X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fdpcm.c;h=b9f5b93ebcb94a2a21d54a1b6ffd097c5ae46dd5;hb=a853388d2fc5be848cca839a9fdf39a97c2d7b0e;hp=df9da948966fe1654e9bfec7ab7b719179402f0f;hpb=62a05b5b009ae49a220e739102b440fc13cb9418;p=ffmpeg diff --git a/libavcodec/dpcm.c b/libavcodec/dpcm.c index df9da948966..b9f5b93ebcb 100644 --- a/libavcodec/dpcm.c +++ b/libavcodec/dpcm.c @@ -1,24 +1,26 @@ /* * 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 + * 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) @@ -35,20 +37,19 @@ * the fourcc 'Axan' in the 'auds' chunk of the AVI header. */ +#include "libavutil/intreadwrite.h" #include "avcodec.h" +#include "bytestream.h" +#include "internal.h" +#include "mathops.h" typedef struct DPCMContext { - int channels; - short roq_square_array[256]; - long sample[2];//for SOL_DPCM - const int *sol_table;//for SOL_DPCM + 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; - -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, @@ -84,15 +85,17 @@ static int interplay_delta_table[] = { }; -static const int 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_old[16] = { + 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15, + -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0 +}; -static const int sol_table_new[16] = - { 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15, - 0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15}; +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 int sol_table_16[128] = { +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, @@ -109,40 +112,40 @@ static const int sol_table_16[128] = { }; - -static int dpcm_decode_init(AVCodecContext *avctx) +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 CODEC_ID_SOL_DPCM: + case AV_CODEC_ID_SOL_DPCM: switch(avctx->codec_tag){ case 1: - s->sol_table=sol_table_old; + s->sol_table = sol_table_old; s->sample[0] = s->sample[1] = 0x80; break; case 2: - s->sol_table=sol_table_new; + s->sol_table = sol_table_new; s->sample[0] = s->sample[1] = 0x80; break; case 3: - s->sol_table=sol_table_16; break; default: av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n"); @@ -154,180 +157,186 @@ static int dpcm_decode_init(AVCodecContext *avctx) 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; + 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; + AVFrame *frame = data; + int out = 0, ret; int predictor[2]; - int channel_number = 0; - short *output_samples = data; - int shift[2]; - unsigned char byte; - short diff; + int ch = 0; + int stereo = avctx->channels - 1; + int16_t *output_samples, *samples_end; + GetByteContext gb; - if (!buf_size) - return 0; + 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 */ + frame->nb_samples = out / avctx->channels; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + output_samples = (int16_t *)frame->data[0]; + samples_end = output_samples + out; - 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; + switch(avctx->codec->id) { + + 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 = 6; /* skip over the stream mask and stream length */ - predictor[0] = LE_16(&buf[in]); - in += 2; - SE_16BIT(predictor[0]) - output_samples[out++] = predictor[0]; - if (s->channels == 2) { - predictor[1] = LE_16(&buf[in]); - in += 2; - SE_16BIT(predictor[1]) - output_samples[out++] = predictor[1]; + 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 */ - channel_number ^= s->channels - 1; + ch ^= stereo; } - break; - case CODEC_ID_XAN_DPCM: - in = 0; - shift[0] = shift[1] = 4; - predictor[0] = LE_16(&buf[in]); - in += 2; - SE_16BIT(predictor[0]); - if (s->channels == 2) { - predictor[1] = LE_16(&buf[in]); - in += 2; - SE_16BIT(predictor[1]); - } + 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); - while (in < buf_size) { - byte = buf[in++]; - diff = (byte & 0xFC) << 8; - if ((byte & 0x03) == 3) - shift[channel_number]++; + ch = 0; + while (output_samples < samples_end) { + int diff = bytestream2_get_byteu(&gb); + int n = diff & 3; + + if (n == 3) + shift[ch]++; else - shift[channel_number] -= (2 * (byte & 3)); + shift[ch] -= (2 * n); + diff = sign_extend((diff &~ 3) << 8, 16); + /* saturate the shifter to a lower limit of 0 */ - if (shift[channel_number] < 0) - shift[channel_number] = 0; + if (shift[ch] < 0) + shift[ch] = 0; - diff >>= shift[channel_number]; - predictor[channel_number] += diff; + diff >>= shift[ch]; + predictor[ch] += diff; - SATURATE_S16(predictor[channel_number]); - output_samples[out++] = predictor[channel_number]; + predictor[ch] = av_clip_int16(predictor[ch]); + *output_samples++ = predictor[ch]; /* toggle channel */ - channel_number ^= s->channels - 1; + ch ^= stereo; } break; - case CODEC_ID_SOL_DPCM: - in = 0; + } + case AV_CODEC_ID_SOL_DPCM: if (avctx->codec_tag != 3) { - while (in < buf_size) { - int n1, n2; - n1 = (buf[in] >> 4) & 0xF; - n2 = buf[in++] & 0xF; - s->sample[0] += s->sol_table[n1]; - if (s->sample[0] < 0) s->sample[0] = 0; - if (s->sample[0] > 255) s->sample[0] = 255; - output_samples[out++] = (s->sample[0] - 128) << 8; - s->sample[s->channels - 1] += s->sol_table[n2]; - if (s->sample[s->channels - 1] < 0) s->sample[s->channels - 1] = 0; - if (s->sample[s->channels - 1] > 255) s->sample[s->channels - 1] = 255; - output_samples[out++] = (s->sample[s->channels - 1] - 128) << 8; + uint8_t *output_samples_u8 = frame->data[0], + *samples_end_u8 = output_samples_u8 + out; + while (output_samples_u8 < samples_end_u8) { + int n = bytestream2_get_byteu(&gb); + + 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 (in < buf_size) { - int n; - n = buf[in++]; - if (n & 0x80) s->sample[channel_number] -= s->sol_table[n & 0x7F]; - else s->sample[channel_number] += s->sol_table[n & 0x7F]; - SATURATE_S16(s->sample[channel_number]); - output_samples[out++] = s->sample[channel_number]; + 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 */ - channel_number ^= s->channels - 1; + ch ^= stereo; } } break; } - *data_size = out * sizeof(short); - return buf_size; -} + *got_frame_ptr = 1; -AVCodec roq_dpcm_decoder = { - "roq_dpcm", - CODEC_TYPE_AUDIO, - CODEC_ID_ROQ_DPCM, - sizeof(DPCMContext), - dpcm_decode_init, - NULL, - NULL, - dpcm_decode_frame, -}; - -AVCodec interplay_dpcm_decoder = { - "interplay_dpcm", - CODEC_TYPE_AUDIO, - CODEC_ID_INTERPLAY_DPCM, - sizeof(DPCMContext), - dpcm_decode_init, - NULL, - NULL, - dpcm_decode_frame, -}; + return avpkt->size; +} -AVCodec xan_dpcm_decoder = { - "xan_dpcm", - CODEC_TYPE_AUDIO, - CODEC_ID_XAN_DPCM, - sizeof(DPCMContext), - dpcm_decode_init, - NULL, - NULL, - dpcm_decode_frame, -}; +#define DPCM_DECODER(id_, name_, long_name_) \ +AVCodec ff_ ## name_ ## _decoder = { \ + .name = #name_, \ + .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ + .type = AVMEDIA_TYPE_AUDIO, \ + .id = id_, \ + .priv_data_size = sizeof(DPCMContext), \ + .init = dpcm_decode_init, \ + .decode = dpcm_decode_frame, \ + .capabilities = AV_CODEC_CAP_DR1, \ +} -AVCodec sol_dpcm_decoder = { - "sol_dpcm", - CODEC_TYPE_AUDIO, - CODEC_ID_SOL_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");