From 8992cb1d0d07edc33d274f6d7924ecdf6f83d994 Mon Sep 17 00:00:00 2001 From: Tim Redfern Date: Thu, 5 Sep 2013 17:57:22 +0100 Subject: making act segmenter --- ffmpeg/libavcodec/binkaudio.c | 359 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 359 insertions(+) create mode 100644 ffmpeg/libavcodec/binkaudio.c (limited to 'ffmpeg/libavcodec/binkaudio.c') diff --git a/ffmpeg/libavcodec/binkaudio.c b/ffmpeg/libavcodec/binkaudio.c new file mode 100644 index 0000000..ef5569a --- /dev/null +++ b/ffmpeg/libavcodec/binkaudio.c @@ -0,0 +1,359 @@ +/* + * Bink Audio decoder + * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org) + * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu) + * + * 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 + * Bink Audio decoder + * + * Technical details here: + * http://wiki.multimedia.cx/index.php?title=Bink_Audio + */ + +#include "libavutil/channel_layout.h" +#include "avcodec.h" +#define BITSTREAM_READER_LE +#include "get_bits.h" +#include "dct.h" +#include "rdft.h" +#include "fmtconvert.h" +#include "internal.h" +#include "libavutil/intfloat.h" + +extern const uint16_t ff_wma_critical_freqs[25]; + +static float quant_table[96]; + +#define MAX_CHANNELS 2 +#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11) + +typedef struct { + GetBitContext gb; + int version_b; ///< Bink version 'b' + int first; + int channels; + int frame_len; ///< transform size (samples) + int overlap_len; ///< overlap size (samples) + int block_size; + int num_bands; + unsigned int *bands; + float root; + DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE]; + float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block + uint8_t *packet_buffer; + union { + RDFTContext rdft; + DCTContext dct; + } trans; +} BinkAudioContext; + + +static av_cold int decode_init(AVCodecContext *avctx) +{ + BinkAudioContext *s = avctx->priv_data; + int sample_rate = avctx->sample_rate; + int sample_rate_half; + int i; + int frame_len_bits; + + /* determine frame length */ + if (avctx->sample_rate < 22050) { + frame_len_bits = 9; + } else if (avctx->sample_rate < 44100) { + frame_len_bits = 10; + } else { + frame_len_bits = 11; + } + + if (avctx->channels < 1 || avctx->channels > MAX_CHANNELS) { + av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels); + return AVERROR_INVALIDDATA; + } + avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO : + AV_CH_LAYOUT_STEREO; + + s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b'; + + if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) { + // audio is already interleaved for the RDFT format variant + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; + sample_rate *= avctx->channels; + s->channels = 1; + if (!s->version_b) + frame_len_bits += av_log2(avctx->channels); + } else { + s->channels = avctx->channels; + avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; + } + + s->frame_len = 1 << frame_len_bits; + s->overlap_len = s->frame_len / 16; + s->block_size = (s->frame_len - s->overlap_len) * s->channels; + sample_rate_half = (sample_rate + 1) / 2; + if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) + s->root = 2.0 / (sqrt(s->frame_len) * 32768.0); + else + s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0); + for (i = 0; i < 96; i++) { + /* constant is result of 0.066399999/log10(M_E) */ + quant_table[i] = expf(i * 0.15289164787221953823f) * s->root; + } + + /* calculate number of bands */ + for (s->num_bands = 1; s->num_bands < 25; s->num_bands++) + if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1]) + break; + + s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands)); + if (!s->bands) + return AVERROR(ENOMEM); + + /* populate bands data */ + s->bands[0] = 2; + for (i = 1; i < s->num_bands; i++) + s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1; + s->bands[s->num_bands] = s->frame_len; + + s->first = 1; + + if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) + ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R); + else if (CONFIG_BINKAUDIO_DCT_DECODER) + ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III); + else + return -1; + + return 0; +} + +static float get_float(GetBitContext *gb) +{ + int power = get_bits(gb, 5); + float f = ldexpf(get_bits_long(gb, 23), power - 23); + if (get_bits1(gb)) + f = -f; + return f; +} + +static const uint8_t rle_length_tab[16] = { + 2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64 +}; + +/** + * Decode Bink Audio block + * @param[out] out Output buffer (must contain s->block_size elements) + * @return 0 on success, negative error code on failure + */ +static int