67 int buf_size = avpkt->
size;
82 "Frame too small (%d bytes). Truncated file?\n", buf_size);
91 samples = (int16_t *)frame->
data[0];
104 refl_rms[1] =
ff_interp(ractx, block_coefs[1], 2,
105 energy <= ractx->old_energy,
107 refl_rms[2] =
ff_interp(ractx, block_coefs[2], 3, 0, energy);
unsigned int lpc_tables[2][10]
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
int ff_t_sqrt(unsigned int x)
Evaluate sqrt(x << 24).
This structure describes decoded (raw) audio or video data.
const int16_t *const ff_lpc_refl_cb[10]
ptrdiff_t const GLvoid * data
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static av_cold int init(AVCodecContext *avctx)
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
#define NBLOCKS
number of subblocks within a block
enum AVSampleFormat sample_fmt
audio sample format
av_cold void ff_audiodsp_init(AudioDSPContext *c)
unsigned int lpc_refl_rms[2]
unsigned int ff_rms(const int *data)
bitstream reader API header.
void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs, int cba_idx, int cb1_idx, int cb2_idx, int gval, int gain)
static av_cold int ra144_decode_init(AVCodecContext *avctx)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
unsigned int * lpc_coef[2]
LPC coefficients: lpc_coef[0] is the coefficients of the current frame and lpc_coef[1] of the previou...
static const int sizes[][2]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
const char * name
Name of the codec implementation.
uint64_t channel_layout
Audio channel layout.
audio channel layout utility functions
int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
void ff_int_to_int16(int16_t *out, const int *inp)
#define BLOCKSIZE
subblock size in 16-bit words
void ff_eval_coefs(int *coefs, const int *refl)
Evaluate the LPC filter coefficients from the reflection coefficients.
Libavcodec external API header.
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
main external API structure.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
AVCodec ff_ra_144_decoder
unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
common internal api header.
unsigned int old_energy
previous frame energy
const int16_t ff_energy_tab[32]
int16_t curr_sblock[50]
The current subblock padded by the last 10 values of the previous one.
static void do_output_subblock(RA144Context *ractx, const int16_t *lpc_coefs, int gval, GetBitContext *gb)
int channels
number of audio channels
Filter the word “frame” indicates either a video frame or a group of audio samples
static int ra144_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Uncompress one block (20 bytes -> 160*2 bytes).
#define FFSWAP(type, a, b)
#define AV_CH_LAYOUT_MONO
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.