CODEC_ID_VBLE,
CODEC_ID_DXTORY,
CODEC_ID_V410,
+ CODEC_ID_XWD,
/* various PCM "codecs" */
CODEC_ID_FIRST_AUDIO = 0x10000, ///< A dummy id pointing at the start of audio codecs
CODEC_ID_ADPCM_EA_MAXIS_XA,
CODEC_ID_ADPCM_IMA_ISS,
CODEC_ID_ADPCM_G722,
+ CODEC_ID_ADPCM_IMA_APC,
/* AMR */
CODEC_ID_AMR_NB = 0x12000,
* 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
/**
* 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
* 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;
/**
* @warning The end of the input buffer avpkt->data should be set to 0 to ensure that
* no overreading happens for damaged MPEG streams.
*
+ * @warning You must not provide a custom get_buffer() when using
+ * avcodec_decode_audio3(). Doing so will override it with
+ * avcodec_default_get_buffer. Use avcodec_decode_audio4() instead,
+ * which does allow the application to provide a custom get_buffer().
+ *
* @note You might have to align the input buffer avpkt->data and output buffer
* samples. The alignment requirements depend on the CPU: On some CPUs it isn't
* necessary at all, on others it won't work at all if not aligned and on others
*/
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
* @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.
projects / ffmpeg / blobdiff