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author | Vladimir Voroshilov <voroshil@gmail.com> | 2009-06-05 11:11:54 +0700 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2011-09-24 21:11:00 +0200 |
commit | b7c7fc335900eb4bfd403115eecfafff23be684d (patch) | |
tree | 4b88718d69d01fb71e1ea5c6b6f2e02eeafab996 /libavcodec | |
parent | 4920a1a913b3baeecbbc28759d22011826b54b50 (diff) | |
download | ffmpeg-b7c7fc335900eb4bfd403115eecfafff23be684d.tar.gz |
Synthesis filter
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/g729dec.c | 41 |
1 files changed, 41 insertions, 0 deletions
diff --git a/libavcodec/g729dec.c b/libavcodec/g729dec.c index db7b5527f5..c1d0905a0d 100644 --- a/libavcodec/g729dec.c +++ b/libavcodec/g729dec.c @@ -34,6 +34,7 @@ #include "g729.h" #include "lsp.h" #include "celp_math.h" +#include "celp_filters.h" #include "acelp_filters.h" #include "acelp_pitch_delay.h" #include "acelp_vectors.h" @@ -57,6 +58,9 @@ */ #define LSFQ_DIFF_MIN 321 +/// interpolation filter length +#define INTERPOL_LEN 11 + /** * minimum gain pitch value (3.8, Equation 47) * 0.2 in (1.14) @@ -111,6 +115,9 @@ typedef struct { int16_t quant_energy[4]; ///< (5.10) past quantized energy + /// previous speech data for LP synthesis filter + int16_t syn_filter_data[10]; + /// (1.14) pitch gain of current and five previous subframes int16_t past_gain_pitch[6]; @@ -283,6 +290,8 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, int pitch_delay_int; // pitch delay, integer part int pitch_delay_3x; // pitch delay, multiplied by 3 int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector + int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector + int j; int is_periodic = 0; // whether one of the subframes is declared as periodic or not if (*data_size < SUBFRAME_SIZE << 2) { @@ -476,10 +485,42 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0], 1 << 13, 14, SUBFRAME_SIZE); + memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t)); + + /* Temporary synth buffer is required since filter needs additional space at top of buffer and, thus, + synthesis can not be done directly to output buffer. This buffer will be reused by future + postprocessing filters. */ + if (ff_celp_lp_synthesis_filter( + synth+10, + &lp[i][1], + ctx->exc + i * SUBFRAME_SIZE, + SUBFRAME_SIZE, + 10, + 1, + 0x800)) { + /* Overflow occured, downscale excitation signal... */ + for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++) + ctx->exc_base[j] >>= 2; + + ff_celp_lp_synthesis_filter( + synth+10, + &lp[i][1], + ctx->exc + i * SUBFRAME_SIZE, + SUBFRAME_SIZE, + 10, + 0, + 0x800); + } + /* Save data (without postfilter) for use in next subframe. */ + memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t)); + if (frame_erasure) ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX); else ctx->pitch_delay_int_prev = pitch_delay_int; + + /* Dumb. Will be replaced by high-pass filter */ + memcpy(out_frame + i * SUBFRAME_SIZE, synth + 10, SUBFRAME_SIZE * sizeof(int16_t)); } ctx->was_periodic = is_periodic; |