ffmpeg/libavcodec/internal.h

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/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* common internal api header.
*/
#ifndef AVCODEC_INTERNAL_H
#define AVCODEC_INTERNAL_H
#include <stdint.h>
#include "libavutil/mathematics.h"
#include "libavutil/pixfmt.h"
#include "avcodec.h"
typedef struct InternalBuffer {
uint8_t *base[AV_NUM_DATA_POINTERS];
uint8_t *data[AV_NUM_DATA_POINTERS];
int linesize[AV_NUM_DATA_POINTERS];
int width;
int height;
enum AVPixelFormat pix_fmt;
uint8_t **extended_data;
int audio_data_size;
int nb_channels;
} InternalBuffer;
typedef struct AVCodecInternal {
/**
* internal buffer count
* used by default get/release/reget_buffer().
*/
int buffer_count;
/**
* internal buffers
* used by default get/release/reget_buffer().
*/
InternalBuffer *buffer;
/**
* Whether the parent AVCodecContext is a copy of the context which had
* init() called on it.
* This is used by multithreading - shared tables and picture pointers
* should be freed from the original context only.
*/
int is_copy;
#if FF_API_OLD_DECODE_AUDIO
/**
* Internal sample count used by avcodec_encode_audio() to fabricate pts.
* Can be removed along with avcodec_encode_audio().
*/
int sample_count;
#endif
/**
* An audio frame with less than required samples has been submitted and
* padded with silence. Reject all subsequent frames.
*/
int last_audio_frame;
Merge remote-tracking branch 'qatar/master' * qatar/master: (27 commits) libxvid: Give more suitable names to libxvid-related files. libxvid: Separate libxvid encoder from libxvid rate control code. jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse(). fate: cosmetics: lowercase some comments fate: Give more consistent names to some RealVideo/RealAudio tests. lavfi: add avfilter_get_audio_buffer_ref_from_arrays(). lavfi: add extended_data to AVFilterBuffer. lavc: check that extended_data is properly set in avcodec_encode_audio2(). lavc: pad last audio frame with silence when needed. samplefmt: add a function for filling a buffer with silence. samplefmt: add a function for copying audio samples. lavr: do not try to copy to uninitialized output audio data. lavr: make avresample_read() with NULL output discard samples. fate: split idroq audio and video into separate tests fate: improve dependencies fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests fate: split some combined tests into separate audio and video tests fate: fix dependencies for probe tests mips: intreadwrite: fix inline asm for gcc 4.8 mips: intreadwrite: remove unnecessary inline asm ... Conflicts: cmdutils.h configure doc/APIchanges doc/filters.texi ffmpeg.c ffplay.c libavcodec/internal.h libavcodec/jpeglsdec.c libavcodec/libschroedingerdec.c libavcodec/libxvid.c libavcodec/libxvid_rc.c libavcodec/utils.c libavcodec/version.h libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/buffersink.h tests/Makefile tests/fate/aac.mak tests/fate/audio.mak tests/fate/demux.mak tests/fate/ea.mak tests/fate/image.mak tests/fate/libavutil.mak tests/fate/lossless-audio.mak tests/fate/lossless-video.mak tests/fate/microsoft.mak tests/fate/qt.mak tests/fate/real.mak tests/fate/screen.mak tests/fate/video.mak tests/fate/voice.mak tests/fate/vqf.mak tests/ref/fate/ea-mad tests/ref/fate/ea-tqi Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-10 02:14:44 +02:00
/**
* temporary buffer used for encoders to store their bitstream
*/
uint8_t *byte_buffer;
unsigned int byte_buffer_size;
void *frame_thread_encoder;
/**
* Number of audio samples to skip at the start of the next decoded frame
*/
int skip_samples;
} AVCodecInternal;
struct AVCodecDefault {
const uint8_t *key;
const uint8_t *value;
};
/**
* Determine whether pix_fmt is a hardware accelerated format.
*/
int ff_is_hwaccel_pix_fmt(enum AVPixelFormat pix_fmt);
/**
* Return the hardware accelerated codec for codec codec_id and
* pixel format pix_fmt.
*
* @param codec_id the codec to match
* @param pix_fmt the pixel format to match
* @return the hardware accelerated codec, or NULL if none was found.
*/
AVHWAccel *ff_find_hwaccel(enum AVCodecID codec_id, enum AVPixelFormat pix_fmt);
/**
* Return the index into tab at which {a,b} match elements {[0],[1]} of tab.
* If there is no such matching pair then size is returned.
*/
int ff_match_2uint16(const uint16_t (*tab)[2], int size, int a, int b);
unsigned int avpriv_toupper4(unsigned int x);
/**
* does needed setup of pkt_pts/pos and such for (re)get_buffer();
*/
void ff_init_buffer_info(AVCodecContext *s, AVFrame *frame);
/**
* Remove and free all side data from packet.
*/
void ff_packet_free_side_data(AVPacket *pkt);
int avpriv_lock_avformat(void);
int avpriv_unlock_avformat(void);
/**
* Maximum size in bytes of extradata.
* This value was chosen such that every bit of the buffer is
* addressable by a 32-bit signed integer as used by get_bits.
*/
#define FF_MAX_EXTRADATA_SIZE ((1 << 28) - FF_INPUT_BUFFER_PADDING_SIZE)
/**
* Check AVPacket size and/or allocate data.
*
* Encoders supporting AVCodec.encode2() can use this as a convenience to
* ensure the output packet data is large enough, whether provided by the user
* or allocated in this function.
*
* @param avctx the AVCodecContext of the encoder
* @param avpkt the AVPacket
* If avpkt->data is already set, avpkt->size is checked
* to ensure it is large enough.
* If avpkt->data is NULL, a new buffer is allocated.
* avpkt->size is set to the specified size.
* All other AVPacket fields will be reset with av_init_packet().
* @param size the minimum required packet size
* @return 0 on success, negative error code on failure
*/
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int size);
int ff_alloc_packet(AVPacket *avpkt, int size);
/**
* Rescale from sample rate to AVCodecContext.time_base.
*/
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx,
int64_t samples)
{
if(samples == AV_NOPTS_VALUE)
return AV_NOPTS_VALUE;
return av_rescale_q(samples, (AVRational){ 1, avctx->sample_rate },
avctx->time_base);
}
int ff_thread_can_start_frame(AVCodecContext *avctx);
int ff_get_logical_cpus(AVCodecContext *avctx);
int avpriv_h264_has_num_reorder_frames(AVCodecContext *avctx);
/**
* Call avcodec_open2 recursively by decrementing counter, unlocking mutex,
* calling the function and then restoring again. Assumes the mutex is
* already locked
*/
int ff_codec_open2_recursive(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options);
/**
* Call avcodec_close recursively, counterpart to avcodec_open2_recursive.
*/
int ff_codec_close_recursive(AVCodecContext *avctx);
#endif /* AVCODEC_INTERNAL_H */