@@ -12,6 +12,7 @@ version <next>:

 - 16.8 floating point pcm decoder

 - 24.0 floating point pcm decoder

 - Apple Pixlet decoder

+- QDMC audio decoder

 version 3.2:

 - libopenmpt demuxer

@@ -1048,6 +1048,7 @@ following image formats are supported:

 @item PCM unsigned 32-bit little-endian  @tab  X  @tab  X

 @item PCM Zork               @tab     @tab  X

 @item QCELP / PureVoice      @tab     @tab  X

+@item QDesign Music Codec 1  @tab     @tab  X

 @item QDesign Music Codec 2  @tab     @tab  X

     @tab There are still some distortions.

 @item RealAudio 1.0 (14.4K)  @tab  X  @tab  X

@@ -466,6 +466,7 @@ OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o                     \

                                           celp_filters.o acelp_vectors.o \

                                           acelp_filters.o

 OBJS-$(CONFIG_QDM2_DECODER)            += qdm2.o

+OBJS-$(CONFIG_QDMC_DECODER)            += qdmc.o

 OBJS-$(CONFIG_QDRAW_DECODER)           += qdrw.o

 OBJS-$(CONFIG_QPEG_DECODER)            += qpeg.o

 OBJS-$(CONFIG_QTRLE_DECODER)           += qtrle.o

@@ -449,6 +449,7 @@ void avcodec_register_all(void)

     REGISTER_DECODER(PAF_AUDIO,         paf_audio);

     REGISTER_DECODER(QCELP,             qcelp);

     REGISTER_DECODER(QDM2,              qdm2);

+    REGISTER_DECODER(QDMC,              qdmc);

     REGISTER_ENCDEC (RA_144,            ra_144);

     REGISTER_DECODER(RA_288,            ra_288);

     REGISTER_DECODER(RALF,              ralf);

new file mode 100644

@@ -0,0 +1,783 @@

+/*

+ * QDMC compatible decoder

+ * Copyright (c) 2017 Paul B Mahol

+ *

+ * 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

+ */

+

+#include <math.h>

+#include <stddef.h>

+#include <stdio.h>

+

+#define BITSTREAM_READER_LE

+

+#include "libavutil/channel_layout.h"

+

+#include "avcodec.h"

+#include "bytestream.h"

+#include "get_bits.h"

+#include "internal.h"

+#include "fft.h"

+

+typedef struct QDMCTone {

+    uint8_t mode;

+    uint8_t phase;

+    uint8_t offset;

+    int16_t freq;

+    int16_t amplitude;

+} QDMCTone;

+

+typedef struct QDMCContext {

+    AVCodecContext *avctx;

+

+    uint8_t frame_bits;

+    int band_index;

+    int frame_size;

+    int subframe_size;

+    int fft_offset;

+    int buffer_offset;

+    int nb_channels;

+    int checksum_size;

+

+    uint8_t noise[2][19][16];

+    QDMCTone tones[5][8192];

+    int nb_tones[5];

+    int cur_tone[5];

+    float alt_sin[5][31];

+    float fft_buffer[4][8192 * 2];

+    float noise2_buffer[4096 * 2];

+    float noise_buffer[4096 * 2];

+    float buffer[2 * 32768];

+    float *buffer_ptr;

+    int rndval;

+

+    DECLARE_ALIGNED(32, FFTComplex, cmplx)[2][512];

+    FFTContext fft_ctx;

+} QDMCContext;

+

+static float sin_table[512];

+static VLC vtable[6];

+

+static const unsigned code_prefix[] = {

+    0x0, 0x1, 0x2, 0x3, 0x4, 0x6, 0x8, 0xA,

+    0xC, 0x10, 0x14, 0x18, 0x1C, 0x24, 0x2C, 0x34,

+    0x3C, 0x4C, 0x5C, 0x6C, 0x7C, 0x9C, 0xBC, 0xDC,

+    0xFC, 0x13C, 0x17C, 0x1BC, 0x1FC, 0x27C, 0x2FC, 0x37C,

+    0x3FC, 0x4FC, 0x5FC, 0x6FC, 0x7FC, 0x9FC, 0xBFC, 0xDFC,

+    0xFFC, 0x13FC, 0x17FC, 0x1BFC, 0x1FFC, 0x27FC, 0x2FFC, 0x37FC,

+    0x3FFC, 0x4FFC, 0x5FFC, 0x6FFC, 0x7FFC, 0x9FFC, 0xBFFC, 0xDFFC,

+    0xFFFC, 0x13FFC, 0x17FFC, 0x1BFFC, 0x1FFFC, 0x27FFC, 0x2FFFC, 0x37FFC,

+    0x3FFFC

+};

