Merge remote-tracking branch 'qatar/master'

* qatar/master:
  rv34: add NEON rv34_idct_add
  rv34: 1-pass inter MB reconstruction
  add SMJPEG muxer
  avformat: split out common SMJPEG code
  pictordec: Use bytestream2 functions
  avconv: use avcodec_encode_audio2()
  pcmenc: use AVCodec.encode2()
  avcodec: bump minor version and add APIChanges for the new audio encoding API
  avcodec: Add avcodec_encode_audio2() as replacement for avcodec_encode_audio()
  avcodec: add a public function, avcodec_fill_audio_frame().
  rv34: Intra 16x16 handling
  rv34: Inter/intra MB code split

Conflicts:
	Changelog
	libavcodec/avcodec.h
	libavcodec/pictordec.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/x86/rv34dsp.asm
	libavformat/version.h

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

1 files changed, 90 insertions, 2 deletions

diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 4e55e0e12b..e690c81e12 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -761,6 +761,11 @@ typedef struct RcOverride{
  * Encoders:
  * The encoder needs to be fed with NULL data at the end of encoding until the
  * encoder no longer returns data.
+ *
+ * NOTE: For encoders implementing the AVCodec.encode2() function, setting this
+ *       flag also means that the encoder must set the pts and duration for
+ *       each output packet. If this flag is not set, the pts and duration will
+ *       be determined by libavcodec from the input frame.
  */
 #define CODEC_CAP_DELAY           0x0020
 /**
@@ -816,6 +821,10 @@ typedef struct RcOverride{
  */
 #define CODEC_CAP_AUTO_THREADS     0x8000
 /**
+ * Audio encoder supports receiving a different number of samples in each call.
+ */
+#define CODEC_CAP_VARIABLE_FRAME_SIZE 0x10000
+/**
  * Codec is lossless.
  */
 #define CODEC_CAP_LOSSLESS         0x80000000
@@ -3314,6 +3323,19 @@ typedef struct AVCodec {
      * Initialize codec static data, called from avcodec_register().
      */
     void (*init_static_data)(struct AVCodec *codec);
+
+    /**
+     * Encode data to an AVPacket.
+     *
+     * @param      avctx          codec context
+     * @param      avpkt          output AVPacket (may contain a user-provided buffer)
+     * @param[in]  frame          AVFrame containing the raw data to be encoded
+     * @param[out] got_packet_ptr encoder sets to 0 or 1 to indicate that a
+     *                            non-empty packet was returned in avpkt.
+     * @return 0 on success, negative error code on failure
+     */
+    int (*encode2)(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame,
+                   int *got_packet_ptr);
 } AVCodec;
 
 /**
@@ -4331,9 +4353,12 @@ int avcodec_decode_subtitle2(AVCodecContext *avctx, AVSubtitle *sub,
  */
 void avsubtitle_free(AVSubtitle *sub);
 
+#if FF_API_OLD_ENCODE_AUDIO
 /**
  * Encode an audio frame from samples into buf.
  *
+ * @deprecated Use avcodec_encode_audio2 instead.
+ *
  * @note The output buffer should be at least FF_MIN_BUFFER_SIZE bytes large.
  * However, for codecs with avctx->frame_size equal to 0 (e.g. PCM) the user
  * will know how much space is needed because it depends on the value passed
@@ -4353,8 +4378,71 @@ void avsubtitle_free(AVSubtitle *sub);
  * @return On error a negative value is returned, on success zero or the number
  * of bytes used to encode the data read from the input buffer.
  */
-int avcodec_encode_audio(AVCodecContext *avctx, uint8_t *buf, int buf_size,
-                         const short *samples);
+int attribute_deprecated avcodec_encode_audio(AVCodecContext *avctx,
+                                              uint8_t *buf, int buf_size,
+                                              const short *samples);
+#endif
+
+/**
+ * Encode a frame of audio.
+ *
+ * Takes input samples from frame and writes the next output packet, if
+ * available, to avpkt. The output packet does not necessarily contain data for
+ * the most recent frame, as encoders can delay, split, and combine input frames
+ * internally as needed.
+ *
+ * @param avctx     codec context
+ * @param avpkt     output AVPacket.
+ *                  The user can supply an output buffer by setting
+ *                  avpkt->data and avpkt->size prior to calling the
+ *                  function, but if the size of the user-provided data is not
+ *                  large enough, encoding will fail. All other AVPacket fields
+ *                  will be reset by the encoder using av_init_packet(). If
+ *                  avpkt->data is NULL, the encoder will allocate it.
+ *                  The encoder will set avpkt->size to the size of the
+ *                  output packet.
+ * @param[in] frame AVFrame containing the raw audio data to be encoded.
+ *                  May be NULL when flushing an encoder that has the
+ *                  CODEC_CAP_DELAY capability set.
+ *                  There are 2 codec capabilities that affect the allowed
+ *                  values of frame->nb_samples.
+ *                  If CODEC_CAP_SMALL_LAST_FRAME is set, then only the final
+ *                  frame may be smaller than avctx->frame_size, and all other
+ *                  frames must be equal to avctx->frame_size.
+ *                  If CODEC_CAP_VARIABLE_FRAME_SIZE is set, then each frame
+ *                  can have any number of samples.
+ *                  If neither is set, frame->nb_samples must be equal to
+ *                  avctx->frame_size for all frames.
+ * @param[out] got_packet_ptr This field is set to 1 by libavcodec if the
+ *                            output packet is non-empty, and to 0 if it is
+ *                            empty. If the function returns an error, the
+ *                            packet can be assumed to be invalid, and the
+ *                            value of got_packet_ptr is undefined and should
+ *                            not be used.
+ * @return          0 on success, negative error code on failure
+ */
+int avcodec_encode_audio2(AVCodecContext *avctx, AVPacket *avpkt,
+                          const AVFrame *frame, int *got_packet_ptr);
+
+/**
+ * Fill audio frame data and linesize.
+ * AVFrame extended_data channel pointers are allocated if necessary for
+ * planar audio.
+ *
+ * @param frame       the AVFrame
+ *                    frame->nb_samples must be set prior to calling the
+ *                    function. This function fills in frame->data,
+ *                    frame->extended_data, frame->linesize[0].
+ * @param nb_channels channel count
+ * @param sample_fmt  sample format
+ * @param buf         buffer to use for frame data
+ * @param buf_size    size of buffer
+ * @param align       plane size sample alignment
+ * @return            0 on success, negative error code on failure
+ */
+int avcodec_fill_audio_frame(AVFrame *frame, int nb_channels,
+                             enum AVSampleFormat sample_fmt, const uint8_t *buf,
+                             int buf_size, int align);
 
 /**
  * Encode a video frame from pict into buf.