Changeset 950110ee in mainline

Timestamp:

2012-07-17T07:33:43Z (13 years ago)

Author:

Jan Vesely <jano.vesely@…>

Branches:

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

Children:

ea6c838

Parents:

389ef25

Message:

hound: convert format while mixing.

Still no resampling.

Location:

uspace/srv/audio/hound

Files:

• audio_format.c (modified) (4 diffs)
• audio_format.h (modified) (1 diff)
• audio_source.c (modified) (2 diffs)

uspace/srv/audio/hound/audio_format.c

@@ -90 +90 @@
 int audio_format_mix(void *dst, const void *src, size_t size, const audio_format_t *f)
 {
-        if (!dst || !src || !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)
+{
+        if (!dst || !src || !sf || !df)
                 return EINVAL;
-        const size_t sample_size = pcm_sample_format_size(f->sample_format);
-        if ((size % sample_size) != 0)
+        const size_t src_frame_size = audio_format_frame_size(sf);
+        if ((src_size % src_frame_size) != 0)
+                return EINVAL;
+
+        const size_t dst_frame_size = audio_format_frame_size(df);
+        if ((src_size % dst_frame_size) != 0)
                 return EINVAL;
 
@@ -101 +110 @@
 #define LOOP_ADD(type, endian, low, high) \
 do { \
-        const unsigned frame_size = audio_format_frame_size(f); \
-        const unsigned frame_count = size / frame_size; \
+        const unsigned frame_count = dst_size / dst_frame_size; \
         for (size_t i = 0; i < frame_count; ++i) { \
-                for (unsigned j = 0; j < f->channels; ++j) { \
+                for (unsigned j = 0; j < df->channels; ++j) { \
                         const float a = \
-                            get_normalized_sample(dst, size, i, j, f);\
+                            get_normalized_sample(dst, src_size, i, j, df);\
                         const float b = \
-                            get_normalized_sample(src, size, i, j, f);\
+                            get_normalized_sample(src, dst_size, i, j, sf);\
                         float c = (a + b); \
                         if (c < -1.0) c = -1.0; \
@@ -115 +123 @@
                         c *= ((float)(type)high - (float)(type)low) / 2; \
                         c += (float)(type)low; \
-                        if (c > (float)(type)high) { \
-                                printf("SCALE HIGH failed\n"); \
-                        } \
-                        if (c < (float)(type)low) { \
-                                printf("SCALE LOW failed\n"); \
-                        } \
                         type *dst_buf = dst; \
-                        const unsigned pos = i * f->channels  + j; \
-                        if (pos < (size / sizeof(type))) \
+                        const unsigned pos = i * df->channels  + j; \
+                        if (pos < (dst_size / sizeof(type))) \
                                 dst_buf[pos] = to((type)c, type, endian); \
                 } \
@@ -129 +131 @@
 } while (0)
 
-        switch (f->sample_format) {
+        switch (df->sample_format) {
         case PCM_SAMPLE_UINT8:
                 LOOP_ADD(uint8_t, le, UINT8_MIN, UINT8_MAX); break;

uspace/srv/audio/hound/audio_format.h

@@ -71 +71 @@
         return audio_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,

uspace/srv/audio/hound/audio_source.c

@@ -120 +120 @@
                 return ENOTSUP;
         }
+        const size_t src_frame_size = audio_format_frame_size(&source->format);
+        const size_t dst_frames = size / audio_format_frame_size(f);
+
         if (source->available_data.position == NULL ||
             source->available_data.size == 0) {
                 int ret = EOVERFLOW; /* In fact this is underflow... */
                 if (source->update_available_data)
-                        ret = source->update_available_data(source, size);
+                        ret = source->update_available_data(source,
+                            dst_frames * src_frame_size);
                 if (ret != EOK) {
                         log_debug("No data to add to %p(%zu)", buffer, size);
@@ -131 +135 @@
         }
 
-        const size_t real_size = min(size, source->available_data.size);
-        const int ret =
-            audio_format_mix(buffer, source->available_data.position, real_size, f);
+        const int ret = audio_format_convert_and_mix(buffer, size,
+               source->available_data.position, source->available_data.size,
+               &source->format, f);
         if (ret != EOK) {
-                log_debug("Mixing failed %p <= %p, %zu",
-                    buffer, source->available_data.position, real_size);
+                log_debug("Mixing failed %p <= %p, frames: %zu",
+                    buffer, source->available_data.position, dst_frames);
                 return ret;
         }
 
-        source->available_data.position += real_size;
-        source->available_data.size -= real_size;
-
-//      log_verbose("Mixing successful %p <= %p, %zu",
-//          buffer, source->available_data.position, real_size);
-
-        buffer += real_size;
-        size -= real_size;
-
-        //TODO update data again
-        if (size)
-                log_warning("not enough data");
-
+        source->available_data.position += (dst_frames * src_frame_size);
+        source->available_data.size -= (dst_frames * src_frame_size);
         return EOK;
 }