[FFmpeg-cvslog] wavpack: switch to planar output
Anton Khirnov
git at videolan.org
Tue May 28 11:07:29 CEST 2013
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Sat May 25 15:31:18 2013 +0200| [528daa399018af74d52192eb1861d2b59d256111] | committer: Anton Khirnov
wavpack: switch to planar output
This simplifies the code and makes it faster.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=528daa399018af74d52192eb1861d2b59d256111
---
libavcodec/wavpack.c | 120 ++++++++++++++++++--------------------------------
1 file changed, 44 insertions(+), 76 deletions(-)
diff --git a/libavcodec/wavpack.c b/libavcodec/wavpack.c
index 4349ccb..2a3e335 100644
--- a/libavcodec/wavpack.c
+++ b/libavcodec/wavpack.c
@@ -518,7 +518,7 @@ static inline int wv_check_crc(WavpackFrameContext *s, uint32_t crc,
}
static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb,
- void *dst, const int type)
+ void *dst_l, void *dst_r, const int type)
{
int i, j, count = 0;
int last, t;
@@ -526,10 +526,12 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb,
int pos = s->pos;
uint32_t crc = s->sc.crc;
uint32_t crc_extra_bits = s->extra_sc.crc;
- int16_t *dst16 = dst;
- int32_t *dst32 = dst;
- float *dstfl = dst;
- const int channel_pad = s->avctx->channels - 2;
+ int16_t *dst16_l = dst_l;
+ int16_t *dst16_r = dst_r;
+ int32_t *dst32_l = dst_l;
+ int32_t *dst32_r = dst_r;
+ float *dstfl_l = dst_l;
+ float *dstfl_r = dst_r;
s->one = s->zero = s->zeroes = 0;
do {
@@ -558,7 +560,7 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb,
B = s->decorr[i].samplesB[pos];
j = (pos + t) & 7;
}
- if (type != AV_SAMPLE_FMT_S16) {
+ if (type != AV_SAMPLE_FMT_S16P) {
L2 = L + ((s->decorr[i].weightA * (int64_t)A + 512) >> 10);
R2 = R + ((s->decorr[i].weightB * (int64_t)B + 512) >> 10);
} else {
@@ -572,13 +574,13 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb,
s->decorr[i].samplesA[j] = L = L2;
s->decorr[i].samplesB[j] = R = R2;
} else if (t == -1) {
- if (type != AV_SAMPLE_FMT_S16)
+ if (type != AV_SAMPLE_FMT_S16P)
L2 = L + ((s->decorr[i].weightA * (int64_t)s->decorr[i].samplesA[0] + 512) >> 10);
else
L2 = L + ((s->decorr[i].weightA * s->decorr[i].samplesA[0] + 512) >> 10);
UPDATE_WEIGHT_CLIP(s->decorr[i].weightA, s->decorr[i].delta, s->decorr[i].samplesA[0], L);
L = L2;
- if (type != AV_SAMPLE_FMT_S16)
+ if (type != AV_SAMPLE_FMT_S16P)
R2 = R + ((s->decorr[i].weightB * (int64_t)L2 + 512) >> 10);
else
R2 = R + ((s->decorr[i].weightB * L2 + 512) >> 10);
@@ -586,7 +588,7 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb,
R = R2;
s->decorr[i].samplesA[0] = R;
} else {
- if (type != AV_SAMPLE_FMT_S16)
+ if (type != AV_SAMPLE_FMT_S16P)
R2 = R + ((s->decorr[i].weightB * (int64_t)s->decorr[i].samplesB[0] + 512) >> 10);
else
R2 = R + ((s->decorr[i].weightB * s->decorr[i].samplesB[0] + 512) >> 10);
@@ -598,7 +600,7 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb,
s->decorr[i].samplesA[0] = R;
}
- if (type != AV_SAMPLE_FMT_S16)
+ if (type != AV_SAMPLE_FMT_S16P)
L2 = L + ((s->decorr[i].weightA * (int64_t)R2 + 512) >> 10);
else
L2 = L + ((s->decorr[i].weightA * R2 + 512) >> 10);
@@ -612,18 +614,15 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb,
L += (R -= (L >> 1));
crc = (crc * 3 + L) * 3 + R;
- if (type == AV_SAMPLE_FMT_FLT) {
- *dstfl++ = wv_get_value_float(s, &crc_extra_bits, L);
