diff options
| author | Nicolas George <nicolas.george@normalesup.org> | 2012-02-15 22:15:45 +0400 |
|---|---|---|
| committer | Nicolas George <nicolas.george@normalesup.org> | 2012-03-05 19:57:27 +0400 |
| commit | 3073aadf2ded5f02f2db7ee151a02f592ea24733 (patch) | |
| tree | 9362cec0db3e674132eb99822742fbcd27b154dc /libavdevice/alsa-audio-dec.c | |
| parent | 1007a805a486a1348a0543ac2dd99d823148d25c (diff) | |
timefilter: internally compute feedback factors.
The feedback factors for the timefilter are directly computed from
the expected period. This commit changes the init function to accept
the period itself and compute the feedback factors internally,
rather than having all client code duplicate the formulas.
This commit also actually fixes the formulas: the current code had
sqrt(2*o), but the correct formula, both theoretically and according
to experimental testing, is sqrt(2)*o.
Furthermore, it adds an exponential to feedback factors larger than
1 with large periods.
Diffstat (limited to 'libavdevice/alsa-audio-dec.c')
| -rw-r--r-- | libavdevice/alsa-audio-dec.c | 5 |
1 files changed, 2 insertions, 3 deletions
diff --git a/libavdevice/alsa-audio-dec.c b/libavdevice/alsa-audio-dec.c index f2dd21655b..62bf42d272 100644 --- a/libavdevice/alsa-audio-dec.c +++ b/libavdevice/alsa-audio-dec.c @@ -59,7 +59,6 @@ static av_cold int audio_read_header(AVFormatContext *s1) AVStream *st; int ret; enum CodecID codec_id; - double o; st = avformat_new_stream(s1, NULL); if (!st) { @@ -81,9 +80,9 @@ static av_cold int audio_read_header(AVFormatContext *s1) st->codec->sample_rate = s->sample_rate; st->codec->channels = s->channels; avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ - o = 2 * M_PI * s->period_size / s->sample_rate * 1.5; // bandwidth: 1.5Hz + /* microseconds instead of seconds, MHz instead of Hz */ s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate, - sqrt(2 * o), o * o); + s->period_size, 1.5E-6); if (!s->timefilter) goto fail; |