[FFmpeg-cvslog] g722: split decoder and encoder into separate files
Justin Ruggles
git at videolan.org
Mon Oct 24 01:58:26 CEST 2011
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Sat Oct 22 17:49:50 2011 -0400| [704721bc9cb47c58ce3dff8ef64ce3fb85283069] | committer: Justin Ruggles
g722: split decoder and encoder into separate files
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=704721bc9cb47c58ce3dff8ef64ce3fb85283069
---
libavcodec/Makefile | 4 +-
libavcodec/g722.c | 427 +-------------------------------------------------
libavcodec/g722.h | 74 +++++++++
libavcodec/g722dec.c | 147 +++++++++++++++++
libavcodec/g722enc.c | 311 ++++++++++++++++++++++++++++++++++++
5 files changed, 542 insertions(+), 421 deletions(-)
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 81881e4..bd4275a 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -495,8 +495,8 @@ OBJS-$(CONFIG_ADPCM_EA_R1_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_EA_R2_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_EA_R3_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_EA_XAS_DECODER) += adpcm.o
-OBJS-$(CONFIG_ADPCM_G722_DECODER) += g722.o
-OBJS-$(CONFIG_ADPCM_G722_ENCODER) += g722.o
+OBJS-$(CONFIG_ADPCM_G722_DECODER) += g722.o g722dec.o
+OBJS-$(CONFIG_ADPCM_G722_ENCODER) += g722.o g722enc.o
OBJS-$(CONFIG_ADPCM_G726_DECODER) += g726.o
OBJS-$(CONFIG_ADPCM_G726_ENCODER) += g726.o
OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER) += adpcm.o adpcm_data.o
diff --git a/libavcodec/g722.c b/libavcodec/g722.c
index 74d5552..7b9ff72 100644
--- a/libavcodec/g722.c
+++ b/libavcodec/g722.c
@@ -36,45 +36,8 @@
* respectively of each byte are ignored.
*/
-#include "avcodec.h"
#include "mathops.h"
-#include "get_bits.h"
-
-#define PREV_SAMPLES_BUF_SIZE 1024
-
-#define FREEZE_INTERVAL 128
-
-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];
-
- struct TrellisNode {
- struct G722Band state;
- uint32_t ssd;
- int path;
- } *node_buf[2], **nodep_buf[2];
-
- struct TrellisPath {
- int value;
- int prev;
- } *paths[2];
-} G722Context;
-
+#include "g722.h"
static const int8_t sign_lookup[2] = { -1, 1 };
@@ -85,7 +48,7 @@ static const int16_t inv_log2_table[32] = {
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 };
+const int16_t ff_g722_high_inv_quant[4] = { -926, -202, 926, 202 };
/**
* low_log_factor_step[index] == wl[rl42[index]]
*/
@@ -93,11 +56,11 @@ 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] = {
+const int16_t ff_g722_low_inv_quant4[16] = {
0, -2557, -1612, -1121, -786, -530, -323, -150,
2557, 1612, 1121, 786, 530, 323, 150, 0
};
-static const int16_t low_inv_quant6[64] = {
+const int16_t ff_g722_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,
@@ -173,10 +136,10 @@ static int inline linear_scale_factor(const int log_factor)
return shift < 0 ? wd1 >> -shift : wd1 << shift;
}
-static void update_low_predictor(struct G722Band *band, const int ilow)
+void ff_g722_update_low_predictor(struct G722Band *band, const int ilow)
{
do_adaptive_prediction(band,
- band->scale_factor * low_inv_quant4[ilow] >> 10);
+ band->scale_factor * ff_g722_low_inv_quant4[ilow] >> 10);
// quantizer adaptation
band->log_factor = av_clip((band->log_factor * 127 >> 7) +
@@ -184,7 +147,7 @@ static void update_low_predictor(struct G722Band *band, const int ilow)