- *dstfl++ = wv_get_value_float(s, &crc_extra_bits, R);
- dstfl += channel_pad;
- } else if (type == AV_SAMPLE_FMT_S32) {
- *dst32++ = wv_get_value_integer(s, &crc_extra_bits, L);
- *dst32++ = wv_get_value_integer(s, &crc_extra_bits, R);
- dst32 += channel_pad;
+ if (type == AV_SAMPLE_FMT_FLTP) {
+ *dstfl_l++ = wv_get_value_float(s, &crc_extra_bits, L);
+ *dstfl_r++ = wv_get_value_float(s, &crc_extra_bits, R);
+ } else if (type == AV_SAMPLE_FMT_S32P) {
+ *dst32_l++ = wv_get_value_integer(s, &crc_extra_bits, L);
+ *dst32_r++ = wv_get_value_integer(s, &crc_extra_bits, R);
} else {
- *dst16++ = wv_get_value_integer(s, &crc_extra_bits, L);
- *dst16++ = wv_get_value_integer(s, &crc_extra_bits, R);
- dst16 += channel_pad;
+ *dst16_l++ = wv_get_value_integer(s, &crc_extra_bits, L);
+ *dst16_r++ = wv_get_value_integer(s, &crc_extra_bits, R);
}
count++;
} while (!last && count < s->samples);
@@ -648,7 +647,6 @@ static inline int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb,
int16_t *dst16 = dst;
int32_t *dst32 = dst;
float *dstfl = dst;
- const int channel_stride = s->avctx->channels;
s->one = s->zero = s->zeroes = 0;
do {
@@ -669,7 +667,7 @@ static inline int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb,
A = s->decorr[i].samplesA[pos];
j = (pos + t) & 7;
}
- if (type != AV_SAMPLE_FMT_S16)
+ if (type != AV_SAMPLE_FMT_S16P)
S = T + ((s->decorr[i].weightA * (int64_t)A + 512) >> 10);
else
S = T + ((s->decorr[i].weightA * A + 512) >> 10);
@@ -680,15 +678,12 @@ static inline int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb,
pos = (pos + 1) & 7;
crc = crc * 3 + S;
- if (type == AV_SAMPLE_FMT_FLT) {
- *dstfl = wv_get_value_float(s, &crc_extra_bits, S);
- dstfl += channel_stride;
- } else if (type == AV_SAMPLE_FMT_S32) {
- *dst32 = wv_get_value_integer(s, &crc_extra_bits, S);
- dst32 += channel_stride;
+ if (type == AV_SAMPLE_FMT_FLTP) {
+ *dstfl++ = wv_get_value_float(s, &crc_extra_bits, S);
+ } else if (type == AV_SAMPLE_FMT_S32P) {
+ *dst32++ = wv_get_value_integer(s, &crc_extra_bits, S);
} else {
- *dst16 = wv_get_value_integer(s, &crc_extra_bits, S);
- dst16 += channel_stride;
+ *dst16++ = wv_get_value_integer(s, &crc_extra_bits, S);
}
count++;
} while (!last && count < s->samples);
@@ -722,9 +717,9 @@ static av_cold int wavpack_decode_init(AVCodecContext *avctx)
s->avctx = avctx;
if (avctx->bits_per_coded_sample <= 16)
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
else
- avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
if (avctx->channels <= 2 && !avctx->channel_layout)
avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO
: AV_CH_LAYOUT_MONO;
@@ -757,13 +752,13 @@ static av_cold int wavpack_decode_end(AVCodecContext *avctx)
}
static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
- void *data, int *got_frame_ptr,
+ uint8_t **data, int *got_frame_ptr,
const uint8_t *buf, int buf_size)
{
WavpackContext *wc = avctx->priv_data;
WavpackFrameContext *s;
GetByteContext gb;
- void *samples = data;
+ void *samples_l, *samples_r;
int samplecount;
int got_terms = 0, got_weights = 0, got_samples = 0,
got_entropy = 0, got_bs = 0, got_float = 0, got_hybrid = 0;
@@ -809,7 +804,6 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
}
s->frame_flags = bytestream2_get_le32(&gb);
bpp = av_get_bytes_per_sample(avctx->sample_fmt);
