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27 #define BITSTREAM_READER_LE
36 #define MAX_BACKWARD_FILTER_ORDER 36
37 #define MAX_BACKWARD_FILTER_LEN 40
38 #define MAX_BACKWARD_FILTER_NONREC 35
40 #define RA288_BLOCK_SIZE 5
41 #define RA288_BLOCKS_PER_FRAME 32
102 float *gain_block = ractx->
gain_hist + 28;
108 for (
i=0;
i < 10;
i++)
116 sumsum =
exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
118 for (
i=0;
i < 5;
i++)
123 sum =
FFMAX(sum, 5.0 / (1<<24));
126 memmove(gain_block, gain_block + 1, 9 *
sizeof(*gain_block));
128 gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
146 int order,
int n,
int non_rec,
float *
out,
147 float *hist,
float *out2,
const float *
window)
163 for (
i=0;
i <= order;
i++) {
164 out2[
i] = out2[
i] * 0.5625 + buffer1[
i];
165 out [
i] = out2[
i] + buffer2[
i];
169 *
out *= 257.0 / 256.0;
176 float *hist,
float *rec,
const float *
window,
177 float *lpc,
const float *
tab,
178 int order,
int n,
int non_rec,
int move_size)
187 memmove(hist, hist + n, move_size*
sizeof(*hist));
191 int *got_frame_ptr,
AVPacket *avpkt)
193 const uint8_t *buf = avpkt->
data;
194 int buf_size = avpkt->
size;
200 if (buf_size < avctx->block_align) {
202 "Error! Input buffer is too small [%d<%d]\n",
221 decode(ractx, gain, cb_coef);
241 .
p.
name =
"real_288",
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
static const int16_t codetable[128][5]
const FFCodec ff_ra_288_decoder
This structure describes decoded (raw) audio or video data.
static int AAC_RENAME() compute_lpc_coefs(const LPC_TYPE *autoc, int max_order, LPC_TYPE *lpc, int lpc_stride, int fail, int normalize)
Levinson-Durbin recursion.
#define MAX_BACKWARD_FILTER_NONREC
static void backward_filter(RA288Context *ractx, float *hist, float *rec, const float *window, float *lpc, const float *tab, int order, int n, int non_rec, int move_size)
Backward synthesis filter, find the LPC coefficients from past speech data.
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static SDL_Window * window
AVCodec p
The public AVCodec.
AVChannelLayout ch_layout
Audio channel layout.
static void decode(RA288Context *ractx, float gain, int cb_coef)
static const struct twinvq_data tab
#define MAX_BACKWARD_FILTER_ORDER
int flags
AV_CODEC_FLAG_*.
static const float gain_window[FFALIGN(38, 16)]
static const float syn_bw_tab[FFALIGN(36, 16)]
synthesis bandwidth broadening table
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
#define LOCAL_ALIGNED(a, t, v,...)
#define FF_CODEC_DECODE_CB(func)
int(* init)(AVBSFContext *ctx)
static void do_hybrid_window(RA288Context *ractx, int order, int n, int non_rec, float *out, float *hist, float *out2, const float *window)
Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
static const float gain_bw_tab[FFALIGN(10, 16)]
gain bandwidth broadening table
#define CODEC_LONG_NAME(str)
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
must be printed separately If there s no standard function for printing the type you the WRITE_1D_FUNC_ARGV macro is a very quick way to create one See libavcodec dv_tablegen c for an example The h file This file should the initialization functions should not do and instead of the variable declarations the generated *_tables h file should be included Since that will be generated in the build the path must be i e not Makefile changes To make the automatic table creation work
#define RA288_BLOCKS_PER_FRAME
float sp_lpc[FFALIGN(36, 16)]
LPC coefficients for speech data (spec: A)
static av_cold int ra288_decode_init(AVCodecContext *avctx)
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
An AVChannelLayout holds information about the channel layout of audio data.
#define DECLARE_ALIGNED(n, t, v)
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats and store the result in a vector of floats.
enum AVSampleFormat sample_fmt
audio sample format
static void convolve(float *tgt, const float *src, int len, int n)
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
int nb_samples
number of audio samples (per channel) described by this frame
#define i(width, name, range_min, range_max)
static int ra288_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
const char * name
Name of the codec implementation.
float gain_lpc[FFALIGN(10, 16)]
LPC coefficients for gain (spec: GB)
static const float syn_window[FFALIGN(111, 16)]
float gain_hist[38]
log-gain history (spec: SBLG).
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
float sp_rec[37]
speech part of the gain autocorrelation (spec: REXP)
float gain_rec[11]
recursive part of the gain autocorrelation (spec: REXPLG)
main external API structure.
the frame and frame reference mechanism is intended to as much as expensive copies of that data while still allowing the filters to produce correct results The data is stored in buffers represented by AVFrame structures Several references can point to the same frame buffer
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define AV_CHANNEL_LAYOUT_MONO
static const float amptable[8]
This structure stores compressed data.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
The exact code depends on how similar the blocks are and how related they are to the block
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
float sp_hist[111]
speech data history (spec: SB).
#define MAX_BACKWARD_FILTER_LEN