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https://gitee.com/openharmony/third_party_ffmpeg
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fixed audio frame buffering problem (should correct problems on some streams) - faster synthesis filter - prototype 'parse_only' support
Originally committed as revision 2173 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Fabrice Bellard
parent
d99ce8d7de
commit
8c5b5683ae
@ -310,7 +310,7 @@ static int decode_init(AVCodecContext * avctx)
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static int init=0;
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int i, j, k;
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if(!init) {
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if (!init && !avctx->parse_only) {
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/* scale factors table for layer 1/2 */
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for(i=0;i<64;i++) {
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int shift, mod;
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@ -737,58 +737,96 @@ static void dct32(int32_t *out, int32_t *tab)
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#if FRAC_BITS <= 15
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#define OUT_SAMPLE(sum)\
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{\
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int sum1;\
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sum1 = (sum + (1 << (OUT_SHIFT - 1))) >> OUT_SHIFT;\
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if (sum1 < -32768)\
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sum1 = -32768;\
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else if (sum1 > 32767)\
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sum1 = 32767;\
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*samples = sum1;\
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samples += incr;\
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static inline int round_sample(int sum)
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{
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int sum1;
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sum1 = (sum + (1 << (OUT_SHIFT - 1))) >> OUT_SHIFT;
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if (sum1 < -32768)
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sum1 = -32768;
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else if (sum1 > 32767)
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sum1 = 32767;
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return sum1;
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}
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#define SUM8(off, op) \
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{ \
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sum op w[0 * 64 + off] * p[0 * 64];\
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sum op w[1 * 64 + off] * p[1 * 64];\
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sum op w[2 * 64 + off] * p[2 * 64];\
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sum op w[3 * 64 + off] * p[3 * 64];\
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sum op w[4 * 64 + off] * p[4 * 64];\
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sum op w[5 * 64 + off] * p[5 * 64];\
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sum op w[6 * 64 + off] * p[6 * 64];\
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sum op w[7 * 64 + off] * p[7 * 64];\
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}
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#if defined(ARCH_POWERPC_405)
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/* signed 16x16 -> 32 multiply add accumulate */
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#define MACS(rt, ra, rb) \
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asm ("maclhw %0, %2, %3" : "=r" (rt) : "0" (rt), "r" (ra), "r" (rb));
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/* signed 16x16 -> 32 multiply */
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#define MULS(ra, rb) \
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({ int __rt; asm ("mullhw %0, %1, %2" : "=r" (__rt) : "r" (ra), "r" (rb)); __rt; })
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#else
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#define OUT_SAMPLE(sum)\
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{\
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int sum1;\
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sum1 = (int)((sum + (int64_t_C(1) << (OUT_SHIFT - 1))) >> OUT_SHIFT);\
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if (sum1 < -32768)\
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sum1 = -32768;\
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else if (sum1 > 32767)\
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sum1 = 32767;\
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*samples = sum1;\
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samples += incr;\
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}
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/* signed 16x16 -> 32 multiply add accumulate */
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#define MACS(rt, ra, rb) rt += (ra) * (rb)
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#define SUM8(off, op) \
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{ \
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sum op MUL64(w[0 * 64 + off], p[0 * 64]);\
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sum op MUL64(w[1 * 64 + off], p[1 * 64]);\
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sum op MUL64(w[2 * 64 + off], p[2 * 64]);\
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sum op MUL64(w[3 * 64 + off], p[3 * 64]);\
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sum op MUL64(w[4 * 64 + off], p[4 * 64]);\
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sum op MUL64(w[5 * 64 + off], p[5 * 64]);\
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sum op MUL64(w[6 * 64 + off], p[6 * 64]);\
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sum op MUL64(w[7 * 64 + off], p[7 * 64]);\
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}
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/* signed 16x16 -> 32 multiply */
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#define MULS(ra, rb) ((ra) * (rb))
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#endif
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#else
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static inline int round_sample(int64_t sum)
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{
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int sum1;
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sum1 = (int)((sum + (int64_t_C(1) << (OUT_SHIFT - 1))) >> OUT_SHIFT);
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if (sum1 < -32768)
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sum1 = -32768;
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else if (sum1 > 32767)
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sum1 = 32767;
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return sum1;
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}
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#define MULS(ra, rb) MUL64(ra, rb)
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#endif
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#define SUM8(sum, op, w, p) \
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{ \
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sum op MULS((w)[0 * 64], p[0 * 64]);\
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sum op MULS((w)[1 * 64], p[1 * 64]);\
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sum op MULS((w)[2 * 64], p[2 * 64]);\
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sum op MULS((w)[3 * 64], p[3 * 64]);\
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sum op MULS((w)[4 * 64], p[4 * 64]);\
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sum op MULS((w)[5 * 64], p[5 * 64]);\
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sum op MULS((w)[6 * 64], p[6 * 64]);\
