62 #define ALAC_EXTRADATA_SIZE 36 117 int sign_modifier = 0;
127 k =
av_log2((history >> 9) + 3);
132 output_buffer[i] = (x >> 1) ^ -(x & 1);
138 history += x * rice_history_mult -
142 if ((history < 128) && (i + 1 < nb_samples)) {
146 k = 7 -
av_log2(history) + ((history + 16) >> 6);
150 if (block_size > 0) {
151 if (block_size >= nb_samples - i) {
153 "invalid zero block size of %d %d %d\n", block_size,
155 block_size = nb_samples - i - 1;
157 memset(&output_buffer[i + 1], 0,
158 block_size *
sizeof(*output_buffer));
161 if (block_size <= 0xffff)
176 int lpc_order,
int lpc_quant)
182 *buffer_out = *error_buffer;
188 memcpy(&buffer_out[1], &error_buffer[1],
189 (nb_samples - 1) *
sizeof(*buffer_out));
193 if (lpc_order == 31) {
196 buffer_out[i] =
sign_extend(buffer_out[i - 1] + error_buffer[i],
203 for (i = 1; i <= lpc_order && i <
nb_samples; i++)
204 buffer_out[i] =
sign_extend(buffer_out[i - 1] + error_buffer[i], bps);
211 int error_val = error_buffer[i];
216 for (j = 0; j < lpc_order; j++)
217 val += (pred[j] - d) * lpc_coefs[j];
218 val = (val + (1 << (lpc_quant - 1))) >> lpc_quant;
219 val += d + error_val;
225 for (j = 0; j < lpc_order && error_val * error_sign > 0; j++) {
229 lpc_coefs[j] -= sign;
231 error_val -= (val >> lpc_quant) * (j + 1);
241 int has_size,
bps, is_compressed, decorr_shift, decorr_left_weight, ret;
242 uint32_t output_samples;
276 }
else if (output_samples != alac->
nb_samples) {
288 int16_t lpc_coefs[2][32];
290 int prediction_type[2];
296 "Compression with rice limit 0");
313 for (i = lpc_order[ch] - 1; i >= 0; i--)
333 if (prediction_type[ch] == 15) {
344 }
else if (prediction_type[ch] > 0) {
346 prediction_type[ch]);
350 bps, lpc_coefs[ch], lpc_order[ch], lpc_quant[ch]);
364 decorr_left_weight = 0;
373 if (decorr_left_weight) {
375 decorr_shift, decorr_left_weight);
390 int16_t *outbuffer = (int16_t *)frame->
extended_data[ch_index + ch];
413 int *got_frame_ptr,
AVPacket *avpkt)
419 int ch, ret, got_end;
438 channels = (element ==
TYPE_CPE) ? 2 : 1;
439 if (ch + channels > alac->
channels ||
491 for (ch = 0; ch < 2; ch++) {
499 buf_size, buf_alloc_fail);
529 "max samples per frame invalid: %"PRIu32
"\n",
537 alac->
rice_limit = bytestream2_get_byteu(&gb);
538 alac->
channels = bytestream2_get_byteu(&gb);
539 bytestream2_get_be16u(&gb);
540 bytestream2_get_be32u(&gb);
541 bytestream2_get_be32u(&gb);
612 {
"extra_bits_bug",
"Force non-standard decoding process",
636 .priv_class = &alac_class
const char const char void * val
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
int nb_samples
number of samples in the current frame
This structure describes decoded (raw) audio or video data.
#define ALAC_EXTRADATA_SIZE
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
static int init_thread_copy(AVCodecContext *avctx)
static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index, int channels)
#define LIBAVUTIL_VERSION_INT
static av_cold int init(AVCodecContext *avctx)
#define AV_OPT_FLAG_AUDIO_PARAM
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={ FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){ in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);} ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map) { switch(av_get_bytes_per_sample(in_fmt)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&HAVE_MMX) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;} if(ctx->out_simd_align_mask) { int planes=out->planar ? out->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;} if(ctx->simd_f &&!ctx->ch_map &&!misaligned){ off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){ if(out->planar==in->planar){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
const char * av_default_item_name(void *ptr)
Return the context name.
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
static const AVOption options[]
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
int32_t * extra_bits_buffer[2]
static int get_sbits(GetBitContext *s, int n)
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static int get_sbits_long(GetBitContext *s, int n)
Read 0-32 bits as a signed integer.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
int32_t * predict_error_buffer[2]
static int get_unary_0_9(GetBitContext *gb)
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
enum AVSampleFormat sample_fmt
audio sample format
uint8_t rice_initial_history
static av_cold int alac_decode_close(AVCodecContext *avctx)
Multithreading support functions.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static int get_bits_count(const GetBitContext *s)
static av_always_inline void bytestream2_skipu(GetByteContext *g, unsigned int size)
static const AVClass alac_class
bitstream reader API header.
int32_t * output_samples_buffer[2]
static int alac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
int extra_bits
number of extra bits beyond 16-bit
static int get_bits_left(GetBitContext *gb)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out, int nb_samples, int bps, int16_t *lpc_coefs, int lpc_order, int lpc_quant)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static int sign_only(int v)
const char * name
Name of the codec implementation.
#define AV_CODEC_CAP_FRAME_THREADS
Codec supports frame-level multithreading.
uint64_t channel_layout
Audio channel layout.
#define ALAC_MAX_CHANNELS
uint32_t max_samples_per_frame
#define ONLY_IF_THREADS_ENABLED(x)
Define a function with only the non-default version specified.
audio channel layout utility functions
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
static int alac_set_info(ALACContext *alac)
uint8_t rice_history_mult
const uint64_t ff_alac_channel_layouts[ALAC_MAX_CHANNELS+1]
static const float pred[4]
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Libavcodec external API header.
int sample_rate
samples per second
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
int ff_thread_get_buffer(AVCodecContext *avctx, ThreadFrame *f, int flags)
Wrapper around get_buffer() for frame-multithreaded codecs.
main external API structure.
static unsigned int get_bits1(GetBitContext *s)
Describe the class of an AVClass context structure.
static void skip_bits(GetBitContext *s, int n)
static unsigned int decode_scalar(GetBitContext *gb, int k, int bps)
#define AV_OPT_FLAG_DECODING_PARAM
a generic parameter which can be set by the user for demuxing or decoding
void(* append_extra_bits[2])(int32_t *buffer[2], int32_t *extra_bits_buffer[2], int extra_bits, int channels, int nb_samples)
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
const uint8_t ff_alac_channel_layout_offsets[ALAC_MAX_CHANNELS][ALAC_MAX_CHANNELS]
static av_const int sign_extend(int val, unsigned bits)
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
void(* decorrelate_stereo)(int32_t *buffer[2], int nb_samples, int decorr_shift, int decorr_left_weight)
common internal api header.
static av_cold int alac_decode_init(AVCodecContext *avctx)
#define FF_ALLOC_OR_GOTO(ctx, p, size, label)
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
static int allocate_buffers(ALACContext *alac)
int channels
number of audio channels
static int rice_decompress(ALACContext *alac, int32_t *output_buffer, int nb_samples, int bps, int rice_history_mult)
uint8_t ** extended_data
pointers to the data planes/channels.
av_cold void ff_alacdsp_init(ALACDSPContext *c)
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.