21 #define BITSTREAM_WRITER_LE
68 qm[0] -= dx[0]; qm[1] -= dx[1]; qm[2] -= dx[2]; qm[3] -= dx[3];
69 qm[4] -= dx[4]; qm[5] -= dx[5]; qm[6] -= dx[6]; qm[7] -= dx[7];
70 }
else if (c->
error > 0) {
71 qm[0] += dx[0]; qm[1] += dx[1]; qm[2] += dx[2]; qm[3] += dx[3];
72 qm[4] += dx[4]; qm[5] += dx[5]; qm[6] += dx[6]; qm[7] += dx[7];
75 sum += dl[0] * qm[0] + dl[1] * qm[1] + dl[2] * qm[2] + dl[3] * qm[3] +
76 dl[4] * qm[4] + dl[5] * qm[5] + dl[6] * qm[6] + dl[7] * qm[7];
78 dx[0] = dx[1]; dx[1] = dx[2]; dx[2] = dx[3]; dx[3] = dx[4];
79 dl[0] = dl[1]; dl[1] = dl[2]; dl[2] = dl[3]; dl[3] = dl[4];
81 dx[4] = ((dl[4] >> 30) | 1);
82 dx[5] = ((dl[5] >> 30) | 2) & ~1;
83 dx[6] = ((dl[6] >> 30) | 2) & ~1;
84 dx[7] = ((dl[7] >> 30) | 4) & ~3;
86 dl[4] = -dl[5]; dl[5] = -dl[6];
87 dl[6] = *in - dl[7]; dl[7] = *
in;
88 dl[5] += dl[6]; dl[4] += dl[5];
90 *in -= (sum >> c->
shift);
102 const int16_t *ptr = (
const int16_t *)frame->
data[0];
106 ret = ptr[sample] >> 8;
117 int ret, i, out_bytes, cur_chan = 0, res = 0, samples = 0;
124 for (i = 0; i < avctx->
channels; i++) {
134 uint32_t k, unary, outval;
140 if (cur_chan < avctx->channels - 1)
147 #define PRED(x, k) (int32_t)((((uint64_t)(x) << (k)) - (x)) >> (k))
156 outval = (value > 0) ? (value << 1) - 1: -value << 1;
160 rice->
sum0 += outval - (rice->
sum0 >> 4);
170 rice->
sum1 += outval - (rice->
sum1 >> 4);
176 unary = 1 + (outval >> k);
182 put_bits(&pb, unary, (1 << unary) - 1);
193 if (cur_chan < avctx->channels - 1)
205 avpkt->
size = out_bytes + 4;
static void av_unused put_bits32(PutBitContext *s, uint32_t value)
Write exactly 32 bits into a bitstream.
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size)
Check AVPacket size and/or allocate data.
This structure describes decoded (raw) audio or video data.
static av_cold int tta_encode_init(AVCodecContext *avctx)
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
#define AV_LOG_WARNING
Something somehow does not look correct.
static av_cold int init(AVCodecContext *avctx)
#define CODEC_CAP_LOSSLESS
Codec is lossless.
void ff_tta_rice_init(TTARice *c, uint32_t k0, uint32_t k1)
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
static av_cold int tta_encode_close(AVCodecContext *avctx)
enum AVSampleFormat sample_fmt
audio sample format
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
#define CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static int32_t get_sample(const AVFrame *frame, int sample, enum AVSampleFormat format)
const char * name
Name of the codec implementation.
Libavcodec external API header.
static int put_bits_count(PutBitContext *s)
const uint32_t ff_tta_shift_1[]
GLsizei GLboolean const GLfloat * value
const uint32_t *const ff_tta_shift_16
uint32_t av_crc(const AVCRC *ctx, uint32_t crc, const uint8_t *buffer, size_t length)
Calculate the CRC of a block.
int frame_size
Number of samples per channel in an audio frame.
AVSampleFormat
Audio sample formats.
int sample_rate
samples per second
const uint8_t ff_tta_filter_configs[]
main external API structure.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
static void ttafilter_process(TTAFilter *c, int32_t *in)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
const AVCRC * av_crc_get_table(AVCRCId crc_id)
Get an initialized standard CRC table.
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
int channels
number of audio channels
static enum AVSampleFormat sample_fmts[]
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
#define av_malloc_array(a, b)
void ff_tta_filter_init(TTAFilter *c, int32_t shift)
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
static int tta_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)