On Sat, Feb 08, 2014 at 08:59:33PM -0500, Andrew Kelley wrote:
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -467,6 +467,80 @@ To fix a 5.1 WAV improperly encoded in AAC's native
> channel order
> +
> +@item attacks
> +@item decays
> +Set list of times in seconds for each channel over which the instantaneous
> level
> +of the input signal is averaged to determine its volume. @var{attacks}
> refers to
What is an "instantaneous level"? A "current level"?
> +@item points
> +Set list of points for transfer function, specified in dB relative to maximum
> +possible signal amplitude. Each key points list need to be defined using the
> +following syntax: @code{x0/y0 x1/y1 x2/y2 ....}
needS
> +@item soft-knee
> +Set amount for which the points at where adjacent line segments on the
> transfer
> +function meet will be rounded. Defaults is 0.01.
This sentence is highly confusing. I have trouble extracting a valid
interpretation from it.
> +@item gain
> +Set additional gain in dB to be applied at all points on the transfer
> function
> +and allows easy adjustment of the overall gain. Default is 0.
allow_
> +@item delay
> +Set delay in seconds. Default is 0. The input audio is analysed immediately,
AnalyZed, let's stick to American English.
> +@item
> +Noise-gate for when the noise is at a lower level than the signal:
> +
> +@item
> +Here is another noise-gate, this time for when the noise is at a higher level
noise gate
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -37,6 +37,7 @@ OBJS-$(CONFIG_CHANNELSPLIT_FILTER) +=
> af_channelsplit.o
> OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
> OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
> OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
> +OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
order
> --- /dev/null
> +++ b/libavfilter/af_compand.c
> @@ -0,0 +1,595 @@
> +
> +#include "libavutil/mem.h"
> +#include "libavutil/avassert.h"
> +#include "libavutil/mathematics.h"
> +#include "libavutil/channel_layout.h"
> +#include "libavutil/opt.h"
> +#include "libavutil/common.h"
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +#include "internal.h"
We order includes nowadays.
> + layouts = ff_all_channel_layouts();
> + if (!layouts)
> + return AVERROR(ENOMEM);
> + ff_set_common_channel_layouts(ctx, layouts);
> +
> + formats = ff_make_format_list(sample_fmts);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_set_common_formats(ctx, formats);
> +
> + formats = ff_all_samplerates();
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_set_common_samplerates(ctx, formats);
This looks like you leak memory if the second or third allocation fails
and you return early.
> +/**
> + * Clip a double value into the amin-amax range.
> + * @param a value to clip
> + * @param amin minimum value of the clip range
> + * @param amax maximum value of the clip range
Vertically aligning the parameter descriptions would make this more
readable.
> +static int compand_delay(AVFilterContext *ctx, AVFrame *frame)
> +{
> + CompandContext *s = ctx->priv;
> + AVFilterLink *inlink = ctx->inputs[0];
> + const int channels =
> av_get_channel_layout_nb_channels(inlink->channel_layout);
> + const int nb_samples = frame->nb_samples;
> + int chan, i, av_uninit(dindex), oindex, av_uninit(count);
The av_uninit look fishy. Why are they necessary?
> +static char *av_strtok(char *s, const char *delim, char **saveptr)
What's the problem with just using strtok()?
> +static int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int
> *linesize, int nb_channels,
> + int nb_samples, enum AVSampleFormat
> sample_fmt, int align)
indentation, long lines
Diego
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