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authorTim Redfern <tim@eclectronics.org>2013-09-05 17:57:22 +0100
committerTim Redfern <tim@eclectronics.org>2013-09-05 17:57:22 +0100
commit8992cb1d0d07edc33d274f6d7924ecdf6f83d994 (patch)
tree3a2c86846b7eec8137c1507e623fc7018f13d453 /ffmpeg/libavcodec/roqaudioenc.c
parent741fb4b9e135cfb161a749db88713229038577bb (diff)
making act segmenter
Diffstat (limited to 'ffmpeg/libavcodec/roqaudioenc.c')
-rw-r--r--ffmpeg/libavcodec/roqaudioenc.c206
1 files changed, 206 insertions, 0 deletions
diff --git a/ffmpeg/libavcodec/roqaudioenc.c b/ffmpeg/libavcodec/roqaudioenc.c
new file mode 100644
index 0000000..b68e3f8
--- /dev/null
+++ b/ffmpeg/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"),
+};