* Copyright (c) 2005 Eric Lasota
* Based on RoQ specs (c)2001 Tim Ferguson
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include "libavutil/intmath.h"
#include "avcodec.h"
#include "bytestream.h"
+#include "internal.h"
-#define ROQ_FIRST_FRAME_SIZE (735*8)
#define ROQ_FRAME_SIZE 735
-
+#define ROQ_HEADER_SIZE 8
#define MAX_DPCM (127*127)
-static unsigned char dpcmValues[MAX_DPCM];
typedef struct
{
short lastSample[2];
-} ROQDPCMContext_t;
+ int input_frames;
+ int buffered_samples;
+ int16_t *frame_buffer;
+ int64_t first_pts;
+} ROQDPCMContext;
+
-static av_cold void roq_dpcm_table_init(void)
+static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
{
- int i;
+ ROQDPCMContext *context = avctx->priv_data;
- /* Create a table of quick DPCM values */
- for (i=0; i<MAX_DPCM; i++) {
- int s= ff_sqrt(i);
- int mid= s*s + s;
- dpcmValues[i]= s + (i>mid);
- }
+#if FF_API_OLD_ENCODE_AUDIO
+ av_freep(&avctx->coded_frame);
+#endif
+ av_freep(&context->frame_buffer);
+
+ return 0;
}
-static int roq_dpcm_encode_init(AVCodecContext *avctx)
+static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
{
- ROQDPCMContext_t *context = avctx->priv_data;
+ 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 -1;
+ return AVERROR(EINVAL);
}
if (avctx->sample_rate != 22050) {
av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
- return -1;
+ return AVERROR(EINVAL);
}
- if (avctx->sample_fmt != SAMPLE_FMT_S16) {
- av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n");
- return -1;
- }
-
- roq_dpcm_table_init();
- avctx->frame_size = ROQ_FIRST_FRAME_SIZE;
+ 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;
+#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
- avctx->coded_frame->key_frame= 1;
+ if (!avctx->coded_frame) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
+#endif
return 0;
+error:
+ roq_dpcm_encode_close(avctx);
+ return ret;
}
static unsigned char dpcm_predict(short *previous, short current)
if (diff >= MAX_DPCM)
result = 127;
- else
- result = dpcmValues[diff];
+ else {
+ result = ff_sqrt(diff);
+ result += diff > result*result+result;
+ }
/* See if this overflows */
retry:
return result;
}
-static int roq_dpcm_encode_frame(AVCodecContext *avctx,
- unsigned char *frame, int buf_size, void *data)
+static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
- int i, samples, stereo, ch;
- short *in;
- unsigned char *out;
-
- ROQDPCMContext_t *context = avctx->priv_data;
+ 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;
+ }
+ in = context->frame_buffer;
+ }
+
if (stereo) {
context->lastSample[0] &= 0xFF00;
context->lastSample[1] &= 0xFF00;
}
- out = frame;
- in = data;
+ if (context->input_frames == 7 || !in)
+ data_size = avctx->channels * context->buffered_samples;
+ else
+ data_size = avctx->channels * avctx->frame_size;
+
+ if ((ret = ff_alloc_packet(avpkt, ROQ_HEADER_SIZE + data_size))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
+ out = avpkt->data;
bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
bytestream_put_byte(&out, 0x10);
- bytestream_put_le32(&out, avctx->frame_size*avctx->channels);
+ bytestream_put_le32(&out, data_size);
if (stereo) {
bytestream_put_byte(&out, (context->lastSample[1])>>8);
bytestream_put_le16(&out, context->lastSample[0]);
/* Write the actual samples */
- samples = avctx->frame_size;
- for (i=0; i<samples; i++)
- for (ch=0; ch<avctx->channels; ch++)
- *out++ = dpcm_predict(&context->lastSample[ch], *in++);
+ for (i = 0; i < data_size; i++)
+ *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
- /* Use smaller frames from now on */
- avctx->frame_size = ROQ_FRAME_SIZE;
-
- /* Return the result size */
- return out - frame;
-}
+ avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
+ avpkt->duration = data_size / avctx->channels;
-static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
-{
- av_freep(&avctx->coded_frame);
+ context->input_frames++;
+ if (!in)
+ context->input_frames = FFMAX(context->input_frames, 8);
+ *got_packet_ptr = 1;
return 0;
}
-AVCodec roq_dpcm_encoder = {
- "roq_dpcm",
- CODEC_TYPE_AUDIO,
- CODEC_ID_ROQ_DPCM,
- sizeof(ROQDPCMContext_t),
- roq_dpcm_encode_init,
- roq_dpcm_encode_frame,
- roq_dpcm_encode_close,
- NULL,
- .long_name = "id RoQ audio",
+AVCodec ff_roq_dpcm_encoder = {
+ .name = "roq_dpcm",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = 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"),
};