Fix rc_eq mem leak.

[ffmpeg] / libavcodec / audioconvert.c

diff --git a/libavcodec/audioconvert.c b/libavcodec/audioconvert.c
index cab743f9d165f7b523778e9afff5fac474c3d446..99029af31b2f5d041fe9ad3038c311b418535577 100644 (file)
--- a/libavcodec/audioconvert.c
+++ b/libavcodec/audioconvert.c
@@ -17,7 +17,6 @@
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
  */
 
 /**
@@ -27,28 +26,95 @@
  */
 
 #include "avcodec.h"
+#include "audioconvert.h"
+
+typedef struct SampleFmtInfo {
+    const char *name;
+    int bits;
+} SampleFmtInfo;
+
+/** this table gives more information about formats */
+static const SampleFmtInfo sample_fmt_info[SAMPLE_FMT_NB] = {
+    [SAMPLE_FMT_U8]  = { .name = "u8",  .bits = 8 },
+    [SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 },
+    [SAMPLE_FMT_S24] = { .name = "s24", .bits = 24 },
+    [SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 },
+    [SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 }
+};
+
+const char *avcodec_get_sample_fmt_name(int sample_fmt)
+{
+    if (sample_fmt < 0 || sample_fmt >= SAMPLE_FMT_NB)
+        return NULL;
+    return sample_fmt_info[sample_fmt].name;
+}
+
+enum SampleFormat avcodec_get_sample_fmt(const char* name)
+{
+    int i;
+
+    for (i=0; i < SAMPLE_FMT_NB; i++)
+        if (!strcmp(sample_fmt_info[i].name, name))
+            return i;
+    return SAMPLE_FMT_NONE;
+}
+
+void avcodec_sample_fmt_string (char *buf, int buf_size, int sample_fmt)
+{
+    /* print header */
+    if (sample_fmt < 0)
+        snprintf (buf, buf_size, "name  " " depth");
+    else if (sample_fmt < SAMPLE_FMT_NB) {
+        SampleFmtInfo info= sample_fmt_info[sample_fmt];
+        snprintf (buf, buf_size, "%-6s" "   %2d ", info.name, info.bits);
+    }
+}
+
+struct AVAudioConvert {
+    int in_channels, out_channels;
+    int fmt_pair;
+};
+
+AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
+                                       enum SampleFormat in_fmt, int in_channels,
+                                       const const float *matrix, int flags)
+{
+    AVAudioConvert *ctx;
+    if (in_channels!=out_channels)
+        return NULL;  /* FIXME: not supported */
+    ctx = av_malloc(sizeof(AVAudioConvert));
+    if (!ctx)
+        return NULL;
+    ctx->in_channels = in_channels;
+    ctx->out_channels = out_channels;
+    ctx->fmt_pair = out_fmt + SAMPLE_FMT_NB*in_fmt;
+    return ctx;
+}
+
+void av_audio_convert_free(AVAudioConvert *ctx)
+{
+    av_free(ctx);
+}
 
-int av_audio_convert(void *maybe_dspcontext_or_something_av_convert_specific,
-                     void *out[6], int out_stride[6], enum SampleFormat out_fmt,
-                     void * in[6], int  in_stride[6], enum SampleFormat  in_fmt, int len){
+int av_audio_convert(AVAudioConvert *ctx,
+                           void * const out[6], const int out_stride[6],
+                     const void * const  in[6], const int  in_stride[6], int len)
+{
     int ch;
-    const int isize= FFMIN( in_fmt+1, 4);
-    const int osize= FFMIN(out_fmt+1, 4);
-    const int fmt_pair= out_fmt + 5*in_fmt;
 
     //FIXME optimize common cases
 
-    for(ch=0; ch<6; ch++){
-        const int is=  in_stride[ch] * isize;
-        const int os= out_stride[ch] * osize;
+    for(ch=0; ch<ctx->out_channels; ch++){
+        const int is=  in_stride[ch];
+        const int os= out_stride[ch];
         uint8_t *pi=  in[ch];
         uint8_t *po= out[ch];
-        uint8_t *end= po + os;
+        uint8_t *end= po + os*len;
         if(!out[ch])
             continue;
 
 #define CONV(ofmt, otype, ifmt, expr)\
-if(fmt_pair == ofmt + 5*ifmt){\
+if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\
     do{\
         *(otype*)po = expr; pi += is; po += os;\
     }while(po < end);\