+

+static const float amplitude_tab[64] = {

+    1.18750000f, 1.68359380f, 2.37500000f, 3.36718750f, 4.75000000f,

+    6.73437500f, 9.50000000f, 13.4687500f, 19.0000000f, 26.9375000f,

+    38.0000000f, 53.8750000f, 76.0000000f, 107.750000f, 152.000000f,

+    215.500000f, 304.000000f, 431.000000f, 608.000000f, 862.000000f,

+    1216.00000f, 1724.00000f, 2432.00000f, 3448.00000f, 4864.00000f,

+    6896.00000f, 9728.00000f, 13792.0000f, 19456.0000f, 27584.0000f,

+    38912.0000f, 55168.0000f, 77824.0000f, 110336.000f, 155648.000f,

+    220672.000f, 311296.000f, 441344.000f, 622592.000f, 882688.000f,

+    1245184.00f, 1765376.00f, 2490368.00f, 3530752.00f, 4980736.00f,

+    7061504.00f, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,

+};

+

+static const uint16_t qdmc_nodes[112] = {

+    0, 1, 2, 4, 6, 8, 12, 16, 24, 32, 48, 56, 64,

+    80, 96, 120, 144, 176, 208, 240, 256,

+    0, 2, 4, 8, 16, 24, 32, 48, 56, 64, 80, 104,

+    128, 160, 208, 256, 0, 0, 0, 0, 0,

+    0, 2, 4, 8, 16, 32, 48, 64, 80, 112, 160, 208,

+    256, 0, 0, 0, 0, 0, 0, 0, 0,

+    0, 4, 8, 16, 32, 48, 64, 96, 144, 208, 256,

+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0,

+    0, 4, 16, 32, 64, 256, 0, 0, 0, 0, 0, 0, 0, 0,

+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0

+};

+

+static const uint8_t noise_bands_size[] = {

+    19, 14, 11, 9, 4, 2, 0

+};

+

+static const uint8_t noise_bands_selector[] = {

+    4, 3, 2, 1, 0, 0, 0,

+};

+

+static const uint8_t noise_value_bits[] = {

+    12, 7, 9, 7, 10, 9, 11, 9, 9, 2, 9, 9, 9, 9,

+    9, 3, 9, 10, 10, 12, 2, 3, 3, 5, 5, 6, 7,

+};

+

+static const uint8_t noise_value_symbols[] = {

+    0, 10, 11, 12, 13, 14, 15, 16, 18, 1, 20, 22, 24,

+    26, 28, 2, 30, 32, 34, 36, 3, 4, 5, 6, 7, 8, 9,

+};

+

+static const uint16_t noise_value_codes[] = {

+    0xC7A, 0x002, 0x0FA, 0x03A, 0x35A, 0x1C2, 0x07A, 0x1FA,

+    0x17A, 0x000, 0x0DA, 0x142, 0x0C2, 0x042, 0x1DA, 0x001,

+    0x05A, 0x15A, 0x27A, 0x47A, 0x003, 0x005, 0x006, 0x012,

+    0x00A, 0x022, 0x01A,

+};

+

+static const uint8_t noise_segment_length_bits[] = {

+    10, 8, 5, 1, 2, 4, 4, 4, 6, 7, 9, 10,

+};

+

+static const uint8_t noise_segment_length_symbols[] = {

+    0, 13, 17, 1, 2, 3, 4, 5, 6, 7, 8, 9,

+};

+

+static const uint16_t noise_segment_length_codes[] = {

+    0x30B, 0x8B, 0x1B, 0x0, 0x1, 0x3, 0x7, 0xF, 0x2b, 0x4B, 0xB, 0x10B,

+};

+

+static const uint8_t freq_diff_bits[] = {

+    18, 2, 4, 4, 5, 4, 4, 5, 5, 4, 5, 5, 5, 5, 6, 6, 6, 6, 6, 7, 7, 6,

+    7, 6, 6, 6, 7, 7, 7, 7, 7, 8, 9, 9, 8, 9, 11, 11, 12, 12, 13, 12,

+    14, 15, 18, 16, 17,

+};