band->scale_factor = linear_scale_factor(band->log_factor - (8 << 11));
}
-static void update_high_predictor(struct G722Band *band, const int dhigh,
+void ff_g722_update_high_predictor(struct G722Band *band, const int dhigh,
const int ihigh)
{
do_adaptive_prediction(band, dhigh);
@@ -195,7 +158,7 @@ static void update_high_predictor(struct G722Band *band, const int dhigh,
band->scale_factor = linear_scale_factor(band->log_factor - (10 << 11));
}
-static void apply_qmf(const int16_t *prev_samples, int *xout1, int *xout2)
+void ff_g722_apply_qmf(const int16_t *prev_samples, int *xout1, int *xout2)
{
int i;
@@ -206,377 +169,3 @@ static void apply_qmf(const int16_t *prev_samples, int *xout1, int *xout2)
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 = AV_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;
-
- if (avctx->trellis) {
- int frontier = 1 << avctx->trellis;
- int max_paths = frontier * FREEZE_INTERVAL;
- int i;
- for (i = 0; i < 2; i++) {
- c->paths[i] = av_mallocz(max_paths * sizeof(**c->paths));
- c->node_buf[i] = av_mallocz(2 * frontier * sizeof(**c->node_buf));
- c->nodep_buf[i] = av_mallocz(2 * frontier * sizeof(**c->nodep_buf));
- }
- }
-
- return 0;
-}
-
-static av_cold int g722_close(AVCodecContext *avctx)
-{
- G722Context *c = avctx->priv_data;
- int i;
- for (i = 0; i < 2; i++) {
- av_freep(&c->paths[i]);
- av_freep(&c->node_buf[i]);
- av_freep(&c->nodep_buf[i]);
- }
- return 0;
-}
-
-#if CONFIG_ADPCM_G722_DECODER
-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_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 ff_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,
-};
-#endif
-
-#if CONFIG_ADPCM_G722_ENCODER
-static const int16_t low_quant[33] = {
- 35, 72, 110, 150, 190, 233, 276, 323,
- 370, 422, 473, 530, 587, 650, 714, 786,
- 858, 940, 1023, 1121, 1219, 1339, 1458, 1612,
- 1765, 1980, 2195, 2557, 2919
-};
-
-static inline void filter_samples(G722Context *c, const int16_t *samples,
- int *xlow, int *xhigh)
-{
- int xout1, xout2;
- c->prev_samples[c->prev_samples_pos++] = samples[0];
- c->prev_samples[c->prev_samples_pos++] = samples[1];
- apply_qmf(c->prev_samples + c->prev_samples_pos - 24, &xout1, &xout2);
- *xlow = xout1 + xout2 >> 13;
- *xhigh = xout1 - xout2 >> 13;
- 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;
- }
-}
-
-static inline int encode_high(const struct G722Band *state, int xhigh)
-{
- int diff = av_clip_int16(xhigh - state->s_predictor);
- int pred = 141 * state->scale_factor >> 8;
- /* = diff >= 0 ? (diff < pred) + 2 : diff >= -pred */
- return ((diff ^ (diff >> (sizeof(diff)*8-1))) < pred) + 2*(diff >= 0);
-}
-
-static inline int encode_low(const struct G722Band* state, int xlow)
-{
- int diff = av_clip_int16(xlow - state->s_predictor);
- /* = diff >= 0 ? diff : -(diff + 1) */
- int limit = diff ^ (diff >> (sizeof(diff)*8-1));
- int i = 0;
- limit = limit + 1 << 10;
- if (limit > low_quant[8] * state->scale_factor)
- i = 9;
- while (i < 29 && limit > low_quant[i] * state->scale_factor)
- i++;
- return (diff < 0 ? (i < 2 ? 63 : 33) : 61) - i;
-}
-
-static int g722_encode_trellis(AVCodecContext *avctx,
- uint8_t *dst, int buf_size, void *data)
-{
- G722Context *c = avctx->priv_data;
- const int16_t *samples = data;
- int i, j, k;
- int frontier = 1 << avctx->trellis;
