adding ffmpeg specific version

1 files changed, 206 insertions, 0 deletions

diff --git a/ffmpeg1/libavcodec/roqaudioenc.c b/ffmpeg1/libavcodec/roqaudioenc.c
new file mode 100644
index 0000000..b68e3f8
--- /dev/null
+++ b/ffmpeg1/libavcodec/roqaudioenc.c
@@ -0,0 +1,206 @@
+/*
+ * 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",
+    .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 },
+    .long_name      = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
+};