/**
* @file
- * external API header
+ * @ingroup libavc
+ * Libavcodec external API header
*/
#include <errno.h>
#include "libavutil/samplefmt.h"
+#include "libavutil/attributes.h"
#include "libavutil/avutil.h"
#include "libavutil/buffer.h"
#include "libavutil/cpu.h"
#include "libavutil/pixfmt.h"
#include "libavutil/rational.h"
-#include "libavcodec/version.h"
+#include "version.h"
+
/**
* @defgroup libavc Encoding/Decoding Library
* @{
AV_CODEC_ID_MSZH,
AV_CODEC_ID_ZLIB,
AV_CODEC_ID_QTRLE,
-#if FF_API_SNOW
- AV_CODEC_ID_SNOW,
-#endif
AV_CODEC_ID_TSCC,
AV_CODEC_ID_ULTI,
AV_CODEC_ID_QDRAW,
AV_CODEC_ID_CLLC,
AV_CODEC_ID_MSS2,
AV_CODEC_ID_VP9,
+ AV_CODEC_ID_AIC,
+ AV_CODEC_ID_ESCAPE130,
+ AV_CODEC_ID_G2M,
+ AV_CODEC_ID_WEBP,
/* various PCM "codecs" */
AV_CODEC_ID_FIRST_AUDIO = 0x10000, ///< A dummy id pointing at the start of audio codecs
AV_CODEC_ID_MLP,
AV_CODEC_ID_GSM_MS, /* as found in WAV */
AV_CODEC_ID_ATRAC3,
+#if FF_API_VOXWARE
AV_CODEC_ID_VOXWARE,
+#endif
AV_CODEC_ID_APE,
AV_CODEC_ID_NELLYMOSER,
AV_CODEC_ID_MUSEPACK8,
AV_CODEC_ID_OPUS,
AV_CODEC_ID_COMFORT_NOISE,
AV_CODEC_ID_TAK,
+ AV_CODEC_ID_METASOUND,
/* subtitle codecs */
AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.
AV_CODEC_ID_MPEG4SYSTEMS = 0x20001, /**< _FAKE_ codec to indicate a MPEG-4 Systems
* stream (only used by libavformat) */
AV_CODEC_ID_FFMETADATA = 0x21000, ///< Dummy codec for streams containing only metadata information.
-
-#if FF_API_CODEC_ID
-#include "old_codec_ids.h"
-#endif
};
-#if FF_API_CODEC_ID
-#define CodecID AVCodecID
-#endif
-
/**
* This struct describes the properties of a single codec described by an
* AVCodecID.
*/
#define AV_CODEC_PROP_LOSSLESS (1 << 2)
-#if FF_API_OLD_DECODE_AUDIO
-/* in bytes */
-#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
-#endif
-
/**
* @ingroup lavc_decoding
* Required number of additionally allocated bytes at the end of the input bitstream for decoding.
ME_X1, ///< reserved for experiments
ME_HEX, ///< hexagon based search
ME_UMH, ///< uneven multi-hexagon search
-#if FF_API_SNOW
- ME_ITER, ///< iterative search
-#endif
ME_TESA, ///< transformed exhaustive search algorithm
};
Note: Not everything is supported yet.
*/
+/**
+ * Allow decoders to produce frames with data planes that are not aligned
+ * to CPU requirements (e.g. due to cropping).
+ */
+#define CODEC_FLAG_UNALIGNED 0x0001
#define CODEC_FLAG_QSCALE 0x0002 ///< Use fixed qscale.
#define CODEC_FLAG_4MV 0x0004 ///< 4 MV per MB allowed / advanced prediction for H.263.
#define CODEC_FLAG_QPEL 0x0010 ///< Use qpel MC.
#define CODEC_FLAG2_LOCAL_HEADER 0x00000008 ///< Place global headers at every keyframe instead of in extradata.
#define CODEC_FLAG2_IGNORE_CROP 0x00010000 ///< Discard cropping information from SPS.
-#if FF_API_MPV_GLOBAL_OPTS
-#define CODEC_FLAG_CBP_RD 0x04000000 ///< Use rate distortion optimization for cbp.
