[FFmpeg-devel] [PATCH] lavfi: prefer nb_samples over size in AVFilterBufferRefAudioProps

Stefano Sabatini stefano.sabatini-lala at poste.it
Mon May 23 00:14:27 CEST 2011


Remove AVFilterBufferRefAudioProps.size, and use nb_samples in
avfilter_get_audio_buffer() and avfilter_default_get_audio_buffer() in
place of size.

This is required as the size in the audio buffer may be aligned, so it
may not contain a well defined number of samples.
---
 libavfilter/avfilter.c |   15 ++++++++-------
 libavfilter/avfilter.h |   11 +++++------
 libavfilter/defaults.c |   11 +++++------
 3 files changed, 18 insertions(+), 19 deletions(-)

diff --git a/libavfilter/avfilter.c b/libavfilter/avfilter.c
index 72e0a87..79326a0 100644
--- a/libavfilter/avfilter.c
+++ b/libavfilter/avfilter.c
@@ -280,10 +280,9 @@ static void ff_dlog_ref(void *ctx, AVFilterBufferRef *ref, int end)
                 av_get_picture_type_char(ref->video->pict_type));
     }
     if (ref->audio) {
-        av_dlog(ctx, " cl:%"PRId64"d sn:%d s:%d sr:%d p:%d",
+        av_dlog(ctx, " cl:%"PRId64"d n:%d r:%d p:%d",
                 ref->audio->channel_layout,
                 ref->audio->nb_samples,
-                ref->audio->size,
                 ref->audio->sample_rate,
                 ref->audio->planar);
     }
@@ -380,16 +379,16 @@ fail:
 }
 
 AVFilterBufferRef *avfilter_get_audio_buffer(AVFilterLink *link, int perms,
-                                             enum AVSampleFormat sample_fmt, int size,
+                                             enum AVSampleFormat sample_fmt, int nb_samples,
                                              int64_t channel_layout, int planar)
 {
     AVFilterBufferRef *ret = NULL;
 
     if (link->dstpad->get_audio_buffer)
-        ret = link->dstpad->get_audio_buffer(link, perms, sample_fmt, size, channel_layout, planar);
+        ret = link->dstpad->get_audio_buffer(link, perms, sample_fmt, nb_samples, channel_layout, planar);
 
     if (!ret)
-        ret = avfilter_default_get_audio_buffer(link, perms, sample_fmt, size, channel_layout, planar);
+        ret = avfilter_default_get_audio_buffer(link, perms, sample_fmt, nb_samples, channel_layout, planar);
 
     if (ret)
         ret->type = AVMEDIA_TYPE_AUDIO;
@@ -520,6 +519,7 @@ void avfilter_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
 {
     void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
     AVFilterPad *dst = link->dstpad;
+    int i;
 
     FF_DPRINTF_START(NULL, filter_samples); ff_dlog_link(NULL, link, 1);
 
@@ -536,14 +536,15 @@ void avfilter_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
 
         link->cur_buf = avfilter_default_get_audio_buffer(link, dst->min_perms,
                                                           samplesref->format,
-                                                          samplesref->audio->size,
+                                                          samplesref->audio->nb_samples,
                                                           samplesref->audio->channel_layout,
                                                           samplesref->audio->planar);
         link->cur_buf->pts                = samplesref->pts;
         link->cur_buf->audio->sample_rate = samplesref->audio->sample_rate;
 
