[deb_ffmpeg.git] / ffmpeg / libavcodec / binkaudio.c

/*
 * 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 "wma_freqs.h"
#include "libavutil/intfloat.h"

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);
        memset(buf + avpkt->size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
        s->packet_buffer = buf;
        memcpy(s->packet_buffer, avpkt->data, avpkt->size);
        if ((ret = init_get_bits8(gb, s->packet_buffer, avpkt->size)) < 0)
            return ret;
        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",
    .long_name      = NULL_IF_CONFIG_SMALL("Bink Audio (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,
};

AVCodec ff_binkaudio_dct_decoder = {
    .name           = "binkaudio_dct",
    .long_name      = NULL_IF_CONFIG_SMALL("Bink Audio (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,
};
Commit	Line	Data
2ba45a60 DM	1	/*
	2	* Bink Audio decoder
	3	* Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
	4	* Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
	5	*
	6	* This file is part of FFmpeg.
	7	*
	8	* FFmpeg is free software; you can redistribute it and/or
	9	* modify it under the terms of the GNU Lesser General Public
	10	* License as published by the Free Software Foundation; either
	11	* version 2.1 of the License, or (at your option) any later version.
	12	*
	13	* FFmpeg is distributed in the hope that it will be useful,
	14	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	15	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	16	* Lesser General Public License for more details.
	17	*
	18	* You should have received a copy of the GNU Lesser General Public
	19	* License along with FFmpeg; if not, write to the Free Software
	20	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	21	*/
	22
	23	/**
	24	* @file
	25	* Bink Audio decoder
	26	*
	27	* Technical details here:
	28	* http://wiki.multimedia.cx/index.php?title=Bink_Audio
	29	*/
	30
	31	#include "libavutil/channel_layout.h"
	32	#include "avcodec.h"
	33	#define BITSTREAM_READER_LE
	34	#include "get_bits.h"
	35	#include "dct.h"
	36	#include "rdft.h"
	37	#include "fmtconvert.h"
	38	#include "internal.h"
	39	#include "wma_freqs.h"
	40	#include "libavutil/intfloat.h"
	41
	42	static float quant_table[96];
	43
	44	#define MAX_CHANNELS 2
	45	#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
	46
	47	typedef struct {
	48	GetBitContext gb;
	49	int version_b; ///< Bink version 'b'
	50	int first;
	51	int channels;
	52	int frame_len; ///< transform size (samples)
	53	int overlap_len; ///< overlap size (samples)
	54	int block_size;
	55	int num_bands;
	56	unsigned int *bands;
	57	float root;
	58	DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
	59	float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
	60	uint8_t *packet_buffer;
	61	union {
	62	RDFTContext rdft;
	63	DCTContext dct;
	64	} trans;
65	} BinkAudioContext;
66
67
68	static av_cold int decode_init(AVCodecContext *avctx)
69	{
70	BinkAudioContext *s = avctx->priv_data;
71	int sample_rate = avctx->sample_rate;
72	int sample_rate_half;
73	int i;
74	int frame_len_bits;
75
76	/* determine frame length */
77	if (avctx->sample_rate < 22050) {
78	frame_len_bits = 9;
79	} else if (avctx->sample_rate < 44100) {
80	frame_len_bits = 10;
81	} else {
82	frame_len_bits = 11;
83	}
84
85	if (avctx->channels < 1 \|\| avctx->channels > MAX_CHANNELS) {
86	av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels);
87	return AVERROR_INVALIDDATA;
88	}
89	avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
90	AV_CH_LAYOUT_STEREO;
91
92	s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';
93
94	if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
95	// audio is already interleaved for the RDFT format variant
96	avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
97	sample_rate *= avctx->channels;
98	s->channels = 1;
99	if (!s->version_b)
100	frame_len_bits += av_log2(avctx->channels);
101	} else {
102	s->channels = avctx->channels;
103	avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
104	}
105
106	s->frame_len = 1 << frame_len_bits;
107	s->overlap_len = s->frame_len / 16;
108	s->block_size = (s->frame_len - s->overlap_len) * s->channels;
109	sample_rate_half = (sample_rate + 1) / 2;
110	if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
111	s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
112	else
113	s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
114	for (i = 0; i < 96; i++) {
115	/* constant is result of 0.066399999/log10(M_E) */
116	quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
117	}
118
119	/* calculate number of bands */
120	for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
121	if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
122	break;
123
124	s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
125	if (!s->bands)
126	return AVERROR(ENOMEM);
127
128	/* populate bands data */
129	s->bands[0] = 2;
130	for (i = 1; i < s->num_bands; i++)
131	s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
132	s->bands[s->num_bands] = s->frame_len;
133
134	s->first = 1;
135
136	if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
137	ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
138	else if (CONFIG_BINKAUDIO_DCT_DECODER)
139	ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
140	else
141	return -1;
142
143	return 0;
144	}
145
146	static float get_float(GetBitContext *gb)
147	{
148	int power = get_bits(gb, 5);
149	float f = ldexpf(get_bits_long(gb, 23), power - 23);
