Fix audio distortion in float -> int32 conversion.

This was noticeable when using sox filter and become prominant when
libavcodec introduced per-codec audio sample formats.
Also, add CLAMP to make code more readable.

diff --git a/src/modules/core/filter_audioconvert.c b/src/modules/core/filter_audioconvert.c
index a38219286f7c10968b2589296f240b3e0d91d398..ad1e943375d5c0dd2c91089ee34d9ba892bbf28e 100644 (file)
--- a/src/modules/core/filter_audioconvert.c
+++ b/src/modules/core/filter_audioconvert.c
@@ -25,6 +25,10 @@
 #include <stdio.h>
 #include <stdlib.h>
 
+#ifndef CLAMP
+#define CLAMP( x, min, max ) (x) < (min) ? (min) : (x) > (max) ? (max) : (x)
+#endif
+
 static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *format, mlt_audio_format requested_format )
 {
        int error = 1;
@@ -102,8 +106,7 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                while ( --i )
                                {
                                        float f = (float)( *q++ ) / 32768.0;
-                                       f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
-                                       *p++ = f;
+                                       *p++ = CLAMP( f, -1.0f, 1.0f );
                                }
                                *audio = buffer;
                                error = 0;
@@ -176,8 +179,7 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                        for ( c = 0; c < channels; c++ )
                                        {
                                                float f = (float)( *( q + c * samples + s ) ) / 2147483648.0;
-                                               f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
-                                               *p++ = f;
+                                               *p++ = CLAMP( f, -1.0f, 1.0f );
                                        }
                                *audio = buffer;
                                error = 0;
@@ -213,7 +215,7 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                        for ( c = 0; c < channels; c++ )
                                        {
                                                float f = *( q + c * samples + s );
-                                               f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
+                                               f = CLAMP( f, -1.0f, 1.0f );
                                                *p++ = 32767 * f;
                                        }
                                *audio = buffer;
@@ -229,8 +231,9 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                while ( --i )
                                {
                                        float f = *q++;
-                                       f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
-                                       *p++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
+                                       f = CLAMP( f, -1.0f, 1.0f );
+                                       int64_t pcm = ( f > 0.0f ? 2147483647LL : 2147483648LL ) * f;
+                                       *p++ = CLAMP( pcm, -2147483648LL, 2147483647LL );
                                }
                                *audio = buffer;
                                error = 0;
@@ -246,8 +249,9 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                        for ( c = 0; c < channels; c++ )
                                        {
                                                float f = *( q + c * samples + s );
-                                               f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
-                                               *p++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
+                                               f = CLAMP( f, -1.0f, 1.0f );
+                                               int64_t pcm = ( f > 0.0f ? 2147483647LL : 2147483648LL ) * f;
+                                               *p++ = CLAMP( pcm, -2147483648LL, 2147483647LL );
                                        }
                                *audio = buffer;
                                error = 0;
@@ -276,7 +280,7 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                        for ( c = 0; c < channels; c++ )
                                        {
                                                float f = *( q + c * samples + s );
-                                               f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
+                                               f = CLAMP( f, -1.0f, 1.0f );
                                                *p++ = ( 127 * f ) + 128;
                                        }
                                *audio = buffer;
@@ -380,7 +384,7 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                while ( --i )
                                {
                                        float f = *q++;
-                                       f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
+                                       f = CLAMP( f, -1.0f , 1.0f );
                                        *p++ = 32767 * f;
                                }
                                *audio = buffer;
@@ -418,8 +422,9 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                        while ( --i )
                                        {
                                                float f = *q;
-                                               f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
-                                               *p++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
+                                               f = CLAMP( f, -1.0f , 1.0f );
+                                               int64_t pcm = ( f > 0.0f ? 2147483647LL : 2147483648LL ) * f;
+                                               *p++ = CLAMP( pcm, -2147483648LL, 2147483647LL );
                                                q += channels;
                                        }
                                }
@@ -436,8 +441,9 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                while ( --i )
                                {
                                        float f = *q++;
-                                       f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
-                                       *p++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
+                                       f = CLAMP( f, -1.0f , 1.0f );
+                                       int64_t pcm = ( f > 0.0f ? 2147483647LL : 2147483648LL ) * f;
+                                       *p++ = CLAMP( pcm, -2147483648LL, 2147483647LL );
                                }
                                *audio = buffer;
                                error = 0;
@@ -452,7 +458,7 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                while ( --i )
                                {
                                        float f = *q++;
-                                       f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
+                                       f = CLAMP( f, -1.0f , 1.0f );
                                        *p++ = ( 127 * f ) + 128;
                                }
                                *audio = buffer;
@@ -537,8 +543,7 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                while ( --i )
                                {
                                        float f = ( (float) *q++ - 128 ) / 256.0f;
-                                       f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
-                                       *p++ = f;
+                                       *p++ = CLAMP( f, -1.0f, 1.0f );
                                }
                                *audio = buffer;
                                error = 0;