-#define CODEC_FLAG_QP_RD 0x08000000 ///< Use rate distortion optimization for qp selectioon.
-#define CODEC_FLAG2_STRICT_GOP 0x00000002 ///< Strictly enforce GOP size.
-#define CODEC_FLAG2_SKIP_RD 0x00004000 ///< RD optimal MB level residual skipping
-#endif
#define CODEC_FLAG2_CHUNKS 0x00008000 ///< Input bitstream might be truncated at a packet boundaries instead of only at frame boundaries.
/* Unsupported options :
* This can be used to prevent truncation of the last audio samples.
*/
#define CODEC_CAP_SMALL_LAST_FRAME 0x0040
+#if FF_API_CAP_VDPAU
/**
* Codec can export data for HW decoding (VDPAU).
*/
#define CODEC_CAP_HWACCEL_VDPAU 0x0080
+#endif
/**
* Codec can output multiple frames per AVPacket
* Normally demuxers return one frame at a time, demuxers which do not do
*/
unsigned int stream_codec_tag;
-#if FF_API_SUB_ID
- /**
- * @deprecated this field is unused
- */
- attribute_deprecated int sub_id;
-#endif
-
void *priv_data;
/**
/**
* picture width / height.
* - encoding: MUST be set by user.
- * - decoding: Set by libavcodec.
- * Note: For compatibility it is possible to set this instead of
- * coded_width/height before decoding.
+ * - decoding: May be set by the user before opening the decoder if known e.g.
+ * from the container. Some decoders will require the dimensions
+ * to be set by the caller. During decoding, the decoder may
+ * overwrite those values as required.
*/
int width, height;
/**
- * Bitstream width / height, may be different from width/height.
+ * Bitstream width / height, may be different from width/height e.g. when
+ * the decoded frame is cropped before being output.
* - encoding: unused
- * - decoding: Set by user before init if known. Codec should override / dynamically change if needed.
+ * - decoding: May be set by the user before opening the decoder if known
+ * e.g. from the container. During decoding, the decoder may
+ * overwrite those values as required.
*/
int coded_width, coded_height;
int b_frame_strategy;
-#if FF_API_MPV_GLOBAL_OPTS
- /**
- * luma single coefficient elimination threshold
- * - encoding: Set by user.
- * - decoding: unused
- */
- attribute_deprecated int luma_elim_threshold;
-
- /**
- * chroma single coeff elimination threshold
- * - encoding: Set by user.
- * - decoding: unused
- */
- attribute_deprecated int chroma_elim_threshold;
-#endif
-
/**
* qscale offset between IP and B-frames
* - encoding: Set by user.
#define FF_CMP_VSAD 8
#define FF_CMP_VSSE 9
#define FF_CMP_NSSE 10
-#if FF_API_SNOW
-#define FF_CMP_W53 11
-#define FF_CMP_W97 12
-#endif
#define FF_CMP_DCTMAX 13
#define FF_CMP_DCT264 14
#define FF_CMP_CHROMA 256
*/
int inter_quant_bias;
-#if FF_API_COLOR_TABLE_ID
- /**
- * color table ID
- * - encoding: unused
- * - decoding: Which clrtable should be used for 8bit RGB images.
- * Tables have to be stored somewhere. FIXME
- */
- attribute_deprecated int color_table_id;
-#endif
-
/**
* slice flags
* - encoding: unused
*/
int noise_reduction;
-#if FF_API_INTER_THRESHOLD
- /**
- * @deprecated this field is unused
- */
- attribute_deprecated int inter_threshold;
-#endif
-
-#if FF_API_MPV_GLOBAL_OPTS
- /**
- * @deprecated use mpegvideo private options instead
- */
- attribute_deprecated int quantizer_noise_shaping;
-#endif
-
/**
* Motion estimation threshold below which no motion estimation is
* performed, but instead the user specified motion vectors are used.
* - decoding: Set by user.
* @deprecated Deprecated in favor of request_channel_layout.
*/
- int request_channels;
+ attribute_deprecated int request_channels;
#endif
/**
/**
* This callback is called at the beginning of each frame to get data
* buffer(s) for it. There may be one contiguous buffer for all the data or
- * there may be a buffer per each data plane or anything in between. Each
- * buffer must be reference-counted using the AVBuffer API.