- struct TrellisNode **nodes[2];
- struct TrellisNode **nodes_next[2];
- int pathn[2] = {0, 0}, froze = -1;
- struct TrellisPath *p[2];
-
- for (i = 0; i < 2; i++) {
- nodes[i] = c->nodep_buf[i];
- nodes_next[i] = c->nodep_buf[i] + frontier;
- memset(c->nodep_buf[i], 0, 2 * frontier * sizeof(*c->nodep_buf));
- nodes[i][0] = c->node_buf[i] + frontier;
- nodes[i][0]->ssd = 0;
- nodes[i][0]->path = 0;
- nodes[i][0]->state = c->band[i];
- }
-
- for (i = 0; i < buf_size >> 1; i++) {
- int xlow, xhigh;
- struct TrellisNode *next[2];
- int heap_pos[2] = {0, 0};
-
- for (j = 0; j < 2; j++) {
- next[j] = c->node_buf[j] + frontier*(i & 1);
- memset(nodes_next[j], 0, frontier * sizeof(**nodes_next));
- }
-
- filter_samples(c, &samples[2*i], &xlow, &xhigh);
-
- for (j = 0; j < frontier && nodes[0][j]; j++) {
- /* Only k >> 2 affects the future adaptive state, therefore testing
- * small steps that don't change k >> 2 is useless, the orignal
- * value from encode_low is better than them. Since we step k
- * in steps of 4, make sure range is a multiple of 4, so that
- * we don't miss the original value from encode_low. */
- int range = j < frontier/2 ? 4 : 0;
- struct TrellisNode *cur_node = nodes[0][j];
-
- int ilow = encode_low(&cur_node->state, xlow);
-
- for (k = ilow - range; k <= ilow + range && k <= 63; k += 4) {
- int decoded, dec_diff, pos;
- uint32_t ssd;
- struct TrellisNode* node;
-
- if (k < 0)
- continue;
-
- decoded = av_clip((cur_node->state.scale_factor *
- low_inv_quant6[k] >> 10)
- + cur_node->state.s_predictor, -16384, 16383);
- dec_diff = xlow - decoded;
-
-#define STORE_NODE(index, UPDATE, VALUE)\
- ssd = cur_node->ssd + dec_diff*dec_diff;\
- /* Check for wraparound. Using 64 bit ssd counters would \
- * be simpler, but is slower on x86 32 bit. */\
- if (ssd < cur_node->ssd)\
- continue;\
- if (heap_pos[index] < frontier) {\
- pos = heap_pos[index]++;\
- assert(pathn[index] < FREEZE_INTERVAL * frontier);\
- node = nodes_next[index][pos] = next[index]++;\
- node->path = pathn[index]++;\
- } else {\
- /* Try to replace one of the leaf nodes with the new \
- * one, but not always testing the same leaf position */\
- pos = (frontier>>1) + (heap_pos[index] & ((frontier>>1) - 1));\
- if (ssd >= nodes_next[index][pos]->ssd)\
- continue;\
- heap_pos[index]++;\
- node = nodes_next[index][pos];\
- }\
- node->ssd = ssd;\
- node->state = cur_node->state;\
- UPDATE;\
- c->paths[index][node->path].value = VALUE;\
- c->paths[index][node->path].prev = cur_node->path;\
- /* Sift the newly inserted node up in the heap to restore \
- * the heap property */\
- while (pos > 0) {\
- int parent = (pos - 1) >> 1;\
- if (nodes_next[index][parent]->ssd <= ssd)\
- break;\
- FFSWAP(struct TrellisNode*, nodes_next[index][parent],\
- nodes_next[index][pos]);\
- pos = parent;\
- }
- STORE_NODE(0, update_low_predictor(&node->state, k >> 2), k);
- }
- }
-
- for (j = 0; j < frontier && nodes[1][j]; j++) {
- int ihigh;
- struct TrellisNode *cur_node = nodes[1][j];
-
- /* We don't try to get any initial guess for ihigh via
- * encode_high - since there's only 4 possible values, test
- * them all. Testing all of these gives a much, much larger
- * gain than testing a larger range around ilow. */
- for (ihigh = 0; ihigh < 4; ihigh++) {
