[FFmpeg-cvslog] r21570 - in trunk: Changelog configure doc/general.texi libavcodec/Makefile libavcodec/allcodecs.c libavcodec/avcodec.h libavcodec/binkaudio.c
pross
subversion
Sun Jan 31 13:51:15 CET 2010
Author: pross
Date: Sun Jan 31 13:51:15 2010
New Revision: 21570
Log:
Bink Audio decoder
Added:
trunk/libavcodec/binkaudio.c
Modified:
trunk/Changelog
trunk/configure
trunk/doc/general.texi
trunk/libavcodec/Makefile
trunk/libavcodec/allcodecs.c
trunk/libavcodec/avcodec.h
Modified: trunk/Changelog
==============================================================================
--- trunk/Changelog Sun Jan 31 13:35:09 2010 (r21569)
+++ trunk/Changelog Sun Jan 31 13:51:15 2010 (r21570)
@@ -51,7 +51,7 @@ version <next>:
- SIPR decoder
- Adobe Filmstrip muxer and demuxer
- RTP depacketization of H.263
-- Bink demuxer
+- Bink demuxer and Bink Audio decoder
Modified: trunk/configure
==============================================================================
--- trunk/configure Sun Jan 31 13:35:09 2010 (r21569)
+++ trunk/configure Sun Jan 31 13:51:15 2010 (r21570)
@@ -1165,6 +1165,8 @@ aac_encoder_select="fft mdct"
ac3_decoder_select="fft mdct ac3_parser"
alac_encoder_select="lpc"
atrac3_decoder_select="fft mdct"
+binkaudio_dct_decoder_select="fft mdct rdft dct"
+binkaudio_rdft_decoder_select="fft mdct rdft"
cavs_decoder_select="golomb"
cook_decoder_select="fft mdct"
cscd_decoder_suggest="zlib"
Modified: trunk/doc/general.texi
==============================================================================
--- trunk/doc/general.texi Sun Jan 31 13:35:09 2010 (r21569)
+++ trunk/doc/general.texi Sun Jan 31 13:51:15 2010 (r21570)
@@ -550,6 +550,8 @@ following image formats are supported:
@tab QuickTime fourcc 'alac'
@item Atrac 1 @tab @tab X
@item Atrac 3 @tab @tab X
+ at item Bink Audio @tab @tab X
+ @tab Used in Bink and Smacker files in many games.
@item Delphine Software International CIN audio @tab @tab X
@tab Codec used in Delphine Software International games.
@item COOK @tab @tab X
Modified: trunk/libavcodec/Makefile
==============================================================================
--- trunk/libavcodec/Makefile Sun Jan 31 13:35:09 2010 (r21569)
+++ trunk/libavcodec/Makefile Sun Jan 31 13:51:15 2010 (r21570)
@@ -65,6 +65,8 @@ OBJS-$(CONFIG_AURA2_DECODER) +
OBJS-$(CONFIG_AVS_DECODER) += avs.o
OBJS-$(CONFIG_BETHSOFTVID_DECODER) += bethsoftvideo.o
OBJS-$(CONFIG_BFI_DECODER) += bfi.o
+OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER) += binkaudio.o wma.o
+OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER) += binkaudio.o wma.o
OBJS-$(CONFIG_BMP_DECODER) += bmp.o msrledec.o
OBJS-$(CONFIG_BMP_ENCODER) += bmpenc.o
OBJS-$(CONFIG_C93_DECODER) += c93.o
Modified: trunk/libavcodec/allcodecs.c
==============================================================================
--- trunk/libavcodec/allcodecs.c Sun Jan 31 13:35:09 2010 (r21569)
+++ trunk/libavcodec/allcodecs.c Sun Jan 31 13:51:15 2010 (r21570)
@@ -213,6 +213,8 @@ void avcodec_register_all(void)
REGISTER_DECODER (APE, ape);
REGISTER_DECODER (ATRAC1, atrac1);
REGISTER_DECODER (ATRAC3, atrac3);
+ REGISTER_DECODER (BINKAUDIO_DCT, binkaudio_dct);
+ REGISTER_DECODER (BINKAUDIO_RDFT, binkaudio_rdft);
REGISTER_DECODER (COOK, cook);
REGISTER_DECODER (DCA, dca);
REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
Modified: trunk/libavcodec/avcodec.h
==============================================================================
--- trunk/libavcodec/avcodec.h Sun Jan 31 13:35:09 2010 (r21569)
+++ trunk/libavcodec/avcodec.h Sun Jan 31 13:51:15 2010 (r21570)
@@ -30,7 +30,7 @@
#include "libavutil/avutil.h"
#define LIBAVCODEC_VERSION_MAJOR 52
-#define LIBAVCODEC_VERSION_MINOR 49
+#define LIBAVCODEC_VERSION_MINOR 50
#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
Added: trunk/libavcodec/binkaudio.c
==============================================================================
--- /dev/null 00:00:00 1970 (empty, because file is newly added)
+++ trunk/libavcodec/binkaudio.c Sun Jan 31 13:51:15 2010 (r21570)
@@ -0,0 +1,303 @@
+/*
+ * Bink Audio decoder
+ * Copyright (c) 2007-2010 Peter Ross (pross at xvid.org)
+ * Copyright (c) 2009 Daniel Verkamp (daniel at 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 libavcodec/binkaudio.c