+ * there may be a buffer per each data plane or anything in between. What
+ * this means is, you may set however many entries in buf[] you feel necessary.
+ * Each buffer must be reference-counted using the AVBuffer API (see description
+ * of buf[] below).
*
* The following fields will be set in the frame before this callback is
* called:
* extended_data must be allocated with av_malloc() and will be freed in
* av_frame_unref().
* * otherwise exended_data must point to data
- * - buf[] must contain references to the buffers that contain the frame
- * data.
+ * - buf[] must contain one or more pointers to AVBufferRef structures. Each of
+ * the frame's data and extended_data pointers must be contained in these. That
+ * is, one AVBufferRef for each allocated chunk of memory, not necessarily one
+ * AVBufferRef per data[] entry. See: av_buffer_create(), av_buffer_alloc(),
+ * and av_buffer_ref().
* - extended_buf and nb_extended_buf must be allocated with av_malloc() by
* this callback and filled with the extra buffers if there are more
* buffers than buf[] can hold. extended_buf will be freed in
#define FF_IDCT_INT 1
#define FF_IDCT_SIMPLE 2
#define FF_IDCT_SIMPLEMMX 3
-#if FF_API_LIBMPEG2
-#define FF_IDCT_LIBMPEG2MMX 4
-#endif
-#if FF_API_MMI
-#define FF_IDCT_MMI 5
-#endif
#define FF_IDCT_ARM 7
#define FF_IDCT_ALTIVEC 8
#define FF_IDCT_SH4 9
#define FF_IDCT_FAAN 20
#define FF_IDCT_SIMPLENEON 22
#define FF_IDCT_SIMPLEALPHA 23
-#if FF_API_IDCT
-#define FF_IDCT_H264 11
-#define FF_IDCT_VP3 12
-#define FF_IDCT_CAVS 15
-#define FF_IDCT_WMV2 19
-#define FF_IDCT_EA 21
-#define FF_IDCT_BINK 24
-#endif
-
-#if FF_API_DSP_MASK
- /**
- * Unused.
- * @deprecated use av_set_cpu_flags_mask() instead.
- */
- attribute_deprecated unsigned dsp_mask;
-#endif
/**
* bits per sample/pixel from the demuxer (needed for huffyuv).
*/
int bits_per_raw_sample;
+#if FF_API_LOWRES
/**
* low resolution decoding, 1-> 1/2 size, 2->1/4 size
* - encoding: unused
* - decoding: Set by user.
+ *
+ * @deprecated use decoder private options instead
*/
attribute_deprecated int lowres;
+#endif
/**
* the picture in the bitstream
#define FF_PROFILE_AAC_HE_V2 28
#define FF_PROFILE_AAC_LD 22
#define FF_PROFILE_AAC_ELD 38
+#define FF_PROFILE_MPEG2_AAC_LOW 128
+#define FF_PROFILE_MPEG2_AAC_HE 131
#define FF_PROFILE_DTS 20
#define FF_PROFILE_DTS_ES 30
#define FF_PROFILE_MPEG4_SIMPLE_STUDIO 14
#define FF_PROFILE_MPEG4_ADVANCED_SIMPLE 15
+#define FF_PROFILE_JPEG2000_CSTREAM_RESTRICTION_0 0
+#define FF_PROFILE_JPEG2000_CSTREAM_RESTRICTION_1 1
+#define FF_PROFILE_JPEG2000_CSTREAM_NO_RESTRICTION 2
+#define FF_PROFILE_JPEG2000_DCINEMA_2K 3
+#define FF_PROFILE_JPEG2000_DCINEMA_4K 4
+
/**
* level
* - encoding: Set by user.
uint8_t* av_packet_get_side_data(AVPacket *pkt, enum AVPacketSideDataType type,
int *size);
+/**
+ * Convenience function to free all the side data stored.
+ * All the other fields stay untouched.