- int dhigh, decoded, dec_diff, pos;
- uint32_t ssd;
- struct TrellisNode* node;
-
- dhigh = cur_node->state.scale_factor *
- high_inv_quant[ihigh] >> 10;
- decoded = av_clip(dhigh + cur_node->state.s_predictor,
- -16384, 16383);
- dec_diff = xhigh - decoded;
-
- STORE_NODE(1, update_high_predictor(&node->state, dhigh, ihigh), ihigh);
- }
- }
-
- for (j = 0; j < 2; j++) {
- FFSWAP(struct TrellisNode**, nodes[j], nodes_next[j]);
-
- if (nodes[j][0]->ssd > (1 << 16)) {
- for (k = 1; k < frontier && nodes[j][k]; k++)
- nodes[j][k]->ssd -= nodes[j][0]->ssd;
- nodes[j][0]->ssd = 0;
- }
- }
-
- if (i == froze + FREEZE_INTERVAL) {
- p[0] = &c->paths[0][nodes[0][0]->path];
- p[1] = &c->paths[1][nodes[1][0]->path];
- for (j = i; j > froze; j--) {
- dst[j] = p[1]->value << 6 | p[0]->value;
- p[0] = &c->paths[0][p[0]->prev];
- p[1] = &c->paths[1][p[1]->prev];
- }
- froze = i;
- pathn[0] = pathn[1] = 0;
- memset(nodes[0] + 1, 0, (frontier - 1)*sizeof(**nodes));
- memset(nodes[1] + 1, 0, (frontier - 1)*sizeof(**nodes));
- }
- }
-
- p[0] = &c->paths[0][nodes[0][0]->path];
- p[1] = &c->paths[1][nodes[1][0]->path];
- for (j = i; j > froze; j--) {
- dst[j] = p[1]->value << 6 | p[0]->value;
- p[0] = &c->paths[0][p[0]->prev];
- p[1] = &c->paths[1][p[1]->prev];
- }
- c->band[0] = nodes[0][0]->state;
- c->band[1] = nodes[1][0]->state;
-
- return i;
-}
-
-static int g722_encode_frame(AVCodecContext *avctx,
- uint8_t *dst, int buf_size, void *data)
-{
- G722Context *c = avctx->priv_data;
- const int16_t *samples = data;
- int i;
-
- if (avctx->trellis)
- return g722_encode_trellis(avctx, dst, buf_size, data);
-
- for (i = 0; i < buf_size >> 1; i++) {
- int xlow, xhigh, ihigh, ilow;
- filter_samples(c, &samples[2*i], &xlow, &xhigh);
- ihigh = encode_high(&c->band[1], xhigh);
- ilow = encode_low(&c->band[0], xlow);
- update_high_predictor(&c->band[1], c->band[1].scale_factor *
- high_inv_quant[ihigh] >> 10, ihigh);
- update_low_predictor(&c->band[0], ilow >> 2);
- *dst++ = ihigh << 6 | ilow;
- }
- return i;
-}
-
-AVCodec ff_adpcm_g722_encoder = {
- .name = "g722",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_ADPCM_G722,
- .priv_data_size = sizeof(G722Context),
- .init = g722_init,
- .close = g722_close,
- .encode = g722_encode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
-};
-#endif
-
diff --git a/libavcodec/g722.h b/libavcodec/g722.h
new file mode 100644
index 0000000..5edb6c8
--- /dev/null
+++ b/libavcodec/g722.h
@@ -0,0 +1,74 @@
+/*
+ * 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 Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_G722_H
+#define AVCODEC_G722_H
+
+#include <stdint.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];
+
+ struct TrellisNode {
+ struct G722Band state;
+ uint32_t ssd;
+ int path;
+ } *node_buf[2], **nodep_buf[2];
+
+ struct TrellisPath {
+ int value;
+ int prev;
+ } *paths[2];
+} G722Context;
+
+extern const int16_t ff_g722_high_inv_quant[4];
+extern const int16_t ff_g722_low_inv_quant4[16];
+extern const int16_t ff_g722_low_inv_quant6[64];
+
+void ff_g722_update_low_predictor(struct G722Band *band, const int ilow);
+
+void ff_g722_update_high_predictor(struct G722Band *band, const int dhigh,
+ const int ihigh);
+
+void ff_g722_apply_qmf(const int16_t *prev_samples, int *xout1, int *xout2);
+
