->bits = bits;
     info->sign = sign;
     info->nchannels = as->nchannels;
-    info->shift = (as->nchannels == 2) + shift;
-    info->align = (1 << info->shift) - 1;
-    info->bytes_per_second = info->freq << info->shift;
+    info->bytes_per_frame = as->nchannels * mul;
+    info->bytes_per_second = info->freq * info->bytes_per_frame;
     info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
 }
 
@@ -344,26 +344,25 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
     }
 
     if (info->sign) {
-        memset (buf, 0x00, len << info->shift);
+        memset(buf, 0x00, len * info->bytes_per_frame);
     }
     else {
         switch (info->bits) {
         case 8:
-            memset (buf, 0x80, len << info->shift);
+            memset(buf, 0x80, len * info->bytes_per_frame);
             break;
 
         case 16:
             {
                 int i;
                 uint16_t *p = buf;
-                int shift = info->nchannels - 1;
                 short s = INT16_MAX;
 
                 if (info->swap_endianness) {
                     s = bswap16 (s);
                 }
 
-                for (i = 0; i < len << shift; i++) {
+                for (i = 0; i < len * info->nchannels; i++) {
                     p[i] = s;
                 }
             }
@@ -373,14 +372,13 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
             {
                 int i;
                 uint32_t *p = buf;
-                int shift = info->nchannels - 1;
                 int32_t s = INT32_MAX;
 
                 if (info->swap_endianness) {
                     s = bswap32 (s);
                 }
 
-                for (i = 0; i < len << shift; i++) {
+                for (i = 0; i < len * info->nchannels; i++) {
                     p[i] = s;
                 }
             }
@@ -558,7 +556,7 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
 
     while (len) {
         st_sample *src = hw->mix_buf->samples + pos;
-        uint8_t *dst = advance(pcm_buf, clipped << hw->info.shift);
+        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_to_clip = MIN(len, samples_till_end_of_buf);
 
@@ -607,7 +605,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
         return 0;
     }
 
-    samples = size >> sw->info.shift;
+    samples = size / sw->info.bytes_per_frame;
     if (!live) {
         return 0;
     }
@@ -642,7 +640,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
 
     sw->clip (buf, sw->buf, ret);
     sw->total_hw_samples_acquired += total;
-    return ret << sw->info.shift;
+    return ret * sw->info.bytes_per_frame;
 }
 
 /*
@@ -715,7 +713,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
     }
 
     wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
-    samples = size >> sw->info.shift;
+    samples = size / sw->info.bytes_per_frame;
 
     dead = hwsamples - live;
     swlim = ((int64_t) dead << 32) / sw->ratio;
@@ -759,13 +757,13 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
     dolog (
         "%s: write size %zu ret %zu total sw %zu\n",
         SW_NAME (sw),
-        size >> sw->info.shift,
+        size / sw->info.bytes_per_frame,
         ret,
         sw->total_hw_samples_mixed
         );
 #endif
 
-    return ret << sw->info.shift;
+    return ret * sw->info.bytes_per_frame;
 }
 
 #ifdef DEBUG_AUDIO
@@ -882,7 +880,7 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
 
 int AUD_get_buffer_size_out (SWVoiceOut *sw)
 {
-    return sw->hw->mix_buf->size << sw->hw->info.shift;
+    return sw->hw->mix_buf->size * sw->hw->info.bytes_per_frame;
 }
 
 void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -998,10 +996,10 @@ static size_t audio_get_avail (SWVoiceIn *sw)
     ldebug (
         "%s: get_avail live %d ret %" PRId64 "\n",
         SW_NAME (sw),
-        live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
+        live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
         );
 
-    return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
+    return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
 }
 
 static size_t audio_get_free(SWVoiceOut *sw)
@@ -1025,10 +1023,11 @@ static size_t audio_get_free(SWVoiceOut *sw)
 #ifdef DEBUG_OUT
     dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
            SW_NAME (sw),
-           live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
+           live, dead, (((int64_t) dead << 32) / sw->ratio) *
+           sw->info.bytes_per_frame);
 #endif
 
-    return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
+    return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
 }
 
 static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
@@ -1047,7 +1046,7 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
             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.shift;
+                size_t bytes = to_write * hw->info.bytes_per_frame;
                 size_t written;
 
                 sw->buf = hw->mix_buf->samples + rpos2;
@@ -1077,10 +1076,11 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
         size_t size, decr, proc;
         void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
 
