diff --git a/Changelog b/Changelog
index aff9ab0..e09bf20 100644
--- a/Changelog
+++ b/Changelog
@@ -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
diff --git a/doc/general.texi b/doc/general.texi
index 084c0a1..a13a8fc 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -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
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 58feb31..44e416e 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -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
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 678f54a..4540ef7 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -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);
diff --git a/libavcodec/qdmc.c b/libavcodec/qdmc.c
new file mode 100644
index 0000000..5559db3
--- /dev/null
+++ b/libavcodec/qdmc.c
@@ -0,0 +1,817 @@
+/*
+ * 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 "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;
+    float *buffer_ptr;
+    int nb_channels;
+
+    int group_size;
+    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];
+    int rndval;
+
+    DECLARE_ALIGNED(32, FFTComplex, cmplx)[2][512];
+    float buffer[2 * 32768];
+
+    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,
+};
+
+static av_cold int qdmc_init_static_data(QDMCContext *s)
+{
+    static int done;
+    int i, ret;
+
+    if (done)
+        return 0;
+
+    ret = ff_init_vlc_sparse(&vtable[0], 12, FF_ARRAY_ELEMS(noise_value_bits),
+                             noise_value_bits, 1, 1, noise_value_codes, 2, 2, noise_value_symbols, 1, 1, INIT_VLC_LE);
+    if (ret < 0)
+        return ret;
+    ret = ff_init_vlc_sparse(&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, INIT_VLC_LE);
+    if (ret < 0)
+        return ret;
+    ret = ff_init_vlc_sparse(&vtable[2], 13, FF_ARRAY_ELEMS(amplitude_bits),
+                             amplitude_bits, 1, 1, amplitude_codes, 2, 2, NULL, 0, 0, INIT_VLC_LE);
+    if (ret < 0)
+        return ret;
+    ret = ff_init_vlc_sparse(&vtable[3], 18, FF_ARRAY_ELEMS(freq_diff_bits),
+                             freq_diff_bits, 1, 1, freq_diff_codes, 4, 4, NULL, 0, 0, INIT_VLC_LE);
+    if (ret < 0)
+        return ret;
+    ret = ff_init_vlc_sparse(&vtable[4], 8, FF_ARRAY_ELEMS(amplitude_diff_bits),
+                             amplitude_diff_bits, 1, 1, amplitude_diff_codes, 1, 1, NULL, 0, 0, INIT_VLC_LE);
+    if (ret < 0)
+        return ret;
+    ret = ff_init_vlc_sparse(&vtable[5], 6, FF_ARRAY_ELEMS(phase_diff_bits),
+                             phase_diff_bits, 1, 1, phase_diff_codes, 1, 1, NULL, 0, 0, INIT_VLC_LE);
+    if (ret < 0)
+        return ret;
+
+    for (i = 0; i < 512; i++)
+        sin_table[i] = sin(2 * i * M_PI * 0.001953125);
+
+    done = 1;
+
+    return 0;
+}
+
+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;
+    uint8_t *extradata;
+    int extradata_size, fft_size, fft_order, ret, size, g, j, x;
+
+    if ((ret = qdmc_init_static_data(s)) < 0)
+        return ret;
+
+    if (!avctx->extradata || (avctx->extradata_size < 48)) {
+        av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    extradata      = avctx->extradata;
+    extradata_size = avctx->extradata_size;
+
+    while (extradata_size > 8) {
+        if (!memcmp(extradata, "frmaQDMC", 8))
+            break;
+        extradata++;
+        extradata_size--;
+    }
+
+    if (extradata_size < 12) {
+        av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
+               extradata_size);
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (memcmp(extradata, "frmaQDMC", 8)) {
+        av_log(avctx, AV_LOG_ERROR, "invalid headers, QDMC not found\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    extradata += 8;
+    extradata_size -= 8;
+
+    size = AV_RB32(extradata);
+    extradata += 4;
+
+    if (size > extradata_size) {
+        av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
+               extradata_size, size);
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
+        av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
+        return AVERROR_INVALIDDATA;
+    }
+    extradata += 8;
+
+    avctx->channels = s->nb_channels = AV_RB32(extradata);
+    extradata += 4;
+    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 = AV_RB32(extradata);
+    extradata += 4;
+
+    avctx->bit_rate = AV_RB32(extradata);
+    extradata += 4;
+
+    s->group_size = AV_RB32(extradata);
+    extradata += 4;
+
+    fft_size = AV_RB32(extradata);
+    fft_order = av_log2(fft_size) + 1;
+    extradata += 4;
+
+    s->checksum_size = AV_RB32(extradata);
+    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 float real_amp(int a)
+{
+    return a >= 0 ? amplitude_tab[a & 0x3F] : 0.0f;
+}
+
+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 = 0.0;
+        if (aindex > 0)
+            amplitude = real_amp(aindex);
+
+        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) - 16384.0) * 0.000030517578 * s->noise2_buffer[j];
+        s->rndval = 214013 * s->rndval + 2531011;
+        rnd_re = ((s->rndval & 0x7FFF) - 16384.0) * 0.000030517578 * 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,
+    .close            = qdmc_decode_close,
+    .decode           = qdmc_decode_frame,
+    .flush            = qdmc_flush,
+    .capabilities     = AV_CODEC_CAP_DR1,
+};
