Remove remains of Libav compatibility

Libav seems rather dead: no release for 2 years, no new git commits in
master for almost a year (with one exception ~6 months ago). From what I
can tell, some developers resigned themselves to the horrifying idea to
post patches to ffmpeg-devel instead, while the rest of the developers
went on to greener pastures.

Libav was a better project than FFmpeg. Unfortunately, FFmpeg won,
because it managed to keep the name and website. Libav was pushed more
and more into obscurity: while there was initially a big push for Libav,
FFmpeg just remained "in place" and visible for most people. FFmpeg was
slowly draining all manpower and energy from Libav. A big part of this
was that FFmpeg stole code from Libav (regular merges of the entire
Libav git tree), making it some sort of Frankenstein mirror of Libav,
think decaying zombie with additional legs ("features") nailed to it.
"Stealing" surely is the wrong word; I'm just aping the language that
some of the FFmpeg members used to use. All that is in the past now, I'm
probably the only person left who is annoyed by this, and with this
commit I'm putting this decade long problem finally to an end. I just
thought I'd express my annoyance about this fucking shitshow one last
time.

The most intrusive change in this commit is the resample filter, which
originally used libavresample. Since the FFmpeg developer refused to
enable libavresample by default for drama reasons, and the API was
slightly different, so the filter used some big preprocessor mess to
make it compatible to libswresample. All that falls away now. The
simplification to the build system is also significant.

1 files changed, 0 insertions, 9 deletions

diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index 8362e918f3..63f9aa0cbe 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -103,10 +103,7 @@ static bool init(struct mp_filter *da, struct mp_codec_params *codec,
     ctx->avframe = av_frame_alloc();
     lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
     lavc_context->codec_id = lavc_codec->id;
-
-#if LIBAVCODEC_VERSION_MICRO >= 100
     lavc_context->pkt_timebase = ctx->codec_timebase;
-#endif
 
     if (opts->downmix && mpopts->audio_output_channels.num_chmaps == 1) {
         lavc_context->request_channel_layout =
@@ -117,10 +114,8 @@ static bool init(struct mp_filter *da, struct mp_codec_params *codec,
     av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc,
                       AV_OPT_SEARCH_CHILDREN);
 
-#if LIBAVCODEC_VERSION_MICRO >= 100
     // Let decoder add AV_FRAME_DATA_SKIP_SAMPLES.
     av_opt_set(lavc_context, "flags2", "+skip_manual", AV_OPT_SEARCH_CHILDREN);
-#endif
 
     mp_set_avopts(da->log, lavc_context, opts->avopts);
 
@@ -199,10 +194,8 @@ static int receive_frame(struct mp_filter *da, struct mp_frame *out)
         MP_ERR(da, "Error decoding audio.\n");
     }
 
-#if LIBAVCODEC_VERSION_MICRO >= 100
     if (priv->avframe->flags & AV_FRAME_FLAG_DISCARD)
         av_frame_unref(priv->avframe);
-#endif
 
     if (!priv->avframe->buf[0])
         return ret;
@@ -224,7 +217,6 @@ static int receive_frame(struct mp_filter *da, struct mp_frame *out)
 
     priv->next_pts = mp_aframe_end_pts(mpframe);
 
-#if LIBAVCODEC_VERSION_MICRO >= 100
     AVFrameSideData *sd =
         av_frame_get_side_data(priv->avframe, AV_FRAME_DATA_SKIP_SAMPLES);
     if (sd && sd->size >= 10) {
@@ -232,7 +224,6 @@ static int receive_frame(struct mp_filter *da, struct mp_frame *out)
         priv->skip_samples += AV_RL32(d + 0);
         priv->trim_samples += AV_RL32(d + 4);
     }
-#endif
 
     if (!priv->preroll_done) {
         // Skip only if this isn't already handled by AV_FRAME_DATA_SKIP_SAMPLES.