December 2018 - ffmpeg-cvslog

avfilter/af_afir: introduce uninit_segment() and use it
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Fri Dec 28 18:45:55 2018 +0100| [30bf54b9f3ac4684470a33014665f98641a33422] | committer: Paul B Mahol avfilter/af_afir: introduce uninit_segment() and use it > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=30bf54b9f3ac46844… --- libavfilter/af_afir.c | 41 +++++++++++++++++++++++------------------ 1 file changed, 23 insertions(+), 18 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index ca0c8811a4..dc32b92dc7 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -593,39 +593,44 @@ static int config_output(AVFilterLink *outlink) return 0; } -static av_cold void uninit(AVFilterContext *ctx) +static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg) { AudioFIRContext *s = ctx->priv; - int ch; - if (s->seg.coeff) { - for (ch = 0; ch < s->nb_coef_channels; ch++) { - av_freep(&s->seg.coeff[ch]); + if (seg->coeff) { + for (int ch = 0; ch < s->nb_coef_channels; ch++) { + av_freep(&seg->coeff[ch]); } } - av_freep(&s->seg.coeff); + av_freep(&seg->coeff); - if (s->seg.rdft) { - for (ch = 0; ch < s->nb_channels; ch++) { - av_rdft_end(s->seg.rdft[ch]); + if (seg->rdft) { + for (int ch = 0; ch < s->nb_channels; ch++) { + av_rdft_end(seg->rdft[ch]); } } - av_freep(&s->seg.rdft); + av_freep(&seg->rdft); - if (s->seg.irdft) { - for (ch = 0; ch < s->nb_channels; ch++) { - av_rdft_end(s->seg.irdft[ch]); + if (seg->irdft) { + for (int ch = 0; ch < s->nb_channels; ch++) { + av_rdft_end(seg->irdft[ch]); } } - av_freep(&s->seg.irdft); + av_freep(&seg->irdft); - av_frame_free(&s->in[1]); + av_frame_free(&seg->block); + av_frame_free(&seg->sum); + av_frame_free(&seg->buffer); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AudioFIRContext *s = ctx->priv; - av_frame_free(&s->seg.block); - av_frame_free(&s->seg.sum); - av_frame_free(&s->seg.buffer); + uninit_segment(ctx, &s->seg); av_freep(&s->fdsp); + av_frame_free(&s->in[1]); for (int i = 0; i < ctx->nb_outputs; i++) av_freep(&ctx->output_pads[i].name);

1 0

avfilter/af_afir: fix minp/maxp range and change default value for maxp
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Fri Dec 28 18:33:14 2018 +0100| [8007e8fc67e3a36ad065415356aa057f01cd83b6] | committer: Paul B Mahol avfilter/af_afir: fix minp/maxp range and change default value for maxp > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8007e8fc67e3a36ad… --- doc/filters.texi | 6 +++--- libavfilter/af_afir.c | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/doc/filters.texi b/doc/filters.texi index 65ce25bc18..45d90f6165 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1232,12 +1232,12 @@ Set video stream frame rate. This option is used only when @var{response} is ena @item minp Set minimal partition size used for convolution. Default is @var{16}. -Allowed range is from @var{16} to @var{65536}. +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{65536}. -Allowed range is from @var{16} to @var{65536}. +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. @end table diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 743f7d2658..ca0c8811a4 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -735,8 +735,8 @@ 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, 65536, AF }, - { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=65536}, 16, 65536, AF }, + { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=16}, 16, 32768, AF }, + { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF }, { NULL } };