+ *
+ * @param pkt packet
+ */
+void av_packet_free_side_data(AVPacket *pkt);
+
+/**
+ * Setup a new reference to the data described by a given packet
+ *
+ * If src is reference-counted, setup dst as a new reference to the
+ * buffer in src. Otherwise allocate a new buffer in dst and copy the
+ * data from src into it.
+ *
+ * All the other fields are copied from src.
+ *
+ * @see av_packet_unref
+ *
+ * @param dst Destination packet
+ * @param src Source packet
+ *
+ * @return 0 on success, a negative AVERROR on error.
+ */
+int av_packet_ref(AVPacket *dst, AVPacket *src);
+
+/**
+ * Wipe the packet.
+ *
+ * Unreference the buffer referenced by the packet and reset the
+ * remaining packet fields to their default values.
+ *
+ * @param pkt The packet to be unreferenced.
+ */
+void av_packet_unref(AVPacket *pkt);
+
+/**
+ * Move every field in src to dst and reset src.
+ *
+ * @see av_packet_unref
+ *
+ * @param src Source packet, will be reset
+ * @param dst Destination packet
+ */
+void av_packet_move_ref(AVPacket *dst, AVPacket *src);
+
+/**
+ * Copy only "properties" fields from src to dst.
+ *
+ * Properties for the purpose of this function are all the fields
+ * beside those related to the packet data (buf, data, size)
+ *
+ * @param dst Destination packet
+ * @param src Source packet
+ *
+ * @return 0 on success AVERROR on failure.
+ *
+ */
+int av_packet_copy_props(AVPacket *dst, const AVPacket *src);
+
/**
* @}
*/
void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height,
int linesize_align[AV_NUM_DATA_POINTERS]);
-#if FF_API_OLD_DECODE_AUDIO
-/**
- * Wrapper function which calls avcodec_decode_audio4.
- *
- * @deprecated Use avcodec_decode_audio4 instead.
- *
- * Decode the audio frame of size avpkt->size from avpkt->data into samples.
- * Some decoders may support multiple frames in a single AVPacket, such
- * decoders would then just decode the first frame. In this case,
- * avcodec_decode_audio3 has to be called again with an AVPacket that contains
- * the remaining data in order to decode the second frame etc.
- * If no frame
- * could be outputted, frame_size_ptr is zero. Otherwise, it is the
- * decompressed frame size in bytes.
- *
- * @warning You must set frame_size_ptr to the allocated size of the
- * output buffer before calling avcodec_decode_audio3().
- *
- * @warning The input buffer must be FF_INPUT_BUFFER_PADDING_SIZE larger than
- * the actual read bytes because some optimized bitstream readers read 32 or 64
- * bits at once and could read over the end.
- *
- * @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
- * it will work but it will have an impact on performance.
- *
- * In practice, avpkt->data should have 4 byte alignment at minimum and
- * samples should be 16 byte aligned unless the CPU doesn't need it
- * (AltiVec and SSE do).
- *
- * @note Codecs which have the CODEC_CAP_DELAY capability set have a delay
- * between input and output, these need to be fed with avpkt->data=NULL,
- * avpkt->size=0 at the end to return the remaining frames.
- *
- * @param avctx the codec context
- * @param[out] samples the output buffer, sample type in avctx->sample_fmt
- * If the sample format is planar, each channel plane will
- * be the same size, with no padding between channels.
- * @param[in,out] frame_size_ptr the output buffer size in bytes
- * @param[in] avpkt The input AVPacket containing the input buffer.
- * You can create such packet with av_init_packet() and by then setting
- * data and size, some decoders might in addition need other fields.
- * All decoders are designed to use the least fields possible though.
- * @return On error a negative value is returned, otherwise the number of bytes
- * used or zero if no frame data was decompressed (used) from the input AVPacket.
- */
-attribute_deprecated int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
- int *frame_size_ptr,
- AVPacket *avpkt);
-#endif
-
/**
* Decode the audio frame of size avpkt->size from avpkt->data into frame.
*
*
* @param avctx the codec context
* @param[out] frame The AVFrame in which to store decoded audio samples.
- * Decoders request a buffer of a particular size by setting
- * AVFrame.nb_samples prior to calling get_buffer(). The
- * decoder may, however, only utilize part of the buffer by
- * setting AVFrame.nb_samples to a smaller value in the
- * output frame.
