December 2018 - ffmpeg-cvslog

lavfi/vaapi: Add constant VAAPI_VPP_BACKGROUND_BLACK
by Zachary Zhou 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Zachary Zhou <zachary.zhou(a)intel.com> | Tue Dec 25 14:03:30 2018 +0800| [b6b3159b04afab4f57ce788a5f4871244385f478] | committer: Mark Thompson lavfi/vaapi: Add constant VAAPI_VPP_BACKGROUND_BLACK Signed-off-by: Zachary Zhou <zachary.zhou(a)intel.com> Signed-off-by: Mark Thompson <sw(a)jkqxz.net> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b6b3159b04afab4f5… --- libavfilter/vaapi_vpp.h | 3 +++ libavfilter/vf_deinterlace_vaapi.c | 2 +- libavfilter/vf_misc_vaapi.c | 2 +- libavfilter/vf_procamp_vaapi.c | 2 +- libavfilter/vf_scale_vaapi.c | 2 +- 5 files changed, 7 insertions(+), 4 deletions(-) diff --git a/libavfilter/vaapi_vpp.h b/libavfilter/vaapi_vpp.h index 0bc31018d4..96f720f07d 100644 --- a/libavfilter/vaapi_vpp.h +++ b/libavfilter/vaapi_vpp.h @@ -27,6 +27,9 @@ #include "avfilter.h" +// ARGB black, for VAProcPipelineParameterBuffer.output_background_color. +#define VAAPI_VPP_BACKGROUND_BLACK 0xff000000 + typedef struct VAAPIVPPContext { const AVClass *class; diff --git a/libavfilter/vf_deinterlace_vaapi.c b/libavfilter/vf_deinterlace_vaapi.c index f7a262d0c6..97aee6588f 100644 --- a/libavfilter/vf_deinterlace_vaapi.c +++ b/libavfilter/vf_deinterlace_vaapi.c @@ -256,7 +256,7 @@ static int deint_vaapi_filter_frame(AVFilterLink *inlink, AVFrame *input_frame) ff_vaapi_vpp_colour_standard(input_frame->colorspace); params.output_region = NULL; - params.output_background_color = 0xff000000; + params.output_background_color = VAAPI_VPP_BACKGROUND_BLACK; params.output_color_standard = params.surface_color_standard; params.pipeline_flags = 0; diff --git a/libavfilter/vf_misc_vaapi.c b/libavfilter/vf_misc_vaapi.c index 30b808a993..e227c9ff6b 100644 --- a/libavfilter/vf_misc_vaapi.c +++ b/libavfilter/vf_misc_vaapi.c @@ -174,7 +174,7 @@ static int misc_vaapi_filter_frame(AVFilterLink *inlink, AVFrame *input_frame) ff_vaapi_vpp_colour_standard(input_frame->colorspace); params.output_region = NULL; - params.output_background_color = 0xff000000; + params.output_background_color = VAAPI_VPP_BACKGROUND_BLACK; params.output_color_standard = params.surface_color_standard; params.pipeline_flags = 0; diff --git a/libavfilter/vf_procamp_vaapi.c b/libavfilter/vf_procamp_vaapi.c index 10eccbe97d..46f3ab6465 100644 --- a/libavfilter/vf_procamp_vaapi.c +++ b/libavfilter/vf_procamp_vaapi.c @@ -171,7 +171,7 @@ static int procamp_vaapi_filter_frame(AVFilterLink *inlink, AVFrame *input_frame ff_vaapi_vpp_colour_standard(input_frame->colorspace); params.output_region = NULL; - params.output_background_color = 0xff000000; + params.output_background_color = VAAPI_VPP_BACKGROUND_BLACK; params.output_color_standard = params.surface_color_standard; params.pipeline_flags = 0; diff --git a/libavfilter/vf_scale_vaapi.c b/libavfilter/vf_scale_vaapi.c index b06ae764bf..3699363140 100644 --- a/libavfilter/vf_scale_vaapi.c +++ b/libavfilter/vf_scale_vaapi.c @@ -133,7 +133,7 @@ static int scale_vaapi_filter_frame(AVFilterLink *inlink, AVFrame *input_frame) ff_vaapi_vpp_colour_standard(input_frame->colorspace); params.output_region = 0; - params.output_background_color = 0xff000000; + params.output_background_color = VAAPI_VPP_BACKGROUND_BLACK; params.output_color_standard = params.surface_color_standard; params.pipeline_flags = 0;