1 0

avfilter/af_afir: introduce init_segment() and use it
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Fri Dec 28 18:31:36 2018 +0100| [c343e81ffd3008e6df9440bf66e2850e41574099] | committer: Paul B Mahol avfilter/af_afir: introduce init_segment() and use it > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c343e81ffd3008e6d… --- libavfilter/af_afir.c | 69 +++++++++++++++++++++++++++++---------------------- 1 file changed, 39 insertions(+), 30 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 077c64cd6a..743f7d2658 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -274,6 +274,42 @@ end: av_free(mag); } +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) + 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->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); + } + + for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) { + seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C); + seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R); + if (!seg->rdft[ch] || !seg->irdft[ch]) + 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); + if (!seg->buffer || !seg->sum || !seg->block) + return AVERROR(ENOMEM); + + return 0; +} + static int convert_coeffs(AVFilterContext *ctx) { AudioFIRContext *s = ctx->priv; @@ -287,36 +323,9 @@ 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)); - s->seg.coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->seg.coeff)); - s->seg.rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*s->seg.rdft)); - s->seg.irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*s->seg.irdft)); - if (!s->seg.coeff || !s->seg.rdft || !s->seg.irdft) - return AVERROR(ENOMEM); - - s->seg.fft_length = (1 << (N + 1)) + 1; - s->seg.part_size = 1 << (N - 1); - s->seg.block_size = FFALIGN(s->seg.fft_length, 32); - s->seg.coeff_size = FFALIGN(s->seg.part_size + 1, 32); - s->seg.nb_partitions = (s->nb_taps + s->seg.part_size - 1) / s->seg.part_size; - - for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { - s->seg.coeff[ch] = av_calloc(s->seg.nb_partitions * s->seg.coeff_size, sizeof(**s->seg.coeff)); - if (!s->seg.coeff[ch]) - return AVERROR(ENOMEM); - } - - for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { - s->seg.rdft[ch] = av_rdft_init(N, DFT_R2C); - s->seg.irdft[ch] = av_rdft_init(N, IDFT_C2R); - if (!s->seg.rdft[ch] || !s->seg.irdft[ch]) - return AVERROR(ENOMEM); - } - - s->seg.sum = ff_get_audio_buffer(ctx->inputs[0], s->seg.fft_length); - s->seg.block = ff_get_audio_buffer(ctx->inputs[0], s->seg.nb_partitions * s->seg.block_size); - s->seg.buffer = ff_get_audio_buffer(ctx->inputs[0], s->seg.part_size); - if (!s->seg.buffer || !s->seg.sum || !s->seg.block) - return AVERROR(ENOMEM); + ret = init_segment(ctx, &s->seg, (s->nb_taps + (1 << N) - 1) / (1 << N), 1 << N); + if (ret < 0) + return ret; ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]); if (ret < 0)

1 0

avfilter/af_afir: move allocation stuff where it belongs
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Fri Dec 28 18:07:13 2018 +0100| [e57053417a090bbdc5cad73d7fa88f72efaf854c] | committer: Paul B Mahol avfilter/af_afir: move allocation stuff where it belongs > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e57053417a090bbdc… --- libavfilter/af_afir.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 89d7ed6c85..077c64cd6a 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -286,6 +286,13 @@ 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)); + + s->seg.coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->seg.coeff)); + s->seg.rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*s->seg.rdft)); + s->seg.irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*s->seg.irdft)); + if (!s->seg.coeff || !s->seg.rdft || !s->seg.irdft) + return AVERROR(ENOMEM); + s->seg.fft_length = (1 << (N + 1)) + 1; s->seg.part_size = 1 << (N - 1); s->seg.block_size = FFALIGN(s->seg.fft_length, 32); @@ -570,12 +577,6 @@ static int config_output(AVFilterLink *outlink) outlink->channel_layout = ctx->inputs[0]->channel_layout; outlink->channels = ctx->inputs[0]->channels; - s->seg.coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->seg.coeff)); - s->seg.rdft = av_calloc(outlink->channels, sizeof(*s->seg.rdft)); - s->seg.irdft = av_calloc(outlink->channels, sizeof(*s->seg.irdft)); - if (!s->seg.coeff || !s->seg.rdft || !s->seg.irdft) - return AVERROR(ENOMEM); - s->nb_channels = outlink->channels; s->nb_coef_channels = ctx->inputs[1]->channels; s->pts = AV_NOPTS_VALUE;

1 0

avfilter/af_afir: introduce AudioFIRSegment structure and use it
by Paul B Mahol 30 Dec '18

