/*
* Assorted DPCM codecs
- * Copyright (c) 2003 The ffmpeg Project.
+ * Copyright (c) 2003 The ffmpeg Project
*
- * This library is free software; you can redistribute it and/or
+ * This file is part of Libav.
+ *
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library 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 this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file: dpcm.c
+ * @file
* Assorted DPCM (differential pulse code modulation) audio codecs
* by Mike Melanson (melanson@pcisys.net)
+ * Xan DPCM decoder by Mario Brito (mbrito@student.dei.uc.pt)
* for more information on the specific data formats, visit:
* http://www.pcisys.net/~melanson/codecs/simpleaudio.html
+ * SOL DPCMs implemented by Konstantin Shishkov
+ *
+ * Note about using the Xan DPCM decoder: Xan DPCM is used in AVI files
+ * found in the Wing Commander IV computer game. These AVI files contain
+ * WAVEFORMAT headers which report the audio format as 0x01: raw PCM.
+ * Clearly incorrect. To detect Xan DPCM, you will probably have to
+ * special-case your AVI demuxer to use Xan DPCM if the file uses 'Xxan'
+ * (Xan video) for its video codec. Alternately, such AVI files also contain
+ * the fourcc 'Axan' in the 'auds' chunk of the AVI header.
*/
+#include "libavutil/intreadwrite.h"
#include "avcodec.h"
+#include "bytestream.h"
+#include "mathops.h"
typedef struct DPCMContext {
- int channels;
- short roq_square_array[256];
- int last_delta[2];
+ AVFrame frame;
+ int16_t roq_square_array[256];
+ int sample[2]; ///< previous sample (for SOL_DPCM)
+ const int8_t *sol_table; ///< delta table for SOL_DPCM
} DPCMContext;
-#define SATURATE_S16(x) if (x < -32768) x = -32768; \
- else if (x > 32767) x = 32767;
-#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000;
-#define LE_16(x) ((((uint8_t*)(x))[1] << 8) | ((uint8_t*)(x))[0])
-#define LE_32(x) ((((uint8_t*)(x))[3] << 24) | \
- (((uint8_t*)(x))[2] << 16) | \
- (((uint8_t*)(x))[1] << 8) | \
- ((uint8_t*)(x))[0])
-
-static int interplay_delta_table[] = {
+static const int16_t interplay_delta_table[] = {
0, 1, 2, 3, 4, 5, 6, 7,
8, 9, 10, 11, 12, 13, 14, 15,
16, 17, 18, 19, 20, 21, 22, 23,
};
-static int dpcm_decode_init(AVCodecContext *avctx)
+static const int8_t sol_table_old[16] = {
+ 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
+ -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0
+};
+
+static const int8_t sol_table_new[16] = {
+ 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
+ 0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15
+};
+
+static const int16_t sol_table_16[128] = {
+ 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
+ 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
+ 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
+ 0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
+ 0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
+ 0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
+ 0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
+ 0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
+ 0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
+ 0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
+ 0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
+ 0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
+ 0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
+};
+
+
+static av_cold int dpcm_decode_init(AVCodecContext *avctx)
{
DPCMContext *s = avctx->priv_data;
int i;
- short square;
- s->channels = avctx->channels;
+ if (avctx->channels < 1 || avctx->channels > 2) {
+ av_log(avctx, AV_LOG_INFO, "invalid number of channels\n");
