CODEC_ID_UTVIDEO,
CODEC_ID_BMV_VIDEO,
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,
#define CH_LAYOUT_STEREO_DOWNMIX AV_CH_LAYOUT_STEREO_DOWNMIX
#endif
+#if FF_API_OLD_DECODE_AUDIO
/* in bytes */
#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
+#endif
/**
* Required number of additionally allocated bytes at the end of the input bitstream for decoding.
/* Codec can export data for HW decoding (XvMC). */
#define CODEC_CAP_HWACCEL 0x0010
/**
- * Codec has a nonzero delay and needs to be fed with avpkt->data=NULL,
+ * Encoder or decoder requires flushing with NULL input at the end in order to
+ * give the complete and correct output.
+ *
+ * NOTE: If this flag is not set, the codec is guaranteed to never be fed with
+ * with NULL data. The user can still send NULL data to the public encode
+ * or decode function, but libavcodec will not pass it along to the codec
+ * unless this flag is set.
+ *
+ * Decoders:
+ * The decoder has a non-zero delay and needs to be fed with avpkt->data=NULL,
* avpkt->size=0 at the end to get the delayed data until the decoder no longer
- * returns frames. If this is not set, the codec is guaranteed to never be fed
- * with NULL data.
+ * returns frames.
+ *
+ * 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 slice-based (or partition-based) multithreading.
*/
#define CODEC_CAP_SLICE_THREADS 0x2000
+/**
+ * Codec supports changed parameters at any point.
+ */
+#define CODEC_CAP_PARAM_CHANGE 0x4000
+/**
+ * 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
enum AVPacketSideDataType {
AV_PKT_DATA_PALETTE,
+ AV_PKT_DATA_NEW_EXTRADATA,
+ AV_PKT_DATA_PARAM_CHANGE,
};
typedef struct AVPacket {
#define AV_PKT_FLAG_KEY 0x0001 ///< The packet contains a keyframe
#define AV_PKT_FLAG_CORRUPT 0x0002 ///< The packet content is corrupted
+/**
+ * An AV_PKT_DATA_PARAM_CHANGE side data packet is laid out as follows:
+ * u32le param_flags
+ * if (param_flags & AV_SIDE_DATA_PARAM_CHANGE_CHANNEL_COUNT)
+ * s32le channel_count
+ * if (param_flags & AV_SIDE_DATA_PARAM_CHANGE_CHANNEL_LAYOUT)
+ * u64le channel_layout
+ * if (param_flags & AV_SIDE_DATA_PARAM_CHANGE_SAMPLE_RATE)
+ * s32le sample_rate
+ * if (param_flags & AV_SIDE_DATA_PARAM_CHANGE_DIMENSIONS)
+ * s32le width
+ * s32le height
+ */
+
+enum AVSideDataParamChangeFlags {
+ AV_SIDE_DATA_PARAM_CHANGE_CHANNEL_COUNT = 0x0001,
+ AV_SIDE_DATA_PARAM_CHANGE_CHANNEL_LAYOUT = 0x0002,
+ AV_SIDE_DATA_PARAM_CHANGE_SAMPLE_RATE = 0x0004,
+ AV_SIDE_DATA_PARAM_CHANGE_DIMENSIONS = 0x0008,
+};
+
/**
* Audio Video Frame.
* New fields can be added to the end of AVFRAME with minor version
#define AV_NUM_DATA_POINTERS 8
#endif
/**
- * pointer to the picture planes.
+ * pointer to the picture/channel planes.
* This might be different from the first allocated byte
- * - encoding:
- * - decoding:
+ * - encoding: Set by user
+ * - decoding: set by AVCodecContext.get_buffer()
*/
uint8_t *data[AV_NUM_DATA_POINTERS];
+
+ /**
+ * Size, in bytes, of the data for each picture/channel plane.
+ *
+ * For audio, only linesize[0] may be set. For planar audio, each channel
+ * plane must be the same size.
+ *
+ * - encoding: Set by user (video only)
+ * - decoding: set by AVCodecContext.get_buffer()
+ */
int linesize[AV_NUM_DATA_POINTERS];
+
/**
* pointer to the first allocated byte of the picture. Can be used in get_buffer/release_buffer.
* This isn't used by libavcodec unless the default get/release_buffer() is used.