30 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Fri Dec 28 17:57:24 2018 +0100| [7312e027d6d3b8aa8220cceacd29c85c87292906] | committer: Paul B Mahol avfilter/af_afir: introduce AudioFIRSegment structure and use it > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7312e027d6d3b8aa8… --- libavfilter/af_afir.c | 161 ++++++++++++++++++++++---------------------------- libavfilter/af_afir.h | 28 +++++---- 2 files changed, 86 insertions(+), 103 deletions(-) diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 65737e05b1..89d7ed6c85 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -60,41 +60,41 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) { AudioFIRContext *s = ctx->priv; const float *src = (const float *)s->in[0]->extended_data[ch]; - float *sum = s->sum[ch]; + float *sum = (float *)s->seg.sum->extended_data[ch]; AVFrame *out = arg; float *block, *dst, *ptr; int n, i, j; - memset(sum, 0, sizeof(*sum) * s->fft_length); - block = s->block[ch] + s->part_index * s->block_size; - memset(block, 0, sizeof(*block) * s->fft_length); + 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); s->fdsp->vector_fmul_scalar(block, src, s->dry_gain, FFALIGN(out->nb_samples, 4)); emms_c(); - av_rdft_calc(s->rdft[ch], block); - block[2 * s->part_size] = block[1]; + av_rdft_calc(s->seg.rdft[ch], block); + block[2 * s->seg.part_size] = block[1]; block[1] = 0; - j = s->part_index; + j = s->seg.part_index; - for (i = 0; i < s->nb_partitions; i++) { - const int coffset = i * s->coeff_size; - const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset; + 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; - block = s->block[ch] + j * s->block_size; - s->fcmul_add(sum, block, (const float *)coeff, s->part_size); + s->fcmul_add(sum, block, (const float *)coeff, s->seg.part_size); if (j == 0) - j = s->nb_partitions; + j = s->seg.nb_partitions; j--; } - sum[1] = sum[2 * s->part_size]; - av_rdft_calc(s->irdft[ch], sum); + sum[1] = sum[2 * s->seg.part_size]; + av_rdft_calc(s->seg.irdft[ch], sum); - dst = (float *)s->buffer->extended_data[ch]; - for (n = 0; n < s->part_size; n++) { + dst = (float *)s->seg.buffer->extended_data[ch]; + for (n = 0; n < s->seg.part_size; n++) { dst[n] += sum[n]; } @@ -102,8 +102,8 @@ 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->buffer->extended_data[ch]; - memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst)); + dst = (float *)s->seg.buffer->extended_data[ch]; + memcpy(dst, sum + s->seg.part_size, s->seg.part_size * sizeof(*dst)); return 0; } @@ -124,7 +124,7 @@ 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->part_index = (s->part_index + 1) % s->nb_partitions; + s->seg.part_index = (s->seg.part_index + 1) % s->seg.nb_partitions; out->pts = s->pts; if (s->pts != AV_NOPTS_VALUE) @@ -286,39 +286,29 @@ 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)); - s->fft_length = (1 << (N + 1)) + 1; - s->part_size = 1 << (N - 1); - s->block_size = FFALIGN(s->fft_length, 32); - s->coeff_size = FFALIGN(s->part_size + 1, 32); - s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size; - - for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { - s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum)); - if (!s->sum[ch]) - return AVERROR(ENOMEM); - } + s->seg.fft_length = (1 << (N + 1)) + 1; + s->seg.part_size = 1 << (N - 1); + s->seg.block_size = FFALIGN(s->seg.fft_length, 32); + s->seg.coeff_size = FFALIGN(s->seg.part_size + 1, 32); + s->seg.nb_partitions = (s->nb_taps + s->seg.part_size - 1) / s->seg.part_size; for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { - s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff)); - if (!s->coeff[ch]) + s->seg.coeff[ch] = av_calloc(s->seg.nb_partitions * s->seg.coeff_size, sizeof(**s->seg.coeff)); + if (!s->seg.coeff[ch]) return AVERROR(ENOMEM); } for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { - s->block[ch] = av_calloc(s->nb_partitions * s->block_size, sizeof(**s->block)); - if (!s->block[ch]) + s->seg.rdft[ch] = av_rdft_init(N, DFT_R2C); + s->seg.irdft[ch] = av_rdft_init(N, IDFT_C2R); + if (!s->seg.rdft[ch] || !s->seg.irdft[ch]) return AVERROR(ENOMEM); } - for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { - s->rdft[ch] = av_rdft_init(N, DFT_R2C); - s->irdft[ch] = av_rdft_init(N, IDFT_C2R); - if (!s->rdft[ch] || !s->irdft[ch]) - return AVERROR(ENOMEM); - } - - s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size); - if (!s->buffer) + s->seg.sum = ff_get_audio_buffer(ctx->inputs[0], s->seg.fft_length); + s->seg.block = ff_get_audio_buffer(ctx->inputs[0], s->seg.nb_partitions * s->seg.block_size); + s->seg.buffer = ff_get_audio_buffer(ctx->inputs[0], s->seg.part_size); + if (!s->seg.buffer || !s->seg.sum || !s->seg.block) return AVERROR(ENOMEM); ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]); @@ -377,40 +367,40 @@ 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 = s->block[ch]; - FFTComplex *coeff = s->coeff[ch]; + float *block = (float *)s->seg.block->extended_data[ch]; + FFTComplex *coeff = s->seg.coeff[ch]; for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++) time[i] = 0; - for (i = 0; i < s->nb_partitions; i++) { - const float scale = 1.f / s->part_size; - const int toffset = i * s->part_size; - const int coffset = i * s->coeff_size; - const int remaining = s->nb_taps - (i * s->part_size); - const int size = remaining >= s->part_size ? s->part_size : remaining; + 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; - memset(block, 0, sizeof(*block) * s->fft_length); + memset(block, 0, sizeof(*block) * s->seg.fft_length); memcpy(block, time + toffset, size * sizeof(*block)); - av_rdft_calc(s->rdft[0], block); + av_rdft_calc(s->seg.rdft[0], block); coeff[coffset].re = block[0] * scale; coeff[coffset].im = 0; - for (n = 1; n < s->part_size; n++) { + 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; } - coeff[coffset + s->part_size].re = block[1] * scale; - coeff[coffset + s->part_size].im = 0; + coeff[coffset + s->seg.part_size].re = block[1] * scale; + coeff[coffset + s->seg.part_size].im = 0; } } 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->nb_partitions); - av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size); - av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->fft_length); + 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; @@ -469,7 +459,7 @@ static int activate(AVFilterContext *ctx) return ret; } - ret = ff_inlink_consume_samples(ctx->inputs[0], s->part_size, s->part_size, &in); + ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg.part_size, s->seg.part_size, &in); if (ret > 0) ret = fir_frame(s, in, outlink); @@ -486,7 +476,7 @@ static int activate(AVFilterContext *ctx) } } - if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->part_size) { + if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg.part_size) { ff_filter_set_ready(ctx, 10); return 0; } @@ -580,12 +570,10 @@ static int config_output(AVFilterLink *outlink) outlink->channel_layout = ctx->inputs[0]->channel_layout; outlink->channels = ctx->inputs[0]->channels; - s->sum = av_calloc(outlink->channels, sizeof(*s->sum)); - s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff)); - s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block)); - s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft)); - s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft)); - if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft) + s->seg.coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->seg.coeff)); + s->seg.rdft = av_calloc(outlink->channels, sizeof(*s->seg.rdft)); + s->seg.irdft = av_calloc(outlink->channels, sizeof(*s->seg.irdft)); + if (!s->seg.coeff || !s->seg.rdft || !s->seg.irdft) return AVERROR(ENOMEM); s->nb_channels = outlink->channels; @@ -600,43 +588,32 @@ static av_cold void uninit(AVFilterContext *ctx) AudioFIRContext *s = ctx->priv; int ch; - if (s->sum) { - for (ch = 0; ch < s->nb_channels; ch++) { - av_freep(&s->sum[ch]); - } - } - av_freep(&s->sum); - - if (s->coeff) { + if (s->seg.coeff) { for (ch = 0; ch < s->nb_coef_channels; ch++) { - av_freep(&s->coeff[ch]); - } - } - av_freep(&s->coeff); - - if (s->block) { - for (ch = 0; ch < s->nb_channels; ch++) { - av_freep(&s->block[ch]); + av_freep(&s->seg.coeff[ch]); } } - av_freep(&s->block); + av_freep(&s->seg.coeff); - if (s->rdft) { + if (s->seg.rdft) { for (ch = 0; ch < s->nb_channels; ch++) { - av_rdft_end(s->rdft[ch]); + av_rdft_end(s->seg.rdft[ch]); } } - av_freep(&s->rdft); + av_freep(&s->seg.rdft); - if (s->irdft) { + if (s->seg.irdft) { for (ch = 0; ch < s->nb_channels; ch++) { - av_rdft_end(s->irdft[ch]); + av_rdft_end(s->seg.irdft[ch]); } } - av_freep(&s->irdft); + av_freep(&s->seg.irdft); av_frame_free(&s->in[1]); - av_frame_free(&s->buffer); + + av_frame_free(&s->seg.block); + av_frame_free(&s->seg.sum); + av_frame_free(&s->seg.buffer); av_freep(&s->fdsp); diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h index 9186e2cfdc..39e4375a47 100644 --- a/libavfilter/af_afir.h +++ b/libavfilter/af_afir.h @@ -31,6 +31,22 @@ #include "formats.h" #include "internal.h" +typedef struct AudioFIRSegment { + int nb_partitions; + int part_index; + int part_size; + int block_size; + int fft_length; + int coeff_size; + + AVFrame *sum; + AVFrame *block; + AVFrame *buffer; + + RDFTContext **rdft, **irdft; + FFTComplex **coeff; +} AudioFIRSegment; + typedef struct AudioFIRContext { const AVClass *class; @@ -53,23 +69,13 @@ typedef struct AudioFIRContext { int eof_coeffs; int have_coeffs; int nb_taps; - int part_size; - int part_index; - int coeff_size; - int block_size; - int nb_partitions; int nb_channels; - int fft_length; int nb_coef_channels; int one2many; - RDFTContext **rdft, **irdft; - float **sum; - float **block; - FFTComplex **coeff; + AudioFIRSegment seg; AVFrame *in[2]; - AVFrame *buffer; AVFrame *video; int64_t pts;

