53 0, 1, 2, 3, 4, 5, 6, 7,
54 8, 9, 10, 11, 12, 13, 14, 15,
55 16, 17, 18, 19, 20, 21, 22, 23,
56 24, 25, 26, 27, 28, 29, 30, 31,
57 32, 33, 34, 35, 36, 37, 38, 39,
58 40, 41, 42, 43, 47, 51, 56, 61,
59 66, 72, 79, 86, 94, 102, 112, 122,
60 133, 145, 158, 173, 189, 206, 225, 245,
61 267, 292, 318, 348, 379, 414, 452, 493,
62 538, 587, 640, 699, 763, 832, 908, 991,
63 1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993,
64 2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008,
65 4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059,
66 8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206,
67 17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589,
68 -29973, -26728, -23186, -19322, -15105, -10503, -5481, -1,
69 1, 1, 5481, 10503, 15105, 19322, 23186, 26728,
70 29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298,
71 -17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597,
72 -8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772,
73 -4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373,
74 -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
75 -1081, -991, -908, -832, -763, -699, -640, -587,
76 -538, -493, -452, -414, -379, -348, -318, -292,
77 -267, -245, -225, -206, -189, -173, -158, -145,
78 -133, -122, -112, -102, -94, -86, -79, -72,
79 -66, -61, -56, -51, -47, -43, -42, -41,
80 -40, -39, -38, -37, -36, -35, -34, -33,
81 -32, -31, -30, -29, -28, -27, -26, -25,
82 -24, -23, -22, -21, -20, -19, -18, -17,
83 -16, -15, -14, -13, -12, -11, -10, -9,
84 -8, -7, -6, -5, -4, -3, -2, -1
89 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
90 -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0
94 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
95 0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15
99 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
100 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
101 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
102 0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
103 0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
104 0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
105 0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
106 0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
107 0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
108 0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
109 0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
110 0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
111 0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
131 for (i = 0; i < 128; i++) {
170 int *got_frame_ptr,
AVPacket *avpkt)
172 int buf_size = avpkt->
size;
179 int16_t *output_samples, *samples_end;
182 if (stereo && (buf_size & 1))
192 out = buf_size - 6 - avctx->
channels;
195 out = buf_size - 2 * avctx->
channels;
215 output_samples = (int16_t *)frame->
data[0];
216 samples_end = output_samples + out;
224 predictor[1] =
sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
225 predictor[0] =
sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
227 predictor[0] =
sign_extend(bytestream2_get_le16u(&gb), 16);
231 while (output_samples < samples_end) {
233 predictor[ch] = av_clip_int16(predictor[ch]);
234 *output_samples++ = predictor[ch];
244 for (ch = 0; ch < avctx->
channels; ch++) {
245 predictor[ch] =
sign_extend(bytestream2_get_le16u(&gb), 16);
246 *output_samples++ = predictor[ch];
250 while (output_samples < samples_end) {
252 predictor[ch] = av_clip_int16(predictor[ch]);
253 *output_samples++ = predictor[ch];
262 int shift[2] = { 4, 4 };
264 for (ch = 0; ch < avctx->
channels; ch++)
265 predictor[ch] =
sign_extend(bytestream2_get_le16u(&gb), 16);
268 while (output_samples < samples_end) {
269 int diff = bytestream2_get_byteu(&gb);
275 shift[ch] -= (2 * n);
283 predictor[ch] += diff;
285 predictor[ch] = av_clip_int16(predictor[ch]);
286 *output_samples++ = predictor[ch];
296 *samples_end_u8 = output_samples_u8 +
out;
297 while (output_samples_u8 < samples_end_u8) {
298 int n = bytestream2_get_byteu(&gb);
302 *output_samples_u8++ = s->
sample[0];
306 *output_samples_u8++ = s->
sample[stereo];
309 while (output_samples < samples_end) {
310 int n = bytestream2_get_byteu(&gb);
314 *output_samples++ = s->
sample[ch];
327 #define DPCM_DECODER(id_, name_, long_name_) \ 328 AVCodec ff_ ## name_ ## _decoder = { \ 330 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ 331 .type = AVMEDIA_TYPE_AUDIO, \ 333 .priv_data_size = sizeof(DPCMContext), \ 334 .init = dpcm_decode_init, \ 335 .decode = dpcm_decode_frame, \ 336 .capabilities = CODEC_CAP_DR1, \ const struct AVCodec * codec
int16_t roq_square_array[256]
This structure describes decoded (raw) audio or video data.
static const int8_t sol_table_old[16]
const int8_t * sol_table
delta table for SOL_DPCM
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
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=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;} 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_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);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)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out
enum AVSampleFormat sample_fmt
audio sample format
static const int16_t interplay_delta_table[]
static av_always_inline void bytestream2_skipu(GetByteContext *g, unsigned int size)
static void predictor(uint8_t *src, int size)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static const int8_t sol_table_new[16]
static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
void av_log(void *avcl, int level, const char *fmt,...)
#define AV_LOG_INFO
Standard information.
Libavcodec external API header.
static const int16_t sol_table_16[128]
main external API structure.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
unsigned int codec_tag
fourcc (LSB first, so "ABCD" -> ('D'<<24) + ('C'<<16) + ('B'<<8) + 'A').
static av_const int sign_extend(int val, unsigned bits)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
#define DPCM_DECODER(id_, name_, long_name_)
common internal api header.
static av_cold int dpcm_decode_init(AVCodecContext *avctx)
int channels
number of audio channels
This structure stores compressed data.
int sample[2]
previous sample (for SOL_DPCM)
int nb_samples
number of audio samples (per channel) described by this frame