*/
int quality;
+#if FF_API_AVFRAME_AGE
/**
- * buffer age (1->was last buffer and dint change, 2->..., ...).
- * Set to INT_MAX if the buffer has not been used yet.
- * - encoding: unused
- * - decoding: MUST be set by get_buffer().
+ * @deprecated unused
*/
- int age;
+ attribute_deprecated int age;
+#endif
/**
* is this picture used as reference
* - decoding: Set by libavcodec.
*/
void *thread_opaque;
+
+ /**
+ * number of audio samples (per channel) described by this frame
+ * - encoding: unused
+ * - decoding: Set by libavcodec
+ */
+ int nb_samples;
+
+ /**
+ * pointers to the data planes/channels.
+ *
+ * For video, this should simply point to data[].
+ *
+ * For planar audio, each channel has a separate data pointer, and
+ * linesize[0] contains the size of each channel buffer.
+ * For packed audio, there is just one data pointer, and linesize[0]
+ * contains the total size of the buffer for all channels.
+ *
+ * Note: Both data and extended_data will always be set by get_buffer(),
+ * but for planar audio with more channels that can fit in data,
+ * extended_data must be used by the decoder in order to access all
+ * channels.
+ *
+ * encoding: unused
+ * decoding: set by AVCodecContext.get_buffer()
+ */
+ uint8_t **extended_data;
+
+ /**
+ * sample aspect ratio for the video frame, 0/1 if unknown\unspecified
+ * - encoding: unused
+ * - decoding: Read by user.
+ */
+ AVRational sample_aspect_ratio;
+
+ /**
+ * width and height of the video frame
+ * - encoding: unused
+ * - decoding: Read by user.
+ */
+ int width, height;
+
+ /**
+ * format of the frame, -1 if unknown or unset
+ * Values correspond to enum PixelFormat for video frames,
+ * enum AVSampleFormat for audio)
+ * - encoding: unused
+ * - decoding: Read by user.
+ */
+ int format;
} AVFrame;
struct AVCodecInternal;
+enum AVFieldOrder {
+ AV_FIELD_UNKNOWN,
+ AV_FIELD_PROGRESSIVE,
+ AV_FIELD_TT, //< Top coded_first, top displayed first
+ AV_FIELD_BB, //< Bottom coded first, bottom displayed first
+ AV_FIELD_TB, //< Top coded first, bottom displayed first
+ AV_FIELD_BT, //< Bottom coded first, top displayed first
+};
+
/**
* main external API structure.
* New fields can be added to the end with minor version bumps.
* Some codecs need additional format info. It is stored here.
* If any muxer uses this then ALL demuxers/parsers AND encoders for the
* specific codec MUST set it correctly otherwise stream copy breaks.
- * In general use of this field by muxers is not recommanded.
+ * In general use of this field by muxers is not recommended.
* - encoding: Set by libavcodec.
* - decoding: Set by libavcodec. (FIXME: Is this OK?)
*/
/**
* Called at the beginning of each frame to get a buffer for it.
- * If pic.reference is set then the frame will be read later by libavcodec.
- * avcodec_align_dimensions2() should be used to find the required width and
- * height, as they normally need to be rounded up to the next multiple of 16.
+ *
+ * The function will set AVFrame.data[], AVFrame.linesize[].
+ * AVFrame.extended_data[] must also be set, but it should be the same as
+ * AVFrame.data[] except for planar audio with more channels than can fit
+ * in AVFrame.data[]. In that case, AVFrame.data[] shall still contain as
+ * many data pointers as it can hold.
+ *
* if CODEC_CAP_DR1 is not set then get_buffer() must call
* avcodec_default_get_buffer() instead of providing buffers allocated by
* some other means.
+ *
+ * AVFrame.data[] should be 32- or 16-byte-aligned unless the CPU doesn't
+ * need it. avcodec_default_get_buffer() aligns the output buffer properly,
+ * but if get_buffer() is overridden then alignment considerations should
+ * be taken into account.
+ *
+ * @see avcodec_default_get_buffer()
+ *
+ * Video:
+ *
+ * If pic.reference is set then the frame will be read later by libavcodec.
+ * avcodec_align_dimensions2() should be used to find the required width and
+ * height, as they normally need to be rounded up to the next multiple of 16.
+ *
* If frame multithreading is used and thread_safe_callbacks is set,
- * it may be called from a different thread, but not from more than one at once.
