Merge remote-tracking branch 'qatar/master'

* qatar/master:
  aac_latm: reconfigure decoder on audio specific config changes
  latmdec: fix audio specific config parsing
  Add avcodec_decode_audio4().
  avcodec: change number of plane pointers from 4 to 8 at next major bump.
  Update developers documentation with coding conventions.
  svq1dec: avoid undefined get_bits(0) call
  ARM: h264dsp_neon cosmetics
  ARM: make some NEON macros reusable
  Do not memcpy raw video frames when using null muxer
  fate: update asf seektest
  vp8: flush buffers on size changes.
  doc: improve general documentation for MacOSX
  asf: use packet dts as approximation of pts
  asf: do not call av_read_frame
  rtsp: Initialize the media_type_mask in the rtp guessing demuxer
  Cleaned up alacenc.c

Conflicts:
	doc/APIchanges
	doc/developer.texi
	libavcodec/8svx.c
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/nellymoserdec.c
	libavcodec/tta.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/asfdec.c
	tests/ref/seek/lavf_asf

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

1 files changed, 61 insertions, 45 deletions

diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 5a2b230d24..a046d991e6 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -471,15 +471,17 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
  * @param   ac          pointer to AACContext, may be null
  * @param   avctx       pointer to AVCCodecContext, used for logging
  * @param   m4ac        pointer to MPEG4AudioConfig, used for parsing
- * @param   data        pointer to AVCodecContext extradata
- * @param   data_size   size of AVCCodecContext extradata
+ * @param   data        pointer to buffer holding an audio specific config
+ * @param   bit_size    size of audio specific config or data in bits
+ * @param   sync_extension look for an appended sync extension
  *
  * @return  Returns error status or number of consumed bits. <0 - error
  */
 static int decode_audio_specific_config(AACContext *ac,
                                         AVCodecContext *avctx,
                                         MPEG4AudioConfig *m4ac,
-                                        const uint8_t *data, int data_size, int asclen)
+                                        const uint8_t *data, int bit_size,
+                                        int sync_extension)
 {
     GetBitContext gb;
     int i;
@@ -489,9 +491,9 @@ static int decode_audio_specific_config(AACContext *ac,
          av_dlog(avctx, "%02x ", avctx->extradata[i]);
     av_dlog(avctx, "\n");
 
-    init_get_bits(&gb, data, data_size * 8);
+    init_get_bits(&gb, data, bit_size);
 
-    if ((i = avpriv_mpeg4audio_get_config(m4ac, data, asclen/8)) < 0)
+    if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0)
         return -1;
     if (m4ac->sampling_index > 12) {
         av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
@@ -591,7 +593,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
     if (avctx->extradata_size > 0) {
         if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
                                          avctx->extradata,
-                                         avctx->extradata_size, 8*avctx->extradata_size) < 0)
+                                         avctx->extradata_size*8, 1) < 0)
             return -1;
     } else {
         int sr, i;
@@ -665,6 +667,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
 
     cbrt_tableinit();
 
+    avcodec_get_frame_defaults(&ac->frame);
+    avctx->coded_frame = &ac->frame;
+
     return 0;
 }
 
@@ -2132,12 +2137,12 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
 }
 
 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
-                                int *data_size, GetBitContext *gb)
+                                int *got_frame_ptr, GetBitContext *gb)
 {
     AACContext *ac = avctx->priv_data;
     ChannelElement *che = NULL, *che_prev = NULL;
     enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
-    int err, elem_id, data_size_tmp;
+    int err, elem_id;
     int samples = 0, multiplier, audio_found = 0;
 
     if (show_bits(gb, 12) == 0xfff) {
@@ -2250,24 +2255,26 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
         avctx->frame_size = samples;
     }
 
-    data_size_tmp = samples * avctx->channels *
-                    av_get_bytes_per_sample(avctx->sample_fmt);
-    if (*data_size < data_size_tmp) {
-        av_log(avctx, AV_LOG_ERROR,
-               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
-               *data_size, data_size_tmp);
-        return -1;
-    }
-    *data_size = data_size_tmp;
-
     if (samples) {
+        /* get output buffer */
+        ac->frame.nb_samples = samples;
+        if ((err = avctx->get_buffer(avctx, &ac->frame)) < 0) {
+            av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+            return err;
+        }
+
         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
-            ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
+            ac->fmt_conv.float_interleave((float *)ac->frame.data[0],
+                                          (const float **)ac->output_data,
                                           samples, avctx->channels);
         else
-            ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
+            ac->fmt_conv.float_to_int16_interleave((int16_t *)ac->frame.data[0],
+                                                   (const float **)ac->output_data,
                                                    samples, avctx->channels);
+
+        *(AVFrame *)data = ac->frame;
     }
+    *got_frame_ptr = !!samples;
 
     if (ac->output_configured && audio_found)
         ac->output_configured = OC_LOCKED;
@@ -2276,7 +2283,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
 }
 
 static int aac_decode_frame(AVCodecContext *avctx, void *data,
-                            int *data_size, AVPacket *avpkt)
+                            int *got_frame_ptr, AVPacket *avpkt)
 {
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
@@ -2287,7 +2294,7 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
 
     init_get_bits(&gb, buf, buf_size * 8);
 
