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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];
119 memset(
c->oldDSCF, 0,
sizeof(
c->oldDSCF));
129 if (
c->maxbands >=
BANDS) {
145 if(vlc_initialized)
return 0;
200 for(
i = 0;
i < 2;
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;
253 keyframe =
c->cur_frame == 0;
256 memset(
c->Q, 0,
sizeof(
c->Q));
257 c->last_bits_used = 0;
268 if(maxband > 32) maxband -= 33;
276 if(maxband >
c->maxbands + 1) {
280 c->last_max_band = maxband;
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++)
301 for(
i = maxband - 1;
i >= 0;
i--)
308 for(
i = maxband;
i <
c->maxbands;
i++)
312 for(
i = 0;
i < 32;
i++)
313 c->oldDSCF[0][
i] =
c->oldDSCF[1][
i] = 1;
316 for(
i = 0;
i < maxband;
i++){
327 for(
i = 0;
i < maxband;
i++){
328 for(
ch = 0;
ch < 2;
ch++){
331 if(
c->oldDSCF[
ch][
i]){
333 c->oldDSCF[
ch][
i] = 0;
340 for(j = 0; j < 2; j++){
347 bands[
i].scf_idx[
ch][j + 1] = ((
bands[
i].scf_idx[
ch][j] + t - 25) & 0x7F) - 6;
354 for(
ch = 0;
ch < 2;
ch++){
386 c->Q[
ch][off + j + 1] = t >> 4;
387 c->Q[
ch][off + j + 0] = (t & 8) ? (t & 0xF) - 16 : (t & 0xF);
397 c->Q[
ch][off + j] = t;
398 cnt = (cnt >> 1) +
FFABS(
c->Q[
ch][off + j]);
405 c->Q[
ch][off + j] <<= res - 9;
408 c->Q[
ch][off + j] -= (1 << (res - 2)) - 1;
419 (int16_t **)
frame->extended_data,
425 if(
c->cur_frame >=
c->frames)
429 c->last_bits_used = buf_size << 3;
431 c->last_bits_used = buf_size << 3;
436 return c->cur_frame ?
c->last_bits_used >> 3 : buf_size;
static const uint8_t mpc8_q6_codes[2][MPC8_Q6_SIZE]
static av_cold int init(AVCodecContext *avctx)
static int get_bits_left(GetBitContext *gb)
uint64_t channel_layout
Audio channel layout.
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
static const int8_t mpc8_res_bits[2][MPC8_RES_SIZE]
static enum AVSampleFormat sample_fmts[]
static const int8_t mpc8_q9up_bits[MPC8_Q9UP_SIZE]
static av_cold void mpc8_decode_flush(AVCodecContext *avctx)
static const int quant_offsets[6]
#define AV_CH_LAYOUT_MONO
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_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), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { 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) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;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)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8: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);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=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) { int planes=out->planar ? out->ch_count :1;unsigned m=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){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
static int get_bits_count(const GetBitContext *s)
av_cold void ff_mpadsp_init(MPADSPContext *s)
static int mpc8_dec_base(GetBitContext *gb, int k, int n)
This structure describes decoded (raw) audio or video data.
