87 if((t << 1) > size) mask = ~mask;
93 0, 640, 1184, 1748, 2298, 2426, 2554, 3066, 3578, 4106, 4618, 5196, 5708
101 static int vlc_initialized = 0;
109 static VLC_TYPE dscf0_table[560][2];
110 static VLC_TYPE dscf1_table[598][2];
113 static VLC_TYPE codes_table[5708][2];
145 if(vlc_initialized)
return 0;
148 band_vlc.
table = band_table;
154 q1_vlc.
table = q1_table;
159 q9up_vlc.
table = q9up_table;
165 scfi_vlc[0].
table = scfi0_table;
170 scfi_vlc[1].
table = scfi1_table;
176 dscf_vlc[0].
table = dscf0_table;
181 dscf_vlc[1].
table = dscf1_table;
187 q3_vlc[0].
table = q3_0_table;
193 q3_vlc[1].
table = q3_1_table;
200 for(i = 0; i < 2; i++){
207 q2_vlc[i].
table = &codes_table[vlc_offsets[2+i]];
213 quant_vlc[0][i].
table = &codes_table[vlc_offsets[4+i]];
218 quant_vlc[1][i].
table = &codes_table[vlc_offsets[6+i]];
223 quant_vlc[2][i].
table = &codes_table[vlc_offsets[8+i]];
228 quant_vlc[3][i].
table = &codes_table[vlc_offsets[10+i]];
229 quant_vlc[3][i].
table_allocated = vlc_offsets[11+i] - vlc_offsets[10+i];
240 int *got_frame_ptr,
AVPacket *avpkt)
244 int buf_size = avpkt->
size;
247 int i, j, k,
ch, cnt, res, t;
250 int maxband, keyframe;
256 memset(c->
Q, 0,
sizeof(c->
Q));
268 if(maxband > 32) maxband -= 33;
284 last[0] = last[1] = 0;
285 for(i = maxband - 1; i >= 0; i--){
286 for(ch = 0; ch < 2; ch++){
288 if(last[ch] > 15) last[
ch] -= 17;
296 for(i = 0; i < maxband; i++)
297 if(bands[i].res[0] || bands[i].res[1])
301 for(i = maxband - 1; i >= 0; i--)
302 if(bands[i].res[0] || bands[i].res[1]){
303 bands[i].
msf = mask & 1;
308 for(i = maxband; i < c->
maxbands; i++)
309 bands[i].res[0] = bands[i].res[1] = 0;
312 for(i = 0; i < 32; i++)
316 for(i = 0; i < maxband; i++){
317 if(bands[i].res[0] || bands[i].res[1]){
318 cnt = !!bands[i].
res[0] + !!bands[i].
res[1] - 1;
320 t =
get_vlc2(gb, scfi_vlc[cnt].
table, scfi_vlc[cnt].bits, 1);
321 if(bands[i].res[0]) bands[i].
scfi[0] = t >> (2 * cnt);
322 if(bands[i].res[1]) bands[i].
scfi[1] = t & 3;
327 for(i = 0; i < maxband; i++){
328 for(ch = 0; ch < 2; ch++){
329 if(!bands[i].res[ch])
continue;
340 for(j = 0; j < 2; j++){
341 if((bands[i].scfi[ch] << j) & 2)
342 bands[i].scf_idx[ch][j + 1] = bands[i].scf_idx[ch][j];
354 for(ch = 0; ch < 2; ch++){
355 res = bands[i].
res[
ch];
367 for(k = 0; k < SAMPLES_PER_BAND / 2; k++, t <<= 1)
368 c->
Q[ch][off + j + k] = (t & 0x20000) ? (
get_bits1(gb) << 1) - 1 : 0;
385 c->
Q[
ch][off + j + 1] = t >> 4;
386 c->
Q[
ch][off + j + 0] = (t & 8) ? (t & 0xF) - 16 : (t & 0xF);
396 c->
Q[
ch][off + j] = t;
397 cnt = (cnt >> 1) +
FFABS(c->
Q[ch][off + j]);
404 c->
Q[
ch][off + j] <<= res - 9;
407 c->
Q[
ch][off + j] -= (1 << (res - 2)) - 1;
static const int quant_offsets[6]
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
static const int8_t mpc8_q5_bits[2][MPC8_Q5_SIZE]
ptrdiff_t const GLvoid * data
static void flush(AVCodecContext *avctx)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static const int8_t mpc8_bands_bits[MPC8_BANDS_SIZE]
static av_cold int init(AVCodecContext *avctx)
int ff_init_vlc_sparse(VLC *vlc_arg, int nb_bits, int nb_codes, const void *bits, int bits_wrap, int bits_size, const void *codes, int codes_wrap, int codes_size, const void *symbols, int symbols_wrap, int symbols_size, int flags)
#define AV_CH_LAYOUT_STEREO
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static const uint8_t mpc8_q8_codes[2][MPC8_Q8_SIZE]
static const int8_t mpc8_q7_bits[2][MPC8_Q7_SIZE]
static const int8_t mpc8_q8_bits[2][MPC8_Q8_SIZE]
static const uint32_t mpc8_cnk_lost[16][33]
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
static const int8_t mpc8_dscf1_bits[MPC8_DSCF1_SIZE]
enum AVSampleFormat sample_fmt
audio sample format
static const int8_t mpc8_q4_syms[MPC8_Q4_SIZE]
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static const uint8_t mpc8_dscf1_codes[MPC8_DSCF1_SIZE]
static int get_bits_count(const GetBitContext *s)
static const uint8_t mpc8_q1_codes[MPC8_Q1_SIZE]
bitstream reader API header.
