Add G.722 ADPCM audio decoder

Originally committed as revision 25086 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 304 insertions, 0 deletions

diff --git a/libavcodec/g722.c b/libavcodec/g722.c
new file mode 100644
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--- /dev/null
+++ b/libavcodec/g722.c
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+/*
+ * G.722 ADPCM audio decoder
+ *
+ * Copyright (c) CMU 1993 Computer Science, Speech Group
+ *                        Chengxiang Lu and Alex Hauptmann
+ * Copyright (c) 2005 Steve Underwood <steveu at coppice.org>
+ * Copyright (c) 2009 Kenan Gillet
+ * Copyright (c) 2010 Martin Storsjo
+ *
+ * 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
+ *
+ * G.722 ADPCM audio codec
+ *
+ * This G.722 decoder is a bit-exact implementation of the ITU G.722
+ * specification for all three specified bitrates - 64000bps, 56000bps
+ * and 48000bps. It passes the ITU tests.
+ *
+ * @note For the 56000bps and 48000bps bitrates, the lowest 1 or 2 bits
+ *       respectively of each byte are ignored.
+ */
+
+#include "avcodec.h"
+#include "mathops.h"
+#include "get_bits.h"
+
+#define PREV_SAMPLES_BUF_SIZE 1024
+
+typedef struct {
+    int16_t prev_samples[PREV_SAMPLES_BUF_SIZE]; ///< memory of past decoded samples
+    int     prev_samples_pos;        ///< the number of values in prev_samples
+
+    /**
+     * The band[0] and band[1] correspond respectively to the lower band and higher band.
+     */
+    struct G722Band {
+        int16_t s_predictor;         ///< predictor output value
+        int32_t s_zero;              ///< previous output signal from zero predictor
+        int8_t  part_reconst_mem[2]; ///< signs of previous partially reconstructed signals
+        int16_t prev_qtzd_reconst;   ///< previous quantized reconstructed signal (internal value, using low_inv_quant4)
+        int16_t pole_mem[2];         ///< second-order pole section coefficient buffer
+        int32_t diff_mem[6];         ///< quantizer difference signal memory
+        int16_t zero_mem[6];         ///< Seventh-order zero section coefficient buffer
+        int16_t log_factor;          ///< delayed 2-logarithmic quantizer factor
+        int16_t scale_factor;        ///< delayed quantizer scale factor
+    } band[2];
+} G722Context;
+
+
+static const int8_t sign_lookup[2] = { -1, 1 };
+
+static const int16_t inv_log2_table[32] = {
+    2048, 2093, 2139, 2186, 2233, 2282, 2332, 2383,
+    2435, 2489, 2543, 2599, 2656, 2714, 2774, 2834,
+    2896, 2960, 3025, 3091, 3158, 3228, 3298, 3371,
+    3444, 3520, 3597, 3676, 3756, 3838, 3922, 4008
+};
+static const int16_t high_log_factor_step[2] = { 798, -214 };
+static const int16_t high_inv_quant[4] = { -926, -202, 926, 202 };
+/**
+ * low_log_factor_step[index] == wl[rl42[index]]
+ */
+static const int16_t low_log_factor_step[16] = {
+     -60, 3042, 1198, 538, 334, 172,  58, -30,
+    3042, 1198,  538, 334, 172,  58, -30, -60
+};
+static const int16_t low_inv_quant4[16] = {
+       0, -2557, -1612, -1121,  -786,  -530,  -323,  -150,
+    2557,  1612,  1121,   786,   530,   323,   150,     0
+};
+
+/**
+ * quadrature mirror filter (QMF) coefficients
+ *
+ * ITU-T G.722 Table 11
+ */
+static const int16_t qmf_coeffs[12] = {
+    3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11,
+};
+
+
+/**
+ * adaptive predictor
+ *
+ * @param cur_diff the dequantized and scaled delta calculated from the
+ *                 current codeword
+ */
+static void do_adaptive_prediction(struct G722Band *band, const int cur_diff)
+{