+

+static const uint32_t freq_diff_codes[] = {

+    0x2AD46, 0x1, 0x0, 0x3, 0xC, 0xA, 0x7, 0x18, 0x12, 0xE, 0x4, 0x16,

+    0xF, 0x1C, 0x8, 0x22, 0x26, 0x2, 0x3B, 0x34, 0x74, 0x1F, 0x14, 0x2B,

+    0x1B, 0x3F, 0x28, 0x54, 0x6, 0x4B, 0xB, 0x68, 0xE8, 0x46, 0xC6, 0x1E8,

+    0x146, 0x346, 0x546, 0x746, 0x1D46, 0xF46, 0xD46, 0x6D46, 0xAD46, 0x2D46,

+    0x1AD46,

+};

+

+static const uint8_t amplitude_bits[] = {

+    13, 7, 8, 9, 10, 10, 10, 10, 10, 9, 8, 7, 6,

+    5, 4, 3, 3, 2, 3, 3, 4, 5, 7, 8, 9, 11, 12, 13,

+};

+

+static const uint16_t amplitude_codes[] = {

+    0x1EC6, 0x6, 0xC2, 0x142, 0x242, 0x246, 0xC6, 0x46, 0x42, 0x146, 0xA2,

+    0x62, 0x26, 0x16, 0xE, 0x5, 0x4, 0x3, 0x0, 0x1, 0xA, 0x12, 0x2, 0x22,

+    0x1C6, 0x2C6, 0x6C6, 0xEC6,

+};

+

+static const uint8_t amplitude_diff_bits[] = {

+    8, 2, 1, 3, 4, 5, 6, 7, 8,

+};

+

+static const uint8_t amplitude_diff_codes[] = {

+    0xFE, 0x0, 0x1, 0x2, 0x6, 0xE, 0x1E, 0x3E, 0x7E,

+};

+

+static const uint8_t phase_diff_bits[] = {

+    6, 2, 2, 4, 4, 6, 5, 4, 2,

+};

+

+static const uint8_t phase_diff_codes[] = {

+    0x35, 0x2, 0x0, 0x1, 0xD, 0x15, 0x5, 0x9, 0x3,

+};

+

+#define INIT_VLC_STATIC_LE(vlc, nb_bits, nb_codes,                 \

+                           bits, bits_wrap, bits_size,             \

+                           codes, codes_wrap, codes_size,          \

+                           symbols, symbols_wrap, symbols_size,    \

+                           static_size)                            \

+    do {                                                           \

+        static VLC_TYPE table[static_size][2];                     \

+        (vlc)->table           = table;                            \

+        (vlc)->table_allocated = static_size;                      \

+        ff_init_vlc_sparse(vlc, nb_bits, nb_codes,                 \

+                           bits, bits_wrap, bits_size,             \

+                           codes, codes_wrap, codes_size,          \

+                           symbols, symbols_wrap, symbols_size,    \

+                           INIT_VLC_LE | INIT_VLC_USE_NEW_STATIC); \

+    } while (0)

+

+static av_cold void qdmc_init_static_data(AVCodec *codec)

+{

+    int i;

+

+    INIT_VLC_STATIC_LE(&vtable[0], 12, FF_ARRAY_ELEMS(noise_value_bits),

+                       noise_value_bits, 1, 1, noise_value_codes, 2, 2, noise_value_symbols, 1, 1, 4096);

+    INIT_VLC_STATIC_LE(&vtable[1], 10, FF_ARRAY_ELEMS(noise_segment_length_bits),

+                       noise_segment_length_bits, 1, 1, noise_segment_length_codes, 2, 2,

+                       noise_segment_length_symbols, 1, 1, 1024);

+    INIT_VLC_STATIC_LE(&vtable[2], 13, FF_ARRAY_ELEMS(amplitude_bits),

+                       amplitude_bits, 1, 1, amplitude_codes, 2, 2, NULL, 0, 0, 8192);

+    INIT_VLC_STATIC_LE(&vtable[3], 18, FF_ARRAY_ELEMS(freq_diff_bits),

+                       freq_diff_bits, 1, 1, freq_diff_codes, 4, 4, NULL, 0, 0, 262144);

+    INIT_VLC_STATIC_LE(&vtable[4], 8, FF_ARRAY_ELEMS(amplitude_diff_bits),

+                       amplitude_diff_bits, 1, 1, amplitude_diff_codes, 1, 1, NULL, 0, 0, 256);

+    INIT_VLC_STATIC_LE(&vtable[5], 6, FF_ARRAY_ELEMS(phase_diff_bits),

+                       phase_diff_bits, 1, 1, phase_diff_codes, 1, 1, NULL, 0, 0, 64);

