xenia-project/FFmpeg
1
0
Fork 0
mirror of https://github.com/xenia-project/FFmpeg.git synced 2024-11-24 12:09:55 +00:00
Code Issues Actions Packages Projects Releases Wiki Activity

lavfi: simplify signature for avfilter_get_audio_buffer() and friends

Browse Source 
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@khirnov.net>
This commit is contained in:
Stefano Sabatini 2011-08-30 23:22:29 +02:00 committed by Anton Khirnov
parent 6735534f19
commit 7ef0adcc2e
3 changed files with 18 additions and 33 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

11
libavfilter/avfilter.c
View File

@ -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;

16
libavfilter/avfilter.h
View File

@ -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

24
libavfilter/defaults.c
View File

@ -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);
}