1 0

avfilter/af_afir: stop using as much threads as stream have it
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Dec 30 17:40:31 2018 +0100| [ba3d983c95b2054d046c9b9b79088634c54a24f0] | committer: Paul B Mahol avfilter/af_afir: stop using as much threads as stream have it > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ba3d983c95b2054d0… --- libavfilter/af_afir.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 3e13edf96a..35fa66bad5 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -56,7 +56,7 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le sum[2 * n] += t[2 * n] * c[2 * n]; } -static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) +static int fir_channel(AVFilterContext *ctx, void *arg, int ch) { AudioFIRContext *s = ctx->priv; const float *in = (const float *)s->in[0]->extended_data[ch]; @@ -138,6 +138,19 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) return 0; } +static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) +{ + AVFrame *out = arg; + const int start = (out->channels * jobnr) / nb_jobs; + const int end = (out->channels * (jobnr+1)) / nb_jobs; + + for (int ch = start; ch < end; ch++) { + fir_channel(ctx, out, ch); + } + + return 0; +} + static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; @@ -152,7 +165,8 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink) if (s->pts == AV_NOPTS_VALUE) s->pts = in->pts; s->in[0] = in; - ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels); + ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels, + ff_filter_get_nb_threads(ctx))); out->pts = s->pts; if (s->pts != AV_NOPTS_VALUE)

1 0

avfilter/af_afir: optimize code a little more
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Dec 30 17:23:53 2018 +0100| [72270d5bd071684944fa0a09d390e818aa76a975] | committer: Paul B Mahol avfilter/af_afir: optimize code a little more Avoid memset. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=72270d5bd07168494… --- libavfilter/af_afir.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 8271c2cfce..3e13edf96a 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -76,7 +76,6 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) seg->output_offset[ch] += s->min_part_size; if (seg->output_offset[ch] == seg->part_size) { seg->output_offset[ch] = 0; - memset(dst, 0, sizeof(*dst) * seg->part_size); } else { memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src)); @@ -119,9 +118,7 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) buf[n] += sum[n]; } - for (n = 0; n < seg->part_size; n++) { - dst[n] += buf[n]; - } + memcpy(dst, buf, seg->part_size * sizeof(*dst)); buf = (float *)seg->buffer->extended_data[ch]; memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));

1 0

avcodec/prores_ks: Fix luma quantization if q >= MAX_STORED_Q
by Alex Mogurenko 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Alex Mogurenko <alex(a)mogurenko.com> | Fri Dec 28 22:30:08 2018 +0200| [e4788ae31b2e9af45d11f4bf4498c075dcc25a6c] | committer: Derek Buitenhuis avcodec/prores_ks: Fix luma quantization if q >= MAX_STORED_Q The problem occurs in slice quant estimation and slice encoding: If the slice quant is larger than MAX_STORED_Q we don't use pre-calculated quant matrices, but generate a new one, but both qmat and qmat_chroma both point to the same table, so the luma table ends up having chroma table values. Add custom_chroma_q the same way as custom_q. Signed-off-by: Derek Buitenhuis <derek.buitenhuis(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e4788ae31b2e9af45… --- libavcodec/proresenc_kostya.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/libavcodec/proresenc_kostya.c b/libavcodec/proresenc_kostya.c index 9a77d24fb6..e045a972f1 100644 --- a/libavcodec/proresenc_kostya.c +++ b/libavcodec/proresenc_kostya.c @@ -222,6 +222,7 @@ typedef struct ProresThreadData { DECLARE_ALIGNED(16, int16_t, blocks)[MAX_PLANES][64 * 4 * MAX_MBS_PER_SLICE]; DECLARE_ALIGNED(16, uint16_t, emu_buf)[16 * 16]; int16_t custom_q[64]; + int16_t custom_chroma_q[64]; struct TrellisNode *nodes; } ProresThreadData; @@ -232,6 +233,7 @@ typedef struct ProresContext { int16_t quants[MAX_STORED_Q][64]; int16_t quants_chroma[MAX_STORED_Q][64]; int16_t custom_q[64]; + int16_t custom_chroma_q[64]; const uint8_t *quant_mat; const uint8_t *quant_chroma_mat; const uint8_t *scantable; @@ -574,7 +576,7 @@ static int encode_slice(AVCodecContext *avctx, const AVFrame *pic, qmat_chroma = ctx->quants_chroma[quant]; } else { qmat = ctx->custom_q; - qmat_chroma = ctx->custom_q; + qmat_chroma = ctx->custom_chroma_q; for (i = 0; i < 64; i++) { qmat[i] = ctx->quant_mat[i] * quant; qmat_chroma[i] = ctx->quant_chroma_mat[i] * quant; @@ -902,7 +904,7 @@ static int find_slice_quant(AVCodecContext *avctx, qmat_chroma = ctx->quants_chroma[q]; } else { qmat = td->custom_q; - qmat_chroma = td->custom_q; + qmat_chroma = td->custom_chroma_q; for (i = 0; i < 64; i++) { qmat[i] = ctx->quant_mat[i] * q; qmat_chroma[i] = ctx->quant_chroma_mat[i] * q;

