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authorMartin Storsjö <martin@martin.st>2010-09-09 19:21:16 +0000
committerMartin Storsjö <martin@martin.st>2010-09-09 19:21:16 +0000
commit58e37eafce10bbcf10b111cafb7ba8586c76ef4b (patch)
tree54bd919cd566792fc1acb80c4cc06425efec892d /libavcodec/g722.c
parenta3f0d2b9674b4bbbdb6eb28a1e600ba324a5853b (diff)
downloadffmpeg-58e37eafce10bbcf10b111cafb7ba8586c76ef4b.tar.gz
Add G.722 ADPCM audio decoder
Originally committed as revision 25086 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/g722.c')
-rw-r--r--libavcodec/g722.c304
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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+++ 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,
+};
+