Merge remote-tracking branch 'qatar/master'

* qatar/master: (27 commits)
  libxvid: Give more suitable names to libxvid-related files.
  libxvid: Separate libxvid encoder from libxvid rate control code.
  jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse().
  fate: cosmetics: lowercase some comments
  fate: Give more consistent names to some RealVideo/RealAudio tests.
  lavfi: add avfilter_get_audio_buffer_ref_from_arrays().
  lavfi: add extended_data to AVFilterBuffer.
  lavc: check that extended_data is properly set in avcodec_encode_audio2().
  lavc: pad last audio frame with silence when needed.
  samplefmt: add a function for filling a buffer with silence.
  samplefmt: add a function for copying audio samples.
  lavr: do not try to copy to uninitialized output audio data.
  lavr: make avresample_read() with NULL output discard samples.
  fate: split idroq audio and video into separate tests
  fate: improve dependencies
  fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests
  fate: split some combined tests into separate audio and video tests
  fate: fix dependencies for probe tests
  mips: intreadwrite: fix inline asm for gcc 4.8
  mips: intreadwrite: remove unnecessary inline asm
  ...

Conflicts:
	cmdutils.h
	configure
	doc/APIchanges
	doc/filters.texi
	ffmpeg.c
	ffplay.c
	libavcodec/internal.h
	libavcodec/jpeglsdec.c
	libavcodec/libschroedingerdec.c
	libavcodec/libxvid.c
	libavcodec/libxvid_rc.c
	libavcodec/utils.c
	libavcodec/version.h
	libavfilter/avfilter.c
	libavfilter/avfilter.h
	libavfilter/buffersink.h
	tests/Makefile
	tests/fate/aac.mak
	tests/fate/audio.mak
	tests/fate/demux.mak
	tests/fate/ea.mak
	tests/fate/image.mak
	tests/fate/libavutil.mak
	tests/fate/lossless-audio.mak
	tests/fate/lossless-video.mak
	tests/fate/microsoft.mak
	tests/fate/qt.mak
	tests/fate/real.mak
	tests/fate/screen.mak
	tests/fate/video.mak
	tests/fate/voice.mak
	tests/fate/vqf.mak
	tests/ref/fate/ea-mad
	tests/ref/fate/ea-tqi

Merged-by: Michael Niedermayer <michaelni@gmx.at>

libavcodec/utils.c

@@ -1001,11 +1001,59 @@ int ff_alloc_packet(AVPacket *avpkt, int size)
     return ff_alloc_packet2(NULL, avpkt, size);
 }
 
+/**
+ * Pad last frame with silence.
+ */
+static int pad_last_frame(AVCodecContext *s, AVFrame **dst, const AVFrame *src)
+{
+    AVFrame *frame = NULL;
+    uint8_t *buf   = NULL;
+    int ret;
+
+    if (!(frame = avcodec_alloc_frame()))
+        return AVERROR(ENOMEM);
+    *frame = *src;
+
+    if ((ret = av_samples_get_buffer_size(&frame->linesize[0], s->channels,
+                                          s->frame_size, s->sample_fmt, 0)) < 0)
+        goto fail;
+
+    if (!(buf = av_malloc(ret))) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    frame->nb_samples = s->frame_size;
+    if ((ret = avcodec_fill_audio_frame(frame, s->channels, s->sample_fmt,
+                                        buf, ret, 0)) < 0)
+        goto fail;
+    if ((ret = av_samples_copy(frame->extended_data, src->extended_data, 0, 0,
+                               src->nb_samples, s->channels, s->sample_fmt)) < 0)
+        goto fail;
+    if ((ret = av_samples_set_silence(frame->extended_data, src->nb_samples,
+                                      frame->nb_samples - src->nb_samples,
+                                      s->channels, s->sample_fmt)) < 0)
+        goto fail;
+
+    *dst = frame;
+
+    return 0;
+
+fail:
+    if (frame->extended_data != frame->data)
+        av_freep(&frame->extended_data);
+    av_freep(&buf);
+    av_freep(&frame);
+    return ret;
+}
+
 int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
                                               AVPacket *avpkt,
                                               const AVFrame *frame,
                                               int *got_packet_ptr)
 {
+    AVFrame tmp;
+    AVFrame *padded_frame = NULL;
     int ret;
     AVPacket user_pkt = *avpkt;
     int needs_realloc = !user_pkt.data;
@@ -1018,12 +1066,38 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
         return 0;
     }
 
+    /* ensure that extended_data is properly set */
+    if (frame && !frame->extended_data) {
+        if (av_sample_fmt_is_planar(avctx->sample_fmt) &&
+            avctx->channels > AV_NUM_DATA_POINTERS) {
+            av_log(avctx, AV_LOG_ERROR, "Encoding to a planar sample format, "
+                   "with more than %d channels, but extended_data is not set.\n",
+                   AV_NUM_DATA_POINTERS);
+            return AVERROR(EINVAL);
+        }
+        av_log(avctx, AV_LOG_WARNING, "extended_data is not set.\n");
+
+        tmp = *frame;
+        tmp.extended_data = tmp.data;
+        frame = &tmp;
+    }
+
     /* check for valid frame size */
     if (frame) {
         if (avctx->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME) {
             if (frame->nb_samples > avctx->frame_size)
                 return AVERROR(EINVAL);
         } else if (!(avctx->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) {
+            if (frame->nb_samples < avctx->frame_size &&
+                !avctx->internal->last_audio_frame) {
+                ret = pad_last_frame(avctx, &padded_frame, frame);
+                if (ret < 0)
+                    return ret;
+
+                frame = padded_frame;
+                avctx->internal->last_audio_frame = 1;
+            }
+
             if (frame->nb_samples != avctx->frame_size)
                 return AVERROR(EINVAL);
         }
@@ -1084,6 +1158,13 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
              here to simplify things */
     avpkt->flags |= AV_PKT_FLAG_KEY;
 
+    if (padded_frame) {
+        av_freep(&padded_frame->data[0]);
+        if (padded_frame->extended_data != padded_frame->data)
+            av_freep(&padded_frame->extended_data);
+        av_freep(&padded_frame);
+    }
+
     return ret;
 }