+

+    for (i = 0; i < 512; i++)

+        sin_table[i] = sin(2.0f * i * M_PI * 0.001953125f);

+}

+

+static void make_noises(QDMCContext *s)

+{

+    int i, j, n0, n1, n2, diff;

+    float *nptr;

+

+    for (j = 0; j < noise_bands_size[s->band_index]; j++) {

+        n0 = qdmc_nodes[j + 21 * s->band_index    ];

+        n1 = qdmc_nodes[j + 21 * s->band_index + 1];

+        n2 = qdmc_nodes[j + 21 * s->band_index + 2];

+        nptr = s->noise_buffer + 256 * j;

+

+        for (i = 0; i + n0 < n1; i++, nptr++)

+            nptr[0] = i / (float)(n1 - n0);

+

+        diff = n2 - n1;

+        nptr = s->noise_buffer + (j << 8) + n1 - n0;

+

+        for (i = n1; i < n2; i++, nptr++, diff--)

+            nptr[0] = diff / (float)(n2 - n1);

+    }

+}

+

+static av_cold int qdmc_decode_init(AVCodecContext *avctx)

+{

+    QDMCContext *s = avctx->priv_data;

+    int fft_size, fft_order, size, g, j, x;

+    GetByteContext b;

+

+    if (!avctx->extradata || (avctx->extradata_size < 48)) {

+        av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");

+        return AVERROR_INVALIDDATA;

+    }

+

+    bytestream2_init(&b, avctx->extradata, avctx->extradata_size);

+

+    while (bytestream2_get_bytes_left(&b) > 8) {

+        if (bytestream2_peek_be64(&b) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |

+                                           (uint64_t)MKBETAG('Q','D','M','C')))

+            break;

+        bytestream2_skipu(&b, 1);

+    }

+    bytestream2_skipu(&b, 8);

+

+    if (bytestream2_get_bytes_left(&b) < 36) {

+        av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",

+               bytestream2_get_bytes_left(&b));

+        return AVERROR_INVALIDDATA;

+    }

+

+    size = bytestream2_get_be32u(&b);

+    if (size > bytestream2_get_bytes_left(&b)) {

+        av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",

+               bytestream2_get_bytes_left(&b), size);

+        return AVERROR_INVALIDDATA;

+    }

+

+    if (bytestream2_get_be32u(&b) != MKBETAG('Q','D','C','A')) {

+        av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");

+        return AVERROR_INVALIDDATA;

+    }

+    bytestream2_skipu(&b, 4);

+

+    avctx->channels = s->nb_channels = bytestream2_get_be32u(&b);

+    if (s->nb_channels <= 0 || s->nb_channels > 2) {

+        av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");

+        return AVERROR_INVALIDDATA;

+    }

+    avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :

+                                                   AV_CH_LAYOUT_MONO;

+

+    avctx->sample_rate = bytestream2_get_be32u(&b);

+    avctx->bit_rate = bytestream2_get_be32u(&b);

+    bytestream2_skipu(&b, 4);

+    fft_size = bytestream2_get_be32u(&b);

+    fft_order = av_log2(fft_size) + 1;

+    s->checksum_size = bytestream2_get_be32u(&b);

+    if (s->checksum_size >= 1U << 28) {

+        av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);

+        return AVERROR_INVALIDDATA;

+    }

+

+    if (avctx->sample_rate >= 32000) {

+        x = 28000;

+        s->frame_bits = 13;

+    } else if (avctx->sample_rate >= 16000) {

+        x = 20000;

+        s->frame_bits = 12;

+    } else {

+        x = 16000;

+        s->frame_bits = 11;

+    }

+    s->frame_size = 1 << s->frame_bits;

+    s->subframe_size = s->frame_size >> 5;

+

+    if (avctx->channels == 2)

+        x = 3 * x / 2;

+    s->band_index = noise_bands_selector[FFMIN(6, llrint(floor(avctx->bit_rate * 3.0 / (double)x + 0.5)))];

+

+    if ((fft_order < 7) || (fft_order > 9)) {

+        avpriv_request_sample(avctx, "Unknown FFT order %d", fft_order);

+        return AVERROR_PATCHWELCOME;

+    }

+

+    if (fft_size != (1 << (fft_order - 1))) {

+        av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", fft_size);

+        return AVERROR_INVALIDDATA;

+    }

+

+    ff_fft_init(&s->fft_ctx, fft_order, 1);

+

+    avctx->sample_fmt = AV_SAMPLE_FMT_S16;

