lavfi: drop planar/packed negotiation support

The planar/packed switch and the packing_formats list is no longer
required, since the planar/packed information is now stored in the sample
format enum.

This is technically a major API break, possibly it should be not too
painful as we marked the audio filtering API as unstable.

3 files changed, 12 insertions, 20 deletions

diff --git a/doc/examples/filtering_audio.c b/doc/examples/filtering_audio.c
index e8e6259635..2207b458fe 100644
--- a/doc/examples/filtering_audio.c
+++ b/doc/examples/filtering_audio.c
@@ -88,7 +88,6 @@ static int init_filters(const char *filters_descr)
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
     const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    const int packing_fmts[]                = { AVFILTER_PACKED, -1 };
     const int64_t *chlayouts                = avfilter_all_channel_layouts;
     AVABufferSinkParams *abuffersink_params;
     const AVFilterLink *outlink;
@@ -98,7 +97,7 @@ static int init_filters(const char *filters_descr)
     /* buffer audio source: the decoded frames from the decoder will be inserted here. */
     if (!dec_ctx->channel_layout)
         dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
-    snprintf(args, sizeof(args), "%d:%d:0x%"PRIx64":packed",
+    snprintf(args, sizeof(args), "%d:%d:0x%"PRIx64,
              dec_ctx->sample_rate, dec_ctx->sample_fmt, dec_ctx->channel_layout);
     ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
                                        args, NULL, filter_graph);
@@ -111,7 +110,6 @@ static int init_filters(const char *filters_descr)
     abuffersink_params = av_abuffersink_params_alloc();
     abuffersink_params->sample_fmts     = sample_fmts;
     abuffersink_params->channel_layouts = chlayouts;
-    abuffersink_params->packing_fmts    = packing_fmts;
     ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
                                        NULL, abuffersink_params, filter_graph);
     av_free(abuffersink_params);
diff --git a/doc/filter_design.txt b/doc/filter_design.txt
index 2b23054a81..4157fd1959 100644
--- a/doc/filter_design.txt
+++ b/doc/filter_design.txt
@@ -15,7 +15,8 @@ Format negotiation
   the list supported formats.
 
   For video links, that means pixel format. For audio links, that means
-  channel layout, sample format and sample packing.
+  channel layout, and sample format (the sample packing is implied by the
+  sample format).
 
   The lists are not just lists, they are references to shared objects. When
   the negotiation mechanism computes the intersection of the formats
diff --git a/doc/filters.texi b/doc/filters.texi
index ba8c9f46fb..bdb9fe23f9 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -146,7 +146,7 @@ Convert the input audio to one of the specified formats. The framework will
 negotiate the most appropriate format to minimize conversions.
 
 The filter accepts three lists of formats, separated by ":", in the form:
-"@var{sample_formats}:@var{channel_layouts}:@var{packing_formats}".
+"@var{sample_formats}:@var{channel_layouts}".
 
 Elements in each list are separated by "," which has to be escaped in the
 filtergraph specification.
@@ -156,9 +156,9 @@ supported formats.
 
 Some examples follow:
 @example
-aformat=u8\\,s16:mono:packed
+aformat=u8\\,s16:mono
 
-aformat=s16:mono\\,stereo:all
+aformat=s16:mono\\,stereo
 @end example
 
 @section amerge
@@ -184,7 +184,7 @@ On the other hand, if both input are in stereo, the output channels will be
 in the default order: a1, a2, b1, b2, and the channel layout will be
 arbitrarily set to 4.0, which may or may not be the expected value.
 
-Both inputs must have the same sample rate, format and packing.
+Both inputs must have the same sample rate, and format.
 
 If inputs do not have the same duration, the output will stop with the
 shortest.
@@ -293,9 +293,6 @@ number of samples (per each channel) contained in the filtered frame
 @item rate
 sample rate for the audio frame
 
-@item planar
-if the packing format is planar, 0 if packed
-
 @item checksum
 Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame
 
@@ -566,7 +563,7 @@ This source is mainly intended for a programmatic use, in particular
 through the interface defined in @file{libavfilter/asrc_abuffer.h}.
 
 It accepts the following mandatory parameters:
-@var{sample_rate}:@var{sample_fmt}:@var{channel_layout}:@var{packing}
+@var{sample_rate}:@var{sample_fmt}:@var{channel_layout}
 
 @table @option
 
@@ -584,23 +581,19 @@ Either a channel layout name from channel_layout_map in
 @file{libavutil/audioconvert.c} or its corresponding integer representation
 from the AV_CH_LAYOUT_* macros in @file{libavutil/audioconvert.h}
 
-@item packing
-Either "packed" or "planar", or their integer representation: 0 or 1
-respectively.
-
 @end table
 
 For example:
 @example
-abuffer=44100:s16:stereo:planar
+abuffer=44100:s16p:stereo
 @end example
 
 will instruct the source to accept planar 16bit signed stereo at 44100Hz.
-Since the sample format with name "s16" corresponds to the number
-1 and the "stereo" channel layout corresponds to the value 0x3, this is
+Since the sample format with name "s16p" corresponds to the number
+6 and the "stereo" channel layout corresponds to the value 0x3, this is
 equivalent to:
 @example
-abuffer=44100:1:0x3:1
+abuffer=44100:6:0x3
 @end example
 
 @section aevalsrc