avcodec: Add avcodec_encode_audio2() as replacement for avcodec_encode_audio()

This allows audio encoders to optionally take an AVFrame as input and write
encoded output to an AVPacket.

This also adds AVCodec.encode2() which will also be usable by video and
subtitle encoders once support is implemented in the public functions.

1 files changed, 70 insertions, 2 deletions

diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index a5071aa438..be1b2021bd 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -741,6 +741,11 @@ 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
 /**
@@ -793,6 +798,10 @@ typedef struct RcOverride{
  * Codec supports avctx->thread_count == 0 (auto).
  */
 #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
 
 //The following defines may change, don't expect compatibility if you use them.
 #define MB_TYPE_INTRA4x4   0x0001
@@ -3246,6 +3255,19 @@ 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;
 
 /**
@@ -4213,9 +4235,12 @@ 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
@@ -4235,8 +4260,51 @@ 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.