virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-18 03:19:31 +02:00
Code Issues Releases Activity
618fb9cc06
FFmpeg/libavcodec/dpcm.c
Michael Niedermayer e4de71677f Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  aac_latm: reconfigure decoder on audio specific config changes
  latmdec: fix audio specific config parsing
  Add avcodec_decode_audio4().
  avcodec: change number of plane pointers from 4 to 8 at next major bump.
  Update developers documentation with coding conventions.
  svq1dec: avoid undefined get_bits(0) call
  ARM: h264dsp_neon cosmetics
  ARM: make some NEON macros reusable
  Do not memcpy raw video frames when using null muxer
  fate: update asf seektest
  vp8: flush buffers on size changes.
  doc: improve general documentation for MacOSX
  asf: use packet dts as approximation of pts
  asf: do not call av_read_frame
  rtsp: Initialize the media_type_mask in the rtp guessing demuxer
  Cleaned up alacenc.c

Conflicts:
	doc/APIchanges
	doc/developer.texi
	libavcodec/8svx.c
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/nellymoserdec.c
	libavcodec/tta.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/asfdec.c
	tests/ref/seek/lavf_asf

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-03 03:00:30 +01:00

339 lines
12 KiB
C
Raw Blame History

/*
* Assorted DPCM codecs
* Copyright (c) 2003 The ffmpeg Project
*
* 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
*/
/**
* @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"
typedef struct DPCMContext {
AVFrame frame;
int channels;
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;
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,
24, 25, 26, 27, 28, 29, 30, 31,
32, 33, 34, 35, 36, 37, 38, 39,
40, 41, 42, 43, 47, 51, 56, 61,
66, 72, 79, 86, 94, 102, 112, 122,
133, 145, 158, 173, 189, 206, 225, 245,
267, 292, 318, 348, 379, 414, 452, 493,
538, 587, 640, 699, 763, 832, 908, 991,
1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993,
2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008,
4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059,
8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206,
17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589,
-29973, -26728, -23186, -19322, -15105, -10503, -5481, -1,
1, 1, 5481, 10503, 15105, 19322, 23186, 26728,
29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298,
-17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597,
-8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772,
-4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373,
-2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
-1081, -991, -908, -832, -763, -699, -640, -587,
-538, -493, -452, -414, -379, -348, -318, -292,
-267, -245, -225, -206, -189, -173, -158, -145,
-133, -122, -112, -102, -94, -86, -79, -72,
-66, -61, -56, -51, -47, -43, -42, -41,
-40, -39, -38, -37, -36, -35, -34, -33,
-32, -31, -30, -29, -28, -27, -26, -25,
-24, -23, -22, -21, -20, -19, -18, -17,
-16, -15, -14, -13, -12, -11, -10, -9,
-8, -7, -6, -5, -4, -3, -2, -1
};
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;
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_INFO, "invalid number of channels\n");
return AVERROR(EINVAL);
}
s->channels = avctx->channels;
s->sample[0] = s->sample[1] = 0;
switch(avctx->codec->id) {
case CODEC_ID_ROQ_DPCM:
/* initialize square table */
for (i = 0; i < 128; i++) {
int16_t square = i * i;
s->roq_square_array[i ] = square;
s->roq_square_array[i + 128] = -square;
}
break;
case 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 == 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 *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
const uint8_t *buf_end = buf + buf_size;
DPCMContext *s = avctx->priv_data;
int out = 0, ret;
int predictor[2];
int ch = 0;
int stereo = s->channels - 1;
int16_t *output_samples;
/* calculate output size */
switch(avctx->codec->id) {
case CODEC_ID_ROQ_DPCM:
out = buf_size - 8;
break;
case CODEC_ID_INTERPLAY_DPCM:
out = buf_size - 6 - s->channels;
break;
case CODEC_ID_XAN_DPCM:
out = buf_size - 2 * s->channels;
break;
case 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 / s->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];
switch(avctx->codec->id) {
case CODEC_ID_ROQ_DPCM:
buf += 6;
if (stereo) {
predictor[1] = (int16_t)(bytestream_get_byte(&buf) << 8);
predictor[0] = (int16_t)(bytestream_get_byte(&buf) << 8);
} else {
predictor[0] = (int16_t)bytestream_get_le16(&buf);
}
/* decode the samples */
while (buf < buf_end) {
predictor[ch] += s->roq_square_array[*buf++];
predictor[ch] = av_clip_int16(predictor[ch]);
*output_samples++ = predictor[ch];
/* toggle channel */
ch ^= stereo;
}
break;
case CODEC_ID_INTERPLAY_DPCM:
buf += 6; /* skip over the stream mask and stream length */
for (ch = 0; ch < s->channels; ch++) {
predictor[ch] = (int16_t)bytestream_get_le16(&buf);
*output_samples++ = predictor[ch];
}
ch = 0;
while (buf < buf_end) {
predictor[ch] += interplay_delta_table[*buf++];
predictor[ch] = av_clip_int16(predictor[ch]);
*output_samples++ = predictor[ch];
/* toggle channel */
ch ^= stereo;
}
break;
case CODEC_ID_XAN_DPCM:
{
int shift[2] = { 4, 4 };
for (ch = 0; ch < s->channels; ch++)
predictor[ch] = (int16_t)bytestream_get_le16(&buf);
ch = 0;
while (buf < buf_end) {
uint8_t n = *buf++;
int16_t diff = (n & 0xFC) << 8;
if ((n & 0x03) == 3)
shift[ch]++;
else
shift[ch] -= (2 * (n & 3));
/* 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 */
ch ^= stereo;
}
break;
}
case CODEC_ID_SOL_DPCM:
if (avctx->codec_tag != 3) {
uint8_t *output_samples_u8 = data;
while (buf < buf_end) {
uint8_t n = *buf++;
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 (buf < buf_end) {
uint8_t n = *buf++;
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;
}
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
#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_), \
}
DPCM_DECODER(CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
DPCM_DECODER(CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
DPCM_DECODER(CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
DPCM_DECODER(CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");
Reference in New Issue View Git Blame Copy Permalink