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;
216 output_samples = (int16_t *)frame->
data[0];
217 samples_end = output_samples + out;
225 predictor[1] =
sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
226 predictor[0] =
sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
228 predictor[0] =
sign_extend(bytestream2_get_le16u(&gb), 16);
232 while (output_samples < samples_end) {
234 predictor[ch] = av_clip_int16(predictor[ch]);
235 *output_samples++ = predictor[ch];
245 for (ch = 0; ch < avctx->
channels; ch++) {
246 predictor[ch] =
sign_extend(bytestream2_get_le16u(&gb), 16);
247 *output_samples++ = predictor[ch];
251 while (output_samples < samples_end) {
253 predictor[ch] = av_clip_int16(predictor[ch]);
254 *output_samples++ = predictor[ch];
263 int shift[2] = { 4, 4 };
265 for (ch = 0; ch < avctx->
channels; ch++)
266 predictor[ch] =
sign_extend(bytestream2_get_le16u(&gb), 16);
269 while (output_samples < samples_end) {
270 int diff = bytestream2_get_byteu(&gb);
276 shift[ch] -= (2 *
n);
284 predictor[ch] +=
diff;
286 predictor[ch] = av_clip_int16(predictor[ch]);
287 *output_samples++ = predictor[ch];
297 *samples_end_u8 = output_samples_u8 +
out;
298 while (output_samples_u8 < samples_end_u8) {
299 int n = bytestream2_get_byteu(&gb);
303 *output_samples_u8++ = s->
sample[0];
307 *output_samples_u8++ = s->
sample[stereo];
310 while (output_samples < samples_end) {
311 int n = bytestream2_get_byteu(&gb);
315 *output_samples++ = s->
sample[ch];
328 #define DPCM_DECODER(id_, name_, long_name_) \
329 AVCodec ff_ ## name_ ## _decoder = { \
331 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
332 .type = AVMEDIA_TYPE_AUDIO, \
334 .priv_data_size = sizeof(DPCMContext), \
335 .init = dpcm_decode_init, \
336 .decode = dpcm_decode_frame, \
337 .capabilities = CODEC_CAP_DR1, \
const struct AVCodec * codec
static int shift(int a, int b)
int16_t roq_square_array[256]
This structure describes decoded (raw) audio or video data.
ptrdiff_t const GLvoid * data
static const int8_t sol_table_old[16]
#define AV_LOG_WARNING
Something somehow does not look correct.
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)
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)
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)
static av_always_inline int diff(const uint32_t a, const uint32_t b)
int channels
number of audio channels
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-> out
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