virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
Code Issues Releases Activity

Merge remote-tracking branch 'qatar/master'

Browse Source 
* qatar/master: (44 commits)
  replacement Indeo 3 decoder
  gsm demuxer: do not allocate packet twice.
  flvenc: use first packet delay as global delay.
  ac3enc: doxygen update.
  imc: return error codes instead of 0 for error conditions.
  imc: return meaningful error codes instead of -1
  imc: do not set channel layout for stereo
  imc: validate channel count
  imc: check for ff_fft_init() failure
  imc: check output buffer size before decoding
  imc: use DSPContext.bswap16_buf() to byte-swap packet data
  rtsp: add allowed_media_types option
  libgsm: add flush function to reset the decoder state when seeking
  libgsm: simplify decoding by using a loop
  gsm: log error message when packet is too small
  libgsmdec: do not needlessly set *data_size to 0
  gsmdec: do not needlessly set *data_size to 0
  gsmdec: add flush function to reset the decoder state when seeking
  libgsmdec: check output buffer size before decoding
  gsmdec: log error message when output buffer is too small.
  ...

Conflicts:
	Changelog
	ffplay.c
	libavcodec/indeo3.c
	libavcodec/mjpeg_parser.c
	libavcodec/vp3.c
	libavformat/cutils.c
	libavformat/id3v2.c
	libavutil/parseutils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2011-11-03 02:01:37 +01:00
commit 988f585fcb
75 changed files with 1878 additions and 3669 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
Changelog
View File

@ -71,6 +71,7 @@ easier to use. The changes are:
- Pulseaudio input device
- Prores encoder
- Video Decoder Acceleration (VDA) HWAccel module.
- replacement Indeo 3 decoder
version 0.8:

10
libavcodec/aac_parser.c
View File

@ -61,9 +61,9 @@ static av_cold int aac_parse_init(AVCodecParserContext *s1)
AVCodecParser ff_aac_parser = {
{ CODEC_ID_AAC },
sizeof(AACAC3ParseContext),
aac_parse_init,
ff_aac_ac3_parse,
ff_parse_close,
.codec_ids = { CODEC_ID_AAC },
.priv_data_size = sizeof(AACAC3ParseContext),
.parser_init = aac_parse_init,
.parser_parse = ff_aac_ac3_parse,
.parser_close = ff_parse_close,
};

10
libavcodec/ac3_parser.c
View File

@ -174,9 +174,9 @@ static av_cold int ac3_parse_init(AVCodecParserContext *s1)
AVCodecParser ff_ac3_parser = {
{ CODEC_ID_AC3, CODEC_ID_EAC3 },
sizeof(AACAC3ParseContext),
ac3_parse_init,
ff_aac_ac3_parse,
ff_parse_close,
.codec_ids = { CODEC_ID_AC3, CODEC_ID_EAC3 },
.priv_data_size = sizeof(AACAC3ParseContext),
.parser_init = ac3_parse_init,
.parser_parse = ff_aac_ac3_parse,
.parser_close = ff_parse_close,
};

