Bink Audio decoder

Originally committed as revision 21570 to svn://svn.ffmpeg.org/ffmpeg/trunk

4 files changed, 308 insertions, 1 deletions

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 94c93dd644..3009aa7f3a 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -65,6 +65,8 @@ OBJS-$(CONFIG_AURA2_DECODER)           += aura.o
 OBJS-$(CONFIG_AVS_DECODER)             += avs.o
 OBJS-$(CONFIG_BETHSOFTVID_DECODER)     += bethsoftvideo.o
 OBJS-$(CONFIG_BFI_DECODER)             += bfi.o
+OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER)   += binkaudio.o wma.o
+OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER)  += binkaudio.o wma.o
 OBJS-$(CONFIG_BMP_DECODER)             += bmp.o msrledec.o
 OBJS-$(CONFIG_BMP_ENCODER)             += bmpenc.o
 OBJS-$(CONFIG_C93_DECODER)             += c93.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 6e2d73dee0..ec9963448c 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -213,6 +213,8 @@ void avcodec_register_all(void)
     REGISTER_DECODER (APE, ape);
     REGISTER_DECODER (ATRAC1, atrac1);
     REGISTER_DECODER (ATRAC3, atrac3);
+    REGISTER_DECODER (BINKAUDIO_DCT, binkaudio_dct);
+    REGISTER_DECODER (BINKAUDIO_RDFT, binkaudio_rdft);
     REGISTER_DECODER (COOK, cook);
     REGISTER_DECODER (DCA, dca);
     REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index e5c86fde44..eafd642123 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -30,7 +30,7 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVCODEC_VERSION_MAJOR 52
-#define LIBAVCODEC_VERSION_MINOR 49
+#define LIBAVCODEC_VERSION_MINOR 50
 #define LIBAVCODEC_VERSION_MICRO  0
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c
new file mode 100644
index 0000000000..8704af7358
--- /dev/null
+++ b/libavcodec/binkaudio.c
@@ -0,0 +1,303 @@
+/*
+ * Bink Audio decoder
+ * Copyright (c) 2007-2010 Peter Ross (pross@xvid.org)
+ * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * 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
+ */
+
+/**
+ * @file libavcodec/binkaudio.c
+ * Bink Audio decoder
+ *
+ * Technical details here:
+ *  http://wiki.multimedia.cx/index.php?title=Bink_Audio
+ */
+
+#include "avcodec.h"
+#define ALT_BITSTREAM_READER_LE
+#include "get_bits.h"
+#include "dsputil.h"
+extern const uint16_t ff_wma_critical_freqs[25];
+
+#define MAX_CHANNELS 2
+#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
+
+typedef struct {
+    AVCodecContext *avctx;
+    GetBitContext gb;
+    DSPContext dsp;
+    int first;
+    int channels;
+    int frame_len;          ///< transform size (samples)
+    int overlap_len;        ///< overlap size (samples)
+    int block_size;
+    int num_bands;
+    unsigned int *bands;
+    float root;
+    DECLARE_ALIGNED_16(FFTSample, coeffs[BINK_BLOCK_MAX_SIZE]);
+    DECLARE_ALIGNED_16(short, previous[BINK_BLOCK_MAX_SIZE / 16]);  ///< coeffs from previous audio block
+    float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
+    union {
+        RDFTContext rdft;
+        DCTContext dct;
+    } trans;
+} BinkAudioContext;
+
+
+static av_cold int decode_init(AVCodecContext *avctx)
+{
+    BinkAudioContext *s = avctx->priv_data;
+    int sample_rate = avctx->sample_rate;
+    int sample_rate_half;
+    int i;
+    int frame_len_bits;
+
+    s->avctx = avctx;
+    dsputil_init(&s->dsp, avctx);
+
+    /* determine frame length */
+    if (avctx->sample_rate < 22050) {
+        frame_len_bits = 9;
+    } else if (avctx->sample_rate < 44100) {
+        frame_len_bits = 10;
+    } else {
+        frame_len_bits = 11;
+    }
+    s->frame_len = 1 << frame_len_bits;
+
+    if (s->channels > MAX_CHANNELS) {
+        av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
+        return -1;
+    }
+
+    if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
+        // audio is already interleaved for the RDFT format variant
+        sample_rate  *= avctx->channels;
+        s->frame_len *= avctx->channels;
+        s->channels = 1;
+        if (avctx->channels == 2)
+            frame_len_bits++;
+    } else {
+        s->channels = avctx->channels;
+    }
+
+    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);
+
+    /* calculate number of bands */
+    for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
+        if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
+            break;
+
+    s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
+    if (!s->bands)
+        return AVERROR(ENOMEM);
+
+    /* populate bands data */
+    s->bands[0] = 1;
+    for (i = 1; i < s->num_bands; i++)
+        s->bands[i] = ff_wma_critical_freqs[i - 1] * (s->frame_len / 2) / sample_rate_half;
+    s->bands[s->num_bands] = s->frame_len / 2;
+
+    s->first = 1;
+    avctx->sample_fmt = SAMPLE_FMT_S16;
+
+    for (i = 0; i < s->channels; i++)
+        s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
+
+    if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