+ return AVERROR(EINVAL);
+ }
+
+ s->sample[0] = s->sample[1] = 0;
switch(avctx->codec->id) {
- case CODEC_ID_ROQ_DPCM:
+ case AV_CODEC_ID_ROQ_DPCM:
/* initialize square table */
for (i = 0; i < 128; i++) {
- square = i * i;
- s->roq_square_array[i] = square;
+ int16_t square = i * i;
+ s->roq_square_array[i ] = square;
s->roq_square_array[i + 128] = -square;
}
break;
+ case AV_CODEC_ID_SOL_DPCM:
+ switch(avctx->codec_tag){
+ case 1:
+ s->sol_table = sol_table_old;
+ s->sample[0] = s->sample[1] = 0x80;
+ break;
+ case 2:
+ s->sol_table = sol_table_new;
+ s->sample[0] = s->sample[1] = 0x80;
+ break;
+ case 3:
+ break;
+ default:
+ av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n");
+ return -1;
+ }
+ break;
+
default:
break;
}
+ if (avctx->codec->id == AV_CODEC_ID_SOL_DPCM && avctx->codec_tag != 3)
+ avctx->sample_fmt = AV_SAMPLE_FMT_U8;
+ else
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
return 0;
}
-static int dpcm_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- uint8_t *buf, int buf_size)
+
+static int dpcm_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
+ int buf_size = avpkt->size;
DPCMContext *s = avctx->priv_data;
- int in, out = 0;
- int i;
+ int out = 0, ret;
int predictor[2];
- int channel_number = 0;
- short *output_samples = data;
- int sequence_number;
+ int ch = 0;
+ int stereo = avctx->channels - 1;
+ int16_t *output_samples, *samples_end;
+ GetByteContext gb;
+
+ if (stereo && (buf_size & 1))
+ buf_size--;
+ bytestream2_init(&gb, avpkt->data, buf_size);
+
+ /* calculate output size */
+ switch(avctx->codec->id) {
+ case AV_CODEC_ID_ROQ_DPCM:
+ out = buf_size - 8;
+ break;
+ case AV_CODEC_ID_INTERPLAY_DPCM:
+ out = buf_size - 6 - avctx->channels;
+ break;
+ case AV_CODEC_ID_XAN_DPCM:
+ out = buf_size - 2 * avctx->channels;
+ break;
+ case AV_CODEC_ID_SOL_DPCM:
+ if (avctx->codec_tag != 3)
+ out = buf_size * 2;
+ else
+ out = buf_size;
+ break;
+ }
+ if (out <= 0) {
+ av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
+ return AVERROR(EINVAL);
+ }
+
+ /* get output buffer */
+ s->frame.nb_samples = out / avctx->channels;
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ output_samples = (int16_t *)s->frame.data[0];
+ samples_end = output_samples + out;
switch(avctx->codec->id) {
- case CODEC_ID_ROQ_DPCM:
- if (s->channels == 1)
- predictor[0] = LE_16(&buf[6]);
- else {
- predictor[0] = buf[7] << 8;
- predictor[1] = buf[6] << 8;
+ case AV_CODEC_ID_ROQ_DPCM:
+ bytestream2_skipu(&gb, 6);
+
+ if (stereo) {
+ predictor[1] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
+ predictor[0] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
+ } else {
+ predictor[0] = sign_extend(bytestream2_get_le16u(&gb), 16);
}
- SE_16BIT(predictor[0]);
- SE_16BIT(predictor[1]);
/* decode the samples */
- for (in = 8, out = 0; in < buf_size; in++, out++) {
- predictor[channel_number] += s->roq_square_array[buf[in]];
- SATURATE_S16(predictor[channel_number]);
- output_samples[out] = predictor[channel_number];
+ while (output_samples < samples_end) {
+ predictor[ch] += s->roq_square_array[bytestream2_get_byteu(&gb)];
+ predictor[ch] = av_clip_int16(predictor[ch]);
+ *output_samples++ = predictor[ch];
/* toggle channel */
- channel_number ^= s->channels - 1;
+ ch ^= stereo;
}
break;
- case CODEC_ID_INTERPLAY_DPCM:
- in = 0;
- sequence_number = LE_16(&buf[in]);
- in += 6; /* skip over the stream mask and stream length */
- if (sequence_number == 1) {
- predictor[0] = LE_16(&buf[in]);
- in += 2;
- SE_16BIT(predictor[0])
- if (s->channels == 2) {
- predictor[1] = LE_16(&buf[in]);
- SE_16BIT(predictor[1])
- in += 2;
- }
- } else {
