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author | Martin Storsjö <martin@martin.st> | 2010-09-23 21:33:29 +0000 |
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committer | Martin Storsjö <martin@martin.st> | 2010-09-23 21:33:29 +0000 |
commit | 21e681ba57735bca225f1f903c7e93ffdb0f86ba (patch) | |
tree | 0b4f8b750c1958fa754a760fedc4c510ac78503f /libavcodec/g722.c | |
parent | 2efcde73fccaeda948e64d5de4af3eb6205fd1c8 (diff) | |
download | ffmpeg-21e681ba57735bca225f1f903c7e93ffdb0f86ba.tar.gz |
Add a G.722 encoder
Originally committed as revision 25164 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/g722.c')
-rw-r--r-- | libavcodec/g722.c | 83 |
1 files changed, 82 insertions, 1 deletions
diff --git a/libavcodec/g722.c b/libavcodec/g722.c index 8707d16719..51a0f39abc 100644 --- a/libavcodec/g722.c +++ b/libavcodec/g722.c @@ -1,5 +1,5 @@ /* - * G.722 ADPCM audio decoder + * G.722 ADPCM audio encoder/decoder * * Copyright (c) CMU 1993 Computer Science, Speech Group * Chengxiang Lu and Alex Hauptmann @@ -219,6 +219,7 @@ static av_cold int g722_init(AVCodecContext * avctx) return 0; } +#if CONFIG_ADPCM_G722_DECODER static const int16_t low_inv_quant5[32] = { -35, -35, -2919, -2195, -1765, -1458, -1219, -1023, -858, -714, -587, -473, -370, -276, -190, -110, @@ -301,4 +302,84 @@ AVCodec adpcm_g722_decoder = { .long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"), .max_lowres = 1, }; +#endif + +#if CONFIG_ADPCM_G722_ENCODER +static const int16_t low_quant[33] = { + 35, 72, 110, 150, 190, 233, 276, 323, + 370, 422, 473, 530, 587, 650, 714, 786, + 858, 940, 1023, 1121, 1219, 1339, 1458, 1612, + 1765, 1980, 2195, 2557, 2919 +}; + +static inline void filter_samples(G722Context *c, const int16_t *samples, + int *xlow, int *xhigh) +{ + int xout1, xout2; + c->prev_samples[c->prev_samples_pos++] = samples[0]; + c->prev_samples[c->prev_samples_pos++] = samples[1]; + apply_qmf(c->prev_samples + c->prev_samples_pos - 24, &xout1, &xout2); + *xlow = xout1 + xout2 >> 13; + *xhigh = xout1 - xout2 >> 13; + if (c->prev_samples_pos >= PREV_SAMPLES_BUF_SIZE) { + memmove(c->prev_samples, + c->prev_samples + c->prev_samples_pos - 22, + 22 * sizeof(c->prev_samples[0])); + c->prev_samples_pos = 22; + } +} + +static inline int encode_high(const struct G722Band *state, int xhigh) +{ + int diff = av_clip_int16(xhigh - state->s_predictor); + int pred = 141 * state->scale_factor >> 8; + /* = diff >= 0 ? (diff < pred) + 2 : diff >= -pred */ + return ((diff ^ (diff >> (sizeof(diff)*8-1))) < pred) + 2*(diff >= 0); +} + +static inline int encode_low(const struct G722Band* state, int xlow) +{ + int diff = av_clip_int16(xlow - state->s_predictor); + /* = diff >= 0 ? diff : -(diff + 1) */ + int limit = diff ^ (diff >> (sizeof(diff)*8-1)); + int i = 0; + limit = limit + 1 << 10; + if (limit > low_quant[8] * state->scale_factor) + i = 9; + while (i < 29 && limit > low_quant[i] * state->scale_factor) + i++; + return (diff < 0 ? (i < 2 ? 63 : 33) : 61) - i; +} + +static int g722_encode_frame(AVCodecContext *avctx, + uint8_t *dst, int buf_size, void *data) +{ + G722Context *c = avctx->priv_data; + const int16_t *samples = data; + int i; + + for (i = 0; i < buf_size >> 1; i++) { + int xlow, xhigh, ihigh, ilow; + filter_samples(c, &samples[2*i], &xlow, &xhigh); + ihigh = encode_high(&c->band[1], xhigh); + ilow = encode_low(&c->band[0], xlow); + update_high_predictor(&c->band[1], c->band[1].scale_factor * + high_inv_quant[ihigh] >> 10, ihigh); + update_low_predictor(&c->band[0], ilow >> 2); + *dst++ = ihigh << 6 | ilow; + } + return i; +} + +AVCodec adpcm_g722_encoder = { + .name = "g722", + .type = AVMEDIA_TYPE_AUDIO, + .id = CODEC_ID_ADPCM_G722, + .priv_data_size = sizeof(G722Context), + .init = g722_init, + .encode = g722_encode_frame, + .long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"), + .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, +}; +#endif |