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https://gitee.com/openharmony/third_party_ffmpeg
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Add functions for decoding >16 bits WavPack files.
Based on patches by Laurent Aimar (fenrir >whirlpool< videolan >dit< org) Originally committed as revision 18667 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Kostya Shishkov
parent
760db32a0c
commit
0638c2ae3c
@ -463,13 +463,142 @@ static int wv_unpack_mono(WavpackContext *s, GetBitContext *gb, int16_t *dst)
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return count;
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}
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static int wv_unpack_stereo_hires(WavpackContext *s, GetBitContext *gb, int32_t *dst)
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{
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int i, j, count = 0;
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int last, t;
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int A, B, L, L2, R, R2, bit;
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int pos = 0;
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uint32_t crc = 0xFFFFFFFF;
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s->one = s->zero = s->zeroes = 0;
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do{
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L = wv_get_value(s, gb, 0, &last);
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if(last) break;
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R = wv_get_value(s, gb, 1, &last);
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if(last) break;
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for(i = 0; i < s->terms; i++){
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t = s->decorr[i].value;
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j = 0;
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if(t > 0){
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if(t > 8){
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if(t & 1){
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A = 2 * s->decorr[i].samplesA[0] - s->decorr[i].samplesA[1];
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B = 2 * s->decorr[i].samplesB[0] - s->decorr[i].samplesB[1];
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}else{
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A = (3 * s->decorr[i].samplesA[0] - s->decorr[i].samplesA[1]) >> 1;
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B = (3 * s->decorr[i].samplesB[0] - s->decorr[i].samplesB[1]) >> 1;
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}
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s->decorr[i].samplesA[1] = s->decorr[i].samplesA[0];
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s->decorr[i].samplesB[1] = s->decorr[i].samplesB[0];
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j = 0;
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}else{
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A = s->decorr[i].samplesA[pos];
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B = s->decorr[i].samplesB[pos];
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j = (pos + t) & 7;
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}
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L2 = L + ((s->decorr[i].weightA * (int64_t)A + 512) >> 10);
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R2 = R + ((s->decorr[i].weightB * (int64_t)B + 512) >> 10);
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if(A && L) s->decorr[i].weightA -= ((((L ^ A) >> 30) & 2) - 1) * s->decorr[i].delta;
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if(B && R) s->decorr[i].weightB -= ((((R ^ B) >> 30) & 2) - 1) * s->decorr[i].delta;
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s->decorr[i].samplesA[j] = L = L2;
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s->decorr[i].samplesB[j] = R = R2;
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}else if(t == -1){
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L2 = L + ((s->decorr[i].weightA * (int64_t)s->decorr[i].samplesA[0] + 512) >> 10);
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UPDATE_WEIGHT_CLIP(s->decorr[i].weightA, s->decorr[i].delta, s->decorr[i].samplesA[0], L);
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L = L2;
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R2 = R + ((s->decorr[i].weightB * (int64_t)L2 + 512) >> 10);
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UPDATE_WEIGHT_CLIP(s->decorr[i].weightB, s->decorr[i].delta, L2, R);
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R = R2;
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s->decorr[i].samplesA[0] = R;
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}else{
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R2 = R + ((s->decorr[i].weightB * (int64_t)s->decorr[i].samplesB[0] + 512) >> 10);
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UPDATE_WEIGHT_CLIP(s->decorr[i].weightB, s->decorr[i].delta, s->decorr[i].samplesB[0], R);
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R = R2;
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if(t == -3){
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R2 = s->decorr[i].samplesA[0];
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s->decorr[i].samplesA[0] = R;
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}
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L2 = L + ((s->decorr[i].weightA * (int64_t)R2 + 512) >> 10);
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UPDATE_WEIGHT_CLIP(s->decorr[i].weightA, s->decorr[i].delta, R2, L);
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L = L2;
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s->decorr[i].samplesB[0] = L;
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}
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}
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pos = (pos + 1) & 7;
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if(s->joint)
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L += (R -= (L >> 1));
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crc = (crc * 3 + L) * 3 + R;
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bit = (L & s->and) | s->or;
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*dst++ = (((L + bit) << s->shift) - bit) << s->post_shift;
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bit = (R & s->and) | s->or;
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*dst++ = (((R + bit) << s->shift) - bit) << s->post_shift;
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count++;
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}while(!last && count < s->samples);
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if(crc != s->CRC){
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av_log(s->avctx, AV_LOG_ERROR, "CRC error\n");
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return -1;
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}
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return count * 2;
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}
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static int wv_unpack_mono_hires(WavpackContext *s, GetBitContext *gb, int32_t *dst)
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{
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int i, j, count = 0;
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int last, t;
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int A, S, T, bit;
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int pos = 0;
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uint32_t crc = 0xFFFFFFFF;
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s->one = s->zero = s->zeroes = 0;
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do{
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T = wv_get_value(s, gb, 0, &last);
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S = 0;
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if(last) break;
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for(i = 0; i < s->terms; i++){
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t = s->decorr[i].value;
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if(t > 8){
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if(t & 1)
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A = 2 * s->decorr[i].samplesA[0] - s->decorr[i].samplesA[1];
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else
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A = (3 * s->decorr[i].samplesA[0] - s->decorr[i].samplesA[1]) >> 1;
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s->decorr[i].samplesA[1] = s->decorr[i].samplesA[0];