95
libavcodec/ac3enc.c
View File

@ -176,6 +176,8 @@ static const int8_t ac3_coupling_start_tab[6][3][19] = {
/**
* Adjust the frame size to make the average bit rate match the target bit rate.
* This is only needed for 11025, 22050, and 44100 sample rates or any E-AC-3.
*
* @param s AC-3 encoder private context
*/
void ff_ac3_adjust_frame_size(AC3EncodeContext *s)
{
@ -190,6 +192,11 @@ void ff_ac3_adjust_frame_size(AC3EncodeContext *s)
}
/**
* Set the initial coupling strategy parameters prior to coupling analysis.
*
* @param s AC-3 encoder private context
*/
void ff_ac3_compute_coupling_strategy(AC3EncodeContext *s)
{
int blk, ch;
@ -258,6 +265,8 @@ void ff_ac3_compute_coupling_strategy(AC3EncodeContext *s)
/**
* Apply stereo rematrixing to coefficients based on rematrixing flags.
*
* @param s AC-3 encoder private context
*/
void ff_ac3_apply_rematrixing(AC3EncodeContext *s)
{
@ -290,7 +299,7 @@ void ff_ac3_apply_rematrixing(AC3EncodeContext *s)
}
/**
/*
* Initialize exponent tables.
*/
static av_cold void exponent_init(AC3EncodeContext *s)
@ -312,7 +321,7 @@ static av_cold void exponent_init(AC3EncodeContext *s)
}
/**
/*
* Extract exponents from the MDCT coefficients.
*/
static void extract_exponents(AC3EncodeContext *s)
@ -341,7 +350,7 @@ static const uint8_t exp_strategy_reuse_tab[4][6] = {
{ EXP_D45, EXP_D25, EXP_D25, EXP_D15, EXP_D15, EXP_D15 }
};
/**
/*
* Calculate exponent strategies for all channels.
* Array arrangement is reversed to simplify the per-channel calculation.
*/
@ -405,6 +414,11 @@ static void compute_exp_strategy(AC3EncodeContext *s)
/**
* Update the exponents so that they are the ones the decoder will decode.
*
* @param[in,out] exp array of exponents for 1 block in 1 channel
* @param nb_exps number of exponents in active bandwidth
* @param exp_strategy exponent strategy for the block
* @param cpl indicates if the block is in the coupling channel
*/
static void encode_exponents_blk_ch(uint8_t *exp, int nb_exps, int exp_strategy,
int cpl)
@ -473,7 +487,7 @@ static void encode_exponents_blk_ch(uint8_t *exp, int nb_exps, int exp_strategy,
}
/**
/*
* Encode exponents from original extracted form to what the decoder will see.
* This copies and groups exponents based on exponent strategy and reduces
* deltas between adjacent exponent groups so that they can be differentially
@ -526,7 +540,7 @@ static void encode_exponents(AC3EncodeContext *s)
}
/**
/*
* Count exponent bits based on bandwidth, coupling, and exponent strategies.
*/
static int count_exponent_bits(AC3EncodeContext *s)
@ -558,6 +572,8 @@ static int count_exponent_bits(AC3EncodeContext *s)
* Group exponents.
* 3 delta-encoded exponents are in each 7-bit group. The number of groups
* varies depending on exponent strategy and bandwidth.
*
* @param s AC-3 encoder private context
*/
void ff_ac3_group_exponents(AC3EncodeContext *s)
{
@ -614,6 +630,8 @@ void ff_ac3_group_exponents(AC3EncodeContext *s)
* Calculate final exponents from the supplied MDCT coefficients and exponent shift.
* Extract exponents from MDCT coefficients, calculate exponent strategies,
* and encode final exponents.
*
* @param s AC-3 encoder private context
*/
void ff_ac3_process_exponents(AC3EncodeContext *s)
{
@ -627,7 +645,7 @@ void ff_ac3_process_exponents(AC3EncodeContext *s)
}
/**
/*
* Count frame bits that are based solely on fixed parameters.
* This only has to be run once when the encoder is initialized.
*/
@ -733,7 +751,7 @@ static void count_frame_bits_fixed(AC3EncodeContext *s)
}
/**
/*
* Initialize bit allocation.
* Set default parameter codes and calculate parameter values.
*/
@ -768,7 +786,7 @@ static void bit_alloc_init(AC3EncodeContext *s)
}
/**
/*
* Count the bits used to encode the frame, minus exponents and mantissas.
* Bits based on fixed parameters have already been counted, so now we just
* have to add the bits based on parameters that change during encoding.
@ -915,7 +933,7 @@ static void count_frame_bits(AC3EncodeContext *s)
}
/**
/*
* Calculate masking curve based on the final exponents.
* Also calculate the power spectral densities to use in future calculations.
*/
@ -945,7 +963,7 @@ static void bit_alloc_masking(AC3EncodeContext *s)
}
/**
/*
* Ensure that bap for each block and channel point to the current bap_buffer.
* They may have been switched during the bit allocation search.
*/
@ -971,6 +989,8 @@ static void reset_block_bap(AC3EncodeContext *s)
* Initialize mantissa counts.
* These are set so that they are padded to the next whole group size when bits
* are counted in compute_mantissa_size.
*
* @param[in,out] mant_cnt running counts for each bap value for each block
*/
static void count_mantissa_bits_init(uint16_t mant_cnt[AC3_MAX_BLOCKS][16])
{
@ -987,6 +1007,12 @@ static void count_mantissa_bits_init(uint16_t mant_cnt[AC3_MAX_BLOCKS][16])
/**
* Update mantissa bit counts for all blocks in 1 channel in a given bandwidth
* range.
*