         /* Copy actual data into new samples buffer */
-        memcpy(link->cur_buf->data[0], samplesref->data[0], samplesref->audio->size);
+        for (i = 0; samplesref->data[i]; i++)
+            memcpy(link->cur_buf->data[i], samplesref->data[i], samplesref->linesize[0]);
 
         avfilter_unref_buffer(samplesref);
     } else
diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h
index cee5bbc..6c7c103 100644
--- a/libavfilter/avfilter.h
+++ b/libavfilter/avfilter.h
@@ -98,8 +98,7 @@ typedef struct AVFilterBuffer {
  */
 typedef struct AVFilterBufferRefAudioProps {
     int64_t channel_layout;     ///< channel layout of audio buffer
-    int nb_samples;             ///< number of audio samples
-    int size;                   ///< audio buffer size
+    int nb_samples;             ///< number of audio samples per channel
     uint32_t sample_rate;       ///< audio buffer sample rate
     int planar;                 ///< audio buffer - planar or packed
 } AVFilterBufferRefAudioProps;
@@ -372,7 +371,7 @@ struct AVFilterPad {
      * Input audio pads only.
      */
     AVFilterBufferRef *(*get_audio_buffer)(AVFilterLink *link, int perms,
-                                           enum AVSampleFormat sample_fmt, int size,
+                                           enum AVSampleFormat sample_fmt, int nb_samples,
                                            int64_t channel_layout, int planar);
 
     /**
@@ -461,7 +460,7 @@ 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 size,
+                                                     enum AVSampleFormat sample_fmt, int nb_samples,
                                                      int64_t channel_layout, int planar);
 
 /**
@@ -679,14 +678,14 @@ avfilter_get_video_buffer_ref_from_arrays(uint8_t *data[4], int linesize[4], int
  *                       be requested
  * @param perms          the required access permissions
  * @param sample_fmt     the format of each sample in the buffer to allocate
- * @param size           the buffer size in bytes
+ * @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 size,
+                                             enum AVSampleFormat sample_fmt, int nb_samples,
                                              int64_t channel_layout, int planar);
 
 /**
diff --git a/libavfilter/defaults.c b/libavfilter/defaults.c
index 9ee23e5..854a105 100644
--- a/libavfilter/defaults.c
+++ b/libavfilter/defaults.c
@@ -79,7 +79,7 @@ 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 size,
+                                                     enum AVSampleFormat sample_fmt, int nb_samples,
                                                      int64_t channel_layout, int planar)
 {
     AVFilterBuffer *samples = av_mallocz(sizeof(AVFilterBuffer));
@@ -98,7 +98,7 @@ AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int per
         goto fail;
 
     ref->audio->channel_layout = channel_layout;
-    ref->audio->size           = size;
+    ref->audio->nb_samples     = nb_samples;
     ref->audio->planar         = planar;
 
     /* make sure the buffer gets read permission or it's useless for output */
@@ -110,8 +110,7 @@ AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int per
     sample_size = av_get_bits_per_sample_fmt(sample_fmt) >>3;
     chans_nb = av_get_channel_layout_nb_channels(channel_layout);
 
-    per_channel_size = size/chans_nb;
-    ref->audio->nb_samples = per_channel_size/sample_size;
+    per_channel_size = nb_samples * sample_size;
 
     /* Set the number of bytes to traverse to reach next sample of a particular channel:
      * For planar, this is simply the sample size.
@@ -122,7 +121,7 @@ AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int per
     memset(&samples->linesize[chans_nb], 0, (8-chans_nb) * sizeof(samples->linesize[0]));
 
     /* Calculate total buffer size, round to multiple of 16 to be SIMD friendly */
-    bufsize = (size + 15)&~15;
+    bufsize = (nb_samples * chans_nb * sample_size + 15)&~15;
     buf = av_malloc(bufsize);
     if (!buf)
         goto fail;
@@ -210,7 +209,7 @@ void avfilter_default_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *sa
 
     if (outlink) {
         outlink->out_buf = avfilter_default_get_audio_buffer(inlink, AV_PERM_WRITE, samplesref->format,
-                                                             samplesref->audio->size,
+                                                             samplesref->audio->nb_samples,
                                                              samplesref->audio->channel_layout,
                                                              samplesref->audio->planar);
         outlink->out_buf->pts                = samplesref->pts;
-- 
1.7.2.3



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