decode_block(BinkAudioContext *s, float **out, int use_dct) +{ + int ch, i, j, k; + float q, quant[25]; + int width, coeff; + GetBitContext *gb = &s->gb; + + if (use_dct) + skip_bits(gb, 2); + + for (ch = 0; ch < s->channels; ch++) { + FFTSample *coeffs = out[ch]; + + if (s->version_b) { + if (get_bits_left(gb) < 64) + return AVERROR_INVALIDDATA; + coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root; + coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root; + } else { + if (get_bits_left(gb) < 58) + return AVERROR_INVALIDDATA; + coeffs[0] = get_float(gb) * s->root; + coeffs[1] = get_float(gb) * s->root; + } + + if (get_bits_left(gb) < s->num_bands * 8) + return AVERROR_INVALIDDATA; + for (i = 0; i < s->num_bands; i++) { + int value = get_bits(gb, 8); + quant[i] = quant_table[FFMIN(value, 95)]; + } + + k = 0; + q = quant[0]; + + // parse coefficients + i = 2; + while (i < s->frame_len) { + if (s->version_b) { + j = i + 16; + } else { + int v = get_bits1(gb); + if (v) { + v = get_bits(gb, 4); + j = i + rle_length_tab[v] * 8; + } else { + j = i + 8; + } + } + + j = FFMIN(j, s->frame_len); + + width = get_bits(gb, 4); + if (width == 0) { + memset(coeffs + i, 0, (j - i) * sizeof(*coeffs)); + i = j; + while (s->bands[k] < i) + q = quant[k++]; + } else { + while (i < j) { + if (s->bands[k] == i) + q = quant[k++]; + coeff = get_bits(gb, width); + if (coeff) { + int v; + v = get_bits1(gb); + if (v) + coeffs[i] = -q * coeff; + else + coeffs[i] = q * coeff; + } else { + coeffs[i] = 0.0f; + } + i++; + } + } + } + + if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) { + coeffs[0] /= 0.5; + s->trans.dct.dct_calc(&s->trans.dct, coeffs); + } + else if (CONFIG_BINKAUDIO_RDFT_DECODER) + s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs); + } + + for (ch = 0; ch < s->channels; ch++) { + int j; + int count = s->overlap_len * s->channels; + if (!s->first) { + j = ch; + for (i = 0; i < s->overlap_len; i++, j += s->channels) + out[ch][i] = (s->previous[ch][i] * (count - j) + + out[ch][i] * j) / count; + } + memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len], + s->overlap_len * sizeof(*s->previous[ch])); + } + + s->first = 0; + + return 0; +} + +static av_cold int decode_end(AVCodecContext *avctx) +{ + BinkAudioContext * s = avctx->priv_data; + av_freep(&s->bands); + av_freep(&s->packet_buffer); + if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) + ff_rdft_end(&s->trans.rdft); + else if (CONFIG_BINKAUDIO_DCT_DECODER) + ff_dct_end(&s->trans.dct); + + return 0; +} + +static void get_bits_align32(GetBitContext *s) +{ + int n = (-get_bits_count(s)) & 31; + if (n) skip_bits(s, n); +} + +static int decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ + BinkAudioContext *s = avctx->priv_data; + AVFrame *frame = data; + GetBitContext *gb = &s->gb; + int ret, consumed = 0; + + if (!get_bits_left(gb)) { + uint8_t *buf; + /* handle end-of-stream */ + if (!avpkt->size) { + *got_frame_ptr = 0; + return 0; + } + if (avpkt->size < 4) { + av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); + return AVERROR_INVALIDDATA; + } + buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE); + if (!buf) + return AVERROR(ENOMEM); + s->packet_buffer = buf; + memcpy(s->packet_buffer, avpkt->data, avpkt->size); + init_get_bits(gb, s->packet_buffer, avpkt->size * 8); + consumed = avpkt->size; + + /* skip reported size */ + skip_bits_long(gb, 32); + } + + /* get output buffer */ + frame->nb_samples = s->frame_len; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + + if (decode_block(s, (float **)frame->extended_data, + avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) { + av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n"); + return AVERROR_INVALIDDATA; + } + get_bits_align32(gb); + + frame->nb_samples = s->block_size / avctx->channels; + *got_frame_ptr = 1; + + return consumed; +} + +AVCodec ff_binkaudio_rdft_decoder = { + .name = "binkaudio_rdft", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_BINKAUDIO_RDFT, + .priv_data_size = sizeof(BinkAudioContext), + .init = decode_init, + .close = decode_end, + .decode = decode_frame, + .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1, + .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)") +}; + +AVCodec ff_binkaudio_dct_decoder = { + .name = "binkaudio_dct", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_BINKAUDIO_DCT, + .priv_data_size = sizeof(BinkAudioContext), + .init = decode_init, + .close = decode_end, + .decode = decode_frame, + .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1, + .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)") +}; -- cgit v1.2.3