+        ff_rdft_init(&s->trans.rdft, frame_len_bits, IRIDFT);
+    else
+        ff_dct_init(&s->trans.dct, frame_len_bits, 0);
+
+    return 0;
+}
+
+static float get_float(GetBitContext *gb)
+{
+    int power = get_bits(gb, 5);
+    float f = ldexpf(get_bits_long(gb, 23), power - 23);
+    if (get_bits1(gb))
+        f = -f;
+    return f;
+}
+
+static const uint8_t rle_length_tab[16] = {
+    2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
+};
+
+/**
+ * Decode Bink Audio block
+ * @param[out] out Output buffer (must contain s->block_size elements)
+ */
+static void decode_block(BinkAudioContext *s, short *out, int use_dct)
+{
+    int ch, i, j, k;
+    float q, quant[25];
+    int width, coeff;
+    GetBitContext *gb = &s->gb;
+
+    if (use_dct)
+        skip_bits(gb, 2);
+
+    for (ch = 0; ch < s->channels; ch++) {
+        FFTSample *coeffs = s->coeffs_ptr[ch];
+        q = 0.0f;
+        coeffs[0] = get_float(gb) * s->root;
+        coeffs[1] = get_float(gb) * s->root;
+
+        for (i = 0; i < s->num_bands; i++) {
+            /* constant is result of 0.066399999/log10(M_E) */
+            int value = get_bits(gb, 8);
+            quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
+        }
+
+        // find band (k)
+        for (k = 0; s->bands[k] < 1; k++) {
+            q = quant[k];
+        }
+
+        // parse coefficients
+        i = 2;
+        while (i < s->frame_len) {
+            if (get_bits1(gb)) {
+                j = i + rle_length_tab[get_bits(gb, 4)] * 8;
+            } else {
+                j = i + 8;
+            }
+
+            j = FFMIN(j, s->frame_len);
+
+            width = get_bits(gb, 4);
+            if (width == 0) {
+                memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
+                i = j;
+                while (s->bands[k] * 2 < i)
+                    q = quant[k++];
+            } else {
+                while (i < j) {
+                    if (s->bands[k] * 2 == i)
+                        q = quant[k++];
+                    coeff = get_bits(gb, width);
+                    if (coeff) {
+                        if (get_bits1(gb))
+                            coeffs[i] = -q * coeff;
+                        else
+                            coeffs[i] =  q * coeff;
+                    } else {
+                        coeffs[i] = 0.0f;
+                    }
+                    i++;
+                }
+            }
+        }
+
+        if (use_dct)
+            ff_dct_calc (&s->trans.dct,  coeffs);
+        else
+            ff_rdft_calc(&s->trans.rdft, coeffs);
+    }
+
+    s->dsp.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, s->frame_len, s->channels);
+
+    if (!s->first) {
+        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;
+        }
+    }
+
+    memcpy(s->previous, out + s->block_size,
+           s->overlap_len * s->channels * sizeof(*out));
+
+    s->first = 0;
+}
+
+static av_cold int decode_end(AVCodecContext *avctx)
+{
+    BinkAudioContext * s = avctx->priv_data;
+    av_freep(&s->bands);
+    if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
+        ff_rdft_end(&s->trans.rdft);
+    else
+        ff_dct_end(&s->trans.dct);
+    return 0;
+}
+
+static void get_bits_align32(GetBitContext *s)
+{
+    int n = (-get_bits_count(s)) & 31;
+    if (n) skip_bits(s, n);
+}
+
+static int decode_frame(AVCodecContext *avctx,
+                        void *data, int *data_size,
+                        AVPacket *avpkt)
+{
+    BinkAudioContext *s = avctx->priv_data;
+    const uint8_t *buf  = avpkt->data;
+    int buf_size        = avpkt->size;
+    short *samples      = data;
+    short *samples_end  = (short*)((uint8_t*)data + *data_size);
+    int reported_size;
+    GetBitContext *gb = &s->gb;
+
+    init_get_bits(gb, buf, buf_size * 8);
+
+    reported_size = get_bits_long(gb, 32);
+    while (get_bits_count(gb) / 8 < buf_size &&
+           samples + s->block_size <= samples_end) {
+        decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);
+        samples += s->block_size;
+        get_bits_align32(gb);
+    }
+
+    *data_size = (uint8_t*)samples - (uint8_t*)data;
+    if (reported_size != *data_size) {
+        av_log(avctx, AV_LOG_WARNING, "reported data size (%d) does not match output data size (%d)\n",
+             reported_size, *data_size);
+    }
+    return buf_size;
+}
+
+AVCodec binkaudio_rdft_decoder = {
+    "binkaudio_rdft",
+    CODEC_TYPE_AUDIO,
+    CODEC_ID_BINKAUDIO_RDFT,
+    sizeof(BinkAudioContext),
+    decode_init,
+    NULL,
+    decode_end,
+    decode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
+};
+
+AVCodec binkaudio_dct_decoder = {
+    "binkaudio_dct",
+    CODEC_TYPE_AUDIO,
+    CODEC_ID_BINKAUDIO_DCT,
+    sizeof(BinkAudioContext),
+    decode_init,
+    NULL,
+    decode_end,
+    decode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
+};