4 * Copyright (c) 2005 Eric Lasota
5 * Based on RoQ specs (c)2001 Tim Ferguson
7 * This file is part of FFmpeg.
9 * FFmpeg is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Lesser General Public
11 * License as published by the Free Software Foundation; either
12 * version 2.1 of the License, or (at your option) any later version.
14 * FFmpeg is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Lesser General Public License for more details.
19 * You should have received a copy of the GNU Lesser General Public
20 * License along with FFmpeg; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 #include "bytestream.h"
29 #define ROQ_FRAME_SIZE 735
30 #define ROQ_HEADER_SIZE 8
32 #define MAX_DPCM (127*127)
40 int16_t *frame_buffer;
45 static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
47 ROQDPCMContext *context = avctx->priv_data;
49 av_freep(&context->frame_buffer);
54 static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
56 ROQDPCMContext *context = avctx->priv_data;
59 if (avctx->channels > 2) {
60 av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
61 return AVERROR(EINVAL);
63 if (avctx->sample_rate != 22050) {
64 av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
65 return AVERROR(EINVAL);
68 avctx->frame_size = ROQ_FRAME_SIZE;
69 avctx->bit_rate = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) *
70 (22050 / ROQ_FRAME_SIZE) * 8;
72 context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels *
73 sizeof(*context->frame_buffer));
74 if (!context->frame_buffer) {
75 ret = AVERROR(ENOMEM);
79 context->lastSample[0] = context->lastSample[1] = 0;
83 roq_dpcm_encode_close(avctx);
87 static unsigned char dpcm_predict(short *previous, short current)
94 diff = current - *previous;
102 result = ff_sqrt(diff);
103 result += diff > result*result+result;
106 /* See if this overflows */
108 diff = result*result;
111 predicted = *previous + diff;
113 /* If it overflows, back off a step */
114 if (predicted > 32767 || predicted < -32768) {
119 /* Add the sign bit */
120 result |= negative << 7; //if (negative) result |= 128;
122 *previous = predicted;
127 static int roq_dpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
128 const AVFrame *frame, int *got_packet_ptr)
130 int i, stereo, data_size, ret;
131 const int16_t *in = frame ? (const int16_t *)frame->data[0] : NULL;
133 ROQDPCMContext *context = avctx->priv_data;
135 stereo = (avctx->channels == 2);
137 if (!in && context->input_frames >= 8)
140 if (in && context->input_frames < 8) {
141 memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels],
142 in, avctx->frame_size * avctx->channels * sizeof(*in));
143 context->buffered_samples += avctx->frame_size;
144 if (context->input_frames == 0)
145 context->first_pts = frame->pts;
146 if (context->input_frames < 7) {
147 context->input_frames++;
151 if (context->input_frames < 8) {
152 in = context->frame_buffer;
156 context->lastSample[0] &= 0xFF00;
157 context->lastSample[1] &= 0xFF00;
160 if (context->input_frames == 7)
161 data_size = avctx->channels * context->buffered_samples;
163 data_size = avctx->channels * avctx->frame_size;
165 if ((ret = ff_alloc_packet2(avctx, avpkt, ROQ_HEADER_SIZE + data_size)) < 0)
169 bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
170 bytestream_put_byte(&out, 0x10);
171 bytestream_put_le32(&out, data_size);
174 bytestream_put_byte(&out, (context->lastSample[1])>>8);
175 bytestream_put_byte(&out, (context->lastSample[0])>>8);
177 bytestream_put_le16(&out, context->lastSample[0]);
179 /* Write the actual samples */
180 for (i = 0; i < data_size; i++)
181 *out++ = dpcm_predict(&context->lastSample[i & 1], *in++);
183 avpkt->pts = context->input_frames <= 7 ? context->first_pts : frame->pts;
184 avpkt->duration = data_size / avctx->channels;
186 context->input_frames++;
188 context->input_frames = FFMAX(context->input_frames, 8);
194 AVCodec ff_roq_dpcm_encoder = {
196 .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
197 .type = AVMEDIA_TYPE_AUDIO,
198 .id = AV_CODEC_ID_ROQ_DPCM,
199 .priv_data_size = sizeof(ROQDPCMContext),
200 .init = roq_dpcm_encode_init,
201 .encode2 = roq_dpcm_encode_frame,
202 .close = roq_dpcm_encode_close,
203 .capabilities = CODEC_CAP_DELAY,
204 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
205 AV_SAMPLE_FMT_NONE },