frsh: Export information about the last RTP contract and VRES

[frescor/ffmpeg.git] / libavcodec / mpegaudio.h

diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h
index 6dd43fce2e6172c0c2ed685bf3a429161ec4bfdf..65a2283fba76a5f2b630c16c577ad4086c845d38 100644 (file)
--- a/libavcodec/mpegaudio.h
+++ b/libavcodec/mpegaudio.h
@@ -19,10 +19,19 @@
  */
 
 /**
- * @file mpegaudio.h
+ * @file libavcodec/mpegaudio.h
  * mpeg audio declarations for both encoder and decoder.
  */
 
+#ifndef AVCODEC_MPEGAUDIO_H
+#define AVCODEC_MPEGAUDIO_H
+
+#include "avcodec.h"
+#include "get_bits.h"
+#include "dsputil.h"
+
+#define CONFIG_AUDIO_NONSHORT 0
+
 /* max frame size, in samples */
 #define MPA_FRAME_SIZE 1152
 
@@ -42,10 +51,9 @@
 #define SAME_HEADER_MASK \
    (0xffe00000 | (3 << 17) | (0xf << 12) | (3 << 10) | (3 << 19))
 
-/* define USE_HIGHPRECISION to have a bit exact (but slower) mpeg
-   audio decoder */
+#define MP3_MASK 0xFFFE0CCF
 
-#ifdef USE_HIGHPRECISION
+#if CONFIG_MPEGAUDIO_HP
 #define FRAC_BITS   23   /* fractional bits for sb_samples and dct */
 #define WFRAC_BITS  16   /* fractional bits for window */
 #else
@@ -53,16 +61,22 @@
 #define WFRAC_BITS  14   /* fractional bits for window */
 #endif
 
-#if defined(USE_HIGHPRECISION) && defined(CONFIG_AUDIO_NONSHORT)
+#define FRAC_ONE    (1 << FRAC_BITS)
+
+#define FIX(a)   ((int)((a) * FRAC_ONE))
+
+#if CONFIG_MPEGAUDIO_HP && CONFIG_AUDIO_NONSHORT
 typedef int32_t OUT_INT;
 #define OUT_MAX INT32_MAX
 #define OUT_MIN INT32_MIN
 #define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 31)
+#define OUT_FMT SAMPLE_FMT_S32
 #else
 typedef int16_t OUT_INT;
 #define OUT_MAX INT16_MAX
 #define OUT_MIN INT16_MIN
 #define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
+#define OUT_FMT SAMPLE_FMT_S16
 #endif
 
 #if FRAC_BITS <= 15
@@ -71,8 +85,58 @@ typedef int16_t MPA_INT;
 typedef int32_t MPA_INT;
 #endif
 
-int l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
-int mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate);
+#define BACKSTEP_SIZE 512
+#define EXTRABYTES 24
+
+struct GranuleDef;
+
+#define MPA_DECODE_HEADER \
+    int frame_size; \
+    int error_protection; \
+    int layer; \
+    int sample_rate; \
+    int sample_rate_index; /* between 0 and 8 */ \
+    int bit_rate; \
+    int nb_channels; \
+    int mode; \
+    int mode_ext; \
+    int lsf;
+
+typedef struct MPADecodeHeader {
+  MPA_DECODE_HEADER
+} MPADecodeHeader;
+
+typedef struct MPADecodeContext {
+    MPA_DECODE_HEADER
+    DECLARE_ALIGNED_8(uint8_t, last_buf[2*BACKSTEP_SIZE + EXTRABYTES]);
+    int last_buf_size;
+    /* next header (used in free format parsing) */
+    uint32_t free_format_next_header;
+    GetBitContext gb;
+    GetBitContext in_gb;
+    DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512 * 2]);
+    int synth_buf_offset[MPA_MAX_CHANNELS];
+    DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]);
+    int32_t mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
+#ifdef DEBUG
+    int frame_count;
+#endif
+    void (*compute_antialias)(struct MPADecodeContext *s, struct GranuleDef *g);
+    int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
+    int dither_state;
+    int error_recognition;
+    AVCodecContext* avctx;
+} MPADecodeContext;
+
+/* layer 3 huffman tables */
+typedef struct HuffTable {
+    int xsize;
+    const uint8_t *bits;
+    const uint16_t *codes;
+} HuffTable;
+
+int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
+int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bitrate);
 void ff_mpa_synth_init(MPA_INT *window);
 void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
                          MPA_INT *window, int *dither_state,
@@ -95,3 +159,5 @@ static inline int ff_mpa_check_header(uint32_t header){
         return -1;
     return 0;
 }
+
+#endif /* AVCODEC_MPEGAUDIO_H */