Merge remote-tracking branch 'qatar/master'

* qatar/master:
  cosmetics: fix some then/than typos
  doxygen: Include libavcodec and libavformat examples into the documentation
  avutil: elaborate documentation for av_get_random_seed
  Add support for aac streams in mp4/mov without extradata.
  aes: whitespace cosmetics
  adler32: whitespace cosmetics
  swscale: fix another yuv range conversion overflow in 16bit scaling.
  Fix cpu flags test program
  opt-test: Add missing braces to silence compiler warnings.
  build: Eliminate obsolete test targets.
  udp: Fix a compilation warning
  swscale: Unbreak build with --enable-small
  base64: add fate test
  aes: improve test program and add fate test
  adler32: make test program more useful and add fate test
  swscale: fix yuv range correction when using 16-bit scaling.
  aacenc: Make chan_map const correct

Conflicts:
	Makefile
	doc/examples/muxing-example.c
	libavformat/udp.c
	libavutil/random_seed.h

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

1 files changed, 42 insertions, 0 deletions

diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index cb8760801a..8a936da59e 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -532,6 +532,22 @@ static void reset_all_predictors(PredictorState *ps)
         reset_predict_state(&ps[i]);
 }
 
+static int sample_rate_idx (int rate)
+{
+         if (92017 <= rate) return 0;
+    else if (75132 <= rate) return 1;
+    else if (55426 <= rate) return 2;
+    else if (46009 <= rate) return 3;
+    else if (37566 <= rate) return 4;
+    else if (27713 <= rate) return 5;
+    else if (23004 <= rate) return 6;
+    else if (18783 <= rate) return 7;
+    else if (13856 <= rate) return 8;
+    else if (11502 <= rate) return 9;
+    else if (9391  <= rate) return 10;
+    else                    return 11;
+}
+
 static void reset_predictor_group(PredictorState *ps, int group_num)
 {
     int i;
@@ -554,10 +570,33 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
     ac->m4ac.sample_rate = avctx->sample_rate;
 
     if (avctx->extradata_size > 0) {
+        avctx->channels    = 0;
+        avctx->frame_size  = 0;
+        avctx->sample_rate = 0;
         if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
                                          avctx->extradata,
                                          avctx->extradata_size) < 0)
             return -1;
+    } else {
+        int sr, i;
+        enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
+
+        sr = sample_rate_idx(avctx->sample_rate);
+        ac->m4ac.sampling_index = sr;
+        ac->m4ac.channels = avctx->channels;
+
+        for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
+            if (ff_mpeg4audio_channels[i] == avctx->channels)
+                break;
+        if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
+            i = 0;
+        }
+        ac->m4ac.chan_config = i;
+
+        if (ac->m4ac.chan_config) {
+            set_default_channel_config(avctx, new_che_pos, ac->m4ac.chan_config);
+            output_configure(ac, ac->che_pos, new_che_pos, ac->m4ac.chan_config, OC_GLOBAL_HDR);
+        }
     }
 
     if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
@@ -2049,6 +2088,7 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
             if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
                 return -7;
         } else if (ac->output_configured != OC_LOCKED) {
+            ac->m4ac.chan_config = 0;
             ac->output_configured = OC_NONE;
         }
         if (ac->output_configured != OC_LOCKED) {
@@ -2516,6 +2556,7 @@ AVCodec ff_aac_decoder = {
     .sample_fmts = (const enum AVSampleFormat[]) {
         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
     },
+    .capabilities = CODEC_CAP_CHANNEL_CONF,
     .channel_layouts = aac_channel_layout,
 };
 
@@ -2536,5 +2577,6 @@ AVCodec ff_aac_latm_decoder = {
     .sample_fmts = (const enum AVSampleFormat[]) {
         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
     },
+    .capabilities = CODEC_CAP_CHANNEL_CONF,
     .channel_layouts = aac_channel_layout,
 };