X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fsonic.c;h=00ae83587373502f0064dd70af1546387b28182e;hb=5e12701c5c881722f28379a15a60d5a20698b10f;hp=79641ae5139c26d1ec0c842349b20201cdc915c4;hpb=98a6fff98c3f2e963783e52fb193529bdc14e040;p=ffmpeg

diff --git a/libavcodec/sonic.c b/libavcodec/sonic.c
index 79641ae5139..00ae8358737 100644
--- a/libavcodec/sonic.c
+++ b/libavcodec/sonic.c
@@ -19,11 +19,11 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 #include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
 #include "golomb.h"
 
 /**
- * @file sonic.c
+ * @file libavcodec/sonic.c
  * Simple free lossless/lossy audio codec
  * Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk)
  * Written and designed by Alex Beregszaszi
@@ -408,7 +408,7 @@ static int predictor_calc_error(int *k, int *state, int order, int error)
     return x;
 }
 
-#ifdef CONFIG_ENCODERS
+#if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER
 // Heavily modified Levinson-Durbin algorithm which
 // copes better with quantization, and calculates the
 // actual whitened result as it goes.
@@ -479,13 +479,12 @@ static void modified_levinson_durbin(int *window, int window_entries,
 
     av_free(state);
 }
-#endif /* CONFIG_ENCODERS */
+#endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */
 
-static int samplerate_table[] =
+static const int samplerate_table[] =
     { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };
 
-#ifdef CONFIG_ENCODERS
-
+#if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER
 static inline int code_samplerate(int samplerate)
 {
     switch (samplerate)
@@ -748,9 +747,9 @@ static int sonic_encode_frame(AVCodecContext *avctx,
     flush_put_bits(&pb);
     return (put_bits_count(&pb)+7)/8;
 }
-#endif //CONFIG_ENCODERS
+#endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */
 
-#ifdef CONFIG_DECODERS
+#if CONFIG_SONIC_DECODER
 static av_cold int sonic_decode_init(AVCodecContext *avctx)
 {
     SonicContext *s = avctx->priv_data;
@@ -828,6 +827,7 @@ static av_cold int sonic_decode_init(AVCodecContext *avctx)
     }
     s->int_samples = av_mallocz(4* s->frame_size);
 
+    avctx->sample_fmt = SAMPLE_FMT_S16;
     return 0;
 }
 
@@ -851,8 +851,10 @@ static av_cold int sonic_decode_close(AVCodecContext *avctx)
 
 static int sonic_decode_frame(AVCodecContext *avctx,
                             void *data, int *data_size,
-                            const uint8_t *buf, int buf_size)
+                            AVPacket *avpkt)
 {
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
     SonicContext *s = avctx->priv_data;
     GetBitContext gb;
     int i, quant, ch, j;
@@ -934,9 +936,9 @@ static int sonic_decode_frame(AVCodecContext *avctx,
 
     return (get_bits_count(&gb)+7)/8;
 }
-#endif
+#endif /* CONFIG_SONIC_DECODER */
 
-#ifdef CONFIG_ENCODERS
+#if CONFIG_SONIC_ENCODER
 AVCodec sonic_encoder = {
     "sonic",
     CODEC_TYPE_AUDIO,
@@ -946,8 +948,11 @@ AVCodec sonic_encoder = {
     sonic_encode_frame,
     sonic_encode_close,
     NULL,
+    .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
 };
+#endif
 
+#if CONFIG_SONIC_LS_ENCODER
 AVCodec sonic_ls_encoder = {
     "sonicls",
     CODEC_TYPE_AUDIO,
@@ -957,10 +962,11 @@ AVCodec sonic_ls_encoder = {
     sonic_encode_frame,
     sonic_encode_close,
     NULL,
+    .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"),
 };
 #endif
 
-#ifdef CONFIG_DECODERS
+#if CONFIG_SONIC_DECODER
 AVCodec sonic_decoder = {
     "sonic",
     CODEC_TYPE_AUDIO,
@@ -970,5 +976,6 @@ AVCodec sonic_decoder = {
     NULL,
     sonic_decode_close,
     sonic_decode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
 };
 #endif