1 0

avfilter/af_afir: implement non-uniform partitioned convolution
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sat Dec 29 11:40:13 2018 +0100| [a53a9f1c8d1ffad80956027ffee5f506c98f28ff] | committer: Paul B Mahol avfilter/af_afir: implement non-uniform partitioned convolution Using multiple frequency delay lines. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a53a9f1c8d1ffad80… --- doc/filters.texi | 4 +- libavfilter/af_afir.c | 180 +++++++++++++++++++++++++++++++++----------------- libavfilter/af_afir.h | 7 +- 3 files changed, 129 insertions(+), 62 deletions(-) diff --git a/doc/filters.texi b/doc/filters.texi index 45d90f6165..b3e2081bcf 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1231,14 +1231,14 @@ Set video stream size. This option is used only when @var{response} is enabled. Set video stream frame rate. This option is used only when @var{response} is enabled. @item minp -Set minimal partition size used for convolution. Default is @var{16}. +Set minimal partition size used for convolution. Default is @var{8192}. Allowed range is from @var{16} to @var{32768}. Lower values decreases latency at cost of higher CPU usage. @item maxp Set maximal partition size used for convolution. Default is @var{8192}. Allowed range is from @var{16} to @var{32768}. -Lower values decreases latency at cost of higher CPU usage. +Lower values may increase CPU usage. @end table @subsection Examples diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 3dda875e35..8271c2cfce 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -59,54 +59,84 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) { AudioFIRContext *s = ctx->priv; - AudioFIRSegment *seg = &s->seg[0]; - const float *src = (const float *)s->in[0]->extended_data[ch]; - float *sum = (float *)seg->sum->extended_data[ch]; + const float *in = (const float *)s->in[0]->extended_data[ch]; AVFrame *out = arg; - float *block, *dst, *ptr; + float *block, *buf, *ptr = (float *)out->extended_data[ch]; int n, i, j; - memset(sum, 0, sizeof(*sum) * seg->fft_length); - block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size; - memset(block, 0, sizeof(*block) * seg->fft_length); + for (int segment = 0; segment < s->nb_segments; segment++) { + AudioFIRSegment *seg = &s->seg[segment]; + float *src = (float *)seg->input->extended_data[ch]; + float *dst = (float *)seg->output->extended_data[ch]; + float *sum = (float *)seg->sum->extended_data[ch]; + + s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(out->nb_samples, 4)); + emms_c(); + + seg->output_offset[ch] += s->min_part_size; + if (seg->output_offset[ch] == seg->part_size) { + seg->output_offset[ch] = 0; + memset(dst, 0, sizeof(*dst) * seg->part_size); + } else { + memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src)); + + dst += seg->output_offset[ch]; + for (n = 0; n < out->nb_samples; n++) { + ptr[n] += dst[n]; + } + continue; + } - s->fdsp->vector_fmul_scalar(block, src, s->dry_gain, FFALIGN(out->nb_samples, 4)); - emms_c(); + memset(sum, 0, sizeof(*sum) * seg->fft_length); + block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size; + memset(block + seg->part_size, 0, sizeof(*block) * (seg->fft_length - seg->part_size)); - av_rdft_calc(seg->rdft[ch], block); - block[2 * seg->part_size] = block[1]; - block[1] = 0; + memcpy(block, src, sizeof(*src) * seg->part_size); - j = seg->part_index[ch]; + av_rdft_calc(seg->rdft[ch], block); + block[2 * seg->part_size] = block[1]; + block[1] = 0; - for (i = 0; i < seg->nb_partitions; i++) { - const int coffset = i * seg->coeff_size; - const float *block = (const float *)seg->block->extended_data[ch] + j * seg->block_size; - const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset; + j = seg->part_index[ch]; - s->fcmul_add(sum, block, (const float *)coeff, seg->part_size); + for (i = 0; i < seg->nb_partitions; i++) { + const int coffset = j * seg->coeff_size; + const float *block = (const float *)seg->block->extended_data[ch] + i * seg->block_size; + const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset; - if (j == 0) - j = seg->nb_partitions; - j--; - } + s->fcmul_add(sum, block, (const float *)coeff, seg->part_size); - sum[1] = sum[2 * seg->part_size]; - av_rdft_calc(seg->irdft[ch], sum); + if (j == 0) + j = seg->nb_partitions; + j--; + } - dst = (float *)seg->buffer->extended_data[ch]; - for (n = 0; n < seg->part_size; n++) { - dst[n] += sum[n]; - } + sum[1] = sum[2 * seg->part_size]; + av_rdft_calc(seg->irdft[ch], sum); - ptr = (float *)out->extended_data[ch]; - s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4)); - emms_c(); + buf = (float *)seg->buffer->extended_data[ch]; + for (n = 0; n < seg->part_size; n++) { + buf[n] += sum[n]; + } - dst = (float *)seg->buffer->extended_data[ch]; - memcpy(dst, sum + seg->part_size, seg->part_size * sizeof(*dst)); + for (n = 0; n < seg->part_size; n++) { + dst[n] += buf[n]; + } + + buf = (float *)seg->buffer->extended_data[ch]; + memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf)); + + seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions; + + memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src)); - seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions; + for (n = 0; n < out->nb_samples; n++) { + ptr[n] += dst[n]; + } + } + + s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(out->nb_samples, 4)); + emms_c(); return 0; } @@ -275,19 +305,28 @@ end: av_free(mag); } -static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_partitions, int part_size) +static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, + int offset, int nb_partitions, int part_size) { + AudioFIRContext *s = ctx->priv; + seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft)); seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft)); if (!seg->rdft || !seg->irdft) return