+ * The decoder will allocate a buffer for the decoded frame by
+ * calling the AVCodecContext.get_buffer2() callback.
+ * When AVCodecContext.refcounted_frames is set to 1, the frame is
+ * reference counted and the returned reference belongs to the
+ * caller. The caller must release the frame using av_frame_unref()
+ * when the frame is no longer needed. The caller may safely write
+ * to the frame if av_frame_is_writable() returns 1.
+ * When AVCodecContext.refcounted_frames is set to 0, the returned
+ * reference belongs to the decoder and is valid only until the
+ * next call to this function or until closing or flushing the
+ * decoder. The caller may not write to it.
* @param[out] got_frame_ptr Zero if no frame could be decoded, otherwise it is
* non-zero.
* @param[in] avpkt The input AVPacket containing the input buffer.
*
* @param avctx the codec context
* @param[out] picture The AVFrame in which the decoded video frame will be stored.
- * Use avcodec_alloc_frame to get an AVFrame, the codec will
- * allocate memory for the actual bitmap.
- * with default get/release_buffer(), the decoder frees/reuses the bitmap as it sees fit.
- * with overridden get/release_buffer() (needs CODEC_CAP_DR1) the user decides into what buffer the decoder
- * decodes and the decoder tells the user once it does not need the data anymore,
- * the user app can at this point free/reuse/keep the memory as it sees fit.
+ * Use av_frame_alloc() to get an AVFrame. The codec will
+ * allocate memory for the actual bitmap by calling the
+ * AVCodecContext.get_buffer2() callback.
+ * When AVCodecContext.refcounted_frames is set to 1, the frame is
+ * reference counted and the returned reference belongs to the
+ * caller. The caller must release the frame using av_frame_unref()
+ * when the frame is no longer needed. The caller may safely write
+ * to the frame if av_frame_is_writable() returns 1.
+ * When AVCodecContext.refcounted_frames is set to 0, the returned
+ * reference belongs to the decoder and is valid only until the
+ * next call to this function or until closing or flushing the
+ * decoder. The caller may not write to it.
*
* @param[in] avpkt The input AVpacket containing the input buffer.
* You can create such packet with av_init_packet() and by then setting
* @{
*/
+enum AVPictureStructure {
+ AV_PICTURE_STRUCTURE_UNKNOWN, //< unknown
+ AV_PICTURE_STRUCTURE_TOP_FIELD, //< coded as top field
+ AV_PICTURE_STRUCTURE_BOTTOM_FIELD, //< coded as bottom field
+ AV_PICTURE_STRUCTURE_FRAME, //< coded as frame
+};
+
typedef struct AVCodecParserContext {
void *priv_data;
struct AVCodecParser *parser;
* For all other types, this is in units of AVCodecContext.time_base.
*/
int duration;
+
+ enum AVFieldOrder field_order;
+
+ /**
+ * Indicate whether a picture is coded as a frame, top field or bottom field.
+ *
+ * For example, H.264 field_pic_flag equal to 0 corresponds to
+ * AV_PICTURE_STRUCTURE_FRAME. An H.264 picture with field_pic_flag
+ * equal to 1 and bottom_field_flag equal to 0 corresponds to
+ * AV_PICTURE_STRUCTURE_TOP_FIELD.
+ */
+ enum AVPictureStructure picture_structure;
+
+ /**
+ * Picture number incremented in presentation or output order.
+ * This field may be reinitialized at the first picture of a new sequence.
+ *
+ * For example, this corresponds to H.264 PicOrderCnt.
+ */
+ int output_picture_number;
} AVCodecParserContext;
typedef struct AVCodecParser {
*/
AVCodec *avcodec_find_encoder_by_name(const char *name);
-#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
- * in buf_size as described below. In that case a lower value can be used.
- *
- * @param avctx the codec context
- * @param[out] buf the output buffer
- * @param[in] buf_size the output buffer size
- * @param[in] samples the input buffer containing the samples
- * The number of samples read from this buffer is frame_size*channels,
- * both of which are defined in avctx.
