FFmpeg
|
wmadec.c
Go to the documentation of this file.
97 av_log(avctx, AV_LOG_WARNING, "Disabling use_variable_block_len, if this fails contact the ffmpeg developers and send us the file\n");
static void wma_lsp_to_curve(WMACodecContext *s, float *out, float *val_max_ptr, int n, float *lsp)
NOTE: We use the same code as Vorbis here.
Definition: wmadec.c:185
uint8_t last_superframe[MAX_CODED_SUPERFRAME_SIZE+FF_INPUT_BUFFER_PADDING_SIZE]
Definition: wma.h:122
About Git write you should know how to use GIT properly Luckily Git comes with excellent documentation git help man git shows you the available git< command > help man git< command > shows information about the subcommand< command > The most comprehensive manual is the website Git Reference visit they are quite exhaustive You do not need a special username or password All you need is to provide a ssh public key to the Git server admin What follows now is a basic introduction to Git and some FFmpeg specific guidelines Read it at least if you are granted commit privileges to the FFmpeg project you are expected to be familiar with these rules I if not You can get git from etc no matter how small Every one of them has been saved from looking like a fool by this many times It s very easy for stray debug output or cosmetic modifications to slip in
Definition: git-howto.txt:5
int ff_wma_run_level_decode(AVCodecContext *avctx, GetBitContext *gb, VLC *vlc, const float *level_table, const uint16_t *run_table, int version, WMACoef *ptr, int offset, int num_coefs, int block_len, int frame_len_bits, int coef_nb_bits)
Decode run level compressed coefficients.
Definition: wma.c:434
av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (%s)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic?ac->func_descr_generic:ac->func_descr)
static void wma_window(WMACodecContext *s, float *out)
Apply MDCT window and add into output.
Definition: wmadec.c:381
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: libavcodec/avcodec.h:1898
initialize output if(nPeaks >3)%at least 3 peaks in spectrum for trying to find f0 nf0peaks
Definition: samplefmt.h:50
float WMACoef
type for decoded coefficients, int16_t would be enough for wma 1/2
Definition: wma.h:56
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: libavcodec/avcodec.h:1242
static int decode_exp_vlc(WMACodecContext *s, int ch)
decode exponents coded with VLC codes
Definition: wmadec.c:317
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: libavcodec/avcodec.h:743
static float pow_m1_4(WMACodecContext *s, float x)
compute x^-0.25 with an exponent and mantissa table.
Definition: wmadec.c:134
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: libavutil/internal.h:123
static void wma_lsp_to_curve_init(WMACodecContext *s, int frame_len)
Definition: wmadec.c:153
Definition: avutil.h:144
static int wma_decode_superframe(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: wmadec.c:802
void(* imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:81
external API header
#define FF_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: libavcodec/avcodec.h:561
int exponent_high_bands[BLOCK_NB_SIZES][HIGH_BAND_MAX_SIZE]
Definition: wma.h:83
int high_band_values[MAX_CHANNELS][HIGH_BAND_MAX_SIZE]
Definition: wma.h:88
Definition: fft.h:62
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:524
Definition: libavcodec/avcodec.h:388
or the Software in violation of any applicable export control laws in any jurisdiction Except as provided by mandatorily applicable UPF has no obligation to provide you with source code to the Software In the event Software contains any source code
Definition: MELODIA - License.txt:24
static int noise(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const char *args, uint8_t **poutbuf, int *poutbuf_size, const uint8_t *buf, int buf_size, int keyframe)
Definition: noise_bsf.c:28
static int wma_decode_frame(WMACodecContext *s, float **samples, int samples_offset)
Definition: wmadec.c:766
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: libavcodec/utils.c:823
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
Definition: float_dsp.h:150
#define init_vlc(vlc, nb_bits, nb_codes,bits, bits_wrap, bits_size,codes, codes_wrap, codes_size,flags)
Definition: get_bits.h:426
static void decode_exp_lsp(WMACodecContext *s, int ch)
decode exponents coded with LSP coefficients (same idea as Vorbis)
Definition: wmadec.c:215
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:379
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Buffer references ownership and permissions
struct WMACodecContext WMACodecContext
common internal api header.
void(* vector_fmul_add)(float *dst, const float *src0, const float *src1, const float *src2, int len)
Calculate the product of two vectors of floats, add a third vector of floats and store the result in ...
Definition: float_dsp.h:121
Definition: libavcodec/avcodec.h:389
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:2
Definition: wma.h:66
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=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);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_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)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out
Definition: audio_convert.c:194
static int decode(AVCodecContext *avctx, void *data, int *got_frame, AVPacket *avpkt)
Definition: crystalhd.c:868
int high_band_coded[MAX_CHANNELS][HIGH_BAND_MAX_SIZE]
Definition: wma.h:87
void(* vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len)
Calculate the product of two vectors of floats, and store the result in a vector of floats...
Definition: float_dsp.h:140
Generated on Mon Nov 18 2024 06:52:03 for FFmpeg by 1.8.11