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j = 0;
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}else{
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A = s->decorr[i].samplesA[pos];
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j = (pos + t) & 7;
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}
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S = T + ((s->decorr[i].weightA * (int64_t)A + 512) >> 10);
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if(A && T) s->decorr[i].weightA -= ((((T ^ A) >> 30) & 2) - 1) * s->decorr[i].delta;
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s->decorr[i].samplesA[j] = T = S;
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}
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pos = (pos + 1) & 7;
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crc = crc * 3 + S;
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bit = (S & s->and) | s->or;
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*dst++ = (((S + bit) << s->shift) - bit) << s->post_shift;
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count++;
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}while(!last && count < s->samples);
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if(crc != s->CRC){
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av_log(s->avctx, AV_LOG_ERROR, "CRC error\n");
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return -1;
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}
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return count;
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}
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static av_cold int wavpack_decode_init(AVCodecContext *avctx)
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{
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WavpackContext *s = avctx->priv_data;
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s->avctx = avctx;
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s->stereo = (avctx->channels == 2);
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avctx->sample_fmt = SAMPLE_FMT_S16;
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if(avctx->bits_per_coded_sample <= 16)
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avctx->sample_fmt = SAMPLE_FMT_S16;
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else
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avctx->sample_fmt = SAMPLE_FMT_S32;
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avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
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return 0;
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@ -483,11 +612,13 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
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int buf_size = avpkt->size;
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WavpackContext *s = avctx->priv_data;
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int16_t *samples = data;
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int32_t *samples32 = data;
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int samplecount;
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int got_terms = 0, got_weights = 0, got_samples = 0, got_entropy = 0, got_bs = 0;
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int got_hybrid = 0;
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const uint8_t* buf_end = buf + buf_size;
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int i, j, id, size, ssize, weights, t;
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int bpp = avctx->bits_per_coded_sample <= 16 ? 2 : 4;
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if (buf_size == 0){
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*data_size = 0;
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@ -504,7 +635,7 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
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return buf_size;
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}
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/* should not happen but who knows */
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if(s->samples * 2 * avctx->channels > *data_size){
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if(s->samples * bpp * avctx->channels > *data_size){
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av_log(avctx, AV_LOG_ERROR, "Packet size is too big to be handled in lavc!\n");
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return -1;
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}
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@ -513,7 +644,7 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
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s->joint = s->frame_flags & WV_JOINT_STEREO;
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s->hybrid = s->frame_flags & WV_HYBRID_MODE;
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s->hybrid_bitrate = s->frame_flags & WV_HYBRID_BITRATE;
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s->post_shift = (s->frame_flags >> 13) & 0x1f;
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s->post_shift = 8 * (bpp-1-(s->frame_flags&0x03)) + ((s->frame_flags >> 13) & 0x1f);
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s->CRC = AV_RL32(buf); buf += 4;
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// parse metadata blocks
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while(buf < buf_end){
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@ -701,11 +832,17 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
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return -1;
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}
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if(s->stereo_in)
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samplecount = wv_unpack_stereo(s, &s->gb, samples);
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else{
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samplecount = wv_unpack_mono(s, &s->gb, samples);
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if(s->stereo){
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if(s->stereo_in){
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if(bpp == 2)
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samplecount = wv_unpack_stereo(s, &s->gb, samples);
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else
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samplecount = wv_unpack_stereo_hires(s, &s->gb, samples32);
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}else{
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if(bpp == 2)
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samplecount = wv_unpack_mono(s, &s->gb, samples);
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else
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samplecount = wv_unpack_mono_hires(s, &s->gb, samples32);
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if(s->stereo && bpp == 2){
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int16_t *dst = samples + samplecount * 2;
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int16_t *src = samples + samplecount;
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int cnt = samplecount;
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@ -714,9 +851,18 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
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*--dst = *src;
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}
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samplecount *= 2;
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}else if(s->stereo){ //32-bit output
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int32_t *dst = samples32 + samplecount * 2;
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int32_t *src = samples32 + samplecount;
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int cnt = samplecount;
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while(cnt--){
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*--dst = *--src;
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*--dst = *src;
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}
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samplecount *= 2;
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}
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}
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*data_size = samplecount * 2;
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*data_size = samplecount * bpp;
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return buf_size;
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}
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