chasing indexing error

1 files changed, 0 insertions, 206 deletions

diff --git a/ffmpeg/libavcodec/roqaudioenc.c b/ffmpeg/libavcodec/roqaudioenc.c
deleted file mode 100644
index b0b76d0..0000000
--- a/ffmpeg/libavcodec/roqaudioenc.c
+++ /dev/null
@@ -1,206 +0,0 @@
-/*
- * RoQ audio encoder
- *
- * Copyright (c) 2005 Eric Lasota
- *    Based on RoQ specs (c)2001 Tim Ferguson
- *
- * 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 "bytestream.h"
-#include "internal.h"
-#include "mathops.h"
-
-#define ROQ_FRAME_SIZE           735
-#define ROQ_HEADER_SIZE   8
-
-#define MAX_DPCM (127*127)
-
-
-typedef struct
-{
-    short lastSample[2];
-    int input_frames;
-    int buffered_samples;
-    int16_t *frame_buffer;
-    int64_t first_pts;
-} ROQDPCMContext;
-
-
-static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
-{
-    ROQDPCMContext *context = avctx->priv_data;
-
-    av_freep(&context->frame_buffer);
-
-    return 0;
-}
-
-static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
-{
-    ROQDPCMContext *context = avctx->priv_data;
-    int ret;
-
-    if (avctx->channels > 2) {
-        av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
-        return AVERROR(EINVAL);
-    }
-    if (avctx->sample_rate != 22050) {
-        av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
-        return AVERROR(EINVAL);
-    }
-
-    avctx->frame_size = ROQ_FRAME_SIZE;
-    avctx->bit_rate   = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *
-                        (22050 / ROQ_FRAME_SIZE) * 8;
-
-    context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *
-                                      sizeof(*context->frame_buffer));
-    if (!context->frame_buffer) {
-        ret = AVERROR(ENOMEM);
-        goto error;
-    }
-
-    context->lastSample[0] = context->lastSample[1] = 0;
-
-    return 0;
-error:
-    roq_dpcm_encode_close(avctx);
-    return ret;
-}
-
-static unsigned char dpcm_predict(short *previous, short current)
-{
-    int diff;
-    int negative;
-    int result;
-    int predicted;
-
-    diff = current - *previous;
-
-    negative = diff<0;
-    diff = FFABS(diff);
-
-    if (diff >= MAX_DPCM)
-        result = 127;
-    else {
-        result = ff_sqrt(diff);
-        result += diff > result*result+result;
-    }
-
-    /* See if this overflows */
- retry:
-    diff = result*result;
-    if (negative)
-        diff = -diff;
-    predicted = *previous + diff;
-
-    /* If it overflows, back off a step */
-    if (predicted > 32767 || predicted < -32768) {
-        result--;
-        goto retry;
-    }
-
-    /* Add the sign bit */
-    result |= negative << 7;   //if (negative) result |= 128;
-
-    *previous = predicted;
-
-    return result;
-}
-
-static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                                 const AVFrame *frame, int *got_packet_ptr)
-{
-    int i, stereo, data_size, ret;
-    const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
-    uint8_t *out;
-    ROQDPCMContext *context = avctx->priv_data;
-
-    stereo = (avctx->channels == 2);
-
-    if (!in && context->input_frames >= 8)
-        return 0;
-
-    if (in && context->input_frames < 8) {
-        memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
-               in, avctx->frame_size * avctx->channels * sizeof(*in));
-        context->buffered_samples += avctx->frame_size;
-        if (context->input_frames == 0)
-            context->first_pts = frame->pts;
-        if (context->input_frames < 7) {
-            context->input_frames++;
-            return 0;
-        }
-    }
-    if (context->input_frames < 8) {
-        in = context->frame_buffer;
-    }
-
-    if (stereo) {
-        context->lastSample[0] &= 0xFF00;
-        context->lastSample[1] &= 0xFF00;
-    }
-
-    if (context->input_frames == 7)
-        data_size = avctx->channels * context->buffered_samples;
-    else
-        data_size = avctx->channels * avctx->frame_size;
-
-    if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size)) < 0)
-        return ret;
-    out = avpkt->data;
-
-    bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
-    bytestream_put_byte(&out, 0x10);
-    bytestream_put_le32(&out, data_size);
-
-    if (stereo) {
-        bytestream_put_byte(&out, (context->lastSample[1])>>8);
-        bytestream_put_byte(&out, (context->lastSample[0])>>8);
-    } else
-        bytestream_put_le16(&out, context->lastSample[0]);
-
-    /* Write the actual samples */
-    for (i = 0; i < data_size; i++)
-        *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
-
-    avpkt->pts      = context->input_frames <= 7 ? context->first_pts : frame->pts;
-    avpkt->duration = data_size / avctx->channels;
-
-    context->input_frames++;
-    if (!in)
-        context->input_frames = FFMAX(context->input_frames, 8);
-
-    *got_packet_ptr = 1;
-    return 0;
-}
-
-AVCodec ff_roq_dpcm_encoder = {
-    .name           = "roq_dpcm",
-    .long_name      = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = AV_CODEC_ID_ROQ_DPCM,
-    .priv_data_size = sizeof(ROQDPCMContext),
-    .init           = roq_dpcm_encode_init,
-    .encode2        = roq_dpcm_encode_frame,
-    .close          = roq_dpcm_encode_close,
-    .capabilities   = CODEC_CAP_DELAY,
-    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
-                                                     AV_SAMPLE_FMT_NONE },
-};