avcodec/cuvid: use pkt_timebase instead of time_base

[ffmpeg] / libavcodec / avcodec.h

diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 06c2b8933227b5dfe15755e91f0c16324350ef31..01f9b29773338138813bd43d9629358ff7a6bd4f 100644 (file)
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -445,9 +445,9 @@ enum AVCodecID {
     AV_CODEC_ID_PCM_S24LE_PLANAR,
     AV_CODEC_ID_PCM_S32LE_PLANAR,
     AV_CODEC_ID_PCM_S16BE_PLANAR,
-    /* new PCM "codecs" should be added right below this line starting with
-     * an explicit value of for example 0x10800
-     */
+
+    AV_CODEC_ID_PCM_S64LE = 0x10800,
+    AV_CODEC_ID_PCM_S64BE,
 
     /* various ADPCM codecs */
     AV_CODEC_ID_ADPCM_IMA_QT = 0x11000,
@@ -2093,22 +2093,23 @@ typedef struct AVCodecContext {
      * - decoding: unused
      */
     int ildct_cmp;
-#define FF_CMP_SAD    0
-#define FF_CMP_SSE    1
-#define FF_CMP_SATD   2
-#define FF_CMP_DCT    3
-#define FF_CMP_PSNR   4
-#define FF_CMP_BIT    5
-#define FF_CMP_RD     6
-#define FF_CMP_ZERO   7
-#define FF_CMP_VSAD   8
-#define FF_CMP_VSSE   9
-#define FF_CMP_NSSE   10
-#define FF_CMP_W53    11
-#define FF_CMP_W97    12
-#define FF_CMP_DCTMAX 13
-#define FF_CMP_DCT264 14
-#define FF_CMP_CHROMA 256
+#define FF_CMP_SAD          0
+#define FF_CMP_SSE          1
+#define FF_CMP_SATD         2
+#define FF_CMP_DCT          3
+#define FF_CMP_PSNR         4
+#define FF_CMP_BIT          5
+#define FF_CMP_RD           6
+#define FF_CMP_ZERO         7
+#define FF_CMP_VSAD         8
+#define FF_CMP_VSSE         9
+#define FF_CMP_NSSE         10
+#define FF_CMP_W53          11
+#define FF_CMP_W97          12
+#define FF_CMP_DCTMAX       13
+#define FF_CMP_DCT264       14
+#define FF_CMP_MEDIAN_SAD   15
+#define FF_CMP_CHROMA       256
 
     /**
      * ME diamond size & shape
@@ -3524,6 +3525,17 @@ typedef struct AVCodecContext {
 #define FF_SUB_TEXT_FMT_ASS_WITH_TIMINGS 1
 #endif
 
+    /**
+     * Audio only. The amount of padding (in samples) appended by the encoder to
+     * the end of the audio. I.e. this number of decoded samples must be
+     * discarded by the caller from the end of the stream to get the original
+     * audio without any trailing padding.
+     *
+     * - decoding: unused
+     * - encoding: unused
+     */
+    int trailing_padding;
+
 } AVCodecContext;
 
 AVRational av_codec_get_pkt_timebase         (const AVCodecContext *avctx);
@@ -5134,7 +5146,10 @@ AVCodecParserContext *av_parser_init(int codec_id);
  * @param poutbuf       set to pointer to parsed buffer or NULL if not yet finished.
  * @param poutbuf_size  set to size of parsed buffer or zero if not yet finished.
  * @param buf           input buffer.
- * @param buf_size      input length, to signal EOF, this should be 0 (so that the last frame can be output).
+ * @param buf_size      buffer size in bytes without the padding. I.e. the full buffer
+                        size is assumed to be buf_size + AV_INPUT_BUFFER_PADDING_SIZE.
+                        To signal EOF, this should be 0 (so that the last frame
+                        can be output).
  * @param pts           input presentation timestamp.
  * @param dts           input decoding timestamp.
  * @param pos           input byte position in stream.
@@ -6000,7 +6015,7 @@ int av_bsf_list_append2(AVBSFList *lst, const char * bsf_name, AVDictionary **op
  * This function will transform @ref AVBSFList to single @ref AVBSFContext,
  * so the whole chain of bitstream filters can be treated as single filter
  * freshly allocated by av_bsf_alloc().
- * If the call is successfull, @ref AVBSFList structure is freed and lst
+ * If the call is successful, @ref AVBSFList structure is freed and lst
  * will be set to NULL. In case of failure, caller is responsible for
  * freeing the structure by av_bsf_list_free()
  *