X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fbinkaudio.c;h=d73ffcdabc36cfdcb8941ac0dd3014e514eb4275;hb=f907615f0813e8499f06a7eebccf1c63fce87c8e;hp=53484654db74f405a980a96a43352fccf8004180;hpb=c73d99e672329c8f2df290736ffc474c360ac4ae;p=ffmpeg diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c index 53484654db7..d73ffcdabc3 100644 --- a/libavcodec/binkaudio.c +++ b/libavcodec/binkaudio.c @@ -1,22 +1,22 @@ /* * Bink Audio decoder - * Copyright (c) 2007-2010 Peter Ross (pross@xvid.org) + * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org) * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu) * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * 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 2.1 of the License, or (at your option) any later version. * - * FFmpeg 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 FFmpeg; 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 */ @@ -29,22 +29,27 @@ */ #include "avcodec.h" -#define ALT_BITSTREAM_READER_LE +#define BITSTREAM_READER_LE #include "get_bits.h" #include "dsputil.h" -#include "fft.h" +#include "dct.h" +#include "rdft.h" #include "fmtconvert.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 { - AVCodecContext *avctx; + AVFrame frame; GetBitContext gb; DSPContext dsp; FmtConvertContext fmt_conv; + int version_b; ///< Bink version 'b' int first; int channels; int frame_len; ///< transform size (samples) @@ -53,9 +58,12 @@ typedef struct { int num_bands; unsigned int *bands; float root; - DECLARE_ALIGNED(16, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE]; - DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block + DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE]; + DECLARE_ALIGNED(16, int16_t, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block + DECLARE_ALIGNED(16, int16_t, current)[BINK_BLOCK_MAX_SIZE / 16]; float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave + float *prev_ptr[MAX_CHANNELS]; ///< pointers to the overlap points in the coeffs array + uint8_t *packet_buffer; union { RDFTContext rdft; DCTContext dct; @@ -71,7 +79,6 @@ static av_cold int decode_init(AVCodecContext *avctx) int i; int frame_len_bits; - s->avctx = avctx; dsputil_init(&s->dsp, avctx); ff_fmt_convert_init(&s->fmt_conv, avctx); @@ -83,28 +90,33 @@ static av_cold int decode_init(AVCodecContext *avctx) } else { frame_len_bits = 11; } - s->frame_len = 1 << frame_len_bits; - if (s->channels > MAX_CHANNELS) { - av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels); + if (avctx->channels > MAX_CHANNELS) { + av_log(avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels); return -1; } + s->version_b = avctx->extradata && avctx->extradata[3] == 'b'; + if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) { // audio is already interleaved for the RDFT format variant sample_rate *= avctx->channels; - s->frame_len *= avctx->channels; s->channels = 1; - if (avctx->channels == 2) - frame_len_bits++; + if (!s->version_b) + frame_len_bits += av_log2(avctx->channels); } else { s->channels = avctx->channels; } + 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; s->root = 2.0 / sqrt(s->frame_len); + 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++) @@ -116,16 +128,18 @@ static av_cold int decode_init(AVCodecContext *avctx) return AVERROR(ENOMEM); /* populate bands data */ - s->bands[0] = 1; + s->bands[0] = 2; for (i = 1; i < s->num_bands; i++) - s->bands[i] = ff_wma_critical_freqs[i - 1] * (s->frame_len / 2) / sample_rate_half; - s->bands[s->num_bands] = s->frame_len / 2; + 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; avctx->sample_fmt = AV_SAMPLE_FMT_S16; - for (i = 0; i < s->channels; i++) + for (i = 0; i < s->channels; i++) { s->coeffs_ptr[i] = s->coeffs + i * s->frame_len; + s->prev_ptr[i] = s->coeffs_ptr[i] + s->frame_len - s->overlap_len; + } if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R); @@ -134,6 +148,9 @@ static av_cold int decode_init(AVCodecContext *avctx) else return -1; + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -150,11 +167,18 @@ static const uint8_t rle_length_tab[16] = { 2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64 }; +#define GET_BITS_SAFE(out, nbits) do { \ + if (get_bits_left(gb) < nbits) \ + return AVERROR_INVALIDDATA; \ + out = get_bits(gb, nbits); \ +} while (0) + /** * 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 void decode_block(BinkAudioContext *s, short *out, int use_dct) +static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct) { int