+ * Bink Audio decoder
+ *
+ * Technical details here:
+ * http://wiki.multimedia.cx/index.php?title=Bink_Audio
+ */
+
+#include "avcodec.h"
+#define ALT_BITSTREAM_READER_LE
+#include "get_bits.h"
+#include "dsputil.h"
+extern const uint16_t ff_wma_critical_freqs[25];
+
+#define MAX_CHANNELS 2
+#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
+
+typedef struct {
+ AVCodecContext *avctx;
+ GetBitContext gb;
+ DSPContext dsp;
+ 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_16(FFTSample, coeffs[BINK_BLOCK_MAX_SIZE]);
+ DECLARE_ALIGNED_16(short, previous[BINK_BLOCK_MAX_SIZE / 16]); ///< coeffs from previous audio block
+ float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
+ 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;
+
+ s->avctx = avctx;
+ dsputil_init(&s->dsp, avctx);
+
+ /* 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;
+ }
+ 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);
+ return -1;
+ }
+
+ 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++;
+ } else {
+ s->channels = avctx->channels;
+ }
+
+ 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);
+
+ /* 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] = 1;
+ 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->first = 1;
+ avctx->sample_fmt = SAMPLE_FMT_S16;
+
+ for (i = 0; i < s->channels; i++)
+ s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
+
+ if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
+ ff_rdft_init(&s->trans.rdft, frame_len_bits, IRIDFT);
+ else
+ ff_dct_init(&s->trans.dct, frame_len_bits, 0);
+
+ 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)
+ */
+static void decode_block(BinkAudioContext *s, short *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 = s->coeffs_ptr[ch];
+ q = 0.0f;
+ coeffs[0] = get_float(gb) * s->root;
+ coeffs[1] = get_float(gb) * s->root;
+
+ 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;
+ }
+
+ // find band (k)
+ for (k = 0; s->bands[k] < 1; k++) {
+ q = quant[k];
+ }
+
+ // parse coefficients
+ i = 2;
+ while (i < s->frame_len) {
+ if (get_bits1(gb)) {
+ j = i + rle_length_tab[get_bits(gb, 4)] * 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] * 2 < i)
+ q = quant[k++];
+ } else {
+ while (i < j) {
+ if (s->bands[k] * 2 == i)
+ q = quant[k++];
+ coeff = get_bits(gb, width);
+ if (coeff) {
+ if (get_bits1(gb))
+ coeffs[i] = -q * coeff;
+ else
+ coeffs[i] = q * coeff;
+ } else {
+ coeffs[i] = 0.0f;
+ }
+ i++;
+ }
+ }
+ }
+
+ if (use_dct)
+ ff_dct_calc (&s->trans.dct, coeffs);
+ else
+ ff_rdft_calc(&s->trans.rdft, coeffs);
+ }
+
+ s->dsp.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, s->frame_len, s->channels);
+
+ if (!s->first) {
+ int count = s->overlap_len * s->channels;
+ int shift = av_log2(count);
+ for (i = 0; i < count; i++) {
+ out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;
+ }
+ }
+
+ memcpy(s->previous, out + s->block_size,
+ s->overlap_len * s->channels * sizeof(*out));
+
+ s->first = 0;
+}
+
+static av_cold int decode_end(AVCodecContext *avctx)
+{
+ BinkAudioContext * s = avctx->priv_data;
+ av_freep(&s->bands);
+ if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
+ ff_rdft_end(&s->trans.rdft);
+ else
+ 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 *data_size,
+ 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;
+ GetBitContext *gb = &s->gb;
+
+ init_get_bits(gb, buf, buf_size * 8);
+
+ 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);
+ }
+
+ *data_size = (uint8_t*)samples - (uint8_t*)data;
+ if (reported_size != *data_size) {
+ av_log(avctx, AV_LOG_WARNING, "reported data size (%d) does not match output data size (%d)\n",
+ reported_size, *data_size);
+ }
+ return buf_size;
+}
+
+AVCodec binkaudio_rdft_decoder = {
+ "binkaudio_rdft",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_BINKAUDIO_RDFT,
+ sizeof(BinkAudioContext),
+ decode_init,
+ NULL,
+ decode_end,
+ decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
+};
+
+AVCodec binkaudio_dct_decoder = {
+ "binkaudio_dct",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_BINKAUDIO_DCT,
+ sizeof(BinkAudioContext),
+ decode_init,
+ NULL,
+ decode_end,
+ decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
+};
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