AVERROR(ENOMEM); - seg->fft_length = part_size * 4 + 1; - seg->part_size = part_size; - seg->block_size = FFALIGN(seg->fft_length, 32); - seg->coeff_size = FFALIGN(seg->part_size + 1, 32); + seg->fft_length = part_size * 4 + 1; + seg->part_size = part_size; + seg->block_size = FFALIGN(seg->fft_length, 32); + seg->coeff_size = FFALIGN(seg->part_size + 1, 32); seg->nb_partitions = nb_partitions; - seg->segment_size = part_size * nb_partitions; + seg->input_size = offset + s->min_part_size; + seg->input_offset = offset; + + seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index)); + seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset)); + if (!seg->part_index || !seg->output_offset) + return AVERROR(ENOMEM); for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) { seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C); @@ -296,15 +335,13 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_parti return AVERROR(ENOMEM); } - seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index)); - if (!seg->part_index) - return AVERROR(ENOMEM); - seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length); seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size); seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size); seg->coeff = ff_get_audio_buffer(ctx->inputs[1], seg->nb_partitions * seg->coeff_size * 2); - if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff) + seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size); + seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size); + if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output) return AVERROR(ENOMEM); return 0; @@ -313,20 +350,37 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_parti static int convert_coeffs(AVFilterContext *ctx) { AudioFIRContext *s = ctx->priv; - int ret, i, ch, n, N; + int left, offset = 0, part_size, max_part_size; + int ret, i, ch, n; float power = 0; s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]); if (s->nb_taps <= 0) return AVERROR(EINVAL); - for (n = av_log2(s->minp); (1 << n) < s->nb_taps; n++); - N = FFMIN(n, av_log2(s->maxp)); + if (s->minp > s->maxp) { + s->maxp = s->minp; + } - s->nb_segments = 1; - ret = init_segment(ctx, &s->seg[0], (s->nb_taps + (1 << N) - 1) / (1 << N), 1 << N); - if (ret < 0) - return ret; + left = s->nb_taps; + part_size = 1 << av_log2(s->minp); + max_part_size = 1 << av_log2(s->maxp); + + s->min_part_size = part_size; + + for (i = 0; left > 0; i++) { + int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0); + int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size); + + s->nb_segments = i + 1; + ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size); + if (ret < 0) + return ret; + offset += nb_partitions * part_size; + left -= nb_partitions * part_size; + part_size *= 2; + part_size = FFMIN(part_size, max_part_size); + } ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]); if (ret < 0) @@ -426,7 +480,11 @@ static int convert_coeffs(AVFilterContext *ctx) av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions); av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size); + av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size); av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length); + av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size); + av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size); + av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset); } } @@ -488,7 +546,7 @@ static int activate(AVFilterContext *ctx) return ret; } - ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg[0].part_size, s->seg[0].part_size, &in); + ret = ff_inlink_consume_samples(ctx->inputs[0], s->min_part_size, s->min_part_size, &in); if (ret > 0) ret = fir_frame(s, in, outlink); @@ -505,7 +563,7 @@ static int activate(AVFilterContext *ctx) } } - if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg[0].part_size) { + if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) { ff_filter_set_ready(ctx, 10); return 0; } @@ -624,12 +682,16 @@ static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg) } av_freep(&seg->irdft); + av_freep(&seg->output_offset); av_freep(&seg->part_index); av_frame_free(&seg->block); av_frame_free(&seg->sum); av_frame_free(&seg->buffer); av_frame_free(&seg->coeff); + av_frame_free(&seg->input); + av_frame_free(&seg->output); + seg->input_size = 0; } static av_cold void uninit(AVFilterContext *ctx) @@ -720,11 +782,11 @@ static av_cold int init(AVFilterContext *ctx) static const AVFilterPad afir_inputs[] = { { - .name = "main", - .type = AVMEDIA_TYPE_AUDIO, + .name = "main", + .type = AVMEDIA_TYPE_AUDIO, },{ - .name = "ir", - .type = AVMEDIA_TYPE_AUDIO, + .name = "ir", + .type = AVMEDIA_TYPE_AUDIO, }, { NULL } }; @@ -751,7 +813,7 @@ static const AVOption afir_options[] = { { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF }, { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF }, { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF }, - { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=16}, 16, 32768, AF }, + { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF }, { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF }, { NULL } }; diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h index f71ff34a0e..f9bec54b8c 100644 --- a/libavfilter/af_afir.h +++ b/libavfilter/af_afir.h @@ -37,14 +37,18 @@ typedef struct AudioFIRSegment { int block_size; int fft_length; int coeff_size; - int segment_size; + int input_size; + int input_offset; + int *output_offset; int *part_index; AVFrame *sum; AVFrame *block; AVFrame *buffer; AVFrame *coeff; + AVFrame *input; + AVFrame *output; RDFTContext **rdft, **irdft; } AudioFIRSegment; @@ -80,6 +84,7 @@ typedef struct AudioFIRContext { AVFrame *in[2]; AVFrame *video; + int min_part_size; int64_t pts; AVFloatDSPContext *fdsp;

