Merge commit '1b3439b3055b083df51d7f7838ecc6b3f708b15c'

* commit '1b3439b3055b083df51d7f7838ecc6b3f708b15c':
  mpegvideo: move frame size dependent memory management to separate functions
  configure: add --toolchain option
  configure: Make the smoothstreaming muxer enable the ismv muxer
  smoothstreaming: Export the mp4 codec tags
  mov: check for EOF in long lasting loops
  avcodec: cleanup utils.c
  binkaudio: remove unneeded GET_BITS_SAFE macro
  binkaudio: use float sample format
  binkaudio: use a different value for the coefficient scale for the DCT codec

Conflicts:
	configure
	libavcodec/mpegvideo.c
	libavcodec/utils.c
	libavformat/Makefile

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

1 files changed, 28 insertions, 49 deletions

diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c
index 662c6f29d6..000895b9e3 100644
--- a/libavcodec/binkaudio.c
+++ b/libavcodec/binkaudio.c
@@ -47,8 +47,6 @@ static float quant_table[96];
 typedef struct {
     AVFrame frame;
     GetBitContext gb;
-    DSPContext dsp;
-    FmtConvertContext fmt_conv;
     int version_b;          ///< Bink version 'b'
     int first;
     int channels;
@@ -59,10 +57,7 @@ typedef struct {
     unsigned int *bands;
     float root;
     DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
-    DECLARE_ALIGNED(16, int16_t, previous)[BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block
-    DECLARE_ALIGNED(16, int16_t, current)[BINK_BLOCK_MAX_SIZE / 16];
-    float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
-    float *prev_ptr[MAX_CHANNELS];   ///< pointers to the overlap points in the coeffs array
+    float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block
     uint8_t *packet_buffer;
     union {
         RDFTContext rdft;
@@ -79,9 +74,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
     int i;
     int frame_len_bits;
 
-    ff_dsputil_init(&s->dsp, avctx);
-    ff_fmt_convert_init(&s->fmt_conv, avctx);
-
     /* determine frame length */
     if (avctx->sample_rate < 22050) {
         frame_len_bits = 9;
@@ -100,19 +92,24 @@ static av_cold int decode_init(AVCodecContext *avctx)
 
     if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
         // audio is already interleaved for the RDFT format variant
+        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
         sample_rate  *= avctx->channels;
         s->channels = 1;
         if (!s->version_b)
             frame_len_bits += av_log2(avctx->channels);
     } else {
         s->channels = avctx->channels;
+        avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
     }
 
     s->frame_len     = 1 << frame_len_bits;
     s->overlap_len   = s->frame_len / 16;
     s->block_size    = (s->frame_len - s->overlap_len) * s->channels;
     sample_rate_half = (sample_rate + 1) / 2;
-    s->root          = 2.0 / sqrt(s->frame_len);
+    if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
+        s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
+    else
+        s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
     for (i = 0; i < 96; i++) {
         /* constant is result of 0.066399999/log10(M_E) */
         quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
@@ -134,12 +131,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
     s->bands[s->num_bands] = s->frame_len;
 
     s->first = 1;
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
-
-    for (i = 0; i < s->channels; i++) {
-        s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
-        s->prev_ptr[i]   = s->coeffs_ptr[i] + s->frame_len - s->overlap_len;
-    }
 
     if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
         ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
@@ -167,18 +158,12 @@ static const uint8_t rle_length_tab[16] = {
     2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
 };
 