- * For codecs which have avctx->frame_size equal to 0 (e.g. PCM) the number of
- * samples read from samples is equal to:
- * buf_size * 8 / (avctx->channels * av_get_bits_per_sample(avctx->codec_id))
- * This also implies that av_get_bits_per_sample() must not return 0 for these
- * codecs.
- * @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 attribute_deprecated avcodec_encode_audio(AVCodecContext *avctx,
- uint8_t *buf, int buf_size,
- const short *samples);
-#endif
-
/**
* Encode a frame of audio.
*
int avcodec_encode_audio2(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr);
-#if FF_API_OLD_ENCODE_VIDEO
-/**
- * @deprecated use avcodec_encode_video2() instead.
- *
- * Encode a video frame from pict into buf.
- * The input picture should be
- * stored using a specific format, namely avctx.pix_fmt.
- *
- * @param avctx the codec context
- * @param[out] buf the output buffer for the bitstream of encoded frame
- * @param[in] buf_size the size of the output buffer in bytes
- * @param[in] pict the input picture to encode
- * @return On error a negative value is returned, on success zero or the number
- * of bytes used from the output buffer.
- */
-attribute_deprecated
-int avcodec_encode_video(AVCodecContext *avctx, uint8_t *buf, int buf_size,
- const AVFrame *pict);
-#endif
-
/**
* Encode a frame of video.
*
* @}
*/
-#if FF_API_AVCODEC_RESAMPLE
-/**
- * @defgroup lavc_resample Audio resampling
- * @ingroup libavc
- * @deprecated use libavresample instead
- *
- * @{
- */
-struct ReSampleContext;
-struct AVResampleContext;
-
-typedef struct ReSampleContext ReSampleContext;
-
-/**
- * Initialize audio resampling context.
- *
- * @param output_channels number of output channels
- * @param input_channels number of input channels
- * @param output_rate output sample rate
- * @param input_rate input sample rate
- * @param sample_fmt_out requested output sample format
- * @param sample_fmt_in input sample format
- * @param filter_length length of each FIR filter in the filterbank relative to the cutoff frequency
- * @param log2_phase_count log2 of the number of entries in the polyphase filterbank
- * @param linear if 1 then the used FIR filter will be linearly interpolated
- between the 2 closest, if 0 the closest will be used
- * @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
- * @return allocated ReSampleContext, NULL if error occurred
- */
-attribute_deprecated
-ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
- int output_rate, int input_rate,
- enum AVSampleFormat sample_fmt_out,
- enum AVSampleFormat sample_fmt_in,
- int filter_length, int log2_phase_count,
- int linear, double cutoff);
-
-attribute_deprecated
-int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
-
-/**
- * Free resample context.
- *
- * @param s a non-NULL pointer to a resample context previously
- * created with av_audio_resample_init()
- */
-attribute_deprecated
-void audio_resample_close(ReSampleContext *s);
-
-
-/**
- * Initialize an audio resampler.
- * Note, if either rate is not an integer then simply scale both rates up so they are.
- * @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq
- * @param log2_phase_count log2 of the number of entries in the polyphase filterbank
- * @param linear If 1 then the used FIR filter will be linearly interpolated
- between the 2 closest, if 0 the closest will be used
- * @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
- */
-attribute_deprecated
-struct AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_length, int log2_phase_count, int linear, double cutoff);
-
-/**
- * Resample an array of samples using a previously configured context.
- * @param src an array of unconsumed samples
- * @param consumed the number of samples of src which have been consumed are returned here
- * @param src_size the number of unconsumed samples available
- * @param dst_size the amount of space in samples available in dst
- * @param update_ctx If this is 0 then the context will not be modified, that way several channels can be resampled with the same context.