-        decr = MIN(size >> hw->info.shift, live);
+        decr = MIN(size / hw->info.bytes_per_frame, live);
         audio_pcm_hw_clip_out(hw, buf, decr);
-        proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
-            hw->info.shift;
+        proc = hw->pcm_ops->put_buffer_out(hw, buf,
+                                           decr * hw->info.bytes_per_frame) /
+            hw->info.bytes_per_frame;
 
         live -= proc;
         clipped += proc;
@@ -1229,16 +1229,16 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
 
     while (samples) {
         size_t proc;
-        size_t size = samples << hw->info.shift;
+        size_t size = samples * hw->info.bytes_per_frame;
         void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
 
-        assert((size & hw->info.align) == 0);
+        assert(size % hw->info.bytes_per_frame == 0);
         if (size == 0) {
             hw->pcm_ops->put_buffer_in(hw, buf, size);
             break;
         }
 
-        proc = MIN(size >> hw->info.shift,
+        proc = MIN(size / hw->info.bytes_per_frame,
                    conv_buf->size - conv_buf->pos);
 
         hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
@@ -1246,7 +1246,7 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
 
         samples -= proc;
         conv += proc;
-        hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift);
+        hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
     }
 
     return conv;
@@ -1320,7 +1320,7 @@ static void audio_run_capture (AudioState *s)
 
             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
                 cb->ops.capture (cb->opaque, cap->buf,
-                                 to_capture << hw->info.shift);
+                                 to_capture * hw->info.bytes_per_frame);
             }
             rpos = (rpos + to_capture) % hw->mix_buf->size;
             live -= to_capture;
@@ -1373,7 +1373,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
     ssize_t start;
 
     if (unlikely(!hw->buf_emul)) {
-        size_t calc_size = hw->conv_buf->size << hw->info.shift;
+        size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame;
         hw->buf_emul = g_malloc(calc_size);
         hw->size_emul = calc_size;
         hw->pos_emul = hw->pending_emul = 0;
@@ -1409,7 +1409,7 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
 void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
 {
     if (unlikely(!hw->buf_emul)) {
-        size_t calc_size = hw->mix_buf->size << hw->info.shift;
+        size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame;
 
         hw->buf_emul = g_malloc(calc_size);
         hw->size_emul = calc_size;
@@ -1826,7 +1826,7 @@ CaptureVoiceOut *AUD_add_capture(
 
         audio_pcm_init_info (&hw->info, as);
 
-        cap->buf = g_malloc0_n(hw->mix_buf->size, 1 << hw->info.shift);
+        cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
 
         hw->clip = mixeng_clip
             [hw->info.nchannels == 2]
@@ -2146,14 +2146,14 @@ size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
     ticks = now - rate->start_ticks;
     bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
-    samples = (bytes - rate->bytes_sent) >> info->shift;
+    samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
     if (samples < 0 || samples > 65536) {
         AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)", samples);
         audio_rate_start(rate);
         samples = 0;
     }
 
-    ret = MIN(samples << info->shift, bytes_avail);
+    ret = MIN(samples * info->bytes_per_frame, bytes_avail);
     rate->bytes_sent += ret;
     return ret;
 }
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index a43f72a495..2539a84285 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -441,7 +441,7 @@ static OSStatus audioDeviceIOProc(
     }
 
     frameCount = core->audioDevicePropertyBufferFrameSize;
-    pending_frames = hw->pending_emul >> hw->info.shift;
+    pending_frames = hw->pending_emul / hw->info.bytes_per_frame;
 
     /* if there are not enough samples, set signal and return */
     if (pending_frames < frameCount) {
@@ -450,7 +450,7 @@ static OSStatus audioDeviceIOProc(
         return 0;
     }
 
-    len = frameCount << hw->info.shift;
+    len = frameCount * hw->info.bytes_per_frame;
     while (len) {
         size_t write_len;
         ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 9504265166..cc7999fc95 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -322,8 +322,8 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
         return;
     }
 
-    len1 = blen1 >> hw->info.shift;
-    len2 = blen2 >> hw->info.shift;
+    len1 = blen1 / hw->info.bytes_per_frame;
+    len2 = blen2 / hw->info.bytes_per_frame;
 
 #ifdef DEBUG_DSOUND
     dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 4cf3714093..6dd4d47887 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -89,7 +89,7 @@ static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
     NoVoiceIn *no = (NoVoiceIn *) hw;
     int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size);
 
-    audio_pcm_info_clear_buf(&hw->info, buf, bytes >> hw->info.shift);
+    audio_pcm_info_clear_buf(&hw->info, buf, bytes / hw->info.bytes_per_frame);
     return bytes;
 }
 
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index cc91956ab7..5c13c3e9d5 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -508,16 +508,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
     oss->nfrags = obt.nfrags;
     oss->fragsize = obt.fragsize;
 