* @param s AC-3 encoder private context
* @param ch channel index
* @param[in,out] mant_cnt running counts for each bap value for each block
* @param start starting coefficient bin
* @param end ending coefficient bin
*/
static void count_mantissa_bits_update_ch(AC3EncodeContext *s, int ch,
uint16_t mant_cnt[AC3_MAX_BLOCKS][16],
@ -1005,7 +1031,7 @@ static void count_mantissa_bits_update_ch(AC3EncodeContext *s, int ch,
}
/**
/*
* Count the number of mantissa bits in the frame based on the bap values.
*/
static int count_mantissa_bits(AC3EncodeContext *s)
@ -1028,6 +1054,9 @@ static int count_mantissa_bits(AC3EncodeContext *s)
* Run the bit allocation with a given SNR offset.
* This calculates the bit allocation pointers that will be used to determine
* the quantization of each mantissa.
*
* @param s AC-3 encoder private context
* @param snr_offset SNR offset, 0 to 1023
* @return the number of bits needed for mantissas if the given SNR offset is
* is used.
*/
@ -1058,7 +1087,7 @@ static int bit_alloc(AC3EncodeContext *s, int snr_offset)
}
/**
/*
* Constant bitrate bit allocation search.
* Find the largest SNR offset that will allow data to fit in the frame.
*/
@ -1107,7 +1136,7 @@ static int cbr_bit_allocation(AC3EncodeContext *s)
}
/**
/*
* Perform bit allocation search.
* Finds the SNR offset value that maximizes quality and fits in the specified
* frame size. Output is the SNR offset and a set of bit allocation pointers
@ -1127,6 +1156,11 @@ int ff_ac3_compute_bit_allocation(AC3EncodeContext *s)
/**
* Symmetric quantization on 'levels' levels.
*
* @param c unquantized coefficient
* @param e exponent
* @param levels number of quantization levels
* @return quantized coefficient
*/
static inline int sym_quant(int c, int e, int levels)
{
@ -1138,6 +1172,11 @@ static inline int sym_quant(int c, int e, int levels)
/**
* Asymmetric quantization on 2^qbits levels.
*
* @param c unquantized coefficient
* @param e exponent
* @param qbits number of quantization bits
* @return quantized coefficient
*/
static inline int asym_quant(int c, int e, int qbits)
{
@ -1154,6 +1193,14 @@ static inline int asym_quant(int c, int e, int qbits)
/**
* Quantize a set of mantissas for a single channel in a single block.
*
* @param s Mantissa count context
* @param fixed_coef unquantized fixed-point coefficients
* @param exp exponents
* @param bap bit allocation pointer indices
* @param[out] qmant quantized coefficients
* @param start_freq starting coefficient bin
* @param end_freq ending coefficient bin
*/
static void quantize_mantissas_blk_ch(AC3Mant *s, int32_t *fixed_coef,
uint8_t *exp, uint8_t *bap,
@ -1246,6 +1293,8 @@ static void quantize_mantissas_blk_ch(AC3Mant *s, int32_t *fixed_coef,
/**
* Quantize mantissas using coefficients, exponents, and bit allocation pointers.
*
* @param s AC-3 encoder private context
*/
void ff_ac3_quantize_mantissas(AC3EncodeContext *s)
{
@ -1273,7 +1322,7 @@ void ff_ac3_quantize_mantissas(AC3EncodeContext *s)
}
/**
/*
* Write the AC-3 frame header to the output bitstream.
*/
static void ac3_output_frame_header(AC3EncodeContext *s)
@ -1329,7 +1378,7 @@ static void ac3_output_frame_header(AC3EncodeContext *s)
}
/**
/*
* Write one audio block to the output bitstream.
*/
static void output_audio_block(AC3EncodeContext *s, int blk)
@ -1557,7 +1606,7 @@ static unsigned int pow_poly(unsigned int a, unsigned int n, unsigned int poly)
}
/**
/*
* Fill the end of the frame with 0's and compute the two CRCs.
*/
static void output_frame_end(AC3EncodeContext *s)
@ -1605,6 +1654,9 @@ static void output_frame_end(AC3EncodeContext *s)
/**
* Write the frame to the output bitstream.
*
* @param s AC-3 encoder private context
* @param frame output data buffer
*/
void ff_ac3_output_frame(AC3EncodeContext *s, unsigned char *frame)
{
@ -1775,6 +1827,8 @@ static void validate_mix_level(void *log_ctx, const char *opt_name,
/**
* Validate metadata options as set by AVOption system.
* These values can optionally be changed per-frame.
*
* @param s AC-3 encoder private context
*/
int ff_ac3_validate_metadata(AC3EncodeContext *s)
{
@ -1957,6 +2011,8 @@ int ff_ac3_validate_metadata(AC3EncodeContext *s)
/**
* Finalize encoding and free any memory allocated by the encoder.
*
* @param avctx Codec context
*/
av_cold int ff_ac3_encode_close(AVCodecContext *avctx)
{
@ -2000,7 +2056,7 @@ av_cold int ff_ac3_encode_close(AVCodecContext *avctx)
}
/**
/*
* Set channel information during initialization.
*/
static av_cold int set_channel_info(AC3EncodeContext *s, int channels,
@ -2170,7 +2226,7 @@ static av_cold int validate_options(AC3EncodeContext *s)
}
/**
/*
* Set bandwidth for all channels.
* The user can optionally supply a cutoff frequency. Otherwise an appropriate
* default value will be used.
@ -2348,9 +2404,6 @@ alloc_fail:
}
/**
* Initialize the encoder.
*/
av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
{
AC3EncodeContext *s = avctx->priv_data;

