summaryrefslogtreecommitdiff
path: root/ffmpeg1/libavcodec/binkaudio.c
diff options
context:
space:
mode:
Diffstat (limited to 'ffmpeg1/libavcodec/binkaudio.c')
-rw-r--r--ffmpeg1/libavcodec/binkaudio.c359
1 files changed, 0 insertions, 359 deletions
diff --git a/ffmpeg1/libavcodec/binkaudio.c b/ffmpeg1/libavcodec/binkaudio.c
deleted file mode 100644
index ef5569a..0000000
--- a/ffmpeg1/libavcodec/binkaudio.c
+++ /dev/null
@@ -1,359 +0,0 @@
-/*
- * 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)")
-};
generated by cgit v1.2.3 (git 2.39.1) at 2025-12-06 21:16:44 +0000