ffmpeg-2.8.5

git-svn-id: svn://kolibrios.org@6147 a494cfbc-eb01-0410-851d-a64ba20cac60

contrib/sdk/sources/ffmpeg/ffmpeg-2.8/libavcodec/adxenc.c

@@ -0,0 +1,182 @@
+/*
+ * ADX ADPCM codecs
+ * Copyright (c) 2001,2003 BERO
+ *
+ * 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 "avcodec.h"
+#include "adx.h"
+#include "bytestream.h"
+#include "internal.h"
+#include "put_bits.h"
+
+/**
+ * @file
+ * SEGA CRI adx codecs.
+ *
+ * Reference documents:
+ * http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
+ * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
+ */
+
+static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
+                       ADXChannelState *prev, int channels)
+{
+    PutBitContext pb;
+    int scale;
+    int i, j;
+    int s0, s1, s2, d;
+    int max = 0;
+    int min = 0;
+
+    s1 = prev->s1;
+    s2 = prev->s2;
+    for (i = 0, j = 0; j < 32; i += channels, j++) {
+        s0 = wav[i];
+        d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
+        if (max < d)
+            max = d;
+        if (min > d)
+            min = d;
+        s2 = s1;
+        s1 = s0;
+    }
+
+    if (max == 0 && min == 0) {
+        prev->s1 = s1;
+        prev->s2 = s2;
+        memset(adx, 0, BLOCK_SIZE);
+        return;
+    }
+
+    if (max / 7 > -min / 8)
+        scale = max / 7;
+    else
+        scale = -min / 8;
+
+    if (scale == 0)
+        scale = 1;
+
+    AV_WB16(adx, scale);
+
+    init_put_bits(&pb, adx + 2, 16);
+
+    s1 = prev->s1;
+    s2 = prev->s2;
+    for (i = 0, j = 0; j < 32; i += channels, j++) {
+        d = ((wav[i] << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
+
+        d = av_clip_intp2(ROUNDED_DIV(d, scale), 3);
+
+        put_sbits(&pb, 4, d);
+
+        s0 = ((d << COEFF_BITS) * scale + c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS;
+        s2 = s1;
+        s1 = s0;
+    }
+    prev->s1 = s1;
+    prev->s2 = s2;
+
+    flush_put_bits(&pb);
+}
+
+#define HEADER_SIZE 36
+
+static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
+{
+    ADXContext *c = avctx->priv_data;
+
+    bytestream_put_be16(&buf, 0x8000);              /* header signature */
+    bytestream_put_be16(&buf, HEADER_SIZE - 4);     /* copyright offset */
+    bytestream_put_byte(&buf, 3);                   /* encoding */
+    bytestream_put_byte(&buf, BLOCK_SIZE);          /* block size */
+    bytestream_put_byte(&buf, 4);                   /* sample size */
+    bytestream_put_byte(&buf, avctx->channels);     /* channels */
+    bytestream_put_be32(&buf, avctx->sample_rate);  /* sample rate */
+    bytestream_put_be32(&buf, 0);                   /* total sample count */
+    bytestream_put_be16(&buf, c->cutoff);           /* cutoff frequency */
+    bytestream_put_byte(&buf, 3);                   /* version */
+    bytestream_put_byte(&buf, 0);                   /* flags */
+    bytestream_put_be32(&buf, 0);                   /* unknown */
+    bytestream_put_be32(&buf, 0);                   /* loop enabled */
+    bytestream_put_be16(&buf, 0);                   /* padding */
+    bytestream_put_buffer(&buf, "(c)CRI", 6);       /* copyright signature */
+
+    return HEADER_SIZE;
+}
+
+static av_cold int adx_encode_init(AVCodecContext *avctx)
+{
+    ADXContext *c = avctx->priv_data;
+
+    if (avctx->channels > 2) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
+        return AVERROR(EINVAL);
+    }
+    avctx->frame_size = BLOCK_SAMPLES;
+
+    /* the cutoff can be adjusted, but this seems to work pretty well */
+    c->cutoff = 500;
+    ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
+
+    return 0;
+}
+
+static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+                            const AVFrame *frame, int *got_packet_ptr)
+{
+    ADXContext *c          = avctx->priv_data;
+    const int16_t *samples = (const int16_t *)frame->data[0];
+    uint8_t *dst;
+    int ch, out_size, ret;
+
+    out_size = BLOCK_SIZE * avctx->channels + !c->header_parsed * HEADER_SIZE;
+    if ((ret = ff_alloc_packet2(avctx, avpkt, out_size, 0)) < 0)
+        return ret;
+    dst = avpkt->data;
+
+    if (!c->header_parsed) {
+        int hdrsize;
+        if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
+            av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
+            return AVERROR(EINVAL);
+        }
+        dst      += hdrsize;
+        c->header_parsed = 1;
+    }
+
+    for (ch = 0; ch < avctx->channels; ch++) {
+        adx_encode(c, dst, samples + ch, &c->prev[ch], avctx->channels);
+        dst += BLOCK_SIZE;
+    }
+
+    *got_packet_ptr = 1;
+    return 0;
+}
+
+AVCodec ff_adpcm_adx_encoder = {
+    .name           = "adpcm_adx",
+    .long_name      = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_ADPCM_ADX,
+    .priv_data_size = sizeof(ADXContext),
+    .init           = adx_encode_init,
+    .encode2        = adx_encode_frame,
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
+                                                      AV_SAMPLE_FMT_NONE },
+};