1 0

avfilter/af_afir: properly split IR into multiple segments
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sat Dec 29 10:57:58 2018 +0100| [300dc45fdca43fe1e308d37be8cded550d7b3a1e] | committer: Paul B Mahol avfilter/af_afir: properly split IR into multiple segments > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=300dc45fdca43fe1e… --- libavfilter/af_afir.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 0d626063b5..3dda875e35 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -387,6 +387,7 @@ static int convert_coeffs(AVFilterContext *ctx) for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; + int toffset = 0; for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++) time[i] = 0; @@ -402,9 +403,8 @@ static int convert_coeffs(AVFilterContext *ctx) for (i = 0; i < seg->nb_partitions; i++) { const float scale = 1.f / seg->part_size; - const int toffset = i * seg->part_size; const int coffset = i * seg->coeff_size; - const int remaining = s->nb_taps - (i * seg->part_size); + const int remaining = s->nb_taps - toffset; const int size = remaining >= seg->part_size ? seg->part_size : remaining; memset(block, 0, sizeof(*block) * seg->fft_length); @@ -420,6 +420,8 @@ static int convert_coeffs(AVFilterContext *ctx) } coeff[coffset + seg->part_size].re = block[1] * scale; coeff[coffset + seg->part_size].im = 0; + + toffset += size; } av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);

1 0

avfilter/af_afir: make part_index values per channel
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sat Dec 29 10:46:25 2018 +0100| [3c3eb4930eb49b3eee76a3ff6f127a78af886345] | committer: Paul B Mahol avfilter/af_afir: make part_index values per channel > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=3c3eb4930eb49b3ee… --- libavfilter/af_afir.c | 18 ++++++++++-------- libavfilter/af_afir.h | 3 ++- 2 files changed, 12 insertions(+), 9 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 8733b76ebe..0d626063b5 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -67,7 +67,7 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) int n, i, j; memset(sum, 0, sizeof(*sum) * seg->fft_length); - block = (float *)seg->block->extended_data[ch] + seg->part_index * seg->block_size; + block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size; memset(block, 0, sizeof(*block) * seg->fft_length); s->fdsp->vector_fmul_scalar(block, src, s->dry_gain, FFALIGN(out->nb_samples, 4)); @@ -77,7 +77,7 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) block[2 * seg->part_size] = block[1]; block[1] = 0; - j = seg->part_index; + j = seg->part_index[ch]; for (i = 0; i < seg->nb_partitions; i++) { const int coffset = i * seg->coeff_size; @@ -106,6 +106,8 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) dst = (float *)seg->buffer->extended_data[ch]; memcpy(dst, sum + seg->part_size, seg->part_size * sizeof(*dst)); + seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions; + return 0; } @@ -125,12 +127,6 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink) s->in[0] = in; ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels); - for (int segment = 0; segment < s->nb_segments; segment++) { - AudioFIRSegment *seg = &s->seg[segment]; - - seg->part_index = (seg->part_index + 1) % seg->nb_partitions; - } - out->pts = s->pts; if (s->pts != AV_NOPTS_VALUE) s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); @@ -300,6 +296,10 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_parti return AVERROR(ENOMEM); } + seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index)); + if (!seg->part_index) + return AVERROR(ENOMEM); + seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length); seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size); seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size); @@ -622,6 +622,8 @@ static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg) } av_freep(&seg->irdft); + av_freep(&seg->part_index); + av_frame_free(&seg->block); av_frame_free(&seg->sum); av_frame_free(&seg->buffer); diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h index 4b35920fd8..f71ff34a0e 100644 --- a/libavfilter/af_afir.h +++ b/libavfilter/af_afir.h @@ -33,13 +33,14 @@ typedef struct AudioFIRSegment { int nb_partitions; - int part_index; int part_size; int block_size; int fft_length; int coeff_size; int segment_size; + int *part_index; + AVFrame *sum; AVFrame *block; AVFrame *buffer;

