Merge remote-tracking branch 'qatar/master'

* qatar/master: (26 commits)
  adxenc: use AVCodec.encode2()
  adxenc: Use the AVFrame in ADXContext for coded_frame
  indeo4: fix out-of-bounds function call.
  configure: Restructure help output.
  configure: Internal-only components should not be command-line selectable.
  vorbisenc: use AVCodec.encode2()
  libvorbis: use AVCodec.encode2()
  libopencore-amrnbenc: use AVCodec.encode2()
  ra144enc: use AVCodec.encode2()
  nellymoserenc: use AVCodec.encode2()
  roqaudioenc: use AVCodec.encode2()
  libspeex: use AVCodec.encode2()
  libvo_amrwbenc: use AVCodec.encode2()
  libvo_aacenc: use AVCodec.encode2()
  wmaenc: use AVCodec.encode2()
  mpegaudioenc: use AVCodec.encode2()
  libmp3lame: use AVCodec.encode2()
  libgsmenc: use AVCodec.encode2()
  libfaac: use AVCodec.encode2()
  g726enc: use AVCodec.encode2()
  ...

Conflicts:
	configure
	libavcodec/Makefile
	libavcodec/ac3enc.c
	libavcodec/adxenc.c
	libavcodec/libgsm.c
	libavcodec/libvorbis.c
	libavcodec/vorbisenc.c
	libavcodec/wmaenc.c
	tests/ref/acodec/g722
	tests/ref/lavf/asf
	tests/ref/lavf/ffm
	tests/ref/lavf/mkv
	tests/ref/lavf/mpg
	tests/ref/lavf/rm
	tests/ref/lavf/ts
	tests/ref/seek/lavf_asf
	tests/ref/seek/lavf_ffm
	tests/ref/seek/lavf_mkv
	tests/ref/seek/lavf_mpg
	tests/ref/seek/lavf_rm
	tests/ref/seek/lavf_ts

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

1 files changed, 31 insertions, 17 deletions

diff --git a/libavcodec/wmaenc.c b/libavcodec/wmaenc.c
index 019176a6d4..0b252663ed 100644
--- a/libavcodec/wmaenc.c
+++ b/libavcodec/wmaenc.c
@@ -20,6 +20,7 @@
  */
 
 #include "avcodec.h"
+#include "internal.h"
 #include "wma.h"
 #include "libavutil/avassert.h"
 
@@ -87,7 +88,12 @@ static int encode_init(AVCodecContext * avctx){
     avctx->bit_rate    = avctx->block_align * 8LL * avctx->sample_rate /
                          s->frame_len;
 //av_log(NULL, AV_LOG_ERROR, "%d %d %d %d\n", s->block_align, avctx->bit_rate, s->frame_len, avctx->sample_rate);
-    avctx->frame_size= s->frame_len;
+    avctx->frame_size = avctx->delay = s->frame_len;
+
+#if FF_API_OLD_ENCODE_AUDIO
+    avctx->coded_frame = &s->frame;
+    avcodec_get_frame_defaults(avctx->coded_frame);
+#endif
 
     return 0;
 }
@@ -341,16 +347,17 @@ static int encode_frame(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
     return put_bits_count(&s->pb)/8 - s->block_align;
 }
 
-static int encode_superframe(AVCodecContext *avctx,
-                            unsigned char *buf, int buf_size, void *data){
+static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
+                             const AVFrame *frame, int *got_packet_ptr)
+{
     WMACodecContext *s = avctx->priv_data;
-    const short *samples = data;
-    int i, total_gain;
+    const int16_t *samples = (const int16_t *)frame->data[0];
+    int i, total_gain, ret;
 
     s->block_len_bits= s->frame_len_bits; //required by non variable block len
     s->block_len = 1 << s->block_len_bits;
 
-    apply_window_and_mdct(avctx, samples, avctx->frame_size);
+    apply_window_and_mdct(avctx, samples, frame->nb_samples);
 
     if (s->ms_stereo) {
         float a, b;
@@ -364,24 +371,25 @@ static int encode_superframe(AVCodecContext *avctx,
         }
     }
 
-    if (buf_size < 2 * MAX_CODED_SUPERFRAME_SIZE) {
-        av_log(avctx, AV_LOG_ERROR, "output buffer size is too small\n");
-        return AVERROR(EINVAL);
+    if ((ret = ff_alloc_packet(avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE))) {
+        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+        return ret;
     }
 
 #if 1
     total_gain= 128;
     for(i=64; i; i>>=1){
-        int error= encode_frame(s, s->coefs, buf, buf_size, total_gain-i);
+        int error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
+                                 total_gain - i);
         if(error<0)
             total_gain-= i;
     }
 #else
     total_gain= 90;
-    best= encode_frame(s, s->coefs, buf, buf_size, total_gain);
+    best = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
     for(i=32; i; i>>=1){
-        int scoreL= encode_frame(s, s->coefs, buf, buf_size, total_gain-i);
-        int scoreR= encode_frame(s, s->coefs, buf, buf_size, total_gain+i);
+        int scoreL = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain - i);
+        int scoreR = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain + i);
         av_log(NULL, AV_LOG_ERROR, "%d %d %d (%d)\n", scoreL, best, scoreR, total_gain);
         if(scoreL < FFMIN(best, scoreR)){
             best = scoreL;
@@ -393,7 +401,7 @@ static int encode_superframe(AVCodecContext *avctx,
     }
 #endif
 
-    encode_frame(s, s->coefs, buf, buf_size, total_gain);
+    encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
     av_assert0((put_bits_count(&s->pb) & 7) == 0);
     i= s->block_align - (put_bits_count(&s->pb)+7)/8;
     av_assert0(i>=0);
@@ -402,7 +410,13 @@ static int encode_superframe(AVCodecContext *avctx,
 
     flush_put_bits(&s->pb);
     av_assert0(put_bits_ptr(&s->pb) - s->pb.buf == s->block_align);
-    return s->block_align;
+
+    if (frame->pts != AV_NOPTS_VALUE)
+        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+
+    avpkt->size = s->block_align;
+    *got_packet_ptr = 1;
+    return 0;
 }
 
 AVCodec ff_wmav1_encoder = {
@@ -411,7 +425,7 @@ AVCodec ff_wmav1_encoder = {
     .id             = CODEC_ID_WMAV1,
     .priv_data_size = sizeof(WMACodecContext),
     .init           = encode_init,
-    .encode         = encode_superframe,
+    .encode2        = encode_superframe,
     .close          = ff_wma_end,
     .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
     .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
@@ -423,7 +437,7 @@ AVCodec ff_wmav2_encoder = {
     .id             = CODEC_ID_WMAV2,
     .priv_data_size = sizeof(WMACodecContext),
     .init           = encode_init,
-    .encode         = encode_superframe,
+    .encode2        = encode_superframe,
     .close          = ff_wma_end,
     .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
     .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),