+

+    for (g = 5; g > 0; g--) {

+        for (j = 0; j < (1 << g) - 1; j++)

+            s->alt_sin[5-g][j] = sin_table[(((j+1) << (8 - g)) & 0x1FF)];

+    }

+

+    make_noises(s);

+

+    return 0;

+}

+

+static av_cold int qdmc_decode_close(AVCodecContext *avctx)

+{

+    QDMCContext *s = avctx->priv_data;

+

+    ff_fft_end(&s->fft_ctx);

+

+    return 0;

+}

+

+static int qdmc_get_vlc(GetBitContext *gb, VLC *table, int flag)

+{

+    int v;

+

+    v = get_vlc2(gb, table->table, table->bits, 1);

+    if (v < 0)

+        return AVERROR_INVALIDDATA;

+    if (v)

+        v = v - 1;

+    else

+        v = get_bits(gb, get_bits(gb, 3) + 1);

+

+    if (flag) {

+        if (v >= FF_ARRAY_ELEMS(code_prefix))

+            return AVERROR_INVALIDDATA;

+

+        v = code_prefix[v] + get_bitsz(gb, v >> 2);

+    }

+

+    return v;

+}

+

+static int skip_label(QDMCContext *s, GetBitContext *gb)

+{

+    uint32_t label = get_bits_long(gb, 32);

+    uint16_t sum = 226, checksum = get_bits(gb, 16);

+    const uint8_t *ptr = gb->buffer + 6;

+    int i;

+

+    if (label != MKTAG('Q', 'M', 'C', 1))

+        return AVERROR_INVALIDDATA;

+

+    for (i = 0; i < s->checksum_size - 6; i++)

+        sum += ptr[i];

+

+    return sum != checksum;

+}

+

+static int read_noise_data(QDMCContext *s, GetBitContext *gb)

+{

+    int ch, j, k, v, idx, band, lastval, newval, len;

+

+    for (ch = 0; ch < s->nb_channels; ch++) {

+        for (band = 0; band < noise_bands_size[s->band_index]; band++) {

+            v = qdmc_get_vlc(gb, &vtable[0], 0);

+            if (v < 0)

+                return AVERROR_INVALIDDATA;

+

+            if (v & 1)

+                v = v + 1;

+            else

+                v = -v;

+

+            lastval = v / 2;

+            s->noise[ch][band][0] = lastval - 1;

+            for (j = 0; j < 15;) {

+                len = qdmc_get_vlc(gb, &vtable[1], 1);

+                if (len < 0)

+                    return AVERROR_INVALIDDATA;

+                len += 1;

+

+                v = qdmc_get_vlc(gb, &vtable[0], 0);

+                if (v < 0)

+                    return AVERROR_INVALIDDATA;

+

+                if (v & 1)

+                    newval = lastval + (v + 1) / 2;

+                else

+                    newval = lastval - v / 2;

+

+                idx = j + 1;

+                if (len + idx > 16)

+                    return AVERROR_INVALIDDATA;

+

+                for (k = 1; idx <= j + len; k++, idx++)

+                    s->noise[ch][band][idx] = lastval + k * (newval - lastval) / len - 1;

+

+                lastval = newval;

+                j += len;

+            }

+        }

+    }

+

+    return 0;

+}

+

+static void add_tone(QDMCContext *s, int group, int offset, int freq, int stereo_mode, int amplitude, int phase)

+{

+    const int index = s->nb_tones[group];

+

+    if (index >= FF_ARRAY_ELEMS(s->tones[group])) {

+        av_log(s->avctx, AV_LOG_WARNING, "Too many tones already in buffer, ignoring tone!\n");

+        return;

+    }

+

+    s->tones[group][index].offset    = offset;

+    s->tones[group][index].freq      = freq;

+    s->tones[group][index].mode      = stereo_mode;

+    s->tones[group][index].amplitude = amplitude;

+    s->tones[group][index].phase     = phase;

+    s->nb_tones[group]++;

+}

+

+static int read_wave_data(QDMCContext *s, GetBitContext *gb)

+{

+    int amp, phase, stereo_mode = 0, i, group, freq, group_size, group_bits;

+    int amp2, phase2, pos2, off;

+

+    for (group = 0; group < 5; group++) {

+        group_size = 1 << (s->frame_bits - group - 1);

+        group_bits = 4 - group;

+        pos2 = 0;

+        off  = 0;

+

+        for (i = 1; ; i = freq + 1) {

+            int v;

+

+            v = qdmc_get_vlc(gb, &vtable[3], 1);