1 0

avfilter/af_afir: make number of segments extendable
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sat Dec 29 10:39:19 2018 +0100| [31c9d693aa708ffdcbd51ed1d3eab35645a20c92] | committer: Paul B Mahol avfilter/af_afir: make number of segments extendable > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=31c9d693aa708ffdc… --- libavfilter/af_afir.c | 75 +++++++++++++++++++++++++++++++-------------------- libavfilter/af_afir.h | 3 ++- 2 files changed, 48 insertions(+), 30 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index e58d70b15a..8733b76ebe 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -59,7 +59,7 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) { AudioFIRContext *s = ctx->priv; - AudioFIRSegment *seg = &s->seg; + AudioFIRSegment *seg = &s->seg[0]; const float *src = (const float *)s->in[0]->extended_data[ch]; float *sum = (float *)seg->sum->extended_data[ch]; AVFrame *out = arg; @@ -125,7 +125,11 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink) s->in[0] = in; ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels); - s->seg.part_index = (s->seg.part_index + 1) % s->seg.nb_partitions; + for (int segment = 0; segment < s->nb_segments; segment++) { + AudioFIRSegment *seg = &s->seg[segment]; + + seg->part_index = (seg->part_index + 1) % seg->nb_partitions; + } out->pts = s->pts; if (s->pts != AV_NOPTS_VALUE) @@ -319,7 +323,8 @@ static int convert_coeffs(AVFilterContext *ctx) for (n = av_log2(s->minp); (1 << n) < s->nb_taps; n++); N = FFMIN(n, av_log2(s->maxp)); - ret = init_segment(ctx, &s->seg, (s->nb_taps + (1 << N) - 1) / (1 << N), 1 << N); + s->nb_segments = 1; + ret = init_segment(ctx, &s->seg[0], (s->nb_taps + (1 << N) - 1) / (1 << N), 1 << N); if (ret < 0) return ret; @@ -377,43 +382,53 @@ static int convert_coeffs(AVFilterContext *ctx) s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4)); } + av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps); + av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments); + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; - float *block = (float *)s->seg.block->extended_data[ch]; - FFTComplex *coeff = (FFTComplex *)s->seg.coeff->extended_data[ch]; for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++) time[i] = 0; - for (i = 0; i < s->seg.nb_partitions; i++) { - const float scale = 1.f / s->seg.part_size; - const int toffset = i * s->seg.part_size; - const int coffset = i * s->seg.coeff_size; - const int remaining = s->nb_taps - (i * s->seg.part_size); - const int size = remaining >= s->seg.part_size ? s->seg.part_size : remaining; + av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch); + + for (int segment = 0; segment < s->nb_segments; segment++) { + AudioFIRSegment *seg = &s->seg[segment]; + float *block = (float *)seg->block->extended_data[ch]; + FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch]; + + av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment); - memset(block, 0, sizeof(*block) * s->seg.fft_length); - memcpy(block, time + toffset, size * sizeof(*block)); + for (i = 0; i < seg->nb_partitions; i++) { + const float scale = 1.f / seg->part_size; + const int toffset = i * seg->part_size; + const int coffset = i * seg->coeff_size; + const int remaining = s->nb_taps - (i * seg->part_size); + const int size = remaining >= seg->part_size ? seg->part_size : remaining; - av_rdft_calc(s->seg.rdft[0], block); + memset(block, 0, sizeof(*block) * seg->fft_length); + memcpy(block, time + toffset, size * sizeof(*block)); - coeff[coffset].re = block[0] * scale; - coeff[coffset].im = 0; - for (n = 1; n < s->seg.part_size; n++) { - coeff[coffset + n].re = block[2 * n] * scale; - coeff[coffset + n].im = block[2 * n + 1] * scale; + av_rdft_calc(seg->rdft[0], block); + + coeff[coffset].re = block[0] * scale; + coeff[coffset].im = 0; + for (n = 1; n < seg->part_size; n++) { + coeff[coffset + n].re = block[2 * n] * scale; + coeff[coffset + n].im = block[2 * n + 1] * scale; + } + coeff[coffset + seg->part_size].re = block[1] * scale; + coeff[coffset + seg->part_size].im = 0; } - coeff[coffset + s->seg.part_size].re = block[1] * scale; - coeff[coffset + s->seg.part_size].im = 0; + + av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions); + av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size); + av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length); } } av_frame_free(&s->in[1]); - av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps); - av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->seg.nb_partitions); - av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->seg.part_size); - av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->seg.fft_length); - s->have_coeffs = 1; return 0; @@ -471,7 +486,7 @@ static int activate(AVFilterContext *ctx) return ret; } - ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg.part_size, s->seg.part_size, &in); + ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg[0].part_size, s->seg[0].part_size, &in); if (ret > 0) ret = fir_frame(s, in, outlink); @@ -488,7 +503,7 @@ static int activate(AVFilterContext *ctx) } } - if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg.part_size) { + if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg[0].part_size) { ff_filter_set_ready(ctx, 10); return 0; } @@ -617,7 +632,9 @@ static av_cold void uninit(AVFilterContext *ctx) { AudioFIRContext *s = ctx->priv; - uninit_segment(ctx, &s->seg); + for (int i = 0; i < s->nb_segments; i++) { + uninit_segment(ctx, &s->seg[i]); + } av_freep(&s->fdsp); av_frame_free(&s->in[1]); diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h index b70d8e6272..4b35920fd8 100644 --- a/libavfilter/af_afir.h +++ b/libavfilter/af_afir.h @@ -74,7 +74,8 @@ typedef struct AudioFIRContext { int nb_coef_channels; int one2many; - AudioFIRSegment seg; + AudioFIRSegment seg[1024]; + int nb_segments; AVFrame *in[2]; AVFrame *video;