ch, i, j, k; float q, quant[25]; @@ -166,45 +190,61 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct) for (ch = 0; ch < s->channels; ch++) { FFTSample *coeffs = s->coeffs_ptr[ch]; - q = 0.0f; - coeffs[0] = get_float(gb) * s->root; - coeffs[1] = get_float(gb) * s->root; + 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++) { - /* constant is result of 0.066399999/log10(M_E) */ int value = get_bits(gb, 8); - quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root; + quant[i] = quant_table[FFMIN(value, 95)]; } - // find band (k) - for (k = 0; s->bands[k] < 1; k++) { - q = quant[k]; - } + k = 0; + q = quant[0]; // parse coefficients i = 2; while (i < s->frame_len) { - if (get_bits1(gb)) { - j = i + rle_length_tab[get_bits(gb, 4)] * 8; + if (s->version_b) { + j = i + 16; } else { - j = i + 8; + int v; + GET_BITS_SAFE(v, 1); + if (v) { + GET_BITS_SAFE(v, 4); + j = i + rle_length_tab[v] * 8; + } else { + j = i + 8; + } } j = FFMIN(j, s->frame_len); - width = get_bits(gb, 4); + GET_BITS_SAFE(width, 4); if (width == 0) { memset(coeffs + i, 0, (j - i) * sizeof(*coeffs)); i = j; - while (s->bands[k] * 2 < i) + while (s->bands[k] < i) q = quant[k++]; } else { while (i < j) { - if (s->bands[k] * 2 == i) + if (s->bands[k] == i) q = quant[k++]; - coeff = get_bits(gb, width); + GET_BITS_SAFE(coeff, width); if (coeff) { - if (get_bits1(gb)) + int v; + GET_BITS_SAFE(v, 1); + if (v) coeffs[i] = -q * coeff; else coeffs[i] = q * coeff; @@ -218,15 +258,19 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct) if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) { coeffs[0] /= 0.5; - ff_dct_calc (&s->trans.dct, coeffs); + s->trans.dct.dct_calc(&s->trans.dct, coeffs); s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len); } else if (CONFIG_BINKAUDIO_RDFT_DECODER) - ff_rdft_calc(&s->trans.rdft, coeffs); + s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs); } + s->fmt_conv.float_to_int16_interleave(s->current, + (const float **)s->prev_ptr, + s->overlap_len, s->channels); s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, - s->frame_len, s->channels); + s->frame_len - s->overlap_len, + s->channels); if (!s->first) { int count = s->overlap_len * s->channels; @@ -236,20 +280,24 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct) } } - memcpy(s->previous, out + s->block_size, - s->overlap_len * s->channels * sizeof(*out)); + memcpy(s->previous, s->current, + s->overlap_len * s->channels * sizeof(*s->previous)); 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 == CODEC_ID_BINKAUDIO_RDFT) ff_rdft_end(&s->trans.rdft); else if (CONFIG_BINKAUDIO_DCT_DECODER) ff_dct_end(&s->trans.dct); + return 0; } @@ -259,52 +307,77 @@ static void get_bits_align32(GetBitContext *s) if (n) skip_bits(s, n); } -static int decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { BinkAudioContext *s = avctx->priv_data; - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; - short *samples = data; - short *samples_end = (short*)((uint8_t*)data + *data_size); - int reported_size; + int16_t *samples; 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); + } - init_get_bits(gb, buf, buf_size * 8); + /* get output buffer */ + s->frame.nb_samples = s->block_size / avctx->channels; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + samples = (int16_t *)s->frame.data[0]; - reported_size = get_bits_long(gb, 32); - while (get_bits_count(gb) / 8 < buf_size && - samples + s->block_size <= samples_end) { - decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT); - samples += s->block_size; - get_bits_align32(gb); + if (decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT)) { + av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n"); + return AVERROR_INVALIDDATA; } + get_bits_align32(gb); + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; - *data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data); - return buf_size; + return consumed; } AVCodec ff_binkaudio_rdft_decoder = { - "binkaudio_rdft", - AVMEDIA_TYPE_AUDIO, - CODEC_ID_BINKAUDIO_RDFT, - sizeof(BinkAudioContext), - decode_init, - NULL, - decode_end, - decode_frame, + .name = "binkaudio_rdft", + .type = AVMEDIA_TYPE_AUDIO, + .id = 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 = { - "binkaudio_dct", - AVMEDIA_TYPE_AUDIO, - CODEC_ID_BINKAUDIO_DCT, - sizeof(BinkAudioContext), - decode_init, - NULL, - decode_end, - decode_frame, + .name = "binkaudio_dct", + .type = AVMEDIA_TYPE_AUDIO, + .id = 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)") };