- for (i = 0; i < s->channels; i++)
- predictor[i] = s->last_delta[i];
+ case AV_CODEC_ID_INTERPLAY_DPCM:
+ bytestream2_skipu(&gb, 6); /* skip over the stream mask and stream length */
+
+ for (ch = 0; ch < avctx->channels; ch++) {
+ predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
+ *output_samples++ = predictor[ch];
}
- while (in < buf_size) {
- predictor[channel_number] += interplay_delta_table[buf[in++]];
- SATURATE_S16(predictor[channel_number]);
- output_samples[out++] = predictor[channel_number];
+ ch = 0;
+ while (output_samples < samples_end) {
+ predictor[ch] += interplay_delta_table[bytestream2_get_byteu(&gb)];
+ predictor[ch] = av_clip_int16(predictor[ch]);
+ *output_samples++ = predictor[ch];
+
+ /* toggle channel */
+ ch ^= stereo;
+ }
+ break;
+
+ case AV_CODEC_ID_XAN_DPCM:
+ {
+ int shift[2] = { 4, 4 };
+
+ for (ch = 0; ch < avctx->channels; ch++)
+ predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
+
+ ch = 0;
+ while (output_samples < samples_end) {
+ int diff = bytestream2_get_byteu(&gb);
+ int n = diff & 3;
+
+ if (n == 3)
+ shift[ch]++;
+ else
+ shift[ch] -= (2 * n);
+ diff = sign_extend((diff &~ 3) << 8, 16);
+
+ /* saturate the shifter to a lower limit of 0 */
+ if (shift[ch] < 0)
+ shift[ch] = 0;
+
+ diff >>= shift[ch];
+ predictor[ch] += diff;
+
+ predictor[ch] = av_clip_int16(predictor[ch]);
+ *output_samples++ = predictor[ch];
/* toggle channel */
- channel_number ^= s->channels - 1;
+ ch ^= stereo;
}
+ break;
+ }
+ case AV_CODEC_ID_SOL_DPCM:
+ if (avctx->codec_tag != 3) {
+ uint8_t *output_samples_u8 = s->frame.data[0],
+ *samples_end_u8 = output_samples_u8 + out;
+ while (output_samples_u8 < samples_end_u8) {
+ int n = bytestream2_get_byteu(&gb);
- /* save predictors for next round */
- for (i = 0; i < s->channels; i++)
- s->last_delta[i] = predictor[i];
+ s->sample[0] += s->sol_table[n >> 4];
+ s->sample[0] = av_clip_uint8(s->sample[0]);
+ *output_samples_u8++ = s->sample[0];
+ s->sample[stereo] += s->sol_table[n & 0x0F];
+ s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
+ *output_samples_u8++ = s->sample[stereo];
+ }
+ } else {
+ while (output_samples < samples_end) {
+ int n = bytestream2_get_byteu(&gb);
+ if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
+ else s->sample[ch] += sol_table_16[n & 0x7F];
+ s->sample[ch] = av_clip_int16(s->sample[ch]);
+ *output_samples++ = s->sample[ch];
+ /* toggle channel */
+ ch ^= stereo;
+ }
+ }
break;
}
- *data_size = out * sizeof(short);
- return buf_size;
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
+
+ return avpkt->size;
}
-AVCodec roq_dpcm_decoder = {
- "roq_dpcm",
- CODEC_TYPE_AUDIO,
- CODEC_ID_ROQ_DPCM,
- sizeof(DPCMContext),
- dpcm_decode_init,
- NULL,
- NULL,
- dpcm_decode_frame,
-};
+#define DPCM_DECODER(id_, name_, long_name_) \
+AVCodec ff_ ## name_ ## _decoder = { \
+ .name = #name_, \
+ .type = AVMEDIA_TYPE_AUDIO, \
+ .id = id_, \
+ .priv_data_size = sizeof(DPCMContext), \
+ .init = dpcm_decode_init, \
+ .decode = dpcm_decode_frame, \
+ .capabilities = CODEC_CAP_DR1, \
+ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
+}
-AVCodec interplay_dpcm_decoder = {
- "interplay_dpcm",
- CODEC_TYPE_AUDIO,
- CODEC_ID_INTERPLAY_DPCM,
- sizeof(DPCMContext),
- dpcm_decode_init,
- NULL,
- NULL,
- dpcm_decode_frame,
-};
+DPCM_DECODER(AV_CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
+DPCM_DECODER(AV_CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
+DPCM_DECODER(AV_CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
+DPCM_DECODER(AV_CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");
projects / ffmpeg / blobdiff