-#define GET_BITS_SAFE(out, nbits) do {  \
-    if (get_bits_left(gb) < nbits)      \
-        return AVERROR_INVALIDDATA;     \
-    out = get_bits(gb, nbits);          \
-} while (0)
-
 /**
  * Decode Bink Audio block
  * @param[out] out Output buffer (must contain s->block_size elements)
  * @return 0 on success, negative error code on failure
  */
-static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
+static int decode_block(BinkAudioContext *s, float **out, int use_dct)
 {
     int ch, i, j, k;
     float q, quant[25];
@@ -189,7 +174,8 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
         skip_bits(gb, 2);
 
     for (ch = 0; ch < s->channels; ch++) {
-        FFTSample *coeffs = s->coeffs_ptr[ch];
+        FFTSample *coeffs = out[ch];
+
         if (s->version_b) {
             if (get_bits_left(gb) < 64)
                 return AVERROR_INVALIDDATA;
@@ -218,10 +204,9 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
             if (s->version_b) {
                 j = i + 16;
             } else {
-                int v;
-                GET_BITS_SAFE(v, 1);
+                int v = get_bits1(gb);
                 if (v) {
-                    GET_BITS_SAFE(v, 4);
+                    v = get_bits(gb, 4);
                     j = i + rle_length_tab[v] * 8;
                 } else {
                     j = i + 8;
@@ -230,7 +215,7 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
 
             j = FFMIN(j, s->frame_len);
 
-            GET_BITS_SAFE(width, 4);
+            width = get_bits(gb, 4);
             if (width == 0) {
                 memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
                 i = j;
@@ -240,10 +225,10 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
                 while (i < j) {
                     if (s->bands[k] == i)
                         q = quant[k++];
-                    GET_BITS_SAFE(coeff, width);
+                    coeff = get_bits(gb, width);
                     if (coeff) {
                         int v;
-                        GET_BITS_SAFE(v, 1);
+                        v = get_bits1(gb);
                         if (v)
                             coeffs[i] = -q * coeff;
                         else
@@ -259,30 +244,24 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
         if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
             coeffs[0] /= 0.5;
             s->trans.dct.dct_calc(&s->trans.dct,  coeffs);
-            s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
         }
         else if (CONFIG_BINKAUDIO_RDFT_DECODER)
             s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
     }
 
-    s->fmt_conv.float_to_int16_interleave(s->current,
-                                          (const float **)s->prev_ptr,
-                                          s->overlap_len, s->channels);
-    s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,
-                                          s->frame_len - s->overlap_len,
-                                          s->channels);
-
-    if (!s->first) {
+    for (ch = 0; ch < s->channels; ch++) {
+        int j;
         int count = s->overlap_len * s->channels;
-        int shift = av_log2(count);
-        for (i = 0; i < count; i++) {
-            out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;
+        if (!s->first) {
+            j = ch;
+            for (i = 0; i < s->overlap_len; i++, j += s->channels)
+                out[ch][i] = (s->previous[ch][i] * (count - j) +
+                                      out[ch][i] *          j) / count;
         }
+        memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
+               s->overlap_len * sizeof(*s->previous[ch]));
     }
 
-    memcpy(s->previous, s->current,
-           s->overlap_len * s->channels * sizeof(*s->previous));
-
     s->first = 0;
 
     return 0;
@@ -311,7 +290,6 @@ static int decode_frame(AVCodecContext *avctx, void *data,
                         int *got_frame_ptr, AVPacket *avpkt)
 {
     BinkAudioContext *s = avctx->priv_data;
-    int16_t *samples;
     GetBitContext *gb = &s->gb;
     int ret, consumed = 0;
 
@@ -339,19 +317,20 @@ static int decode_frame(AVCodecContext *avctx, void *data,
     }
 
     /* get output buffer */
-    s->frame.nb_samples = s->block_size / avctx->channels;
+    s->frame.nb_samples = s->frame_len;
     if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
-    samples = (int16_t *)s->frame.data[0];
 
-    if (decode_block(s, samples, avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
+    if (decode_block(s, (float **)s->frame.extended_data,
+                     avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
         av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
         return AVERROR_INVALIDDATA;
     }
     get_bits_align32(gb);
 
+    s->frame.nb_samples = s->block_size / avctx->channels;
     *got_frame_ptr   = 1;
     *(AVFrame *)data = s->frame;