43 0, 512, 1024, 1536, 2052, 2564, 3076, 3588, 4100, 4612, 5124,
44 5636, 6164, 6676, 7224
54 static int vlc_initialized = 0;
59 static VLC_TYPE quant_tables[7224][2];
96 if(vlc_initialized)
return 0;
98 scfi_vlc.
table = scfi_table;
106 dscf_vlc.
table = dscf_table;
114 hdr_vlc.
table = hdr_table;
123 for(j = 0; j < 2; j++){
125 quant_vlc[i][j].
table_allocated = quant_offsets[i*2 + j + 1] - quant_offsets[i*2 + j];
153 for(i = 0; i < SAMPLES_PER_BAND/3; i++){
162 for(i = 0; i < SAMPLES_PER_BAND/2; i++){
168 case 3:
case 4:
case 5:
case 6:
case 7:
173 case 8:
case 9:
case 10:
case 11:
case 12:
174 case 13:
case 14:
case 15:
case 16:
case 17:
175 t = (1 << (idx - 2)) - 1;
193 int *got_frame_ptr,
AVPacket *avpkt)
203 int off, ret, last_frame, skip;
204 int bits_used, bits_avail;
206 memset(bands, 0,
sizeof(*bands) * (c->
maxbands + 1));
208 buf_size = avpkt->
size & ~3;
214 if (buf_size != avpkt->
size) {
216 "extra bytes at the end will be skipped.\n");
239 for(ch = 0; ch < 2; ch++){
243 else bands[i].
res[
ch] = bands[i-1].
res[
ch] + t;
244 if (bands[i].res[ch] < -1 || bands[i].res[ch] > 17) {
250 if(bands[i].res[0] || bands[i].res[1]){
256 for(i = 0; i <=
mb; i++)
257 for(ch = 0; ch < 2; ch++)
260 for(i = 0; i <=
mb; i++){
261 for(ch = 0; ch < 2; ch++){
262 if(bands[i].res[ch]){
265 switch(bands[i].scfi[ch]){
287 memset(c->
Q, 0,
sizeof(c->
Q));
290 for(ch = 0; ch < 2; ch++)
298 bits_avail = buf_size * 8;
299 if (!last_frame && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))) {
300 av_log(avctx,
AV_LOG_ERROR,
"Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
void(* bswap_buf)(uint32_t *dst, const uint32_t *src, int w)
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 mpc7_idx32[]
ptrdiff_t const GLvoid * data
static void flush(AVCodecContext *avctx)
static const uint16_t mpc7_quant_vlc[MPC7_QUANT_VLC_TABLES][2][64 *2]
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
static void skip_bits_long(GetBitContext *s, int n)
static av_cold int init(AVCodecContext *avctx)
void av_fast_padded_malloc(void *ptr, unsigned int *size, size_t min_size)
Same behaviour av_fast_malloc but the buffer has additional AV_INPUT_BUFFER_PADDING_SIZE at the end w...
#define AV_CH_LAYOUT_STEREO
static const uint8_t mpc7_quant_vlc_sizes[MPC7_QUANT_VLC_TABLES *2]
static int mpc7_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
enum AVSampleFormat sample_fmt
audio sample format
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static int get_bits_count(const GetBitContext *s)
bitstream reader API header.
static const int8_t mpc7_idx50[]
static const int8_t mpc7_idx31[]
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define init_vlc(vlc, nb_bits, nb_codes,bits, bits_wrap, bits_size,codes, codes_wrap, codes_size,flags)
static const uint8_t mpc7_hdr[MPC7_HDR_SIZE *2]
#define MPC7_QUANT_VLC_TABLES
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 const uint8_t mpc7_quant_vlc_off[MPC7_QUANT_VLC_TABLES]
const char * name
Name of the codec implementation.
static const uint8_t mpc7_scfi[MPC7_SCFI_SIZE *2]
uint64_t channel_layout
Audio channel layout.
static av_cold int mpc7_decode_init(AVCodecContext *avctx)
common internal API header
audio channel layout utility functions
static void mpc7_decode_flush(AVCodecContext *avctx)
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
#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.
static const int8_t mpc7_idx51[]
av_cold void ff_mpc_init(void)
main external API structure.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
Fill samples for given subband.
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 int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
static av_cold int mpc7_decode_close(AVCodecContext *avctx)
common internal api header.
static int ref[MAX_W *MAX_W]
#define INIT_VLC_USE_NEW_STATIC
static const uint16_t quant_offsets[MPC7_QUANT_VLC_TABLES *2+1]
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_YASM &&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
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
static const int8_t mpc7_idx30[]
int channels
number of audio channels
VLC_TYPE(* table)[2]
code, bits
static enum AVSampleFormat sample_fmts[]
#define LOCAL_ALIGNED_16(t, v,...)
static int decode(AVCodecContext *avctx, AVFrame *frame, int *got_frame, AVPacket *pkt)
uint8_t ** extended_data
pointers to the data planes/channels.
static const uint8_t mpc7_dscf[MPC7_DSCF_SIZE *2]
static int get_scale_idx(GetBitContext *gb, int ref)
Subband structure - hold all variables for each subband.
This structure stores compressed data.
av_cold void ff_mpadsp_init(MPADSPContext *s)
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 VLC quant_vlc[MPC7_QUANT_VLC_TABLES][2]