@@ -519,8 +519,8 @@ 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

@@ -45,7 +45,7 @@ static const AVOption eac3_options[] = {

 /* other metadata options */

 {"copyright", "Copyright Bit", OFFSET(copyright), AV_OPT_TYPE_INT, {.dbl = AC3ENC_OPT_NONE }, AC3ENC_OPT_NONE, 1, AC3ENC_PARAM},

 {"dialnorm", "Dialogue Level (dB)", OFFSET(dialogue_level), AV_OPT_TYPE_INT, {.dbl = -31 }, -31, -1, AC3ENC_PARAM},

-{"dsur_mode", "Dolby Surround Mode", OFFSET(dolby_surround_mode), AV_OPT_TYPE_INT, {.dbl = AC3ENC_OPT_NONE }, AC3ENC_OPT_NONE, AC3ENC_OPT_MODE_OFF, AC3ENC_PARAM, "dsur_mode"},

+{"dsur_mode", "Dolby Surround Mode", OFFSET(dolby_surround_mode), AV_OPT_TYPE_INT, {.dbl = AC3ENC_OPT_NONE }, AC3ENC_OPT_NONE, AC3ENC_OPT_MODE_ON, AC3ENC_PARAM, "dsur_mode"},

     {"notindicated", "Not Indicated (default)",    0, AV_OPT_TYPE_CONST, {.dbl = AC3ENC_OPT_NOT_INDICATED }, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"},

     {"on",           "Dolby Surround Encoded",     0, AV_OPT_TYPE_CONST, {.dbl = AC3ENC_OPT_MODE_ON       }, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"},

     {"off",          "Not Dolby Surround Encoded", 0, AV_OPT_TYPE_CONST, {.dbl = AC3ENC_OPT_MODE_OFF      }, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"},

@@ -59,11 +59,11 @@ static const AVOption eac3_options[] = {

 {"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, AC3ENC_PARAM},

 {"loro_cmixlev", "Lo/Ro Center Mix Level", OFFSET(loro_center_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, AC3ENC_PARAM},

 {"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, AC3ENC_PARAM},

-{"dsurex_mode", "Dolby Surround EX Mode", OFFSET(dolby_surround_ex_mode), AV_OPT_TYPE_INT, {.dbl = AC3ENC_OPT_NONE }, AC3ENC_OPT_NONE, AC3ENC_OPT_MODE_OFF, AC3ENC_PARAM, "dsurex_mode"},

+{"dsurex_mode", "Dolby Surround EX Mode", OFFSET(dolby_surround_ex_mode), AV_OPT_TYPE_INT, {.dbl = AC3ENC_OPT_NONE }, AC3ENC_OPT_NONE, AC3ENC_OPT_MODE_ON, AC3ENC_PARAM, "dsurex_mode"},

     {"notindicated", "Not Indicated (default)",       0, AV_OPT_TYPE_CONST, {.dbl = AC3ENC_OPT_NOT_INDICATED }, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"},

     {"on",           "Dolby Surround EX Encoded",     0, AV_OPT_TYPE_CONST, {.dbl = AC3ENC_OPT_MODE_ON       }, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"},

     {"off",          "Not Dolby Surround EX Encoded", 0, AV_OPT_TYPE_CONST, {.dbl = AC3ENC_OPT_MODE_OFF      }, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"},

-{"dheadphone_mode", "Dolby Headphone Mode", OFFSET(dolby_headphone_mode), AV_OPT_TYPE_INT, {.dbl = AC3ENC_OPT_NONE }, AC3ENC_OPT_NONE, AC3ENC_OPT_MODE_OFF, AC3ENC_PARAM, "dheadphone_mode"},

+{"dheadphone_mode", "Dolby Headphone Mode", OFFSET(dolby_headphone_mode), AV_OPT_TYPE_INT, {.dbl = AC3ENC_OPT_NONE }, AC3ENC_OPT_NONE, AC3ENC_OPT_MODE_ON, AC3ENC_PARAM, "dheadphone_mode"},

     {"notindicated", "Not Indicated (default)",     0, AV_OPT_TYPE_CONST, {.dbl = AC3ENC_OPT_NOT_INDICATED }, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"},

     {"on",           "Dolby Headphone Encoded",     0, AV_OPT_TYPE_CONST, {.dbl = AC3ENC_OPT_MODE_ON       }, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"},

     {"off",          "Not Dolby Headphone Encoded", 0, AV_OPT_TYPE_CONST, {.dbl = AC3ENC_OPT_MODE_OFF      }, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"},

@@ -29,11 +29,11 @@

 #endif

 /* MpegEncContext */

-#define Y_DC_SCALE               0xac

-#define C_DC_SCALE               0xb0

-#define AC_PRED                  0xb4

-#define BLOCK_LAST_INDEX         0xb8

-#define H263_AIC                 0xe8

-#define INTER_SCANTAB_RASTER_END 0x130

+#define Y_DC_SCALE               0xa8

+#define C_DC_SCALE               0xac

+#define AC_PRED                  0xb0

+#define BLOCK_LAST_INDEX         0xb4

+#define H263_AIC                 0xe4

+#define INTER_SCANTAB_RASTER_END 0x12c

 #endif /* AVCODEC_ARM_ASM_OFFSETS_H */

@@ -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

-

new file mode 100644

@@ -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 */

new file mode 100644

@@ -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,

+};

new file mode 100644

@@ -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));