On Fri, Feb 14, 2014 at 02:56:46AM -0500, Andrew Kelley wrote:
> This patch adds the `compand` audio filter from ffmpeg master branch
> (currently at 7f0f47b3df) adapted to work with libav.
>
> The following changes are made:
>
> * use float instead of double
> * use strtok_r instead of av_strtok
> * remove asserts
> * use AVFrame instead of manually allocating memory
> ---
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> --- 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
> +increase of volume and @var{decays} refers to decrease of volume. For most
> +situations, the attack time (response to the audio getting louder) should be
> +shorter than the decay time because the human ear is more sensitive to sudden
> +loud audio than sudden soft audio. Typical value for attack is 0.3 seconds
> and
> +for decay 0.8 seconds.
In the last sentence you suddenly write attack/decay instead of attackS/decayS.
Also, either write "A typical value" or "Typical values".
> +@item points
> +Set list of points for transfer function, specified in dB relative to maximum
> +possible signal amplitude.
for the, to the
> Each key points list need to be defined using the following syntax:
needS
> +The input values must be in strictly increasing order but the transfer
> function
> +does not have to be monotonically rising. The point @code{0/0} is assumed but
> +may be overridden (by @code{0/out-dBn}). Typical values for the transfer
> +function are @code{-70/-70 -60/-20}.
> +
> +@item soft-knee
> +Set amount for which the points at where adjacent line segments on the
> transfer
> +function meet will be rounded.
This sentence makes me stumble, maybe
The points at where adjacent line segments on the transfer function
meet will be rounded by this amount.
> Defaults is 0.01.
Default_
> +@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.
I'd split the sentence, as I stumbled again. Maybe
Set additional gain in dB to be applied at all points on the transfer
function. This allows easy adjustment of the overall gain. Default is 0.
> +@item delay
> +Set delay in seconds. Default is 0. The input audio is analysed immediately,
analyZed
The rest of the documentation is written in American English.
> --- 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
This list is sorted alphabetically, please keep it sorted.
> --- /dev/null
> +++ b/libavfilter/af_compand.c
> @@ -0,0 +1,548 @@
> + * You should have received a copy of the GNU Lesser General Public
> + * License along with Libav; if not, write to the Free Software
> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301
> USA
> + *
> + */
nit: stray empty line
> +static const AVOption compand_options[] = {
> + { "attacks", "set time over which increase of volume is determined",
> OFFSET(attacks), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
> + { "decays", "set time over which decrease of volume is determined",
> OFFSET(decays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
> + { "points", "set points of transfer function", OFFSET(points),
> AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
> + { "soft-knee", "set soft-knee", OFFSET(curve_dB), AV_OPT_TYPE_FLOAT,
> {.dbl=0.01}, 0.01, 900, A },
> + { "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_FLOAT,
> {.dbl=0}, -900, 900, A },
> + { "volume", "set initial volume", OFFSET(initial_volume),
> AV_OPT_TYPE_FLOAT, {.dbl=0}, -900, 0, A },
> + { "delay", "set delay for samples before sending them to volume
> adjuster", OFFSET(delay), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 20, A },
> + { NULL }
> +};
spaces inside {}, spaces around =
> + s->pts += av_rescale_q(nb_samples - i, (AVRational){1,
> inlink->sample_rate}, inlink->time_base);
> + s->pts += av_rescale_q(frame->nb_samples, (AVRational){1,
> outlink->sample_rate}, outlink->time_base);
same
> + in1 = cx - L(3).x;
> + out1 = cy - L(3).y;
> + in2 = L(2).x - L(3).x;
> + out2 = L(2).y - L(3).y;
align the =
> + L(3).a = (out2 / in2 - out1 / in1) / (in2-in1);
spaces around -
> + s->delay_frame->format = outlink->format;
> + s->delay_frame->nb_samples = s->delay_samples;
> + s->delay_frame->channel_layout = outlink->channel_layout;
align the =
Diego
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