33 #define BITSTREAM_READER_LE 44 #define MAX_CHANNELS 2 45 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11) 109 sample_rate_half = (sample_rate + 1) / 2;
114 for (i = 0; i < 96; i++) {
156 2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
174 for (ch = 0; ch < s->
channels; ch++) {
201 while (i < s->frame_len) {
218 memset(coeffs + i, 0, (j - i) *
sizeof(*coeffs));
220 while (s->
bands[k] < i)
224 if (s->
bands[k] == i)
231 coeffs[i] = -q * coeff;
233 coeffs[i] = q * coeff;
250 for (ch = 0; ch < s->
channels; ch++) {
256 out[ch][i] = (s->
previous[ch][i] * (count - j) +
257 out[ch][i] * j) / count;
288 int *got_frame_ptr,
AVPacket *avpkt)
293 int ret, consumed = 0;
302 if (avpkt->
size < 4) {
312 consumed = avpkt->
size;
339 .
name =
"binkaudio_rdft",
351 .
name =
"binkaudio_dct",
av_cold void ff_rdft_end(RDFTContext *s)
const struct AVCodec * codec
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
static float get_float(GetBitContext *gb)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static av_cold int decode_end(AVCodecContext *avctx)
static void skip_bits_long(GetBitContext *s, int n)
static av_always_inline float av_int2float(uint32_t i)
Reinterpret a 32-bit integer as a float.
static const uint8_t rle_length_tab[16]
#define DECLARE_ALIGNED(n, t, v)
const uint16_t ff_wma_critical_freqs[25]
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_dlog(ac->avr, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out
#define AV_CH_LAYOUT_STEREO
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
union BinkAudioContext::@11 trans
enum AVSampleFormat sample_fmt
audio sample format
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
static int get_bits_count(const GetBitContext *s)
bitstream reader API header.
float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE/16]
coeffs from previous audio block
static int get_bits_left(GetBitContext *gb)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define BINK_BLOCK_MAX_SIZE
#define CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static void get_bits_align32(GetBitContext *s)
static int decode_block(BinkAudioContext *s, float **out, int use_dct)
Decode Bink Audio block.
void av_log(void *avcl, int level, const char *fmt,...)
const char * name
Name of the codec implementation.
uint64_t channel_layout
Audio channel layout.
#define FF_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
audio channel layout utility functions
void(* dct_calc)(struct DCTContext *s, FFTSample *data)
static float quant_table[96]
static av_cold int decode_init(AVCodecContext *avctx)
void(* rdft_calc)(struct RDFTContext *s, FFTSample *z)
AVCodec ff_binkaudio_rdft_decoder
int overlap_len
overlap size (samples)
Libavcodec external API header.
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
int sample_rate
samples per second
AVCodec ff_binkaudio_dct_decoder
main external API structure.
static void close(AVCodecParserContext *s)
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static unsigned int get_bits1(GetBitContext *s)
static void skip_bits(GetBitContext *s, int n)
av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
Set up DCT.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
int frame_len
transform size (samples)
int version_b
Bink version 'b'.
common internal api header.
void * av_realloc(void *ptr, size_t size)
Allocate or reallocate a block of memory.
static av_cold int init(AVCodecParserContext *s)
int channels
number of audio channels
av_cold void ff_dct_end(DCTContext *s)
uint8_t ** extended_data
pointers to the data planes/channels.
#define AV_CH_LAYOUT_MONO
av_cold int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans)
Set up a real FFT.
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame