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mpegaudiodsp.h
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void ff_mpadsp_apply_window_float(float *synth_buf, float *window, int *dither_state, float *samples, int incr)
struct MPADSPContext MPADSPContext
int32_t ff_mpa_synth_window_fixed[]
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
float ff_mdct_win_float[8][MDCT_BUF_SIZE]
void ff_mpadsp_init_mipsdspr1(MPADSPContext *s)
Definition: mpegaudiodsp_mips_fixed.c:903
void ff_mpa_synth_init_fixed(int32_t *window)
void ff_mpa_synth_init_float(float *window)
void ff_imdct36_blocks_fixed(int *out, int *buf, int *in, int count, int switch_point, int block_type)
void ff_mpa_synth_filter_float(MPADSPContext *s, float *synth_buf_ptr, int *synth_buf_offset, float *window, int *dither_state, float *samples, int incr, float *sb_samples)
void ff_imdct36_blocks_float(float *out, float *buf, float *in, int count, int switch_point, int block_type)
overlapping window(triangular window to avoid too much overlapping) ovidx
void(* apply_window_fixed)(int32_t *synth_buf, int32_t *window, int *dither_state, int16_t *samples, int incr)
Definition: mpegaudiodsp.h:28
#define MDCT_BUF_SIZE
For SSE implementation, MDCT_BUF_SIZE/2 should be 128-bit aligned.
Definition: mpegaudiodsp.h:84
void(* imdct36_blocks_float)(float *out, float *buf, float *in, int count, int switch_point, int block_type)
Definition: mpegaudiodsp.h:33
float ff_mpa_synth_window_float[]
void ff_mpa_synth_filter_fixed(MPADSPContext *s, int32_t *synth_buf_ptr, int *synth_buf_offset, int32_t *window, int *dither_state, int16_t *samples, int incr, int32_t *sb_samples)
void ff_mpadsp_init_mipsfpu(MPADSPContext *s)
Definition: mpegaudiodsp_mips_float.c:1245
void ff_init_mpadsp_tabs_fixed(void)
Definition: mpegaudiodsp.h:25
void(* dct32_float)(float *dst, const float *src)
Definition: mpegaudiodsp.h:30
void ff_mpadsp_apply_window_fixed(int32_t *synth_buf, int32_t *window, int *dither_state, int16_t *samples, int incr)
int ff_mdct_win_fixed[8][MDCT_BUF_SIZE]
void(* dct32_fixed)(int *dst, const int *src)
Definition: mpegaudiodsp.h:31
common internal and external API header
void(* apply_window_float)(float *synth_buf, float *window, int *dither_state, float *samples, int incr)
Definition: mpegaudiodsp.h:26
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:2
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
void ff_mpadsp_init_altivec(MPADSPContext *s)
Definition: mpegaudiodec_altivec.c:127
void ff_init_mpadsp_tabs_float(void)
void(* imdct36_blocks_fixed)(int *out, int *buf, int *in, int count, int switch_point, int block_type)
Definition: mpegaudiodsp.h:35
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