+    int sg[2], limit, i, cur_qtzd_reconst;
+
+    const int cur_part_reconst = band->s_zero + cur_diff < 0;
+
+    sg[0] = sign_lookup[cur_part_reconst != band->part_reconst_mem[0]];
+    sg[1] = sign_lookup[cur_part_reconst == band->part_reconst_mem[1]];
+    band->part_reconst_mem[1] = band->part_reconst_mem[0];
+    band->part_reconst_mem[0] = cur_part_reconst;
+
+    band->pole_mem[1] = av_clip((sg[0] * av_clip(band->pole_mem[0], -8191, 8191) >> 5) +
+                                (sg[1] << 7) + (band->pole_mem[1] * 127 >> 7), -12288, 12288);
+
+    limit = 15360 - band->pole_mem[1];
+    band->pole_mem[0] = av_clip(-192 * sg[0] + (band->pole_mem[0] * 255 >> 8), -limit, limit);
+
+
+    if (cur_diff) {
+        for (i = 0; i < 6; i++)
+            band->zero_mem[i] = ((band->zero_mem[i]*255) >> 8) +
+                                ((band->diff_mem[i]^cur_diff) < 0 ? -128 : 128);
+    } else
+        for (i = 0; i < 6; i++)
+            band->zero_mem[i] = (band->zero_mem[i]*255) >> 8;
+
+    for (i = 5; i > 0; i--)
+        band->diff_mem[i] = band->diff_mem[i-1];
+    band->diff_mem[0] = av_clip_int16(cur_diff << 1);
+
+    band->s_zero = 0;
+    for (i = 5; i >= 0; i--)
+        band->s_zero += (band->zero_mem[i]*band->diff_mem[i]) >> 15;
+
+
+    cur_qtzd_reconst = av_clip_int16((band->s_predictor + cur_diff) << 1);
+    band->s_predictor = av_clip_int16(band->s_zero +
+                                      (band->pole_mem[0] * cur_qtzd_reconst >> 15) +
+                                      (band->pole_mem[1] * band->prev_qtzd_reconst >> 15));
+    band->prev_qtzd_reconst = cur_qtzd_reconst;
+}
+
+static int inline linear_scale_factor(const int log_factor)
+{
+    const int wd1 = inv_log2_table[(log_factor >> 6) & 31];
+    const int shift = log_factor >> 11;
+    return shift < 0 ? wd1 >> -shift : wd1 << shift;
+}
+
+static void update_low_predictor(struct G722Band *band, const int ilow)
+{
+    do_adaptive_prediction(band,
+                           band->scale_factor * low_inv_quant4[ilow] >> 10);
+
+    // quantizer adaptation
+    band->log_factor   = av_clip((band->log_factor * 127 >> 7) +
+                                 low_log_factor_step[ilow], 0, 18432);
+    band->scale_factor = linear_scale_factor(band->log_factor - (8 << 11));
+}
+
+static void update_high_predictor(struct G722Band *band, const int dhigh,
+                                  const int ihigh)
+{
+    do_adaptive_prediction(band, dhigh);
+
+    // quantizer adaptation
+    band->log_factor   = av_clip((band->log_factor * 127 >> 7) +
+                                 high_log_factor_step[ihigh&1], 0, 22528);
+    band->scale_factor = linear_scale_factor(band->log_factor - (10 << 11));
+}
+
+static void apply_qmf(const int16_t *prev_samples, int *xout1, int *xout2)
+{
+    int i;
+
+    *xout1 = 0;
+    *xout2 = 0;
+    for (i = 0; i < 12; i++) {
+        MAC16(*xout2, prev_samples[2*i  ], qmf_coeffs[i   ]);
+        MAC16(*xout1, prev_samples[2*i+1], qmf_coeffs[11-i]);
+    }
+}
+
+static av_cold int g722_init(AVCodecContext * avctx)
+{
+    G722Context *c = avctx->priv_data;
+
+    if (avctx->channels != 1) {
+        av_log(avctx, AV_LOG_ERROR, "Only mono tracks are allowed.\n");
+        return AVERROR_INVALIDDATA;
+    }
+    avctx->sample_fmt = SAMPLE_FMT_S16;
+
+    switch (avctx->bits_per_coded_sample) {
+    case 8:
+    case 7:
+    case 6:
+        break;
+    default:
+        av_log(avctx, AV_LOG_WARNING, "Unsupported bits_per_coded_sample [%d], "
+                                      "assuming 8\n",