18
libavcodec/ac3enc_fixed.c
View File

@ -41,6 +41,8 @@ static const AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_ite
/**
* Finalize MDCT and free allocated memory.
*
* @param s AC-3 encoder private context
*/
av_cold void AC3_NAME(mdct_end)(AC3EncodeContext *s)
{
@ -50,7 +52,9 @@ av_cold void AC3_NAME(mdct_end)(AC3EncodeContext *s)
/**
* Initialize MDCT tables.
* @param nbits log2(MDCT size)
*
* @param s AC-3 encoder private context
* @return 0 on success, negative error code on failure
*/
av_cold int AC3_NAME(mdct_init)(AC3EncodeContext *s)
{
@ -60,7 +64,7 @@ av_cold int AC3_NAME(mdct_init)(AC3EncodeContext *s)
}
/**
/*
* Apply KBD window to input samples prior to MDCT.
*/
static void apply_window(DSPContext *dsp, int16_t *output, const int16_t *input,
@ -70,11 +74,9 @@ static void apply_window(DSPContext *dsp, int16_t *output, const int16_t *input,
}
/**
/*
* Normalize the input samples to use the maximum available precision.
* This assumes signed 16-bit input samples.
*
* @return exponent shift
*/
static int normalize_samples(AC3EncodeContext *s)
{
@ -87,7 +89,7 @@ static int normalize_samples(AC3EncodeContext *s)
}
/**
/*
* Scale MDCT coefficients to 25-bit signed fixed-point.
*/
static void scale_coefficients(AC3EncodeContext *s)
@ -110,7 +112,7 @@ static void sum_square_butterfly(AC3EncodeContext *s, int64_t sum[4],
s->ac3dsp.sum_square_butterfly_int32(sum, coef0, coef1, len);
}
/**
/*
* Clip MDCT coefficients to allowable range.
*/
static void clip_coefficients(DSPContext *dsp, int32_t *coef, unsigned int len)
@ -119,7 +121,7 @@ static void clip_coefficients(DSPContext *dsp, int32_t *coef, unsigned int len)
}
/**
/*
* Calculate a single coupling coordinate.
*/
static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl)

16
libavcodec/ac3enc_float.c
View File

@ -44,6 +44,8 @@ static const AVClass ac3enc_class = { "AC-3 Encoder", av_default_item_name,
/**
* Finalize MDCT and free allocated memory.
*
* @param s AC-3 encoder private context
*/
av_cold void ff_ac3_float_mdct_end(AC3EncodeContext *s)
{
@ -54,7 +56,9 @@ av_cold void ff_ac3_float_mdct_end(AC3EncodeContext *s)
/**
* Initialize MDCT tables.
* @param nbits log2(MDCT size)
*
* @param s AC-3 encoder private context
* @return 0 on success, negative error code on failure
*/
av_cold int ff_ac3_float_mdct_init(AC3EncodeContext *s)
{
@ -78,7 +82,7 @@ av_cold int ff_ac3_float_mdct_init(AC3EncodeContext *s)
}
/**
/*
* Apply KBD window to input samples prior to MDCT.
*/
static void apply_window(DSPContext *dsp, float *output, const float *input,
@ -88,7 +92,7 @@ static void apply_window(DSPContext *dsp, float *output, const float *input,
}
/**
/*
* Normalize the input samples.
* Not needed for the floating-point encoder.
*/
@ -98,7 +102,7 @@ static int normalize_samples(AC3EncodeContext *s)
}
/**
/*
* Scale MDCT coefficients from float to 24-bit fixed-point.
*/
static void scale_coefficients(AC3EncodeContext *s)
@ -117,7 +121,7 @@ static void sum_square_butterfly(AC3EncodeContext *s, float sum[4],
s->ac3dsp.sum_square_butterfly_float(sum, coef0, coef1, len);
}
/**
/*
* Clip MDCT coefficients to allowable range.
*/
static void clip_coefficients(DSPContext *dsp, float *coef, unsigned int len)
@ -126,7 +130,7 @@ static void clip_coefficients(DSPContext *dsp, float *coef, unsigned int len)
}
/**
/*
* Calculate a single coupling coordinate.
*/
static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl)

11
libavcodec/ac3enc_template.c
View File

@ -67,7 +67,7 @@ alloc_fail:
}
/**
/*
* Deinterleave input samples.
* Channels are reordered from Libav's default order to AC-3 order.
*/
@ -96,7 +96,7 @@ static void deinterleave_input_samples(AC3EncodeContext *s,
}
/**
/*
* Apply the MDCT to input samples to generate frequency coefficients.
* This applies the KBD window and normalizes the input to reduce precision
* loss due to fixed-point calculations.
@ -123,7 +123,7 @@ static void apply_mdct(AC3EncodeContext *s)
}
/**
/*
* Calculate coupling channel and coupling coordinates.
*/
static void apply_channel_coupling(AC3EncodeContext *s)
@ -331,7 +331,7 @@ static void apply_channel_coupling(AC3EncodeContext *s)
}
/**
/*
* Determine rematrixing flags for each block and band.
*/
static void compute_rematrixing_strategy(AC3EncodeContext *s)
@ -386,9 +386,6 @@ static void compute_rematrixing_strategy(AC3EncodeContext *s)
}
/**
* Encode a single AC-3 frame.
*/
int AC3_NAME(encode_frame)(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
{