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tools: add target_dec_fate.sh
by Michael Niedermayer 28 Dec '18

28 Dec '18

ffmpeg | branch: master | Michael Niedermayer <michael(a)niedermayer.cc> | Mon Sep 10 01:45:04 2018 +0200| [fccba32b4c4f16682b2c34a4c84449a84f91d4d9] | committer: Michael Niedermayer tools: add target_dec_fate.sh Script to download and test ossfuzz testcases This also includes a list of such testcases. I intend to subsequently fill this list with the cases we have fixed in the past Signed-off-by: Michael Niedermayer <michael(a)niedermayer.cc> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=fccba32b4c4f16682… --- tools/target_dec_fate.list | 6 ++++ tools/target_dec_fate.sh | 78 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 84 insertions(+) diff --git a/tools/target_dec_fate.list b/tools/target_dec_fate.list new file mode 100644 index 0000000000..e0509aaa64 --- /dev/null +++ b/tools/target_dec_fate.list @@ -0,0 +1,6 @@ +544/clusterfuzz-testcase-5936536407244800.f8bd9b24_8ba77916_70c2c7be_3df6a2ea_96cd9f14 target_dec_interplay_video_fuzzer +724/clusterfuzz-testcase-6738249571631104 target_dec_pictor_fuzzer +1214/clusterfuzz-testcase-minimized-6130606599569408 target_dec_h264_fuzzer +5264/clusterfuzz-testcase-minimized-4621956621008896 target_dec_indeo5_fuzzer +7279/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G2M_fuzzer-5977332473921536 target_dec_g2m_fuzzer +10053/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PROSUMER_fuzzer-5636993883570176 target_dec_prosumer_fuzzer diff --git a/tools/target_dec_fate.sh b/tools/target_dec_fate.sh new file mode 100755 index 0000000000..1fdfdcaaea --- /dev/null +++ b/tools/target_dec_fate.sh @@ -0,0 +1,78 @@ +#!/bin/sh +# +# * Copyright (C) 2018 Michael Niedermayer (michaelni(a)gmx.at) +# * +# * This file is part of FFmpeg. +# * +# * FFmpeg is free software; you can redistribute it and/or modify +# * it under the terms of the GNU General Public License as published by +# * the Free Software Foundation; either version 2 of the License, or +# * (at your option) any later version. +# * +# * FFmpeg is distributed in the hope that it will be useful, +# * but WITHOUT ANY WARRANTY; without even the implied warranty of +# * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +# * GNU General Public License for more details. +# * +# * You should have received a copy of the GNU General Public License +# * along with FFmpeg; if not, write to the Free Software +# * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + +set -e + +LC_ALL=C +export LC_ALL + +LIST=target_dec_fate.list + +show_help(){ + cat <<EOF +Usage: ./target_dec_fate.sh <directory> [<test to run>] + +directory the directory into which sample files will be downloaded +test to run the number of the issue to test +Note, some test samples may not yet be available to the public, also this +script will not download samples which are already in the directory. So you +may want to preserve its content between runs. +EOF + exit 0 +} + +test -z "$1" && show_help +test ! -d "$1" && echo $1 is not an accessable directory && show_help +test ! -f target_dec_fate.sh && echo $0 Must be run from its location && show_help +grep 'CONFIG_OSSFUZZ 0' ../config.h && echo not configured for ossfuzz && show_help + +#Download testcases +while read -r LINE; do + ISSUE_NUM=`echo $LINE | sed 's#/.*##'` + FILE_ID=`echo $LINE | sed 's#.*/clusterfuzz-testcase[a-zA-Z0-9_-]*-$[0-9]*$.*#\1#'` + FILE=`echo $LINE | sed 's# .*##'` + if test -f "$1/$FILE" ; then + echo exists $FILE + else + echo downloading $FILE + mkdir -p "$1/$ISSUE_NUM" + wget -O "$1/$FILE" "https://oss-fuzz.com/download?testcase_id=$FILE_ID" || rm "$1/$FILE" + fi +done < "$LIST" + +#Find which fuzzers we need to build +TOOLS= +while read -r LINE; do + TOOL_ID=`echo $LINE | sed 's#[^ ]* ##'` + TOOLS="$TOOLS tools/$TOOL_ID" +done < "$LIST" + +cd .. +#Build fuzzers +make -j4 $TOOLS + +#Run testcases +while read -r LINE; do + TOOL_ID=`echo $LINE | sed 's#[^ ]* ##'` + FILE=`echo $LINE | sed 's# .*##'` + tools/$TOOL_ID $1/$FILE +done < "tools/$LIST" + +echo OK

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Makefile: Allow "make clean" to delete compat/atomics/pthread/stdatomic.o
by Carl Eugen Hoyos 28 Dec '18

28 Dec '18

ffmpeg | branch: master | Carl Eugen Hoyos <ceffmpeg(a)gmail.com> | Fri Dec 28 14:04:10 2018 +0100| [8d422e7fef9744e0b582a34217dab9352a8b2a8a] | committer: Carl Eugen Hoyos Makefile: Allow "make clean" to delete compat/atomics/pthread/stdatomic.o Reported-by: Eric Thomas > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8d422e7fef9744e0b… --- Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Makefile b/Makefile index 4bf1dfedcf..59b26c65c9 100644 --- a/Makefile +++ b/Makefile @@ -135,7 +135,7 @@ uninstall-data: clean:: $(RM) $(CLEANSUFFIXES) - $(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES)) + $(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES)) $(addprefix compat/*/*/,$(CLEANSUFFIXES)) $(RM) -r coverage-html $(RM) -rf coverage.info coverage.info.in lcov