+            if (v < 0)

+                return AVERROR_INVALIDDATA;

+

+            freq = i + v;

+            while (freq >= group_size - 1) {

+                freq += 2 - group_size;

+                pos2 += group_size;

+                off  += 1 << group_bits;

+            }

+

+            if (pos2 >= s->frame_size)

+                break;

+

+            if (s->nb_channels > 1)

+                stereo_mode = get_bits(gb, 2);

+

+            amp   = qdmc_get_vlc(gb, &vtable[2], 0);

+            if (amp < 0)

+                return AVERROR_INVALIDDATA;

+            phase = get_bits(gb, 3);

+

+            if (stereo_mode > 1) {

+                amp2   = qdmc_get_vlc(gb, &vtable[4], 0);

+                if (amp2 < 0)

+                    return AVERROR_INVALIDDATA;

+                amp2   = amp - amp2;

+

+                phase2 = qdmc_get_vlc(gb, &vtable[5], 0);

+                if (phase2 < 0)

+                    return AVERROR_INVALIDDATA;

+                phase2 = phase - phase2;

+

+                if (phase2 < 0)

+                    phase2 += 8;

+            }

+

+            if ((freq >> group_bits) + 1 < s->subframe_size) {

+                add_tone(s, group, off, freq, stereo_mode & 1, amp, phase);

+                if (stereo_mode > 1)

+                    add_tone(s, group, off, freq, ~stereo_mode & 1, amp2, phase2);

+            }

+        }

+    }

+

+    return 0;

+}

+

+static void lin_calc(QDMCContext *s, float amplitude, int node1, int node2, int index)

+{

+    int subframe_size, i, j, k, length;

+    float scale, *noise_ptr;

+

+    scale = 0.5 * amplitude;

+    subframe_size = s->subframe_size;

+    if (subframe_size >= node2)

+        subframe_size = node2;

+    length = (subframe_size - node1) & 0xFFFC;

+    j = node1;

+    noise_ptr = &s->noise_buffer[256 * index];

+

+    for (i = 0; i < length; i += 4, j+= 4, noise_ptr += 4) {

+        s->noise2_buffer[j    ] += scale * noise_ptr[0];

+        s->noise2_buffer[j + 1] += scale * noise_ptr[1];

+        s->noise2_buffer[j + 2] += scale * noise_ptr[2];

+        s->noise2_buffer[j + 3] += scale * noise_ptr[3];

+    }

+

+    k = length + node1;

+    noise_ptr = s->noise_buffer + length + (index << 8);

+    for (i = length; i < subframe_size - node1; i++, k++, noise_ptr++)

+        s->noise2_buffer[k] += scale * noise_ptr[0];

+}

+

+static void add_noise(QDMCContext *s, int ch, int current_subframe)

+{

+    int i, j, aindex;

+    float amplitude;

+    float *im = &s->fft_buffer[0 + ch][s->fft_offset + s->subframe_size * current_subframe];

+    float *re = &s->fft_buffer[2 + ch][s->fft_offset + s->subframe_size * current_subframe];

+

+    memset(s->noise2_buffer, 0, 4 * s->subframe_size);

+

+    for (i = 0; i < noise_bands_size[s->band_index]; i++) {

+        if (qdmc_nodes[i + 21 * s->band_index] > s->subframe_size - 1)

+            break;

+

+        aindex = s->noise[ch][i][current_subframe / 2];

+        amplitude = aindex > 0 ? amplitude_tab[aindex & 0x3F] : 0.0f;

+

+        lin_calc(s, amplitude, qdmc_nodes[21 * s->band_index + i],

+                 qdmc_nodes[21 * s->band_index + i + 2], i);

+    }

+

+    for (j = 2; j < s->subframe_size - 1; j++) {

+        float rnd_re, rnd_im;

+

+        s->rndval = 214013 * s->rndval + 2531011;

+        rnd_im = ((s->rndval & 0x7FFF) - 16384.0f) * 0.000030517578f * s->noise2_buffer[j];

+        s->rndval = 214013 * s->rndval + 2531011;

+        rnd_re = ((s->rndval & 0x7FFF) - 16384.0f) * 0.000030517578f * s->noise2_buffer[j];

+        im[j  ] += rnd_im;

+        re[j  ] += rnd_re;

+        im[j+1] -= rnd_im;

+        re[j+1] -= rnd_re;

+    }

+}

+

+static void add_wave(QDMCContext *s, int offset, int freqs, int group, int stereo_mode, int amp, int phase)