+#endif /* AVCODEC_G722_H */
diff --git a/libavcodec/g722dec.c b/libavcodec/g722dec.c
new file mode 100644
index 0000000..51f87c9
--- /dev/null
+++ b/libavcodec/g722dec.c
@@ -0,0 +1,147 @@
+/*
+ * 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 Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * G.722 ADPCM audio decoder
+ *
+ * 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 "get_bits.h"
+#include "g722.h"
+
+static av_cold int g722_decode_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 = AV_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_quants[3] = { ff_g722_low_inv_quant6,
+ low_inv_quant5,
+ ff_g722_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);
+
+ ff_g722_update_low_predictor(&c->band[0], ilow >> (2 - skip));
+
+ if (!avctx->lowres) {
+ const int dhigh = c->band[1].scale_factor *
+ ff_g722_high_inv_quant[ihigh] >> 10;
+ const int rhigh = av_clip(dhigh + c->band[1].s_predictor,
+ -16384, 16383);
+ int xout1, xout2;
+
+ ff_g722_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;
+ ff_g722_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 ff_adpcm_g722_decoder = {
+ .name = "g722",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_ADPCM_G722,
+ .priv_data_size = sizeof(G722Context),
+ .init = g722_decode_init,
+ .decode = g722_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
+ .max_lowres = 1,
+};
diff --git a/libavcodec/g722enc.c b/libavcodec/g722enc.c
new file mode 100644
index 0000000..bdc30d5
--- /dev/null
+++ b/libavcodec/g722enc.c
@@ -0,0 +1,311 @@
+/*
+ * 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 Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * G.722 ADPCM audio encoder
+ */
+
+#include "avcodec.h"
+#include "g722.h"
+
+#define FREEZE_INTERVAL 128
+
+static av_cold int g722_encode_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;
+ }
+
+ c->band[0].scale_factor = 8;
+ c->band[1].scale_factor = 2;
+ c->prev_samples_pos = 22;
+
+ if (avctx->trellis) {
+ int frontier = 1 << avctx->trellis;
+ int max_paths = frontier * FREEZE_INTERVAL;
+ int i;
+ for (i = 0; i < 2; i++) {
+ c->paths[i] = av_mallocz(max_paths * sizeof(**c->paths));
+ c->node_buf[i] = av_mallocz(2 * frontier * sizeof(**c->node_buf));
+ c->nodep_buf[i] = av_mallocz(2 * frontier * sizeof(**c->nodep_buf));
+ }
+ }
+
+ return 0;
+}
+
+static av_cold int g722_encode_close(AVCodecContext *avctx)
+{
+ G722Context *c = avctx->priv_data;
+ int i;
+ for (i = 0; i < 2; i++) {
+ av_freep(&c->paths[i]);
+ av_freep(&c->node_buf[i]);
+ av_freep(&c->nodep_buf[i]);
+ }
+ return 0;
+}
+
+static const int16_t low_quant[33] = {
+ 35, 72, 110, 150, 190, 233, 276, 323,
+ 370, 422, 473, 530, 587, 650, 714, 786,
+ 858, 940, 1023, 1121, 1219, 1339, 1458, 1612,
+ 1765, 1980, 2195, 2557, 2919
+};
+
+static inline void filter_samples(G722Context *c, const int16_t *samples,
+ int *xlow, int *xhigh)
+{
+ int xout1, xout2;
+ c->prev_samples[c->prev_samples_pos++] = samples[0];
+ c->prev_samples[c->prev_samples_pos++] = samples[1];
+ ff_g722_apply_qmf(c->prev_samples + c->prev_samples_pos - 24, &xout1, &xout2);
+ *xlow = xout1 + xout2 >> 13;
+ *xhigh = xout1 - xout2 >> 13;
+ 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;
+ }
+}
+
+static inline int encode_high(const struct G722Band *state, int xhigh)
+{