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avformat/wvdec: detect and error out on WavPack DSD files
by David Bryant 28 Dec '18

28 Dec '18

ffmpeg | branch: master | David Bryant <david(a)wavpack.com> | Tue Nov 20 21:00:47 2018 -0800| [db109373d87b1fa5fe9f3d027d1bb752f725b74a] | committer: Paul B Mahol avformat/wvdec: detect and error out on WavPack DSD files Not currently supported. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=db109373d87b1fa5f… --- libavformat/wvdec.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/libavformat/wvdec.c b/libavformat/wvdec.c index b6932e65af..ddcb9002cb 100644 --- a/libavformat/wvdec.c +++ b/libavformat/wvdec.c @@ -40,6 +40,7 @@ enum WV_FLAGS { WV_HBAL = 0x0400, WV_MCINIT = 0x0800, WV_MCEND = 0x1000, + WV_DSD = 0x80000000, }; static const int wv_rates[16] = { @@ -97,6 +98,11 @@ static int wv_read_block_header(AVFormatContext *ctx, AVIOContext *pb) return ret; } + if (wc->header.flags & WV_DSD) { + avpriv_report_missing_feature(ctx, "WV DSD"); + return AVERROR_PATCHWELCOME; + } + if (wc->header.version < 0x402 || wc->header.version > 0x410) { avpriv_report_missing_feature(ctx, "WV version 0x%03X", wc->header.version);

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avcodec/wavpack: fix decoding of files with many channels
by Paul B Mahol 28 Dec '18

28 Dec '18

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Fri Dec 28 12:16:39 2018 +0100| [cfa7709d05b1ad6cdafe3d20c91d38d7db503b3e] | committer: Paul B Mahol avcodec/wavpack: fix decoding of files with many channels Fixes decoding of Run_The_Race_-_3rd_Order_Ambisonic_SN3D.wv > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=cfa7709d05b1ad6cd… --- libavcodec/wavpack.c | 18 ++++++++++++++---- libavformat/wvdec.c | 9 ++++++++- 2 files changed, 22 insertions(+), 5 deletions(-) diff --git a/libavcodec/wavpack.c b/libavcodec/wavpack.c index 8306ec020f..d0242809fe 100644 --- a/libavcodec/wavpack.c +++ b/libavcodec/wavpack.c @@ -940,13 +940,23 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no, case 3: chmask = bytestream2_get_le32(&gb); break; - case 5: + case 4: size = bytestream2_get_byte(&gb); - if (avctx->channels != size) + chan |= (bytestream2_get_byte(&gb) & 0xF) << 8; + chan += 1; + if (avctx->channels != chan) av_log(avctx, AV_LOG_WARNING, "%i channels signalled" - " instead of %i.\n", size, avctx->channels); + " instead of %i.\n", chan, avctx->channels); + chmask = bytestream2_get_le24(&gb); + break; + case 5: + size = bytestream2_get_byte(&gb); chan |= (bytestream2_get_byte(&gb) & 0xF) << 8; - chmask = bytestream2_get_le16(&gb); + chan += 1; + if (avctx->channels != chan) + av_log(avctx, AV_LOG_WARNING, "%i channels signalled" + " instead of %i.\n", chan, avctx->channels); + chmask = bytestream2_get_le32(&gb); break; default: av_log(avctx, AV_LOG_ERROR, "Invalid channel info size %d\n", diff --git a/libavformat/wvdec.c b/libavformat/wvdec.c index 82526563ec..b6932e65af 100644 --- a/libavformat/wvdec.c +++ b/libavformat/wvdec.c @@ -153,11 +153,18 @@ static int wv_read_block_header(AVFormatContext *ctx, AVIOContext *pb) case 3: chmask = avio_rl32(pb); break; - case 5: + case 4: avio_skip(pb, 1); chan |= (avio_r8(pb) & 0xF) << 8; + chan += 1; chmask = avio_rl24(pb); break; + case 5: + avio_skip(pb, 1); + chan |= (avio_r8(pb) & 0xF) << 8; + chan += 1; + chmask = avio_rl32(pb); + break; default: av_log(ctx, AV_LOG_ERROR, "Invalid channel info size %d\n", size);