+{

+    int j, group_bits, pos, pindex;

+    float im, re, amplitude, level, *imptr, *reptr;

+

+    if (s->nb_channels == 1)

+        stereo_mode = 0;

+

+    group_bits = 4 - group;

+    pos = freqs >> (4 - group);

+    amplitude = amplitude_tab[amp & 0x3F];

+    imptr = &s->fft_buffer[    stereo_mode][s->fft_offset + s->subframe_size * offset + pos];

+    reptr = &s->fft_buffer[2 + stereo_mode][s->fft_offset + s->subframe_size * offset + pos];

+    pindex = (phase << 6) - ((2 * (freqs >> (4 - group)) + 1) << 7);

+    for (j = 0; j < (1 << (group_bits + 1)) - 1; j++) {

+        pindex += (2 * freqs + 1) << (7 - group_bits);

+        level = amplitude * s->alt_sin[group][j];

+        im = level * sin_table[ pindex        & 0x1FF];

+        re = level * sin_table[(pindex + 128) & 0x1FF];

+        imptr[0] += im;

+        imptr[1] -= im;

+        reptr[0] += re;

+        reptr[1] -= re;

+        imptr += s->subframe_size;

+        reptr += s->subframe_size;

+        if (imptr >= &s->fft_buffer[stereo_mode][2 * s->frame_size]) {

+            imptr = &s->fft_buffer[0 + stereo_mode][pos];

+            reptr = &s->fft_buffer[2 + stereo_mode][pos];

+        }

+    }

+}

+

+static void add_wave0(QDMCContext *s, int offset, int freqs, int stereo_mode, int amp, int phase)

+{

+    float level, im, re;

+    int pos;

+

+    if (s->nb_channels == 1)

+        stereo_mode = 0;

+

+    level = amplitude_tab[amp & 0x3F];

+    im = level * sin_table[ (phase << 6)        & 0x1FF];

+    re = level * sin_table[((phase << 6) + 128) & 0x1FF];

+    pos = s->fft_offset + freqs + s->subframe_size * offset;

+    s->fft_buffer[    stereo_mode][pos    ] += im;

+    s->fft_buffer[2 + stereo_mode][pos    ] += re;

+    s->fft_buffer[    stereo_mode][pos + 1] -= im;

+    s->fft_buffer[2 + stereo_mode][pos + 1] -= re;

+}

+

+static void add_waves(QDMCContext *s, int current_subframe)

+{

+    int w, g;

+

+    for (g = 0; g < 4; g++) {

+        for (w = s->cur_tone[g]; w < s->nb_tones[g]; w++) {

+            QDMCTone *t = &s->tones[g][w];

+

+            if (current_subframe < t->offset)

+                break;

+            add_wave(s, t->offset, t->freq, g, t->mode, t->amplitude, t->phase);

+        }

+        s->cur_tone[g] = w;

+    }

+    for (w = s->cur_tone[4]; w < s->nb_tones[4]; w++) {

+        QDMCTone *t = &s->tones[4][w];

+

+        if (current_subframe < t->offset)

+            break;

+        add_wave0(s, t->offset, t->freq, t->mode, t->amplitude, t->phase);

+    }

+    s->cur_tone[4] = w;

+}

+

+static int decode_frame(QDMCContext *s, GetBitContext *gb, int16_t *out)

+{

+    int ret, ch, i, n;

+

+    if (skip_label(s, gb))

+        return AVERROR_INVALIDDATA;

+

+    s->fft_offset = s->frame_size - s->fft_offset;

+    s->buffer_ptr = &s->buffer[s->nb_channels * s->buffer_offset];

+

+    ret = read_noise_data(s, gb);

+    if (ret < 0)

+        return ret;

+

+    ret = read_wave_data(s, gb);

+    if (ret < 0)

+        return ret;

+

+    for (n = 0; n < 32; n++) {

+        float *r;

+

+        for (ch = 0; ch < s->nb_channels; ch++)

+            add_noise(s, ch, n);

+

+        add_waves(s, n);

+

+        for (ch = 0; ch < s->nb_channels; ch++) {

+            for (i = 0; i < s->subframe_size; i++) {

+                s->cmplx[ch][i].re = s->fft_buffer[ch + 2][s->fft_offset + n * s->subframe_size + i];

+                s->cmplx[ch][i].im = s->fft_buffer[ch + 0][s->fft_offset + n * s->subframe_size + i];

+                s->cmplx[ch][s->subframe_size + i].re = 0;