+                                      avctx->bits_per_coded_sample);
+    case 0:
+        avctx->bits_per_coded_sample = 8;
+        break;
+    }
+
+    c->band[0].scale_factor = 8;
+    c->band[1].scale_factor = 2;
+    c->prev_samples_pos = 22;
+
+    if (avctx->lowres)
+        avctx->sample_rate /= 2;
+
+    return 0;
+}
+
+static const int16_t low_inv_quant5[32] = {
+     -35,   -35, -2919, -2195, -1765, -1458, -1219, -1023,
+    -858,  -714,  -587,  -473,  -370,  -276,  -190,  -110,
+    2919,  2195,  1765,  1458,  1219,  1023,   858,   714,
+     587,   473,   370,   276,   190,   110,    35,   -35
+};
+static const int16_t low_inv_quant6[64] = {
+     -17,   -17,   -17,   -17, -3101, -2738, -2376, -2088,
+   -1873, -1689, -1535, -1399, -1279, -1170, -1072,  -982,
+    -899,  -822,  -750,  -682,  -618,  -558,  -501,  -447,
+    -396,  -347,  -300,  -254,  -211,  -170,  -130,   -91,
+    3101,  2738,  2376,  2088,  1873,  1689,  1535,  1399,
+    1279,  1170,  1072,   982,   899,   822,   750,   682,
+     618,   558,   501,   447,   396,   347,   300,   254,
+     211,   170,   130,    91,    54,    17,   -54,   -17
+};
+
+static const int16_t *low_inv_quants[3] = { low_inv_quant6, low_inv_quant5,
+                                 low_inv_quant4 };
+
+static int g722_decode_frame(AVCodecContext *avctx, void *data,
+                             int *data_size, AVPacket *avpkt)
+{
+    G722Context *c = avctx->priv_data;
+    int16_t *out_buf = data;
+    int j, out_len = 0;
+    const int skip = 8 - avctx->bits_per_coded_sample;
+    const int16_t *quantizer_table = low_inv_quants[skip];
+    GetBitContext gb;
+
+    init_get_bits(&gb, avpkt->data, avpkt->size * 8);
+
+    for (j = 0; j < avpkt->size; j++) {
+        int ilow, ihigh, rlow;
+
+        ihigh = get_bits(&gb, 2);
+        ilow = get_bits(&gb, 6 - skip);
+        skip_bits(&gb, skip);
+
+        rlow = av_clip((c->band[0].scale_factor * quantizer_table[ilow] >> 10)
+                      + c->band[0].s_predictor, -16384, 16383);
+
+        update_low_predictor(&c->band[0], ilow >> (2 - skip));
+
+        if (!avctx->lowres) {
+            const int dhigh = c->band[1].scale_factor *
+                              high_inv_quant[ihigh] >> 10;
+            const int rhigh = av_clip(dhigh + c->band[1].s_predictor,
+                                      -16384, 16383);
+            int xout1, xout2;
+
+            update_high_predictor(&c->band[1], dhigh, ihigh);
+
+            c->prev_samples[c->prev_samples_pos++] = rlow + rhigh;
+            c->prev_samples[c->prev_samples_pos++] = rlow - rhigh;
+            apply_qmf(c->prev_samples + c->prev_samples_pos - 24,
+                      &xout1, &xout2);
+            out_buf[out_len++] = av_clip_int16(xout1 >> 12);
+            out_buf[out_len++] = av_clip_int16(xout2 >> 12);
+            if (c->prev_samples_pos >= PREV_SAMPLES_BUF_SIZE) {
+                memmove(c->prev_samples,
+                        c->prev_samples + c->prev_samples_pos - 22,
+                        22 * sizeof(c->prev_samples[0]));
+                c->prev_samples_pos = 22;
+            }
+        } else
+            out_buf[out_len++] = rlow;
+    }
+    *data_size = out_len << 1;
+    return avpkt->size;
+}
+
+AVCodec adpcm_g722_decoder = {
+    .name           = "g722",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_ADPCM_G722,
+    .priv_data_size = sizeof(G722Context),
+    .init           = g722_init,
+    .decode         = g722_decode_frame,
+    .long_name      = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
+    .max_lowres     = 1,
+};
+