Merge remote-tracking branch 'remotes/kraxel/tags/audio-20200316-pull…

…-request' into staging

audio: float fixes

# gpg: Signature made Mon 16 Mar 2020 11:30:00 GMT
# gpg:                using RSA key 4CB6D8EED3E87138
# 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-20200316-pull-request:
  audio: add audiodev format=f32 option documentation
  audio: fix saturation nonlinearity in clip_* functions
  audio: change mixing engine float range to [-1.f, 1.f]
  audio: consistency changes
  audio: change naming scheme of FLOAT_CONV macros
  qapi/audio: add documentation for AudioFormat

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

audio/mixeng.c

@@ -268,17 +268,17 @@ f_sample *mixeng_clip[2][2][2][3] = {
 };
 
 #ifdef FLOAT_MIXENG
-#define FLOAT_CONV_TO(x) (x)
-#define FLOAT_CONV_FROM(x) (x)
+#define CONV_NATURAL_FLOAT(x) (x)
+#define CLIP_NATURAL_FLOAT(x) (x)
 #else
-static const float float_scale = UINT_MAX;
-#define FLOAT_CONV_TO(x) ((x) * float_scale)
+static const float float_scale = UINT_MAX / 2.f;
+#define CONV_NATURAL_FLOAT(x) ((x) * float_scale)
 
 #ifdef RECIPROCAL
-static const float float_scale_reciprocal = 1.f / UINT_MAX;
-#define FLOAT_CONV_FROM(x) ((x) * float_scale_reciprocal)
+static const float float_scale_reciprocal = 2.f / UINT_MAX;
+#define CLIP_NATURAL_FLOAT(x) ((x) * float_scale_reciprocal)
 #else
-#define FLOAT_CONV_FROM(x) ((x) / float_scale)
+#define CLIP_NATURAL_FLOAT(x) ((x) / float_scale)
 #endif
 #endif
 
@@ -288,7 +288,7 @@ static void conv_natural_float_to_mono(struct st_sample *dst, const void *src,
     float *in = (float *)src;
 
     while (samples--) {
-        dst->r = dst->l = FLOAT_CONV_TO(*in++);
+        dst->r = dst->l = CONV_NATURAL_FLOAT(*in++);
         dst++;
     }
 }
@@ -299,8 +299,8 @@ static void conv_natural_float_to_stereo(struct st_sample *dst, const void *src,
     float *in = (float *)src;
 
     while (samples--) {
-        dst->l = FLOAT_CONV_TO(*in++);
-        dst->r = FLOAT_CONV_TO(*in++);
+        dst->l = CONV_NATURAL_FLOAT(*in++);
+        dst->r = CONV_NATURAL_FLOAT(*in++);
         dst++;
     }
 }
@@ -316,7 +316,7 @@ static void clip_natural_float_from_mono(void *dst, const struct st_sample *src,
     float *out = (float *)dst;
 
     while (samples--) {
-        *out++ = FLOAT_CONV_FROM(src->l) + FLOAT_CONV_FROM(src->r);
+        *out++ = CLIP_NATURAL_FLOAT(src->l + src->r);
         src++;
     }
 }
@@ -327,8 +327,8 @@ static void clip_natural_float_from_stereo(
     float *out = (float *)dst;
 
     while (samples--) {
-        *out++ = FLOAT_CONV_FROM(src->l);
-        *out++ = FLOAT_CONV_FROM(src->r);
+        *out++ = CLIP_NATURAL_FLOAT(src->l);
+        *out++ = CLIP_NATURAL_FLOAT(src->r);
         src++;
     }
 }

audio/mixeng_template.h

@@ -41,32 +41,31 @@ static inline mixeng_real glue (conv_, ET) (IN_T v)
 
 #ifdef RECIPROCAL
 #ifdef SIGNED
-    return nv * (1.f / (mixeng_real) (IN_MAX - IN_MIN));
+    return nv * (2.f / ((mixeng_real)IN_MAX - IN_MIN));
 #else
-    return (nv - HALF) * (1.f / (mixeng_real) IN_MAX);
+    return (nv - HALF) * (2.f / (mixeng_real)IN_MAX);
 #endif
 #else  /* !RECIPROCAL */
 #ifdef SIGNED
-    return nv / (mixeng_real) ((mixeng_real) IN_MAX - IN_MIN);
+    return nv / (((mixeng_real)IN_MAX - IN_MIN) / 2.f);
 #else
-    return (nv - HALF) / (mixeng_real) IN_MAX;
+    return (nv - HALF) / ((mixeng_real)IN_MAX / 2.f);
 #endif
 #endif
 }
 
 static inline IN_T glue (clip_, ET) (mixeng_real v)
 {
-    if (v >= 0.5) {
+    if (v >= 1.f) {
         return IN_MAX;
-    }
-    else if (v < -0.5) {
+    } else if (v < -1.f) {
         return IN_MIN;
     }
 
 #ifdef SIGNED
-    return ENDIAN_CONVERT ((IN_T) (v * ((mixeng_real) IN_MAX - IN_MIN)));
+    return ENDIAN_CONVERT((IN_T)(v * (((mixeng_real)IN_MAX - IN_MIN) / 2.f)));
 #else
-    return ENDIAN_CONVERT ((IN_T) ((v * IN_MAX) + HALF));
+    return ENDIAN_CONVERT((IN_T)((v * ((mixeng_real)IN_MAX / 2.f)) + HALF));
 #endif
 }
 
@@ -84,10 +83,9 @@ static inline int64_t glue (conv_, ET) (IN_T v)
 
 static inline IN_T glue (clip_, ET) (int64_t v)
 {
-    if (v >= 0x7f000000) {
+    if (v >= 0x7fffffffLL) {
         return IN_MAX;
-    }
-    else if (v < -2147483648LL) {
+    } else if (v < -2147483648LL) {
         return IN_MIN;
     }
 

qapi/audio.json

@@ -273,6 +273,20 @@
 #
 # An enumeration of possible audio formats.
 #
+# @u8: unsigned 8 bit integer
+#
+# @s8: signed 8 bit integer
+#
+# @u16: unsigned 16 bit integer
+#
+# @s16: signed 16 bit integer
+#
+# @u32: unsigned 32 bit integer
+#
+# @s32: signed 32 bit integer
+#
+# @f32: single precision floating-point (since 5.0)
+#
 # Since: 4.0
 ##
 { 'enum': 'AudioFormat',

qemu-options.hx

@@ -551,7 +551,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
     "                in|out.frequency= frequency to use with fixed settings\n"
     "                in|out.channels= number of channels to use with fixed settings\n"
     "                in|out.format= sample format to use with fixed settings\n"
-    "                valid values: s8, s16, s32, u8, u16, u32\n"
+    "                valid values: s8, s16, s32, u8, u16, u32, f32\n"
     "                in|out.voices= number of voices to use\n"
     "                in|out.buffer-length= length of buffer in microseconds\n"
     "-audiodev none,id=id,[,prop[=value][,...]]\n"
@@ -647,7 +647,7 @@ SRST
     ``in|out.format=format``
         Specify the sample format to use when using fixed-settings.
         Valid values are: ``s8``, ``s16``, ``s32``, ``u8``, ``u16``,
-        ``u32``. Default is ``s16``.
+        ``u32``, ``f32``. Default is ``s16``.
 
     ``in|out.voices=voices``
         Specify the number of voices to use. Default is 1.