[FFmpeg-cvslog] lavfi: simplify signature for avfilter_get_audio_buffer() and friends
Stefano Sabatini
git at videolan.org
Thu May 10 23:33:12 CEST 2012
ffmpeg | branch: master | Stefano Sabatini <stefano.sabatini-lala at poste.it> | Tue Aug 30 23:22:29 2011 +0200| [7ef0adcc2e800cb1357d4d5d7ce878c0f9a36c01] | committer: Anton Khirnov
lavfi: simplify signature for avfilter_get_audio_buffer() and friends
The additional parameters are just complicating the function interface.
Assume that a requested samples buffer will *always* have the format
specified in the requested link.
This breaks audio filtering API and ABI in theory, but since it's
unusable right now this shouldn't be a problem.
Signed-off-by: Anton Khirnov <anton at khirnov.net>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7ef0adcc2e800cb1357d4d5d7ce878c0f9a36c01
---
libavfilter/avfilter.c | 11 ++++-------
libavfilter/avfilter.h | 16 ++++------------
libavfilter/defaults.c | 24 ++++++++++--------------
3 files changed, 18 insertions(+), 33 deletions(-)
diff --git a/libavfilter/avfilter.c b/libavfilter/avfilter.c
index 9f81be7..e301ddb 100644
--- a/libavfilter/avfilter.c
+++ b/libavfilter/avfilter.c
@@ -367,16 +367,15 @@ fail:
}
AVFilterBufferRef *avfilter_get_audio_buffer(AVFilterLink *link, int perms,
- enum AVSampleFormat sample_fmt, int nb_samples,
- uint64_t channel_layout)
+ int nb_samples)
{
AVFilterBufferRef *ret = NULL;
if (link->dstpad->get_audio_buffer)
- ret = link->dstpad->get_audio_buffer(link, perms, sample_fmt, nb_samples, channel_layout);
+ ret = link->dstpad->get_audio_buffer(link, perms, nb_samples);
if (!ret)
- ret = avfilter_default_get_audio_buffer(link, perms, sample_fmt, nb_samples, channel_layout);
+ ret = avfilter_default_get_audio_buffer(link, perms, nb_samples);
if (ret)
ret->type = AVMEDIA_TYPE_AUDIO;
@@ -593,9 +592,7 @@ void avfilter_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
link->cur_buf = avfilter_default_get_audio_buffer(link, dst->min_perms,
- samplesref->format,
- samplesref->audio->nb_samples,
- samplesref->audio->channel_layout);
+ samplesref->audio->nb_samples);
link->cur_buf->pts = samplesref->pts;
link->cur_buf->audio->sample_rate = samplesref->audio->sample_rate;
diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h
index 19ac057..cf95b4b 100644
--- a/libavfilter/avfilter.h
+++ b/libavfilter/avfilter.h
@@ -387,8 +387,7 @@ struct AVFilterPad {
* Input audio pads only.
*/
AVFilterBufferRef *(*get_audio_buffer)(AVFilterLink *link, int perms,
- enum AVSampleFormat sample_fmt, int nb_samples,
- uint64_t channel_layout);
+ int nb_samples);
/**
* Callback called after the slices of a frame are completely sent. If
@@ -473,9 +472,7 @@ AVFilterBufferRef *avfilter_default_get_video_buffer(AVFilterLink *link,
/** default handler for get_audio_buffer() for audio inputs */
AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int perms,
- enum AVSampleFormat sample_fmt,
- int nb_samples,
- uint64_t channel_layout);
+ int nb_samples);
/**
* A helper for query_formats() which sets all links to the same list of
@@ -505,8 +502,7 @@ AVFilterBufferRef *avfilter_null_get_video_buffer(AVFilterLink *link,
/** get_audio_buffer() handler for filters which simply pass audio along */
AVFilterBufferRef *avfilter_null_get_audio_buffer(AVFilterLink *link, int perms,
- enum AVSampleFormat sample_fmt, int nb_samples,
- uint64_t channel_layout);
+ int nb_samples);
/**
* Filter definition. This defines the pads a filter contains, and all the
@@ -689,16 +685,12 @@ avfilter_get_video_buffer_ref_from_arrays(uint8_t *data[4], int linesize[4], int
* @param link the output link to the filter from which the buffer will
* be requested
* @param perms the required access permissions
- * @param sample_fmt the format of each sample in the buffer to allocate
* @param nb_samples the number of samples per channel
- * @param channel_layout the number and type of channels per sample in the buffer to allocate
- * @param planar audio data layout - planar or packed
* @return A reference to the samples. This must be unreferenced with
* avfilter_unref_buffer when you are finished with it.
*/
AVFilterBufferRef *avfilter_get_audio_buffer(AVFilterLink *link, int perms,
- enum AVSampleFormat sample_fmt, int nb_samples,
- uint64_t channel_layout);
+ int nb_samples);
/**
* Create an audio buffer reference wrapped around an already
diff --git a/libavfilter/defaults.c b/libavfilter/defaults.c
index 7c75ab9..df05c06 100644
--- a/libavfilter/defaults.c
+++ b/libavfilter/defaults.c
@@ -58,25 +58,24 @@ AVFilterBufferRef *avfilter_default_get_video_buffer(AVFilterLink *link, int per
}
AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int perms,
- enum AVSampleFormat sample_fmt, int nb_samples,
- uint64_t channel_layout)
+ int nb_samples)
{
AVFilterBufferRef *samplesref = NULL;
uint8_t **data;
- int planar = av_sample_fmt_is_planar(sample_fmt);
- int nb_channels = av_get_channel_layout_nb_channels(channel_layout);
+ int planar = av_sample_fmt_is_planar(link->format);
+ int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
int planes = planar ? nb_channels : 1;
int linesize;
if (!(data = av_mallocz(sizeof(*data) * planes)))
goto fail;
- if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, sample_fmt, 0) < 0)
+ if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0)
goto fail;
samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms,
- nb_samples, sample_fmt,
- channel_layout);
+ nb_samples, link->format,
+ link->channel_layout);
if (!samplesref)
goto fail;
@@ -142,9 +141,8 @@ void avfilter_default_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *sa
outlink = inlink->dst->outputs[0];
if (outlink) {
- outlink->out_buf = avfilter_default_get_audio_buffer(inlink, AV_PERM_WRITE, samplesref->format,
- samplesref->audio->nb_samples,
- samplesref->audio->channel_layout);
+ outlink->out_buf = avfilter_default_get_audio_buffer(inlink, AV_PERM_WRITE,
+ samplesref->audio->nb_samples);
outlink->out_buf->pts = samplesref->pts;
outlink->out_buf->audio->sample_rate = samplesref->audio->sample_rate;
avfilter_filter_samples(outlink, avfilter_ref_buffer(outlink->out_buf, ~0));
@@ -246,9 +244,7 @@ AVFilterBufferRef *avfilter_null_get_video_buffer(AVFilterLink *link, int perms,
}
AVFilterBufferRef *avfilter_null_get_audio_buffer(AVFilterLink *link, int perms,
- enum AVSampleFormat sample_fmt, int nb_samples,
- uint64_t channel_layout)
+ int nb_samples)
{
- return avfilter_get_audio_buffer(link->dst->outputs[0], perms, sample_fmt,
- nb_samples, channel_layout);
+ return avfilter_get_audio_buffer(link->dst->outputs[0], perms, nb_samples);
}
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