11
libavcodec/cavs_parser.c
View File

@ -98,10 +98,9 @@ static int cavsvideo_parse(AVCodecParserContext *s,
}
AVCodecParser ff_cavsvideo_parser = {
{ CODEC_ID_CAVS },
sizeof(ParseContext1),
NULL,
cavsvideo_parse,
ff_parse1_close,
ff_mpeg4video_split,
.codec_ids = { CODEC_ID_CAVS },
.priv_data_size = sizeof(ParseContext1),
.parser_parse = cavsvideo_parse,
.parser_close = ff_parse1_close,
.split = ff_mpeg4video_split,
};

10
libavcodec/dca_parser.c
View File

@ -124,9 +124,9 @@ static int dca_parse(AVCodecParserContext * s,
}
AVCodecParser ff_dca_parser = {
{CODEC_ID_DTS},
sizeof(DCAParseContext),
dca_parse_init,
dca_parse,
ff_parse_close,
.codec_ids = { CODEC_ID_DTS },
.priv_data_size = sizeof(DCAParseContext),
.parser_init = dca_parse_init,
.parser_parse = dca_parse,
.parser_close = ff_parse_close,
};

9
libavcodec/dirac_parser.c
View File

@ -248,9 +248,8 @@ static void dirac_parse_close(AVCodecParserContext *s)
}
AVCodecParser ff_dirac_parser = {
{ CODEC_ID_DIRAC },
sizeof(DiracParseContext),
NULL,
dirac_parse,
dirac_parse_close,
.codec_ids = { CODEC_ID_DIRAC },
.priv_data_size = sizeof(DiracParseContext),
.parser_parse = dirac_parse,
.parser_close = dirac_parse_close,
};

9
libavcodec/dnxhd_parser.c
View File

@ -87,9 +87,8 @@ static int dnxhd_parse(AVCodecParserContext *s,
}
AVCodecParser ff_dnxhd_parser = {
{ CODEC_ID_DNXHD },
sizeof(ParseContext),
NULL,
dnxhd_parse,
ff_parse_close,
.codec_ids = { CODEC_ID_DNXHD },
.priv_data_size = sizeof(ParseContext),
.parser_parse = dnxhd_parse,
.parser_close = ff_parse_close,
};

2
libavcodec/dvbsub.c
View File

@ -1,5 +1,5 @@
/*
* DVB subtitle encoding for ffmpeg
* DVB subtitle encoding
* Copyright (c) 2005 Fabrice Bellard
*
* This file is part of FFmpeg.

10
libavcodec/dvbsub_parser.c
View File

@ -172,9 +172,9 @@ static av_cold void dvbsub_parse_close(AVCodecParserContext *s)
}
AVCodecParser ff_dvbsub_parser = {
{ CODEC_ID_DVB_SUBTITLE },
sizeof(DVBSubParseContext),
dvbsub_parse_init,
dvbsub_parse,
dvbsub_parse_close,
.codec_ids = { CODEC_ID_DVB_SUBTITLE },
.priv_data_size = sizeof(DVBSubParseContext),
.parser_init = dvbsub_parse_init,
.parser_parse = dvbsub_parse,
.parser_close = dvbsub_parse_close,
};

2
libavcodec/dvbsubdec.c
View File

@ -1,5 +1,5 @@
/*
* DVB subtitle decoding for ffmpeg
* DVB subtitle decoding
* Copyright (c) 2005 Ian Caulfield
*
* This file is part of FFmpeg.

12
libavcodec/dvdsub_parser.c
View File

@ -1,5 +1,5 @@
/*
* DVD subtitle decoding for ffmpeg
* DVD subtitle decoding
* Copyright (c) 2005 Fabrice Bellard
*
* This file is part of FFmpeg.
@ -77,9 +77,9 @@ static av_cold void dvdsub_parse_close(AVCodecParserContext *s)
}
AVCodecParser ff_dvdsub_parser = {
{ CODEC_ID_DVD_SUBTITLE },
sizeof(DVDSubParseContext),
dvdsub_parse_init,
dvdsub_parse,
dvdsub_parse_close,
.codec_ids = { CODEC_ID_DVD_SUBTITLE },
.priv_data_size = sizeof(DVDSubParseContext),
.parser_init = dvdsub_parse_init,
.parser_parse = dvdsub_parse,
.parser_close = dvdsub_parse_close,
};

2
libavcodec/dvdsubdec.c
View File

@ -1,5 +1,5 @@
/*
* DVD subtitle decoding for ffmpeg
* DVD subtitle decoding
* Copyright (c) 2005 Fabrice Bellard
*
* This file is part of FFmpeg.