static const int8_t mpc8_dscf0_bits[MPC8_DSCF0_SIZE]
static const int8_t mpc8_scfi1_bits[MPC8_SCFI1_SIZE]
static const int8_t mpc8_huffq2[5 *5 *5]
static int get_bits_left(GetBitContext *gb)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static const int8_t mpc8_q1_bits[MPC8_Q1_SIZE]
static const uint8_t mpc8_dscf0_codes[MPC8_DSCF0_SIZE]
static const int8_t mpc8_idx50[125]
static const uint16_t mask[17]
#define init_vlc(vlc, nb_bits, nb_codes,bits, bits_wrap, bits_size,codes, codes_wrap, codes_size,flags)
static av_cold int mpc8_decode_init(AVCodecContext *avctx)
static const struct endianess table[]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static VLC quant_vlc[4][2]
static const int8_t mpc8_scfi0_bits[MPC8_SCFI0_SIZE]
static const int8_t mpc8_q3_bits[MPC8_Q3_SIZE]
const char * name
Name of the codec implementation.
uint64_t channel_layout
Audio channel layout.
static const uint8_t mpc8_bands_codes[MPC8_BANDS_SIZE]
static av_cold void mpc8_decode_flush(AVCodecContext *avctx)
static const int8_t mpc8_idx51[125]
audio channel layout utility functions
static int mpc8_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static int mpc8_get_mod_golomb(GetBitContext *gb, int m)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
static const uint8_t mpc8_q5_codes[2][MPC8_Q5_SIZE]
static const uint8_t mpc8_cnk_len[16][33]
static const uint8_t mpc8_q3_codes[MPC8_Q3_SIZE]
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Libavcodec external API header.
Musepack decoder MPEG Audio Layer 1/2 -like codec with frames of 1152 samples divided into 32 subband...
AVSampleFormat
Audio sample formats.
av_cold void ff_mpc_init(void)
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
main external API structure.
static const float bands[]
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
static unsigned int get_bits1(GetBitContext *s)
static void skip_bits(GetBitContext *s, int n)
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
static const uint32_t mpc8_cnk[16][32]
static int mpc8_dec_base(GetBitContext *gb, int k, int n)
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
static const uint16_t vlc_offsets[13]
static const unsigned int mpc8_thres[]
static const int8_t mpc8_q3_syms[MPC8_Q3_SIZE]
static const uint8_t mpc8_scfi1_codes[MPC8_SCFI1_SIZE]
static const int q3_offsets[2]
static const int8_t mpc8_q9up_bits[MPC8_Q9UP_SIZE]
common internal api header.
static const int8_t mpc8_idx52[125]
static const uint8_t mpc8_q9up_codes[MPC8_Q9UP_SIZE]
#define INIT_VLC_USE_NEW_STATIC
static const uint8_t mpc8_q2_codes[2][MPC8_Q2_SIZE]
static const int8_t mpc8_q2_bits[2][MPC8_Q2_SIZE]
static const uint8_t mpc8_res_codes[2][MPC8_RES_SIZE]
int channels
number of audio channels
VLC_TYPE(* table)[2]
code, bits
static int mpc8_get_mask(GetBitContext *gb, int size, int t)
static const int8_t mpc8_res_bits[2][MPC8_RES_SIZE]
static const uint8_t mpc8_q6_codes[2][MPC8_Q6_SIZE]
static enum AVSampleFormat sample_fmts[]
static const int8_t mpc8_q6_bits[2][MPC8_Q6_SIZE]
static const uint8_t mpc8_q4_codes[MPC8_Q4_SIZE]
uint8_t ** extended_data
pointers to the data planes/channels.
#define AV_CH_LAYOUT_MONO
Subband structure - hold all variables for each subband.
This structure stores compressed data.
av_cold void ff_mpadsp_init(MPADSPContext *s)
static int mpc8_dec_enum(GetBitContext *gb, int k, int n)
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
static const uint8_t mpc8_q7_codes[2][MPC8_Q7_SIZE]
static const uint8_t mpc8_scfi0_codes[MPC8_SCFI0_SIZE]
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
static const int8_t mpc8_q4_bits[MPC8_Q4_SIZE]