From 651da919153e385f0769238c091109c06a142ca6 Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Sat, 21 Jan 2023 19:25:41 +0100 Subject: avcodec: add WavArc decoder --- Changelog | 1 + doc/general_contents.texi | 1 + libavcodec/Makefile | 1 + libavcodec/allcodecs.c | 1 + libavcodec/codec_desc.c | 7 + libavcodec/codec_id.h | 1 + libavcodec/version.h | 2 +- libavcodec/wavarc.c | 460 ++++++++++++++++++++++++++++++++++++++++++++++ 8 files changed, 473 insertions(+), 1 deletion(-) create mode 100644 libavcodec/wavarc.c diff --git a/Changelog b/Changelog index cdbe43eac1..2d2206e1ba 100644 --- a/Changelog +++ b/Changelog @@ -37,6 +37,7 @@ version : - XMD ADPCM decoder and demuxer - media100 to mjpegb bsf - ffmpeg CLI new option: -fix_sub_duration_heartbeat +- WavArc decoder version 5.1: diff --git a/doc/general_contents.texi b/doc/general_contents.texi index 87e180c979..84df3432cf 100644 --- a/doc/general_contents.texi +++ b/doc/general_contents.texi @@ -1356,6 +1356,7 @@ following image formats are supported: @item Vorbis @tab E @tab X @tab A native but very primitive encoder exists. @item Voxware MetaSound @tab @tab X +@item Waveform Archiver @tab @tab X @item WavPack @tab X @tab X @item Westwood Audio (SND1) @tab @tab X @item Windows Media Audio 1 @tab X @tab X diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 1fb963f820..4971832ff4 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -781,6 +781,7 @@ OBJS-$(CONFIG_VP9_V4L2M2M_DECODER) += v4l2_m2m_dec.o OBJS-$(CONFIG_VQA_DECODER) += vqavideo.o OBJS-$(CONFIG_VQC_DECODER) += vqcdec.o OBJS-$(CONFIG_WADY_DPCM_DECODER) += dpcm.o +OBJS-$(CONFIG_WAVARC_DECODER) += wavarc.o OBJS-$(CONFIG_WAVPACK_DECODER) += wavpack.o wavpackdata.o dsd.o OBJS-$(CONFIG_WAVPACK_ENCODER) += wavpackdata.o wavpackenc.o OBJS-$(CONFIG_WBMP_DECODER) += wbmpdec.o diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c index ff82423a88..b80b6983e9 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -538,6 +538,7 @@ extern const FFCodec ff_twinvq_decoder; extern const FFCodec ff_vmdaudio_decoder; extern const FFCodec ff_vorbis_encoder; extern const FFCodec ff_vorbis_decoder; +extern const FFCodec ff_wavarc_decoder; extern const FFCodec ff_wavpack_encoder; extern const FFCodec ff_wavpack_decoder; extern const FFCodec ff_wmalossless_decoder; diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c index 8ab228d846..57d0f98211 100644 --- a/libavcodec/codec_desc.c +++ b/libavcodec/codec_desc.c @@ -3353,6 +3353,13 @@ static const AVCodecDescriptor codec_descriptors[] = { .long_name = NULL_IF_CONFIG_SMALL("FTR Voice"), .props = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSY, }, + { + .id = AV_CODEC_ID_WAVARC, + .type = AVMEDIA_TYPE_AUDIO, + .name = "wavarc", + .long_name = NULL_IF_CONFIG_SMALL("Waveform Archiver"), + .props = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSLESS, + }, /* subtitle codecs */ { diff --git a/libavcodec/codec_id.h b/libavcodec/codec_id.h index 0c574c9619..ad1131b464 100644 --- a/libavcodec/codec_id.h +++ b/libavcodec/codec_id.h @@ -536,6 +536,7 @@ enum AVCodecID { AV_CODEC_ID_MISC4, AV_CODEC_ID_APAC, AV_CODEC_ID_FTR, + AV_CODEC_ID_WAVARC, /* subtitle codecs */ AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs. diff --git a/libavcodec/version.h b/libavcodec/version.h index 499c6bb175..310c80eeef 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -29,7 +29,7 @@ #include "version_major.h" -#define LIBAVCODEC_VERSION_MINOR 61 +#define LIBAVCODEC_VERSION_MINOR 62 #define LIBAVCODEC_VERSION_MICRO 100 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ diff --git a/libavcodec/wavarc.c