Changeset ea6c838 in mainline for uspace/lib/pcm

Timestamp:

2012-07-17T08:26:49Z (13 years ago)

Author:

Jan Vesely <jano.vesely@…>

Branches:

lfn, master, serial, ticket/834-toolchain-update, topic/msim-upgrade, topic/simplify-dev-export

Children:

e6bba8f

Parents:

950110ee

Message:

Create libpcm.

Move pcm related functions and definitions there.
Make other stuff use this library.
Rename most of it on the way.

Location:

uspace/lib/pcm

Files:

• Makefile (added)
• include/pcm/format.h (moved) (moved from uspace/srv/audio/hound/audio_format.h ) (4 diffs)
• include/pcm/sample_format.h (moved) (moved from uspace/lib/c/include/pcm_sample_format.h )
• src/format.c (moved) (moved from uspace/srv/audio/hound/audio_format.c ) (4 diffs)

uspace/lib/pcm/include/pcm/format.h

@@ -34 +34 @@
  */
 
-#ifndef AUDIO_FORMAT_H_
-#define AUDIO_FORMAT_H_
+#ifndef PCM_FORMAT_H_
+#define PCM_FORMAT_H_
 
 #include <assert.h>
 #include <bool.h>
-#include <pcm_sample_format.h>
-
+#include <pcm/sample_format.h>
 
 typedef struct {
@@ -46 +45 @@
         unsigned sampling_rate;
         pcm_sample_format_t sample_format;
-} audio_format_t;
+} pcm_format_t;
 
-static const audio_format_t AUDIO_FORMAT_DEFAULT = {
+static const pcm_format_t AUDIO_FORMAT_DEFAULT = {
         .channels = 2,
         .sampling_rate = 44100,
@@ -54 +53 @@
         };
 
-static const audio_format_t AUDIO_FORMAT_ANY = {
+static const pcm_format_t AUDIO_FORMAT_ANY = {
         .channels = 0,
         .sampling_rate = 0,
@@ -60 +59 @@
         };
 
-
-static inline size_t audio_format_frame_size(const audio_format_t *a)
+static inline size_t pcm_format_frame_size(const pcm_format_t *a)
 {
         return a->channels * pcm_sample_format_size(a->sample_format);
 }
 
-bool audio_format_same(const audio_format_t *a, const audio_format_t* b);
-static inline bool audio_format_is_any(const audio_format_t *f)
+bool pcm_format_same(const pcm_format_t *a, const pcm_format_t* b);
+static inline bool pcm_format_is_any(const pcm_format_t *f)
 {
-        return audio_format_same(f, &AUDIO_FORMAT_ANY);
+        return pcm_format_same(f, &AUDIO_FORMAT_ANY);
 }
-int audio_format_convert_and_mix(void *dst, size_t dst_size, const void *src,
-    size_t src_size, const audio_format_t *sf, const audio_format_t *df);
-int audio_format_mix(void *dst, const void *src, size_t size, const audio_format_t *f);
-int audio_format_convert(audio_format_t a, void* srca, size_t sizea,
-    audio_format_t b, void* srcb, size_t *sizeb);
+int pcm_format_convert_and_mix(void *dst, size_t dst_size, const void *src,
+    size_t src_size, const pcm_format_t *sf, const pcm_format_t *df);
+int pcm_format_mix(void *dst, const void *src, size_t size, const pcm_format_t *f);
+int pcm_format_convert(pcm_format_t a, void* srca, size_t sizea,
+    pcm_format_t b, void* srcb, size_t *sizeb);
 
 #endif

uspace/lib/pcm/src/format.c

@@ -40 +40 @@
 #include <stdio.h>
 
-#include "audio_format.h"
+#include "format.h"
 
 #define uint8_t_le2host(x) (x)
@@ -76 +76 @@
 
 static float get_normalized_sample(const void *buffer, size_t size,
-    unsigned frame, unsigned channel, const audio_format_t *f);
-
-bool audio_format_same(const audio_format_t *a, const audio_format_t* b)
+    unsigned frame, unsigned channel, const pcm_format_t *f);
+
+bool pcm_format_same(const pcm_format_t *a, const pcm_format_t* b)
 {
         assert(a);
@@ -88 +88 @@
 }
 
-int audio_format_mix(void *dst, const void *src, size_t size, const audio_format_t *f)
-{
-        return audio_format_convert_and_mix(dst, size, src, size, f, f);
-}
-int audio_format_convert_and_mix(void *dst, size_t dst_size, const void *src,
-    size_t src_size, const audio_format_t *sf, const audio_format_t *df)
+int pcm_format_mix(void *dst, const void *src, size_t size, const pcm_format_t *f)
+{
+        return pcm_format_convert_and_mix(dst, size, src, size, f, f);
+}
+int pcm_format_convert_and_mix(void *dst, size_t dst_size, const void *src,
+    size_t src_size, const pcm_format_t *sf, const pcm_format_t *df)
 {
         if (!dst || !src || !sf || !df)
                 return EINVAL;
-        const size_t src_frame_size = audio_format_frame_size(sf);
+        const size_t src_frame_size = pcm_format_frame_size(sf);
         if ((src_size % src_frame_size) != 0)
                 return EINVAL;
 
-        const size_t dst_frame_size = audio_format_frame_size(df);
+        const size_t dst_frame_size = pcm_format_frame_size(df);
         if ((src_size % dst_frame_size) != 0)
                 return EINVAL;
@@ -170 +170 @@
 /** Converts all sample formats to float <-1,1> */
 static float get_normalized_sample(const void *buffer, size_t size,
-    unsigned frame, unsigned channel, const audio_format_t *f)
+    unsigned frame, unsigned channel, const pcm_format_t *f)
 {
         assert(f);