-    if (obt.nfrags * obt.fragsize & hw->info.align) {
+    if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
         dolog ("warning: Misaligned DAC buffer, size %d, alignment %d\n",
-               obt.nfrags * obt.fragsize, hw->info.align + 1);
+               obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
     }
 
-    oss->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+    oss->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
 
     oss->mmapped = 0;
     if (oopts->has_try_mmap && oopts->try_mmap) {
-        hw->size_emul = oss->samples << hw->info.shift;
+        hw->size_emul = oss->samples * hw->info.bytes_per_frame;
         hw->buf_emul = mmap(
             NULL,
             hw->size_emul,
@@ -652,12 +652,12 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     oss->nfrags = obt.nfrags;
     oss->fragsize = obt.fragsize;
 
-    if (obt.nfrags * obt.fragsize & hw->info.align) {
+    if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
         dolog ("warning: Misaligned ADC buffer, size %d, alignment %d\n",
-               obt.nfrags * obt.fragsize, hw->info.align + 1);
+               obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
     }
 
-    oss->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+    oss->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
 
     oss->fd = fd;
     oss->dev = dev;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index fcd8ffa810..ea15c49ad7 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -133,8 +133,9 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
         out->fpos = 0;
     }
 
-    *size = audio_rate_get_bytes(&hw->info, &out->rate,
-                                 (out->fsize - out->fpos) << hw->info.shift);
+    *size = audio_rate_get_bytes(
+        &hw->info, &out->rate,
+        (out->fsize - out->fpos) * hw->info.bytes_per_frame);
     return out->frame + out->fpos;
 }
 
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 9b9f75effc..01062ec426 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -43,14 +43,14 @@ static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
 {
     WAVVoiceOut *wav = (WAVVoiceOut *) hw;
     int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
-    assert(bytes >> hw->info.shift << hw->info.shift == bytes);
+    assert(bytes % hw->info.bytes_per_frame == 0);
 
     if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
         dolog("wav_write_out: fwrite of %zu bytes failed\nReaons: %s\n",
               bytes, strerror(errno));
     }
 
-    wav->total_samples += bytes >> hw->info.shift;
+    wav->total_samples += bytes / hw->info.bytes_per_frame;
     return bytes;
 }
 
@@ -133,7 +133,7 @@ static void wav_fini_out (HWVoiceOut *hw)
     WAVVoiceOut *wav = (WAVVoiceOut *) hw;
     uint8_t rlen[4];
     uint8_t dlen[4];
-    uint32_t datalen = wav->total_samples << hw->info.shift;
+    uint32_t datalen = wav->total_samples * hw->info.bytes_per_frame;
     uint32_t rifflen = datalen + 36;
 
     if (!wav->f) {
-- 
2.22.0



  parent reply index

Thread overview: 27+ messages / expand[flat|nested]  mbox.gz  Atom feed  top
2019-08-25 18:46 [Qemu-devel] [PATCH 00/25] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 01/25] audio: api for mixeng code free backends Kővágó, Zoltán
2019-08-27  5:43   ` Gerd Hoffmann
2019-08-25 18:46 ` [Qemu-devel] [PATCH 02/25] alsaaudio: port to the new audio backend api Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 03/25] coreaudio: " Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 04/25] dsoundaudio: " Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 05/25] noaudio: " Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 06/25] ossaudio: " Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 07/25] paaudio: " Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 08/25] sdlaudio: " Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 10/25] wavaudio: " Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 11/25] audio: remove remains of the old " Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 12/25] audio: unify input and output mixeng buffer management Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 13/25] audio: remove hw->samples, buffer_size_in/out pcm_ops Kővágó, Zoltán
2019-08-27  5:48   ` Gerd Hoffmann
2019-08-25 18:46 ` [Qemu-devel] [PATCH 14/25] audio: common rate control code for timer based outputs Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 15/25] audio: split ctl_* functions into enable_* and volume_* Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 16/25] audio: add mixeng option (documentation) Kővágó, Zoltán
2019-08-26 13:35   ` Eric Blake
2019-08-26 20:04     ` Zoltán Kővágó
2019-08-25 18:46 ` [Qemu-devel] [PATCH 17/25] audio: make mixeng optional Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 19/25] audio: support more than two channels in volume setting Kővágó, Zoltán
2019-08-25 18:46 ` Kővágó, Zoltán [this message]
2019-08-25 18:46 ` [Qemu-devel] [PATCH 21/25] audio: basic support for multichannel audio Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 22/25] paaudio: channel-map option Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 23/25] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
2019-08-25 18:46 ` [Qemu-devel] [PATCH 25/25] usbaudio: change playback counters to 64 bit Kővágó, Zoltán

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