- * Does not need to be reentrant.
+ * it may be called from a different thread, but not from more than one at
+ * once. Does not need to be reentrant.
+ *
+ * @see release_buffer(), reget_buffer()
+ * @see avcodec_align_dimensions2()
+ *
+ * Audio:
+ *
+ * Decoders request a buffer of a particular size by setting
+ * AVFrame.nb_samples prior to calling get_buffer(). The decoder may,
+ * however, utilize only part of the buffer by setting AVFrame.nb_samples
+ * to a smaller value in the output frame.
+ *
+ * Decoders cannot use the buffer after returning from
+ * avcodec_decode_audio4(), so they will not call release_buffer(), as it
+ * is assumed to be released immediately upon return.
+ *
+ * As a convenience, av_samples_get_buffer_size() and
+ * av_samples_fill_arrays() in libavutil may be used by custom get_buffer()
+ * functions to find the required data size and to fill data pointers and
+ * linesize. In AVFrame.linesize, only linesize[0] may be set for audio
+ * since all planes must be the same size.
+ *
+ * @see av_samples_get_buffer_size(), av_samples_fill_arrays()
+ *
* - encoding: unused
* - decoding: Set by libavcodec, user can override.
*/
#if FF_API_X264_GLOBAL_OPTS
/**
- * Influences how often B-frames are used.
+ * Influence how often B-frames are used.
* - encoding: Set by user.
* - decoding: unused
*/
int mv0_threshold;
/**
- * Adjusts sensitivity of b_frame_strategy 1.
+ * Adjust sensitivity of b_frame_strategy 1.
* - encoding: Set by user.
* - decoding: unused
*/
#if FF_API_FLAC_GLOBAL_OPTS
/**
- * Determines which LPC analysis algorithm to use.
+ * Determine which LPC analysis algorithm to use.
* - encoding: Set by user
* - decoding: unused
*/
* libavcodec functions.
*/
struct AVCodecInternal *internal;
+
+ /** Field order
+ * - encoding: set by libavcodec
+ * - decoding: Set by libavcodec
+ */
+ enum AVFieldOrder field_order;
} AVCodecContext;
/**
* 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;
/**
* @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 occured
+ * @return allocated ReSampleContext, NULL if error occurred
*/
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate,
*/
int avcodec_open2(AVCodecContext *avctx, AVCodec *codec, AVDictionary **options);
+#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,
* @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
*
* @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
* @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.
*/
-int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
+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.
+ *
+ * Some decoders may support multiple frames in a single AVPacket. Such
+ * decoders would then just decode the first frame. In this case,
+ * avcodec_decode_audio4 has to be called again with an AVPacket containing
+ * the remaining data in order to decode the second frame, etc...
+ * Even if no frames are returned, the packet needs to be fed to the decoder
+ * with remaining data until it is completely consumed or an error occurs.
+ *
+ * @warning The input buffer, avpkt->data 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.
+ *
+ * @note You might have to align the input buffer. The alignment requirements
+ * depend on the CPU and the decoder.
+ *
+ * @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.
+ * @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.
+ * At least avpkt->data and avpkt->size should be set. Some
+ * decoders might also require additional fields to be set.
+ * @return A negative error code is returned if an error occurred during
+ * decoding, otherwise the number of bytes consumed from the input
+ * AVPacket is returned.
+ */
+int avcodec_decode_audio4(AVCodecContext *avctx, AVFrame *frame,
+ int *got_frame_ptr, AVPacket *avpkt);
/**
* Decode the video frame of size avpkt->size from avpkt->data into picture.
AVPacket *avpkt);
/**
- * Frees all allocated data in the given subtitle struct.
+ * Free all allocated data in the given subtitle struct.
*
* @param sub AVSubtitle to free.
*/
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 PCM audio 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.
+ * 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] 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 PCM audio the number of samples read from samples is equal to
- * buf_size * input_sample_size / output_sample_size.
+ * 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 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.
unsigned int av_xiphlacing(unsigned char *s, unsigned int v);
/**
- * Logs a generic warning message about a missing feature. This function is
+ * Log a generic warning message about a missing feature. This function is
* intended to be used internally by Libav (libavcodec, libavformat, etc.)
* only, and would normally not be used by applications.
* @param[in] avc a pointer to an arbitrary struct of which the first field is