#define init_vlc(vlc, nb_bits, nb_codes, bits, bits_wrap, bits_size, codes, codes_wrap, codes_size, flags)
static const uint16_t table[]
static const int8_t mpc8_q8_bits[2][MPC8_Q8_SIZE]
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
static const int8_t mpc8_q4_bits[MPC8_Q4_SIZE]
static void skip_bits(GetBitContext *s, int n)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static const int8_t mpc8_q4_syms[MPC8_Q4_SIZE]
static const int8_t mpc8_scfi1_bits[MPC8_SCFI1_SIZE]
static const int8_t mpc8_bands_bits[MPC8_BANDS_SIZE]
static int mpc8_get_mod_golomb(GetBitContext *gb, int m)
static int mpc8_get_mask(GetBitContext *gb, int size, int t)
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels)
#define AV_CH_LAYOUT_STEREO
static const uint32_t mpc8_cnk_lost[16][33]
static const uint8_t mpc8_cnk_len[16][33]
s EdgeDetect Foobar g libavfilter vf_edgedetect c libavfilter vf_foobar c edit libavfilter and add an entry for foobar following the pattern of the other filters edit libavfilter allfilters and add an entry for foobar following the pattern of the other filters configure make j< whatever > ffmpeg ffmpeg i you should get a foobar png with Lena edge detected That s your new playground is ready Some little details about what s going which in turn will define variables for the build system and the C
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
static const uint16_t mask[17]
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
static const uint16_t vlc_offsets[13]
static const int8_t mpc8_idx52[125]
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static const uint8_t mpc8_dscf0_codes[MPC8_DSCF0_SIZE]
av_cold void ff_mpc_init(void)
static const uint8_t mpc8_q8_codes[2][MPC8_Q8_SIZE]
static int mpc8_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static const float bands[]
static const uint8_t mpc8_q2_codes[2][MPC8_Q2_SIZE]
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
static const int8_t mpc8_q1_bits[MPC8_Q1_SIZE]
static void flush(AVCodecContext *avctx)
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
static const uint8_t mpc8_q1_codes[MPC8_Q1_SIZE]
static unsigned int get_bits1(GetBitContext *s)
static const int8_t mpc8_q5_bits[2][MPC8_Q5_SIZE]
#define INIT_VLC_USE_NEW_STATIC
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)
static const int8_t mpc8_huffq2[5 *5 *5]
static VLC quant_vlc[4][2]
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
static const uint8_t mpc8_scfi0_codes[MPC8_SCFI0_SIZE]
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
static const unsigned int mpc8_thres[]
enum AVSampleFormat sample_fmt
audio sample format
static const uint8_t mpc8_q9up_codes[MPC8_Q9UP_SIZE]
static const uint32_t mpc8_cnk[16][32]
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
static const uint8_t mpc8_scfi1_codes[MPC8_SCFI1_SIZE]
int channels
number of audio channels
#define i(width, name, range_min, range_max)
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
static const int8_t mpc8_q3_bits[MPC8_Q3_SIZE]
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
AVSampleFormat
Audio sample formats.
static int mpc8_dec_enum(GetBitContext *gb, int k, int n)
static const uint8_t mpc8_q7_codes[2][MPC8_Q7_SIZE]
const char * name
Name of the codec implementation.
static const int8_t mpc8_q2_bits[2][MPC8_Q2_SIZE]
static const uint8_t mpc8_bands_codes[MPC8_BANDS_SIZE]
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
static const int q3_offsets[2]
static const int8_t mpc8_q3_syms[MPC8_Q3_SIZE]
static av_cold int mpc8_decode_init(AVCodecContext *avctx)
static const uint8_t mpc8_q4_codes[MPC8_Q4_SIZE]
static const uint8_t mpc8_dscf1_codes[MPC8_DSCF1_SIZE]
static const int8_t mpc8_dscf1_bits[MPC8_DSCF1_SIZE]
static const uint8_t mpc8_q3_codes[MPC8_Q3_SIZE]
main external API structure.
Subband structure - hold all variables for each subband.
static const int8_t mpc8_idx50[125]
static const uint8_t mpc8_q5_codes[2][MPC8_Q5_SIZE]
static const int8_t mpc8_dscf0_bits[MPC8_DSCF0_SIZE]
static const uint8_t mpc8_res_codes[2][MPC8_RES_SIZE]
static const int8_t mpc8_q7_bits[2][MPC8_Q7_SIZE]
#define avpriv_request_sample(...)
This structure stores compressed data.
static const int8_t mpc8_scfi0_bits[MPC8_SCFI0_SIZE]
static const int8_t mpc8_idx51[125]
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
VLC_TYPE(* table)[2]
code, bits
static const int8_t mpc8_q6_bits[2][MPC8_Q6_SIZE]