- samples = (uint8_t *)samples + bpp * wc->ch_offset;
orig_bpp = ((s->frame_flags & 0x03) + 1) << 3;
s->stereo = !(s->frame_flags & WV_MONO);
@@ -822,6 +816,10 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
s->hybrid_minclip = ((-1LL << (orig_bpp - 1)));
s->CRC = bytestream2_get_le32(&gb);
+ samples_l = data[wc->ch_offset];
+ if (s->stereo)
+ samples_r = data[wc->ch_offset + 1];
+
if (wc->mkv_mode)
bytestream2_skip(&gb, 4); // skip block size;
@@ -1111,11 +1109,11 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
av_log(avctx, AV_LOG_ERROR, "Packed samples not found\n");
return AVERROR_INVALIDDATA;
}
- if (!got_float && avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+ if (!got_float && avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
av_log(avctx, AV_LOG_ERROR, "Float information not found\n");
return AVERROR_INVALIDDATA;
}
- if (s->got_extra_bits && avctx->sample_fmt != AV_SAMPLE_FMT_FLT) {
+ if (s->got_extra_bits && avctx->sample_fmt != AV_SAMPLE_FMT_FLTP) {
const int size = get_bits_left(&s->gb_extra_bits);
const int wanted = s->samples * s->extra_bits << s->stereo_in;
if (size < wanted) {
@@ -1125,46 +1123,16 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
}
if (s->stereo_in) {
- samplecount = wv_unpack_stereo(s, &s->gb, samples, avctx->sample_fmt);
+ samplecount = wv_unpack_stereo(s, &s->gb, samples_l, samples_r, avctx->sample_fmt);
if (samplecount < 0)
return samplecount;
-
- samplecount >>= 1;
} else {
- const int channel_stride = avctx->channels;
-
- samplecount = wv_unpack_mono(s, &s->gb, samples, avctx->sample_fmt);
+ samplecount = wv_unpack_mono(s, &s->gb, samples_l, avctx->sample_fmt);
if (samplecount < 0)
return samplecount;
- if (s->stereo && avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
- int16_t *dst = (int16_t *)samples + 1;
- int16_t *src = (int16_t *)samples;
- int cnt = samplecount;
- while (cnt--) {
- *dst = *src;
- src += channel_stride;
- dst += channel_stride;
- }
- } else if (s->stereo && avctx->sample_fmt == AV_SAMPLE_FMT_S32) {
- int32_t *dst = (int32_t *)samples + 1;
- int32_t *src = (int32_t *)samples;
- int cnt = samplecount;
- while (cnt--) {
- *dst = *src;
- src += channel_stride;
- dst += channel_stride;
- }
- } else if (s->stereo) {
- float *dst = (float *)samples + 1;
- float *src = (float *)samples;
- int cnt = samplecount;
- while (cnt--) {
- *dst = *src;
- src += channel_stride;
- dst += channel_stride;
- }
- }
+ if (s->stereo)
+ memcpy(samples_r, samples_l, bpp * s->samples);
}
*got_frame_ptr = 1;
@@ -1218,11 +1186,11 @@ static int wavpack_decode_frame(AVCodecContext *avctx, void *data,
}
if (frame_flags & 0x80) {
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
} else if ((frame_flags & 0x03) <= 1) {
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
} else {
- avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
avctx->bits_per_raw_sample = ((frame_flags & 0x03) + 1) << 3;
}
@@ -1255,7 +1223,7 @@ static int wavpack_decode_frame(AVCodecContext *avctx, void *data,
return AVERROR_INVALIDDATA;
}
if ((samplecount = wavpack_decode_block(avctx, s->block,
- frame->data[0], got_frame_ptr,
+ frame->extended_data, got_frame_ptr,
buf, frame_size)) < 0) {
wavpack_decode_flush(avctx);
return samplecount;
More information about the ffmpeg-cvslog
mailing list