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sum op MULS((w)[7 * 64], p[7 * 64]);\
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}
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#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
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{ \
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int tmp;\
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tmp = p[0 * 64];\
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sum1 op1 MULS((w1)[0 * 64], tmp);\
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sum2 op2 MULS((w2)[0 * 64], tmp);\
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tmp = p[1 * 64];\
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sum1 op1 MULS((w1)[1 * 64], tmp);\
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sum2 op2 MULS((w2)[1 * 64], tmp);\
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tmp = p[2 * 64];\
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sum1 op1 MULS((w1)[2 * 64], tmp);\
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sum2 op2 MULS((w2)[2 * 64], tmp);\
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tmp = p[3 * 64];\
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sum1 op1 MULS((w1)[3 * 64], tmp);\
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sum2 op2 MULS((w2)[3 * 64], tmp);\
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tmp = p[4 * 64];\
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sum1 op1 MULS((w1)[4 * 64], tmp);\
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sum2 op2 MULS((w2)[4 * 64], tmp);\
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tmp = p[5 * 64];\
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sum1 op1 MULS((w1)[5 * 64], tmp);\
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sum2 op2 MULS((w2)[5 * 64], tmp);\
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tmp = p[6 * 64];\
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sum1 op1 MULS((w1)[6 * 64], tmp);\
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sum2 op2 MULS((w2)[6 * 64], tmp);\
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tmp = p[7 * 64];\
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sum1 op1 MULS((w1)[7 * 64], tmp);\
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sum2 op2 MULS((w2)[7 * 64], tmp);\
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}
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/* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
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32 samples. */
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/* XXX: optimize by avoiding ring buffer usage */
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@ -797,15 +835,16 @@ static void synth_filter(MPADecodeContext *s1,
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int32_t sb_samples[SBLIMIT])
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{
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int32_t tmp[32];
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register MPA_INT *synth_buf, *p;
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register MPA_INT *w;
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register MPA_INT *synth_buf;
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const register MPA_INT *w, *w2, *p;
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int j, offset, v;
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int16_t *samples2;
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#if FRAC_BITS <= 15
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int sum;
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int sum, sum2;
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#else
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int64_t sum;
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int64_t sum, sum2;
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#endif
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dct32(tmp, sb_samples);
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offset = s1->synth_buf_offset[ch];
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@ -826,32 +865,42 @@ static void synth_filter(MPADecodeContext *s1,
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/* copy to avoid wrap */
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memcpy(synth_buf + 512, synth_buf, 32 * sizeof(MPA_INT));
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samples2 = samples + 31 * incr;
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w = window;
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for(j=0;j<16;j++) {
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sum = 0;
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p = synth_buf + 16 + j; /* 0-15 */
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SUM8(0, +=);
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p = synth_buf + 48 - j; /* 32-47 */
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SUM8(32, -=);
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OUT_SAMPLE(sum);
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w++;
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}
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p = synth_buf + 32; /* 48 */
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w2 = window + 31;
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sum = 0;
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SUM8(32, -=);
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OUT_SAMPLE(sum);
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p = synth_buf + 16;
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SUM8(sum, +=, w, p);
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p = synth_buf + 48;
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SUM8(sum, -=, w + 32, p);
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*samples = round_sample(sum);
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samples += incr;
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w++;
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for(j=17;j<32;j++) {
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/* we calculate two samples at the same time to avoid one memory
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access per two sample */
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for(j=1;j<16;j++) {
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sum = 0;
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p = synth_buf + 48 - j; /* 17-31 */
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SUM8(0, -=);
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p = synth_buf + 16 + j; /* 49-63 */
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SUM8(32, -=);
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OUT_SAMPLE(sum);
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sum2 = 0;
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p = synth_buf + 16 + j;
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SUM8P2(sum, +=, sum2, -=, w, w2, p);
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p = synth_buf + 48 - j;
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SUM8P2(sum, -=, sum2, -=, w + 32, w2 + 32, p);
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*samples = round_sample(sum);
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samples += incr;
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*samples2 = round_sample(sum2);
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samples2 -= incr;
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w++;
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w2--;
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}
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p = synth_buf + 32;
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sum = 0;
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SUM8(sum, -=, w + 32, p);
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*samples = round_sample(sum);
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offset = (offset - 32) & 511;
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s1->synth_buf_offset[ch] = offset;
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}