2
libavcodec/dvdsubenc.c
View File

@ -1,5 +1,5 @@
/*
* DVD subtitle encoding for ffmpeg
* DVD subtitle encoding
* Copyright (c) 2005 Wolfram Gloger
*
* This file is part of FFmpeg.

10
libavcodec/flac_parser.c
View File

@ -674,9 +674,9 @@ static void flac_parse_close(AVCodecParserContext *c)
}
AVCodecParser ff_flac_parser = {
{ CODEC_ID_FLAC },
sizeof(FLACParseContext),
flac_parse_init,
flac_parse,
flac_parse_close,
.codec_ids = { CODEC_ID_FLAC },
.priv_data_size = sizeof(FLACParseContext),
.parser_init = flac_parse_init,
.parser_parse = flac_parse,
.parser_close = flac_parse_close,
};

160
libavcodec/g726.c
View File

@ -22,7 +22,10 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <limits.h>
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "put_bits.h"
@ -71,6 +74,7 @@ typedef struct G726Tables {
} G726Tables;
typedef struct G726Context {
AVClass *class;
G726Tables tbls; /**< static tables needed for computation */
Float11 sr[2]; /**< prev. reconstructed samples */
@ -266,11 +270,11 @@ static int16_t g726_decode(G726Context* c, int I)
return av_clip(re_signal << 2, -0xffff, 0xffff);
}
static av_cold int g726_reset(G726Context* c, int index)
static av_cold int g726_reset(G726Context *c)
{
int i;
c->tbls = G726Tables_pool[index];
c->tbls = G726Tables_pool[c->code_size - 2];
for (i=0; i<2; i++) {
c->sr[i].mant = 1<<5;
c->pk[i] = 1;
@ -295,65 +299,59 @@ static int16_t g726_encode(G726Context* c, int16_t sig)
g726_decode(c, i);
return i;
}
#endif
/* Interfacing to the libavcodec */
static av_cold int g726_init(AVCodecContext * avctx)
static av_cold int g726_encode_init(AVCodecContext *avctx)
{
G726Context* c = avctx->priv_data;
unsigned int index;
if (avctx->sample_rate <= 0) {
av_log(avctx, AV_LOG_ERROR, "Samplerate is invalid\n");
return -1;
if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL &&
avctx->sample_rate != 8000) {
av_log(avctx, AV_LOG_ERROR, "Sample rates other than 8kHz are not "
"allowed when the compliance level is higher than unofficial. "
"Resample or reduce the compliance level.\n");
return AVERROR(EINVAL);
}
av_assert0(avctx->sample_rate > 0);
index = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate - 2;
if (avctx->bit_rate % avctx->sample_rate && avctx->codec->encode) {
av_log(avctx, AV_LOG_ERROR, "Bitrate - Samplerate combination is invalid\n");
return -1;
}
if(avctx->channels != 1){
av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
return -1;
return AVERROR(EINVAL);
}
if(index>3){
av_log(avctx, AV_LOG_ERROR, "Unsupported number of bits %d\n", index+2);
return -1;
}
g726_reset(c, index);
c->code_size = index+2;
if (avctx->bit_rate)
c->code_size = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate;
c->code_size = av_clip(c->code_size, 2, 5);
avctx->bit_rate = c->code_size * avctx->sample_rate;
avctx->bits_per_coded_sample = c->code_size;
g726_reset(c);
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
avctx->coded_frame->key_frame = 1;
if (avctx->codec->decode)
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
/* select a frame size that will end on a byte boundary and have a size of
approximately 1024 bytes */
if (avctx->codec->encode)
avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[index];
avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2];
return 0;
}
static av_cold int g726_close(AVCodecContext *avctx)
static av_cold int g726_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
}
#if CONFIG_ADPCM_G726_ENCODER
static int g726_encode_frame(AVCodecContext *avctx,
uint8_t *dst, int buf_size, void *data)
{
G726Context *c = avctx->priv_data;
const short *samples = data;
const int16_t *samples = data;
PutBitContext pb;
int i;
@ -366,8 +364,72 @@ static int g726_encode_frame(AVCodecContext *avctx,
return put_bits_count(&pb)>>3;
}
#define OFFSET(x) offsetof(G726Context, x)
#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
{ "code_size", "Bits per code", OFFSET(code_size), AV_OPT_TYPE_INT, { 4 }, 2, 5, AE },
{ NULL },
};
static const AVClass class = {
.class_name = "g726",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static const AVCodecDefault defaults[] = {
{ "b", "0" },
{ NULL },
};
AVCodec ff_adpcm_g726_encoder = {
.name = "g726",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ADPCM_G726,
.priv_data_size = sizeof(G726Context),
.init = g726_encode_init,
.encode = g726_encode_frame,
.close = g726_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
.priv_class = &class,
.defaults = defaults,
};
#endif
#if CONFIG_ADPCM_G726_DECODER
static av_cold int g726_decode_init(AVCodecContext *avctx)
{
G726Context* c = avctx->priv_data;
if (avctx->strict_std_compliance >= FF_COMPLIANCE_STRICT &&
avctx->sample_rate != 8000) {
av_log(avctx, AV_LOG_ERROR, "Only 8kHz sample rate is allowed when "
"the compliance level is strict. Reduce the compliance level "
"if you wish to decode the stream anyway.\n");
return AVERROR(EINVAL);
}
if(avctx->channels != 1){
av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
return AVERROR(EINVAL);
}
c->code_size = avctx->bits_per_coded_sample;
if (c->code_size < 2 || c->code_size > 5) {
av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size);
return AVERROR(EINVAL);
}
g726_reset(c);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
static int g726_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
@ -375,43 +437,43 @@ static int g726_decode_frame(AVCodecContext *avctx,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
G726Context *c = avctx->priv_data;
short *samples = data;
int16_t *samples = data;
GetBitContext gb;
int out_samples, out_size;
out_samples = buf_size * 8 / c->code_size;
out_size = out_samples * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
}
init_get_bits(&gb, buf, buf_size * 8);
while (get_bits_count(&gb) + c->code_size <= buf_size*8)
while (out_samples--)
*samples++ = g726_decode(c, get_bits(&gb, c->code_size));
if(buf_size*8 != get_bits_count(&gb))
if (get_bits_left(&gb) > 0)
av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
*data_size = (uint8_t*)samples - (uint8_t*)data;
*data_size = out_size;
return buf_size;
}
#if CONFIG_ADPCM_G726_ENCODER
AVCodec ff_adpcm_g726_encoder = {
.name = "g726",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ADPCM_G726,
.priv_data_size = sizeof(G726Context),
.init = g726_init,
.encode = g726_encode_frame,
.close = g726_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
};
#endif
static void g726_decode_flush(AVCodecContext *avctx)
{
G726Context *c = avctx->priv_data;
g726_reset(c);
}
AVCodec ff_adpcm_g726_decoder = {
.name = "g726",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ADPCM_G726,
.priv_data_size = sizeof(G726Context),
.init = g726_init,
.close = g726_close,
.init = g726_decode_init,
.decode = g726_decode_frame,
.flush = g726_decode_flush,
.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
};
#endif