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configure: Make sure libpostproc can be found if -rpath-link doesn't work.
by Carl Eugen Hoyos 27 Dec '18

27 Dec '18

ffmpeg | branch: master | Carl Eugen Hoyos <ceffmpeg(a)gmail.com> | Sat Dec 22 00:04:43 2018 +0100| [1a6cca19898c5e65d0e3cb9b9b293146b8268dcc] | committer: Carl Eugen Hoyos configure: Make sure libpostproc can be found if -rpath-link doesn't work. Solaris ld takes "-rpath-link=libpostproc" as indication to search in "-link=libpostproc". > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=1a6cca19898c5e65d… --- configure | 2 +- tools/target_dec_fuzzer.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/configure b/configure index 2205751402..04d3417200 100755 --- a/configure +++ b/configure @@ -6535,7 +6535,7 @@ EOF # add some linker flags check_ldflags -Wl,--warn-common -check_ldflags -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample +check_ldflags -Wl,-rpath-link=:libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample enabled rpath && add_ldexeflags -Wl,-rpath,$libdir && add_ldsoflags -Wl,-rpath,$libdir test_ldflags -Wl,-Bsymbolic && append SHFLAGS -Wl,-Bsymbolic diff --git a/tools/target_dec_fuzzer.c b/tools/target_dec_fuzzer.c index a94d5e6a92..2a6d525b73 100644 --- a/tools/target_dec_fuzzer.c +++ b/tools/target_dec_fuzzer.c @@ -30,7 +30,7 @@ * build the fuzz target. Choose the value of FFMPEG_CODEC (e.g. AV_CODEC_ID_DVD_SUBTITLE) and choose one of FUZZ_FFMPEG_VIDEO, FUZZ_FFMPEG_AUDIO, FUZZ_FFMPEG_SUBTITLE. - clang -fsanitize=address -fsanitize-coverage=trace-pc-guard,trace-cmp tools/target_dec_fuzzer.c -o target_dec_fuzzer -I. -DFFMPEG_CODEC=AV_CODEC_ID_MPEG1VIDEO -DFUZZ_FFMPEG_VIDEO ../../libfuzzer/libFuzzer.a -Llibavcodec -Llibavdevice -Llibavfilter -Llibavformat -Llibavresample -Llibavutil -Llibpostproc -Llibswscale -Llibswresample -Wl,--as-needed -Wl,-z,noexecstack -Wl,--warn-common -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample -lavdevice -lavfilter -lavformat -lavcodec -lswresample -lswscale -lavutil -ldl -lxcb -lxcb-shm -lxcb -lxcb-xfixes -lxcb -lxcb-shape -lxcb -lX11 -lasound -lm -lbz2 -lz -pthread + clang -fsanitize=address -fsanitize-coverage=trace-pc-guard,trace-cmp tools/target_dec_fuzzer.c -o target_dec_fuzzer -I. -DFFMPEG_CODEC=AV_CODEC_ID_MPEG1VIDEO -DFUZZ_FFMPEG_VIDEO ../../libfuzzer/libFuzzer.a -Llibavcodec -Llibavdevice -Llibavfilter -Llibavformat -Llibavresample -Llibavutil -Llibpostproc -Llibswscale -Llibswresample -Wl,--as-needed -Wl,-z,noexecstack -Wl,--warn-common -Wl,-rpath-link=:libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample -lavdevice -lavfilter -lavformat -lavcodec -lswresample -lswscale -lavutil -ldl -lxcb -lxcb-shm -lxcb -lxcb-xfixes -lxcb -lxcb-shape -lxcb -lX11 -lasound -lm -lbz2 -lz -pthread * create a corpus directory and put some samples there (empty dir is ok too): mkdir CORPUS && cp some-files CORPUS

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