Merge remote branch 'qatar/master'

* qatar/master:
  mpegaudiodec: group #includes more sanely
  mpegaudio: remove #if 0 blocks
  ffmpeg.c: reset avoptions after each input/output file.
  ffmpeg.c: store per-output stream sws flags.
  mpegaudio: remove CONFIG_MPEGAUDIO_HP option
  mpegtsenc: Clear st->priv_data when freeing it
  udp: Fix receiving RTP data over multicast
  rtpproto: Remove an unused variable
  regtest: fix wma tests
NOT pulled:  mpegaudio: remove CONFIG_AUDIO_NONSHORT
  regtest: separate flags for encoding and decoding

Conflicts:
	ffmpeg.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 1 insertions, 94 deletions

diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c
index 5d420dca3e..d7bc89f837 100644
--- a/libavcodec/mpegaudiodec.c
+++ b/libavcodec/mpegaudiodec.c
@@ -28,18 +28,16 @@
 #include "avcodec.h"
 #include "get_bits.h"
 #include "dsputil.h"
+#include "mathops.h"
 
 /*
  * TODO:
- *  - in low precision mode, use more 16 bit multiplies in synth filter
  *  - test lsf / mpeg25 extensively.
  */
 
 #include "mpegaudio.h"
 #include "mpegaudiodecheader.h"
 
-#include "mathops.h"
-
 #if CONFIG_FLOAT
 #   define SHR(a,b)       ((a)*(1.0f/(1<<(b))))
 #   define compute_antialias compute_antialias_float
@@ -248,14 +246,6 @@ static inline int l3_unscale(int value, int exponent)
 
 static int dev_4_3_coefs[DEV_ORDER];
 
-#if 0 /* unused */
-static int pow_mult3[3] = {
-    POW_FIX(1.0),
-    POW_FIX(1.25992104989487316476),
-    POW_FIX(1.58740105196819947474),
-};
-#endif
-
 static av_cold void int_pow_init(void)
 {
     int i, a;
@@ -267,53 +257,6 @@ static av_cold void int_pow_init(void)
     }
 }
 
-#if 0 /* unused, remove? */
-/* return the mantissa and the binary exponent */
-static int int_pow(int i, int *exp_ptr)
-{
-    int e, er, eq, j;
-    int a, a1;
-
-    /* renormalize */
-    a = i;
-    e = POW_FRAC_BITS;
-    while (a < (1 << (POW_FRAC_BITS - 1))) {
-        a = a << 1;
-        e--;
-    }
-    a -= (1 << POW_FRAC_BITS);
-    a1 = 0;
-    for(j = DEV_ORDER - 1; j >= 0; j--)
-        a1 = POW_MULL(a, dev_4_3_coefs[j] + a1);
-    a = (1 << POW_FRAC_BITS) + a1;
-    /* exponent compute (exact) */
-    e = e * 4;
-    er = e % 3;
-    eq = e / 3;
-    a = POW_MULL(a, pow_mult3[er]);
-    while (a >= 2 * POW_FRAC_ONE) {
-        a = a >> 1;
-        eq++;
-    }
-    /* convert to float */
-    while (a < POW_FRAC_ONE) {
-        a = a << 1;
-        eq--;
-    }
-    /* now POW_FRAC_ONE <= a < 2 * POW_FRAC_ONE */
-#if POW_FRAC_BITS > FRAC_BITS
-    a = (a + (1 << (POW_FRAC_BITS - FRAC_BITS - 1))) >> (POW_FRAC_BITS - FRAC_BITS);
-    /* correct overflow */
-    if (a >= 2 * (1 << FRAC_BITS)) {
-        a = a >> 1;
-        eq++;
-    }
-#endif
-    *exp_ptr = eq;
-    return a;
-}
-#endif
-
 static av_cold int decode_init(AVCodecContext * avctx)
 {
     MPADecodeContext *s = avctx->priv_data;
@@ -540,24 +483,6 @@ static inline float round_sample(float *sum)
 
 #define MLSS(rt, ra, rb) rt-=(ra)*(rb)
 
-#elif FRAC_BITS <= 15
-
-static inline int round_sample(int *sum)
-{
-    int sum1;
-    sum1 = (*sum) >> OUT_SHIFT;
-    *sum &= (1<<OUT_SHIFT)-1;
-    return av_clip(sum1, OUT_MIN, OUT_MAX);
-}
-
-/* signed 16x16 -> 32 multiply add accumulate */
-#define MACS(rt, ra, rb) MAC16(rt, ra, rb)
-
-/* signed 16x16 -> 32 multiply */
-#define MULS(ra, rb) MUL16(ra, rb)
-
-#define MLSS(rt, ra, rb) MLS16(rt, ra, rb)
-
 #else
 
 static inline int round_sample(int64_t *sum)
@@ -624,8 +549,6 @@ void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window)
         v = ff_mpa_enwindow[i];
 #if CONFIG_FLOAT
         v *= 1.0 / (1LL<<(16 + FRAC_BITS));
-#elif WFRAC_BITS < 16
-        v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
 #endif
         window[i] = v;
         if ((i & 63) != 0)
@@ -652,8 +575,6 @@ static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
     OUT_INT *samples2;
 #if CONFIG_FLOAT
     float sum, sum2;
-#elif FRAC_BITS <= 15
-    int sum, sum2;
 #else
     int64_t sum, sum2;
 #endif
@@ -710,25 +631,11 @@ void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
 {
     register MPA_INT *synth_buf;
     int offset;
-#if FRAC_BITS <= 15
-    int32_t tmp[32];
-    int j;
-#endif
 
     offset = *synth_buf_offset;
     synth_buf = synth_buf_ptr + offset;
 
-#if FRAC_BITS <= 15
-    dct32(tmp, sb_samples);
-    for(j=0;j<32;j++) {
-        /* NOTE: can cause a loss in precision if very high amplitude
-           sound */
-        synth_buf[j] = av_clip_int16(tmp[j]);
-    }
-#else
     dct32(synth_buf, sb_samples);
-#endif
-
     apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
 
     offset = (offset - 32) & 511;