1 0

avfilter/af_afir: use AVFrame for coeff too
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sat Dec 29 10:17:39 2018 +0100| [6095356d5b1ebac0b09d94b8061bdddbdd49daea] | committer: Paul B Mahol avfilter/af_afir: use AVFrame for coeff too > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=6095356d5b1ebac0b… --- libavfilter/af_afir.c | 25 +++++++------------------ libavfilter/af_afir.h | 3 ++- 2 files changed, 9 insertions(+), 19 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 76ef21098a..e58d70b15a 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -82,7 +82,7 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) for (i = 0; i < seg->nb_partitions; i++) { const int coffset = i * seg->coeff_size; const float *block = (const float *)seg->block->extended_data[ch] + j * seg->block_size; - const FFTComplex *coeff = seg->coeff[ch * !s->one2many] + coffset; + const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset; s->fcmul_add(sum, block, (const float *)coeff, seg->part_size); @@ -277,10 +277,9 @@ end: static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_partitions, int part_size) { - seg->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*seg->coeff)); seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft)); seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft)); - if (!seg->coeff || !seg->rdft || !seg->irdft) + if (!seg->rdft || !seg->irdft) return AVERROR(ENOMEM); seg->fft_length = part_size * 4 + 1; @@ -288,12 +287,7 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_parti seg->block_size = FFALIGN(seg->fft_length, 32); seg->coeff_size = FFALIGN(seg->part_size + 1, 32); seg->nb_partitions = nb_partitions; - - for (int ch = 0; ch < ctx->inputs[1]->channels; ch++) { - seg->coeff[ch] = av_calloc(seg->nb_partitions * seg->coeff_size, sizeof(**seg->coeff)); - if (!seg->coeff[ch]) - return AVERROR(ENOMEM); - } + seg->segment_size = part_size * nb_partitions; for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) { seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C); @@ -305,7 +299,8 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_parti seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length); seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size); seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size); - if (!seg->buffer || !seg->sum || !seg->block) + seg->coeff = ff_get_audio_buffer(ctx->inputs[1], seg->nb_partitions * seg->coeff_size * 2); + if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff) return AVERROR(ENOMEM); return 0; @@ -385,7 +380,7 @@ static int convert_coeffs(AVFilterContext *ctx) for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; float *block = (float *)s->seg.block->extended_data[ch]; - FFTComplex *coeff = s->seg.coeff[ch]; + FFTComplex *coeff = (FFTComplex *)s->seg.coeff->extended_data[ch]; for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++) time[i] = 0; @@ -598,13 +593,6 @@ static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg) { AudioFIRContext *s = ctx->priv; - if (seg->coeff) { - for (int ch = 0; ch < s->nb_coef_channels; ch++) { - av_freep(&seg->coeff[ch]); - } - } - av_freep(&seg->coeff); - if (seg->rdft) { for (int ch = 0; ch < s->nb_channels; ch++) { av_rdft_end(seg->rdft[ch]); @@ -622,6 +610,7 @@ static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg) av_frame_free(&seg->block); av_frame_free(&seg->sum); av_frame_free(&seg->buffer); + av_frame_free(&seg->coeff); } static av_cold void uninit(AVFilterContext *ctx) diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h index 39e4375a47..b70d8e6272 100644 --- a/libavfilter/af_afir.h +++ b/libavfilter/af_afir.h @@ -38,13 +38,14 @@ typedef struct AudioFIRSegment { int block_size; int fft_length; int coeff_size; + int segment_size; AVFrame *sum; AVFrame *block; AVFrame *buffer; + AVFrame *coeff; RDFTContext **rdft, **irdft; - FFTComplex **coeff; } AudioFIRSegment; typedef struct AudioFIRContext {