+ int diff = av_clip_int16(xhigh - state->s_predictor);
+ int pred = 141 * state->scale_factor >> 8;
+ /* = diff >= 0 ? (diff < pred) + 2 : diff >= -pred */
+ return ((diff ^ (diff >> (sizeof(diff)*8-1))) < pred) + 2*(diff >= 0);
+}
+
+static inline int encode_low(const struct G722Band* state, int xlow)
+{
+ int diff = av_clip_int16(xlow - state->s_predictor);
+ /* = diff >= 0 ? diff : -(diff + 1) */
+ int limit = diff ^ (diff >> (sizeof(diff)*8-1));
+ int i = 0;
+ limit = limit + 1 << 10;
+ if (limit > low_quant[8] * state->scale_factor)
+ i = 9;
+ while (i < 29 && limit > low_quant[i] * state->scale_factor)
+ i++;
+ return (diff < 0 ? (i < 2 ? 63 : 33) : 61) - i;
+}
+
+static int g722_encode_trellis(AVCodecContext *avctx,
+ uint8_t *dst, int buf_size, void *data)
+{
+ G722Context *c = avctx->priv_data;
+ const int16_t *samples = data;
+ int i, j, k;
+ int frontier = 1 << avctx->trellis;
+ struct TrellisNode **nodes[2];
+ struct TrellisNode **nodes_next[2];
+ int pathn[2] = {0, 0}, froze = -1;
+ struct TrellisPath *p[2];
+
+ for (i = 0; i < 2; i++) {
+ nodes[i] = c->nodep_buf[i];
+ nodes_next[i] = c->nodep_buf[i] + frontier;
+ memset(c->nodep_buf[i], 0, 2 * frontier * sizeof(*c->nodep_buf));
+ nodes[i][0] = c->node_buf[i] + frontier;
+ nodes[i][0]->ssd = 0;
+ nodes[i][0]->path = 0;
+ nodes[i][0]->state = c->band[i];
+ }
+
+ for (i = 0; i < buf_size >> 1; i++) {
+ int xlow, xhigh;
+ struct TrellisNode *next[2];
+ int heap_pos[2] = {0, 0};
+
+ for (j = 0; j < 2; j++) {
+ next[j] = c->node_buf[j] + frontier*(i & 1);
+ memset(nodes_next[j], 0, frontier * sizeof(**nodes_next));
+ }
+
+ filter_samples(c, &samples[2*i], &xlow, &xhigh);
+
+ for (j = 0; j < frontier && nodes[0][j]; j++) {
+ /* Only k >> 2 affects the future adaptive state, therefore testing
+ * small steps that don't change k >> 2 is useless, the orignal
+ * value from encode_low is better than them. Since we step k
+ * in steps of 4, make sure range is a multiple of 4, so that
+ * we don't miss the original value from encode_low. */
+ int range = j < frontier/2 ? 4 : 0;
+ struct TrellisNode *cur_node = nodes[0][j];
+
+ int ilow = encode_low(&cur_node->state, xlow);
+
+ for (k = ilow - range; k <= ilow + range && k <= 63; k += 4) {
+ int decoded, dec_diff, pos;
+ uint32_t ssd;
+ struct TrellisNode* node;
+
+ if (k < 0)
+ continue;
+
+ decoded = av_clip((cur_node->state.scale_factor *
+ ff_g722_low_inv_quant6[k] >> 10)
+ + cur_node->state.s_predictor, -16384, 16383);
+ dec_diff = xlow - decoded;
+
+#define STORE_NODE(index, UPDATE, VALUE)\
+ ssd = cur_node->ssd + dec_diff*dec_diff;\
+ /* Check for wraparound. Using 64 bit ssd counters would \
+ * be simpler, but is slower on x86 32 bit. */\
+ if (ssd < cur_node->ssd)\
+ continue;\
+ if (heap_pos[index] < frontier) {\
+ pos = heap_pos[index]++;\
+ assert(pathn[index] < FREEZE_INTERVAL * frontier);\
+ node = nodes_next[index][pos] = next[index]++;\
+ node->path = pathn[index]++;\
+ } else {\
+ /* Try to replace one of the leaf nodes with the new \
+ * one, but not always testing the same leaf position */\
+ pos = (frontier>>1) + (heap_pos[index] & ((frontier>>1) - 1));\
+ if (ssd >= nodes_next[index][pos]->ssd)\
+ continue;\
+ heap_pos[index]++;\
+ node = nodes_next[index][pos];\
+ }\
+ node->ssd = ssd;\
+ node->state = cur_node->state;\