23
libavcodec/gsmdec.c
View File

@ -58,13 +58,18 @@ static int gsm_decode_frame(AVCodecContext *avctx, void *data,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *samples = data;
int frame_bytes = 2 * avctx->frame_size;
int frame_bytes = avctx->frame_size *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < frame_bytes)
return -1;
*data_size = 0;
if(buf_size < avctx->block_align)
if (*data_size < frame_bytes) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
}
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
switch (avctx->codec_id) {
case CODEC_ID_GSM:
@ -84,6 +89,12 @@ static int gsm_decode_frame(AVCodecContext *avctx, void *data,
return avctx->block_align;
}
static void gsm_flush(AVCodecContext *avctx)
{
GSMContext *s = avctx->priv_data;
memset(s, 0, sizeof(*s));
}
AVCodec ff_gsm_decoder = {
.name = "gsm",
.type = AVMEDIA_TYPE_AUDIO,
@ -91,6 +102,7 @@ AVCodec ff_gsm_decoder = {
.priv_data_size = sizeof(GSMContext),
.init = gsm_init,
.decode = gsm_decode_frame,
.flush = gsm_flush,
.long_name = NULL_IF_CONFIG_SMALL("GSM"),
};
@ -101,5 +113,6 @@ AVCodec ff_gsm_ms_decoder = {
.priv_data_size = sizeof(GSMContext),
.init = gsm_init,
.decode = gsm_decode_frame,
.flush = gsm_flush,
.long_name = NULL_IF_CONFIG_SMALL("GSM Microsoft variant"),
};

9
libavcodec/h261_parser.c
View File

@ -86,9 +86,8 @@ static int h261_parse(AVCodecParserContext *s,
}
AVCodecParser ff_h261_parser = {
{ CODEC_ID_H261 },
sizeof(ParseContext),
NULL,
h261_parse,
ff_parse_close,
.codec_ids = { CODEC_ID_H261 },
.priv_data_size = sizeof(ParseContext),
.parser_parse = h261_parse,
.parser_close = ff_parse_close,
};

2
libavcodec/h263.c
View File

@ -1,5 +1,5 @@
/*
* H263/MPEG4 backend for ffmpeg encoder and decoder
* H263/MPEG4 backend for encoder and decoder
* Copyright (c) 2000,2001 Fabrice Bellard
* H263+ support.
* Copyright (c) 2001 Juan J. Sierralta P