150	if (get_bits1(gb))
151	f = -f;
152	return f;
153	}
154
155	static const uint8_t rle_length_tab[16] = {
156	2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
157	};
158
159	/**
160	* Decode Bink Audio block
161	* @param[out] out Output buffer (must contain s->block_size elements)
162	* @return 0 on success, negative error code on failure
163	*/
164	static int decode_block(BinkAudioContext s, float *out, int use_dct)
165	{
166	int ch, i, j, k;
167	float q, quant[25];
168	int width, coeff;
169	GetBitContext *gb = &s->gb;
170
171	if (use_dct)
172	skip_bits(gb, 2);
173
174	for (ch = 0; ch < s->channels; ch++) {
175	FFTSample *coeffs = out[ch];
176
177	if (s->version_b) {
178	if (get_bits_left(gb) < 64)
179	return AVERROR_INVALIDDATA;
180	coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
181	coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
182	} else {
183	if (get_bits_left(gb) < 58)
184	return AVERROR_INVALIDDATA;
185	coeffs[0] = get_float(gb) * s->root;
186	coeffs[1] = get_float(gb) * s->root;
187	}
188
189	if (get_bits_left(gb) < s->num_bands * 8)
190	return AVERROR_INVALIDDATA;
191	for (i = 0; i < s->num_bands; i++) {
192	int value = get_bits(gb, 8);
193	quant[i] = quant_table[FFMIN(value, 95)];
194	}
195
196	k = 0;
197	q = quant[0];
198
199	// parse coefficients
200	i = 2;
201	while (i < s->frame_len) {
202	if (s->version_b) {
203	j = i + 16;
204	} else {
205	int v = get_bits1(gb);
206	if (v) {
207	v = get_bits(gb, 4);
208	j = i + rle_length_tab[v] * 8;
209	} else {
210	j = i + 8;
211	}
212	}
213
214	j = FFMIN(j, s->frame_len);
215
216	width = get_bits(gb, 4);
217	if (width == 0) {
218	memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
219	i = j;
220	while (s->bands[k] < i)
221	q = quant[k++];
222	} else {
223	while (i < j) {
224	if (s->bands[k] == i)
225	q = quant[k++];
226	coeff = get_bits(gb, width);
227	if (coeff) {
228	int v;
229	v = get_bits1(gb);
230	if (v)
231	coeffs[i] = -q * coeff;
232	else
233	coeffs[i] = q * coeff;
234	} else {
235	coeffs[i] = 0.0f;
236	}
237	i++;
238	}
239	}
240	}
241
242	if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
243	coeffs[0] /= 0.5;
244	s->trans.dct.dct_calc(&s->trans.dct, coeffs);
245	}
246	else if (CONFIG_BINKAUDIO_RDFT_DECODER)
247	s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
248	}
249
250	for (ch = 0; ch < s->channels; ch++) {
251	int j;
252	int count = s->overlap_len * s->channels;
253	if (!s->first) {
254	j = ch;
255	for (i = 0; i < s->overlap_len; i++, j += s->channels)
256	out[ch][i] = (s->previous[ch][i] * (count - j) +
257	out[ch][i] * j) / count;
258	}
259	memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
260	s->overlap_len * sizeof(*s->previous[ch]));
261	}
262
263	s->first = 0;
264
265	return 0;
266	}
267
268	static av_cold int decode_end(AVCodecContext *avctx)
269	{
270	BinkAudioContext * s = avctx->priv_data;
271	av_freep(&s->bands);
272	av_freep(&s->packet_buffer);
273	if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
274	ff_rdft_end(&s->trans.rdft);
275	else if (CONFIG_BINKAUDIO_DCT_DECODER)
276	ff_dct_end(&s->trans.dct);
277
278	return 0;
279	}
280
281	static void get_bits_align32(GetBitContext *s)
282	{
283	int n = (-get_bits_count(s)) & 31;
284	if (n) skip_bits(s, n);
285	}
286
287	static int decode_frame(AVCodecContext avctx, void data,
288	int got_frame_ptr, AVPacket avpkt)
289	{
290	BinkAudioContext *s = avctx->priv_data;
291	AVFrame *frame = data;
292	GetBitContext *gb = &s->gb;
293	int ret, consumed = 0;
294
295	if (!get_bits_left(gb)) {
296	uint8_t *buf;
297	/* handle end-of-stream */
298	if (!avpkt->size) {
299	*got_frame_ptr = 0;
300	return 0;
301	}
302	if (avpkt->size < 4) {
303	av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
304	return AVERROR_INVALIDDATA;
305	}
306	buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE);
307	if (!buf)
308	return AVERROR(ENOMEM);
309	memset(buf + avpkt->size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
310	s->packet_buffer = buf;
311	memcpy(s->packet_buffer, avpkt->data, avpkt->size);
312	if ((ret = init_get_bits8(gb, s->packet_buffer, avpkt->size)) < 0)
313	return ret;
314	consumed = avpkt->size;
315
316	/* skip reported size */
317	skip_bits_long(gb, 32);
318	}
319
320	/* get output buffer */
321	frame->nb_samples = s->frame_len;
322	if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
323	return ret;
324
325	if (decode_block(s, (float **)frame->extended_data,
326	avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
327	av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
328	return AVERROR_INVALIDDATA;
329	}
330	get_bits_align32(gb);
331
332	frame->nb_samples = s->block_size / avctx->channels;
333	*got_frame_ptr = 1;
334
335	return consumed;
336	}
337
338	AVCodec ff_binkaudio_rdft_decoder = {
339	.name = "binkaudio_rdft",
340	.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),
341	.type = AVMEDIA_TYPE_AUDIO,
342	.id = AV_CODEC_ID_BINKAUDIO_RDFT,
343	.priv_data_size = sizeof(BinkAudioContext),
344	.init = decode_init,
345	.close = decode_end,
346	.decode = decode_frame,
347	.capabilities = CODEC_CAP_DELAY \| CODEC_CAP_DR1,
348	};
349
350	AVCodec ff_binkaudio_dct_decoder = {
351	.name = "binkaudio_dct",
352	.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),
353	.type = AVMEDIA_TYPE_AUDIO,
354	.id = AV_CODEC_ID_BINKAUDIO_DCT,
355	.priv_data_size = sizeof(BinkAudioContext),
356	.init = decode_init,
357	.close = decode_end,
358	.decode = decode_frame,
359	.capabilities = CODEC_CAP_DELAY \| CODEC_CAP_DR1,
360	};