1 0

avfilter/af_afir: use segment pointer
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Fri Dec 28 21:04:50 2018 +0100| [1ba909fabef276212d2abec93a1a3b6aacf0269f] | committer: Paul B Mahol avfilter/af_afir: use segment pointer > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=1ba909fabef276212… --- libavfilter/af_afir.c | 39 ++++++++++++++++++++------------------- 1 file changed, 20 insertions(+), 19 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index dc32b92dc7..76ef21098a 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -59,42 +59,43 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) { AudioFIRContext *s = ctx->priv; + AudioFIRSegment *seg = &s->seg; const float *src = (const float *)s->in[0]->extended_data[ch]; - float *sum = (float *)s->seg.sum->extended_data[ch]; + float *sum = (float *)seg->sum->extended_data[ch]; AVFrame *out = arg; float *block, *dst, *ptr; int n, i, j; - memset(sum, 0, sizeof(*sum) * s->seg.fft_length); - block = (float *)s->seg.block->extended_data[ch] + s->seg.part_index * s->seg.block_size; - memset(block, 0, sizeof(*block) * s->seg.fft_length); + memset(sum, 0, sizeof(*sum) * seg->fft_length); + block = (float *)seg->block->extended_data[ch] + seg->part_index * seg->block_size; + memset(block, 0, sizeof(*block) * seg->fft_length); s->fdsp->vector_fmul_scalar(block, src, s->dry_gain, FFALIGN(out->nb_samples, 4)); emms_c(); - av_rdft_calc(s->seg.rdft[ch], block); - block[2 * s->seg.part_size] = block[1]; + av_rdft_calc(seg->rdft[ch], block); + block[2 * seg->part_size] = block[1]; block[1] = 0; - j = s->seg.part_index; + j = seg->part_index; - for (i = 0; i < s->seg.nb_partitions; i++) { - const int coffset = i * s->seg.coeff_size; - const float *block = (const float *)s->seg.block->extended_data[ch] + j * s->seg.block_size; - const FFTComplex *coeff = s->seg.coeff[ch * !s->one2many] + coffset; + for (i = 0; i < seg->nb_partitions; i++) { + const int coffset = i * seg->coeff_size; + const float *block = (const float *)seg->block->extended_data[ch] + j * seg->block_size; + const FFTComplex *coeff = seg->coeff[ch * !s->one2many] + coffset; - s->fcmul_add(sum, block, (const float *)coeff, s->seg.part_size); + s->fcmul_add(sum, block, (const float *)coeff, seg->part_size); if (j == 0) - j = s->seg.nb_partitions; + j = seg->nb_partitions; j--; } - sum[1] = sum[2 * s->seg.part_size]; - av_rdft_calc(s->seg.irdft[ch], sum); + sum[1] = sum[2 * seg->part_size]; + av_rdft_calc(seg->irdft[ch], sum); - dst = (float *)s->seg.buffer->extended_data[ch]; - for (n = 0; n < s->seg.part_size; n++) { + dst = (float *)seg->buffer->extended_data[ch]; + for (n = 0; n < seg->part_size; n++) { dst[n] += sum[n]; } @@ -102,8 +103,8 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4)); emms_c(); - dst = (float *)s->seg.buffer->extended_data[ch]; - memcpy(dst, sum + s->seg.part_size, s->seg.part_size * sizeof(*dst)); + dst = (float *)seg->buffer->extended_data[ch]; + memcpy(dst, sum + seg->part_size, seg->part_size * sizeof(*dst)); return 0; }

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