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@ -1157,6 +1206,47 @@ static int decode_header(MPADecodeContext *s, uint32_t header)
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return 0;
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}
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/* useful helper to get mpeg audio stream infos. Return -1 if error in
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header */
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int mp_decode_header(int *sample_rate_ptr,
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int *nb_channels_ptr,
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int *coded_frame_size_ptr,
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int *decoded_frame_size_ptr,
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uint32_t head)
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{
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MPADecodeContext s1, *s = &s1;
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int decoded_frame_size;
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if (check_header(head) != 0)
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return -1;
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if (decode_header(s, head) != 0) {
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return -1;
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}
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switch(s->layer) {
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case 1:
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decoded_frame_size = 384;
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break;
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case 2:
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decoded_frame_size = 1152;
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break;
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default:
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case 3:
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if (s->lsf)
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decoded_frame_size = 576;
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else
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decoded_frame_size = 1152;
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break;
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}
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*sample_rate_ptr = s->sample_rate;
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*nb_channels_ptr = s->nb_channels;
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*coded_frame_size_ptr = s->frame_size;
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*decoded_frame_size_ptr = decoded_frame_size * 2 * s->nb_channels;
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return 0;
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}
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/* return the number of decoded frames */
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static int mp_decode_layer1(MPADecodeContext *s)
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{
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@ -2391,7 +2481,20 @@ static int decode_frame(AVCodecContext * avctx,
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avctx->sample_rate = s->sample_rate;
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avctx->channels = s->nb_channels;
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avctx->bit_rate = s->bit_rate;
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avctx->frame_size = s->frame_size;
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switch(s->layer) {
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case 1:
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avctx->frame_size = 384;
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break;
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case 2:
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avctx->frame_size = 1152;
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break;
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case 3:
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if (s->lsf)
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avctx->frame_size = 576;
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else
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avctx->frame_size = 1152;
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break;
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}
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}
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}
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} else if (s->frame_size == -1) {
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@ -2457,15 +2560,22 @@ static int decode_frame(AVCodecContext * avctx,
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buf_ptr += len;
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s->inbuf_ptr += len;
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buf_size -= len;
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} else {
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out_size = mp_decode_frame(s, out_samples);
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}
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next_data:
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if (s->frame_size > 0 &&
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(s->inbuf_ptr - s->inbuf) >= s->frame_size) {
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if (avctx->parse_only) {
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/* simply return the frame data */
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*(uint8_t **)data = s->inbuf;
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out_size = s->inbuf_ptr - s->inbuf;
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} else {
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out_size = mp_decode_frame(s, out_samples);
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}
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s->inbuf_ptr = s->inbuf;
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s->frame_size = 0;
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*data_size = out_size;
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break;
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}
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next_data:
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;
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}
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return buf_ptr - buf;
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}
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@ -2480,6 +2590,7 @@ AVCodec mp2_decoder =
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NULL,
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NULL,
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decode_frame,
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CODEC_CAP_PARSE_ONLY,
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};
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AVCodec mp3_decoder =
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@ -2492,15 +2603,5 @@ AVCodec mp3_decoder =
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NULL,
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NULL,
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decode_frame,
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CODEC_CAP_PARSE_ONLY,
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};
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#undef C1
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#undef C2
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#undef C3
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#undef C4
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#undef C5
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#undef C6
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#undef C7
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#undef C8
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#undef FRAC_BITS
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#undef HEADER_SIZE
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