-    if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
+    if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb)) < 0)
         return err;
 
     buf_consumed = (get_bits_count(&gb) + 7) >> 3;
@@ -2340,30 +2347,40 @@ static inline uint32_t latm_get_value(GetBitContext *b)
 static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
                                              GetBitContext *gb, int asclen)
 {
-    AVCodecContext *avctx = latmctx->aac_ctx.avctx;
-    AACContext *ac= &latmctx->aac_ctx;
-    MPEG4AudioConfig m4ac=ac->m4ac;
-    int  config_start_bit = get_bits_count(gb);
-    int     bits_consumed, esize;
+    AACContext *ac        = &latmctx->aac_ctx;
+    AVCodecContext *avctx = ac->avctx;
+    MPEG4AudioConfig m4ac = {0};
+    int config_start_bit  = get_bits_count(gb);
+    int sync_extension    = 0;
+    int bits_consumed, esize;
+
+    if (asclen) {
+        sync_extension = 1;
+        asclen         = FFMIN(asclen, get_bits_left(gb));
+    } else
+        asclen         = get_bits_left(gb);
 
     if (config_start_bit % 8) {
         av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
                                "config not byte aligned.\n", 1);
         return AVERROR_INVALIDDATA;
-    } else {
-        bits_consumed =
-            decode_audio_specific_config(ac, avctx, &m4ac,
+    }
+    bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
                                          gb->buffer + (config_start_bit / 8),
-                                         get_bits_left(gb) / 8, asclen);
+                                         asclen, sync_extension);
 
-        if (bits_consumed < 0)
-            return AVERROR_INVALIDDATA;
-        if(ac->m4ac.sample_rate != m4ac.sample_rate || m4ac.chan_config != ac->m4ac.chan_config)
-            ac->m4ac= m4ac;
+    if (bits_consumed < 0)
+        return AVERROR_INVALIDDATA;
+
+    if (ac->m4ac.sample_rate != m4ac.sample_rate ||
+        ac->m4ac.chan_config != m4ac.chan_config) {
+
+        av_log(avctx, AV_LOG_INFO, "audio config changed\n");
+        latmctx->initialized = 0;
 
         esize = (bits_consumed+7) / 8;
 
-        if (avctx->extradata_size <= esize) {
+        if (avctx->extradata_size < esize) {
             av_free(avctx->extradata);
             avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
             if (!avctx->extradata)
@@ -2373,9 +2390,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
         avctx->extradata_size = esize;
         memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
         memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
-
-        skip_bits_long(gb, bits_consumed);
     }
+    skip_bits_long(gb, bits_consumed);
 
     return bits_consumed;
 }
@@ -2512,8 +2528,8 @@ static int read_audio_mux_element(struct LATMContext *latmctx,
 }
 
 
-static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
-                             AVPacket *avpkt)
+static int latm_decode_frame(AVCodecContext *avctx, void *out,
+                             int *got_frame_ptr, AVPacket *avpkt)
 {
     struct LATMContext *latmctx = avctx->priv_data;
     int                 muxlength, err;
@@ -2535,12 +2551,12 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
 
     if (!latmctx->initialized) {
         if (!avctx->extradata) {
-            *out_size = 0;
+            *got_frame_ptr = 0;
             return avpkt->size;
         } else {
             if ((err = decode_audio_specific_config(
                     &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.m4ac,
-                    avctx->extradata, avctx->extradata_size, 8*avctx->extradata_size)) < 0)
+                    avctx->extradata, avctx->extradata_size*8, 1)) < 0)
                 return err;
             latmctx->initialized = 1;
         }
@@ -2553,7 +2569,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
         return AVERROR_INVALIDDATA;
     }
 
-    if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
+    if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb)) < 0)
         return err;
 
     return muxlength;
@@ -2583,7 +2599,7 @@ 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,
+    .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
     .channel_layouts = aac_channel_layout,
 };
 
@@ -2604,7 +2620,7 @@ 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,
+    .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
     .channel_layouts = aac_channel_layout,
     .flush = flush,
 };