- * @return the number of samples written in dst or -1 if an error occurred
- */
-attribute_deprecated
-int av_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx);
-
-
-/**
- * Compensate samplerate/timestamp drift. The compensation is done by changing
- * the resampler parameters, so no audible clicks or similar distortions occur
- * @param compensation_distance distance in output samples over which the compensation should be performed
- * @param sample_delta number of output samples which should be output less
- *
- * example: av_resample_compensate(c, 10, 500)
- * here instead of 510 samples only 500 samples would be output
- *
- * note, due to rounding the actual compensation might be slightly different,
- * especially if the compensation_distance is large and the in_rate used during init is small
- */
-attribute_deprecated
-void av_resample_compensate(struct AVResampleContext *c, int sample_delta, int compensation_distance);
-attribute_deprecated
-void av_resample_close(struct AVResampleContext *c);
-
-/**
- * @}
- */
-#endif
-
/**
* @addtogroup lavc_picture
* @{
int avcodec_get_pix_fmt_loss(enum AVPixelFormat dst_pix_fmt, enum AVPixelFormat src_pix_fmt,
int has_alpha);
-#if FF_API_FIND_BEST_PIX_FMT
-/**
- * @deprecated use avcodec_find_best_pix_fmt2() instead.
- *
- * Find the best pixel format to convert to given a certain source pixel
- * format. When converting from one pixel format to another, information loss
- * may occur. For example, when converting from RGB24 to GRAY, the color
- * information will be lost. Similarly, other losses occur when converting from
- * some formats to other formats. avcodec_find_best_pix_fmt() searches which of
- * the given pixel formats should be used to suffer the least amount of loss.
- * The pixel formats from which it chooses one, are determined by the
- * pix_fmt_mask parameter.
- *
- * @code
- * src_pix_fmt = AV_PIX_FMT_YUV420P;
- * pix_fmt_mask = (1 << AV_PIX_FMT_YUV422P) || (1 << AV_PIX_FMT_RGB24);
- * dst_pix_fmt = avcodec_find_best_pix_fmt(pix_fmt_mask, src_pix_fmt, alpha, &loss);
- * @endcode
- *
- * @param[in] pix_fmt_mask bitmask determining which pixel format to choose from
- * @param[in] src_pix_fmt source pixel format
- * @param[in] has_alpha Whether the source pixel format alpha channel is used.
- * @param[out] loss_ptr Combination of flags informing you what kind of losses will occur.
- * @return The best pixel format to convert to or -1 if none was found.
- */
-attribute_deprecated
-enum AVPixelFormat avcodec_find_best_pix_fmt(int64_t pix_fmt_mask, enum AVPixelFormat src_pix_fmt,
- int has_alpha, int *loss_ptr);
-#endif /* FF_API_FIND_BEST_PIX_FMT */
-
/**
* Find the best pixel format to convert to given a certain source pixel
* format. When converting from one pixel format to another, information loss
int buf_size, int align);
/**
- * Flush buffers, should be called when seeking or when switching to a different stream.
+ * Reset the internal decoder state / flush internal buffers. Should be called
+ * e.g. when seeking or when switching to a different stream.
+ *
+ * @note when refcounted frames are not used (i.e. avctx->refcounted_frames is 0),
+ * this invalidates the frames previously returned from the decoder. When
+ * refcounted frames are used, the decoder just releases any references it might
+ * keep internally, but the caller's reference remains valid.
*/
void avcodec_flush_buffers(AVCodecContext *avctx);
*/
unsigned int av_xiphlacing(unsigned char *s, unsigned int v);
+#if FF_API_MISSING_SAMPLE
/**
* Log a generic warning message about a missing feature. This function is
* intended to be used internally by Libav (libavcodec, libavformat, etc.)
* If want_sample is non-zero, additional verbage will be added to the log
* message which tells the user how to report samples to the development
* mailing list.
+ * @deprecated Use avpriv_report_missing_feature() instead.
*/
+attribute_deprecated
void av_log_missing_feature(void *avc, const char *feature, int want_sample);
/**
* @param[in] avc a pointer to an arbitrary struct of which the first field is
* a pointer to an AVClass struct
* @param[in] msg string containing an optional message, or NULL if no message
+ * @deprecated Use avpriv_request_sample() instead.
*/
+attribute_deprecated
void av_log_ask_for_sample(void *avc, const char *msg, ...) av_printf_format(2, 3);
+#endif /* FF_API_MISSING_SAMPLE */
/**
* Register the hardware accelerator hwaccel.
projects / ffmpeg / blobdiff