+                s->cmplx[ch][s->subframe_size + i].im = 0;

+            }

+        }

+

+        for (ch = 0; ch < s->nb_channels; ch++) {

+            s->fft_ctx.fft_permute(&s->fft_ctx, s->cmplx[ch]);

+            s->fft_ctx.fft_calc(&s->fft_ctx, s->cmplx[ch]);

+        }

+

+        r = &s->buffer_ptr[s->nb_channels * n * s->subframe_size];

+        for (i = 0; i < 2 * s->subframe_size; i++) {

+            for (ch = 0; ch < s->nb_channels; ch++) {

+                *r++ += s->cmplx[ch][i].re;

+            }

+        }

+

+        r = &s->buffer_ptr[n * s->subframe_size * s->nb_channels];

+        for (i = 0; i < s->nb_channels * s->subframe_size; i++) {

+            out[i] = av_clipf(r[i], INT16_MIN, INT16_MAX);

+        }

+        out += s->subframe_size * s->nb_channels;

+

+        for (ch = 0; ch < s->nb_channels; ch++) {

+            memset(s->fft_buffer[ch+0] + s->fft_offset + n * s->subframe_size, 0, 4 * s->subframe_size);

+            memset(s->fft_buffer[ch+2] + s->fft_offset + n * s->subframe_size, 0, 4 * s->subframe_size);

+        }

+        memset(s->buffer + s->nb_channels * (n * s->subframe_size + s->frame_size + s->buffer_offset), 0, 4 * s->subframe_size * s->nb_channels);

+    }

+

+    s->buffer_offset += s->frame_size;

+    if (s->buffer_offset >= 32768 - s->frame_size) {

+        memcpy(s->buffer, &s->buffer[s->nb_channels * s->buffer_offset], 4 * s->frame_size * s->nb_channels);

+        s->buffer_offset = 0;

+    }

+

+    return 0;

+}

+

+static av_cold void qdmc_flush(AVCodecContext *avctx)

+{

+    QDMCContext *s = avctx->priv_data;

+

+    memset(s->buffer, 0, sizeof(s->buffer));

+    memset(s->fft_buffer, 0, sizeof(s->fft_buffer));

+    s->fft_offset = 0;

+    s->buffer_offset = 0;

+}

+

+static int qdmc_decode_frame(AVCodecContext *avctx, void *data,

+                             int *got_frame_ptr, AVPacket *avpkt)

+{

+    QDMCContext *s = avctx->priv_data;

+    AVFrame *frame = data;

+    GetBitContext gb;

+    int ret;

+

+    if (!avpkt->data)

+        return 0;

+    if (avpkt->size < s->checksum_size)

+        return AVERROR_INVALIDDATA;

+

+    s->avctx = avctx;

+    frame->nb_samples = s->frame_size;

+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

+        return ret;

+

+    if ((ret = init_get_bits8(&gb, avpkt->data, s->checksum_size)) < 0)

+        return ret;

+

+    memset(s->nb_tones, 0, sizeof(s->nb_tones));

+    memset(s->cur_tone, 0, sizeof(s->cur_tone));

+

+    ret = decode_frame(s, &gb, (int16_t *)frame->data[0]);

+    if (ret >= 0) {

+        *got_frame_ptr = 1;

+        return s->checksum_size;

+    }

+    qdmc_flush(avctx);

+    return ret;

+}

+

+AVCodec ff_qdmc_decoder = {

+    .name             = "qdmc",

+    .long_name        = NULL_IF_CONFIG_SMALL("QDesign Music Codec 1"),

+    .type             = AVMEDIA_TYPE_AUDIO,

+    .id               = AV_CODEC_ID_QDMC,

+    .priv_data_size   = sizeof(QDMCContext),

+    .init             = qdmc_decode_init,

+    .init_static_data = qdmc_init_static_data,

+    .close            = qdmc_decode_close,

+    .decode           = qdmc_decode_frame,

+    .flush            = qdmc_flush,

+    .capabilities     = AV_CODEC_CAP_DR1,

+};

@@ -28,8 +28,8 @@

 #include "libavutil/version.h"

 #define LIBAVCODEC_VERSION_MAJOR  57

-#define LIBAVCODEC_VERSION_MINOR  70

-#define LIBAVCODEC_VERSION_MICRO 101

+#define LIBAVCODEC_VERSION_MINOR  71

+#define LIBAVCODEC_VERSION_MICRO 100

 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \

                                                LIBAVCODEC_VERSION_MINOR, \