+ UPDATE;\
+ c->paths[index][node->path].value = VALUE;\
+ c->paths[index][node->path].prev = cur_node->path;\
+ /* Sift the newly inserted node up in the heap to restore \
+ * the heap property */\
+ while (pos > 0) {\
+ int parent = (pos - 1) >> 1;\
+ if (nodes_next[index][parent]->ssd <= ssd)\
+ break;\
+ FFSWAP(struct TrellisNode*, nodes_next[index][parent],\
+ nodes_next[index][pos]);\
+ pos = parent;\
+ }
+ STORE_NODE(0, ff_g722_update_low_predictor(&node->state, k >> 2), k);
+ }
+ }
+
+ for (j = 0; j < frontier && nodes[1][j]; j++) {
+ int ihigh;
+ struct TrellisNode *cur_node = nodes[1][j];
+
+ /* We don't try to get any initial guess for ihigh via
+ * encode_high - since there's only 4 possible values, test
+ * them all. Testing all of these gives a much, much larger
+ * gain than testing a larger range around ilow. */
+ for (ihigh = 0; ihigh < 4; ihigh++) {
+ int dhigh, decoded, dec_diff, pos;
+ uint32_t ssd;
+ struct TrellisNode* node;
+
+ dhigh = cur_node->state.scale_factor *
+ ff_g722_high_inv_quant[ihigh] >> 10;
+ decoded = av_clip(dhigh + cur_node->state.s_predictor,
+ -16384, 16383);
+ dec_diff = xhigh - decoded;
+
+ STORE_NODE(1, ff_g722_update_high_predictor(&node->state, dhigh, ihigh), ihigh);
+ }
+ }
+
+ for (j = 0; j < 2; j++) {
+ FFSWAP(struct TrellisNode**, nodes[j], nodes_next[j]);
+
+ if (nodes[j][0]->ssd > (1 << 16)) {
+ for (k = 1; k < frontier && nodes[j][k]; k++)
+ nodes[j][k]->ssd -= nodes[j][0]->ssd;
+ nodes[j][0]->ssd = 0;
+ }
+ }
+
+ if (i == froze + FREEZE_INTERVAL) {
+ p[0] = &c->paths[0][nodes[0][0]->path];
+ p[1] = &c->paths[1][nodes[1][0]->path];
+ for (j = i; j > froze; j--) {
+ dst[j] = p[1]->value << 6 | p[0]->value;
+ p[0] = &c->paths[0][p[0]->prev];
+ p[1] = &c->paths[1][p[1]->prev];
+ }
+ froze = i;
+ pathn[0] = pathn[1] = 0;
+ memset(nodes[0] + 1, 0, (frontier - 1)*sizeof(**nodes));
+ memset(nodes[1] + 1, 0, (frontier - 1)*sizeof(**nodes));
+ }
+ }
+
+ p[0] = &c->paths[0][nodes[0][0]->path];
+ p[1] = &c->paths[1][nodes[1][0]->path];
+ for (j = i; j > froze; j--) {
+ dst[j] = p[1]->value << 6 | p[0]->value;
+ p[0] = &c->paths[0][p[0]->prev];
+ p[1] = &c->paths[1][p[1]->prev];
+ }
+ c->band[0] = nodes[0][0]->state;
+ c->band[1] = nodes[1][0]->state;
+
+ return i;
+}
+
+static int g722_encode_frame(AVCodecContext *avctx,
+ uint8_t *dst, int buf_size, void *data)
+{
+ G722Context *c = avctx->priv_data;
+ const int16_t *samples = data;
+ int i;
+
+ if (avctx->trellis)
+ return g722_encode_trellis(avctx, dst, buf_size, data);
+
+ for (i = 0; i < buf_size >> 1; i++) {
+ int xlow, xhigh, ihigh, ilow;
+ filter_samples(c, &samples[2*i], &xlow, &xhigh);
+ ihigh = encode_high(&c->band[1], xhigh);
+ ilow = encode_low(&c->band[0], xlow);
+ ff_g722_update_high_predictor(&c->band[1], c->band[1].scale_factor *
+ ff_g722_high_inv_quant[ihigh] >> 10, ihigh);
+ ff_g722_update_low_predictor(&c->band[0], ilow >> 2);
+ *dst++ = ihigh << 6 | ilow;
+ }
+ return i;
+}
+
+AVCodec ff_adpcm_g722_encoder = {
+ .name = "g722",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_ADPCM_G722,
+ .priv_data_size = sizeof(G722Context),
+ .init = g722_encode_init,
+ .close = g722_encode_close,
+ .encode = g722_encode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+};
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