b/libavcodec/wavarc.c new file mode 100644 index 0000000000..898c3c2055 --- /dev/null +++ b/libavcodec/wavarc.c @@ -0,0 +1,460 @@ +/* + * WavArc audio decoder + * Copyright (c) 2023 Paul B Mahol + * + * 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 + */ + +#include "libavutil/internal.h" +#include "libavutil/intreadwrite.h" +#include "avcodec.h" +#include "codec_internal.h" +#include "decode.h" +#include "get_bits.h" +#include "bytestream.h" +#include "mathops.h" +#include "unary.h" + +typedef struct WavArcContext { + GetBitContext gb; + + int shift; + int nb_samples; + int offset; + + int eof; + int skip; + uint8_t *bitstream; + int64_t max_framesize; + int bitstream_size; + int bitstream_index; + + int pred[2][70]; + int filter[2][70]; + int samples[2][640]; +} WavArcContext; + +static av_cold int wavarc_init(AVCodecContext *avctx) +{ + WavArcContext *s = avctx->priv_data; + + if (avctx->extradata_size < 44) + return AVERROR_INVALIDDATA; + if (AV_RL32(avctx->extradata + 16) != MKTAG('R','I','F','F')) + return AVERROR_INVALIDDATA; + if (AV_RL32(avctx->extradata + 24) != MKTAG('W','A','V','E')) + return AVERROR_INVALIDDATA; + if (AV_RL32(avctx->extradata + 28) != MKTAG('f','m','t',' ')) + return AVERROR_INVALIDDATA; + if (AV_RL16(avctx->extradata + 38) != 1 && + AV_RL16(avctx->extradata + 38) != 2) + return AVERROR_INVALIDDATA; + + av_channel_layout_uninit(&avctx->ch_layout); + av_channel_layout_default(&avctx->ch_layout, AV_RL16(avctx->extradata + 38)); + avctx->sample_rate = AV_RL32(avctx->extradata + 40); + + switch (avctx->extradata[36]) { + case 0: avctx->sample_fmt = AV_SAMPLE_FMT_U8P; break; + case 1: avctx->sample_fmt = AV_SAMPLE_FMT_S16P; break; + } + + s->shift = 0; + switch (avctx->codec_tag) { + case MKTAG('1','D','I','F'): + s->nb_samples = 256; + s->offset = 4; + break; + case MKTAG('2','S','L','P'): + case MKTAG('3','N','L','P'): + case MKTAG('4','A','L','P'): + s->nb_samples = 570; + s->offset = 70; + break; + default: + return AVERROR_INVALIDDATA; + } + + s->max_framesize = s->nb_samples * 16; + s->bitstream = av_calloc(s->max_framesize, sizeof(*s->bitstream)); + if (!s->bitstream) + return AVERROR(ENOMEM); + + return 0; +} + +static unsigned get_urice(GetBitContext *gb, int k) +{ + unsigned x = get_unary(gb, 1, get_bits_left(gb)); + unsigned y = get_bits_long(gb, k); + unsigned z = (x << k) | y; + + return z; +} + +static int get_srice(GetBitContext *gb, int k) +{ + unsigned z = get_urice(gb, k); + + return (z & 1) ? ~((int)(z >> 1)) : z >> 1; +} + +static void do_stereo(WavArcContext *s, int ch, int correlated, int len) +{ + const int nb_samples = s->nb_samples; + const int shift = s->shift; + + if (ch == 0) { + if (correlated) { + for (int n = 0; n < len; n++) { + s->samples[0][n] = s->samples[0][nb_samples + n] >> shift; + s->samples[1][n] = s->pred[1][n] >> shift; + } + } else { + for (int n = 0; n < len; n++) { + s->samples[0][n] = s->samples[0][nb_samples + n] >> shift; + s->samples[1][n] = s->pred[0][n] >> shift; + } + } + } else { + if (correlated) { + for (int n = 0; n < nb_samples; n++) + s->samples[1][n + len] += s->samples[0][n + len]; + } + for (int n = 0; n < len; n++) { + s->pred[0][n] = s->samples[1][nb_samples + n]; + s->pred[1][n] = s->pred[0][n] - s->samples[0][nb_samples + n]; + } + } +} + +static int decode_1dif(AVCodecContext *avctx, + WavArcContext *s, GetBitContext *gb) +{ + int ch, finished, fill, correlated; + + ch = 0; + finished = 0; + while (!finished) { + int *samples = s->samples[ch]; + int k, block_type; + + if (get_bits_left(gb) <= 0) + return AVERROR_INVALIDDATA; + + block_type = get_urice(gb, 1); + if (block_type < 4 && block_type >= 0) { + k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P); + k = get_urice(gb, k) + 1; + } + + switch (block_type) { + case 8: + s->eof = 1; + return AVERROR_EOF; + case 7: + s->nb_samples = get_bits(gb, 8); + continue; + case 6: + s->shift = get_urice(gb, 2); + continue; + case 5: + if (avctx->sample_fmt == AV_SAMPLE_FMT_U8P) { + fill = (int8_t)get_bits(gb, 8); + fill -= 0x80; + } else { + fill = (int16_t)get_bits(gb, 16); + fill -= 0x8000; + } + + for (int n = 0; n < s->nb_samples; n++) + samples[n + 4] = fill; + finished = 1; + break; + case 4: + for (int n = 0; n < s->nb_samples; n++) + samples[n + 4] = 0; + finished = 1; + break; + case 3: + for (int n = 0; n < s->nb_samples; n++) + samples[n + 4] = get_srice(gb, k) + (samples[n + 3] - samples[n + 2]) * 3 + + samples[n + 1]; + finished = 1; + break; + case 2: + for (int n = 0; n < s->nb_samples; n++) + samples[n + 4] = get_srice(gb, k) + (samples[n + 3] * 2 - samples[n + 2]); + finished = 1; + break; + case 1: + for (int n = 0; n < s->nb_samples; n++) + samples[n + 4] = get_srice(gb, k) + samples[n + 3]; + finished = 1; + break; + case 0: + for (int n = 0; n < s->nb_samples; n++) + samples[n + 4] = get_srice(gb, k); + finished = 1; + break; + default: + return AVERROR_INVALIDDATA; + } + + if (finished == 1 && avctx->ch_layout.nb_channels == 2) { + if (ch == 0) + correlated = get_bits1(gb); + finished = ch != 0; + do_stereo(s, ch, correlated, 4); + ch = 1; + } + } + + if (avctx->ch_layout.nb_channels == 1) { + for (int n = 0; n < 4; n++) + s->samples[0][n] = s->samples[0][s->nb_samples + n] >> s->shift; + } + + return 0; +} + +static int decode_2slp(AVCodecContext *avctx, + WavArcContext *s, GetBitContext *gb) +{ + int ch, finished, fill, correlated, order; + + ch = 0; + finished = 0; + while (!finished) { + int *samples = s->samples[ch]; + int k, block_type; + + if (get_bits_left(gb) <= 0) + return AVERROR_INVALIDDATA; + + block_type = get_urice(gb, 1); + if (block_type < 5 && block_type >= 0) { + k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P); + k = get_urice(gb, k) + 1; + } + + switch (block_type) { + case 9: + s->eof = 1; + return AVERROR_EOF; + case 8: + s->nb_samples = get_urice(gb, 8); + continue; + case 7: + s->shift = get_urice(gb, 2); + continue; + case 6: + if (avctx->sample_fmt == AV_SAMPLE_FMT_U8P) { + fill = (int8_t)get_bits(gb, 8); + fill -= 0x80; + } else { + fill = (int16_t)get_bits(gb, 16); + fill -= 0x8000; + } + + for (int n = 0; n < s->nb_samples; n++) + samples[n + 70] = fill; + finished = 1; + break; + case 5: + for (int n = 0; n < s->nb_samples; n++) + samples[n + 70] = 0; + finished = 1; + break; + case 4: + for (int n = 0; n < s->nb_samples; n++) + samples[n + 70] = get_srice(gb, k) + (samples[n + 69] - samples[n + 68]) * 3 + + samples[n + 67]; + finished = 1; + break; + case 3: + for (int n = 0; n < s->nb_samples; n++) + samples[n + 70] = get_srice(gb, k) + (samples[n + 69] * 2 - samples[n + 68]); + finished = 1; + break; + case 2: + for (int n = 0; n < s->nb_samples; n++) + samples[n + 70] = get_srice(gb, k); + finished = 1; + break; + case 1: + for (int n = 0; n < s->nb_samples; n++) + samples[n + 70] = get_srice(gb, k) + samples[n + 69]; + finished = 1; + break; + case 0: + order = get_urice(gb, 2); + for (int o = 0; o < order; o++) + s->filter[ch][o] = get_srice(gb, 2); + for (int n = 0; n < s->nb_samples; n++) { + int sum = 15; + + for (int o = 0; o < order; o++) + sum += s->filter[ch][o] * samples[n + 70 - o - 1]; + + samples[n + 70] = get_srice(gb, k) + (sum >> 4); + } + finished = 1; + break; + default: + return AVERROR_INVALIDDATA; + } + + if (finished == 1 && avctx->ch_layout.nb_channels == 2) { + if (ch == 0) + correlated = get_bits1(gb); + finished = ch != 0; + do_stereo(s, ch, correlated, 70); + ch = 1; + } + } + + if (avctx->ch_layout.nb_channels == 1) { + for (int n = 0; n < 70; n++) + s->samples[0][n] = s->samples[0][s->nb_samples + n] >> s->shift; + } + + return 0; +} + +static int wavarc_decode(AVCodecContext *avctx, AVFrame *frame, + int *got_frame_ptr, AVPacket *pkt) +{ + WavArcContext *s = avctx->priv_data; + GetBitContext *gb = &s->gb; + int buf_size, input_buf_size; + const uint8_t *buf; + int ret, n; + + if ((!pkt->size && !s->bitstream_size) || s->nb_samples == 0 || s->eof) { + *got_frame_ptr = 0; + return pkt->size; + } + + buf_size = FFMIN(pkt->size, s->max_framesize - s->bitstream_size); + input_buf_size = buf_size; + if (s->bitstream_index + s->bitstream_size + buf_size + AV_INPUT_BUFFER_PADDING_SIZE > s->max_framesize) { + memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size); + s->bitstream_index = 0; + } + if (pkt->data) + memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], pkt->data, buf_size); + buf = &s->bitstream[s->bitstream_index]; + buf_size += s->bitstream_size; + s->bitstream_size = buf_size; + if (buf_size < s->max_framesize && pkt->data) { + *got_frame_ptr = 0; + return input_buf_size; + } + + if ((ret = init_get_bits8(gb, buf, buf_size)) < 0) + return ret; + skip_bits(gb, s->skip); + + switch (avctx->codec_tag) { + case MKTAG('1','D','I','F'): + ret = decode_1dif(avctx, s, gb); + break; + case MKTAG('2','S','L','P'): + case MKTAG('3','N','L','P'): + case MKTAG('4','A','L','P'): + ret = decode_2slp(avctx, s, gb); + break; + default: + ret = AVERROR_INVALIDDATA; + } + + if (ret < 0) + goto fail; + + s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8); + n = get_bits_count(gb) / 8; + + if (n > buf_size) { +fail: + s->bitstream_size = 0; + s->bitstream_index = 0; + return ret; + } + + frame->nb_samples = s->nb_samples; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + + switch (avctx->sample_fmt) { + case AV_SAMPLE_FMT_U8P: + for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) { + uint8_t *dst = (uint8_t *)frame->extended_data[ch]; + const int *src = s->samples[ch] + s->offset; + + for (int n = 0; n < frame->nb_samples; n++) + dst[n] = src[n] * (1 << s->shift); + } + break; + case AV_SAMPLE_FMT_S16P: + for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) { + int16_t *dst = (int16_t *)frame->extended_data[ch]; + const int *src = s->samples[ch] + s->offset; + + for (int n = 0; n < frame->nb_samples; n++) + dst[n] = src[n] * (1 << s->shift); + } + break; + } + + *got_frame_ptr = 1; + + if (s->bitstream_size) { + s->bitstream_index += n; + s->bitstream_size -= n; + return input_buf_size; + } + + return n; +} + +static av_cold int wavarc_close(AVCodecContext *avctx) +{ + WavArcContext *s = avctx->priv_data; + + av_freep(&s->bitstream); + s->bitstream_size = 0; + + return 0; +} + +const FFCodec ff_wavarc_decoder = { + .p.name = "wavarc", + CODEC_LONG_NAME("Waveform Archiver"), + .p.type = AVMEDIA_TYPE_AUDIO, + .p.id = AV_CODEC_ID_WAVARC, + .priv_data_size = sizeof(WavArcContext), + .init = wavarc_init, + FF_CODEC_DECODE_CB(wavarc_decode), + .close = wavarc_close, + .p.capabilities = AV_CODEC_CAP_DR1 | + AV_CODEC_CAP_SUBFRAMES | + AV_CODEC_CAP_DELAY, + .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P, + AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_NONE }, +}; 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