9
libavcodec/h263_parser.c
View File

@ -88,9 +88,8 @@ static int h263_parse(AVCodecParserContext *s,
}
AVCodecParser ff_h263_parser = {
{ CODEC_ID_H263 },
sizeof(ParseContext),
NULL,
h263_parse,
ff_parse_close,
.codec_ids = { CODEC_ID_H263 },
.priv_data_size = sizeof(ParseContext),
.parser_parse = h263_parse,
.parser_close = ff_parse_close,
};

12
libavcodec/h264_parser.c
View File

@ -375,10 +375,10 @@ static int init(AVCodecParserContext *s)
}
AVCodecParser ff_h264_parser = {
{ CODEC_ID_H264 },
sizeof(H264Context),
init,
h264_parse,
close,
h264_split,
.codec_ids = { CODEC_ID_H264 },
.priv_data_size = sizeof(H264Context),
.parser_init = init,
.parser_parse = h264_parse,
.parser_close = close,
.split = h264_split,
};

2
libavcodec/idcinvideo.c
View File

@ -36,7 +36,7 @@
* a little more compression by exploiting the fact that adjacent pixels
* tend to be similar.
*
* Note that this decoder could use ffmpeg's optimized VLC facilities
* Note that this decoder could use libavcodec's optimized VLC facilities
* rather than naive, tree-based Huffman decoding. However, there are 256
* Huffman tables. Plus, the VLC bit coding order is right -> left instead
* or left -> right, so all of the bits would have to be reversed. Further,

51
libavcodec/imc.c
View File

@ -104,10 +104,15 @@ static VLC_TYPE vlc_tables[VLC_TABLES_SIZE][2];
static av_cold int imc_decode_init(AVCodecContext * avctx)
{
int i, j;
int i, j, ret;
IMCContext *q = avctx->priv_data;
double r1, r2;
if (avctx->channels != 1) {
av_log_ask_for_sample(avctx, "Number of channels is not supported\n");
return AVERROR_PATCHWELCOME;
}
q->decoder_reset = 1;
for(i = 0; i < BANDS; i++)
@ -156,10 +161,13 @@ static av_cold int imc_decode_init(AVCodecContext * avctx)
}
q->one_div_log2 = 1/log(2);
ff_fft_init(&q->fft, 7, 1);
if ((ret = ff_fft_init(&q->fft, 7, 1))) {
av_log(avctx, AV_LOG_INFO, "FFT init failed\n");
return ret;
}
dsputil_init(&q->dsp, avctx);
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
return 0;
}
@ -336,7 +344,7 @@ static int bit_allocation (IMCContext* q, int stream_format_code, int freebits,
indx = 2;
if (indx == -1)
return -1;
return AVERROR_INVALIDDATA;
q->flcoeffs4[i] = q->flcoeffs4[i] + xTab[(indx*2 + (q->flcoeffs1[i] < highest)) * 2 + flag];
}
@ -595,7 +603,7 @@ static int inverse_quant_coeff (IMCContext* q, int stream_format_code) {
middle_value = max_size >> 1;
if (q->codewords[j] >= max_size || q->codewords[j] < 0)
return -1;
return AVERROR_INVALIDDATA;
if (cw_len >= 4){
quantizer = imc_quantizer2[(stream_format_code & 2) >> 1];
@ -628,7 +636,7 @@ static int imc_get_coeffs (IMCContext* q) {
if (get_bits_count(&q->gb) + cw_len > 512){
//av_log(NULL,0,"Band %i coeff %i cw_len %i\n",i,j,cw_len);
return -1;
return AVERROR_INVALIDDATA;
}
if(cw_len && (!q->bandFlagsBuf[i] || !q->skipFlags[j]))
@ -651,18 +659,24 @@ static int imc_decode_frame(AVCodecContext * avctx,
IMCContext *q = avctx->priv_data;
int stream_format_code;
int imc_hdr, i, j;
int imc_hdr, i, j, out_size, ret;
int flag;
int bits, summer;
int counter, bitscount;
uint16_t buf16[IMC_BLOCK_SIZE / 2];
LOCAL_ALIGNED_16(uint16_t, buf16, [IMC_BLOCK_SIZE / 2]);
if (buf_size < IMC_BLOCK_SIZE) {
av_log(avctx, AV_LOG_ERROR, "imc frame too small!\n");
return -1;
return AVERROR_INVALIDDATA;
}
for(i = 0; i < IMC_BLOCK_SIZE / 2; i++)
buf16[i] = av_bswap16(((const uint16_t*)buf)[i]);
out_size = COEFFS * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
}
q->dsp.bswap16_buf(buf16, (const uint16_t*)buf, IMC_BLOCK_SIZE / 2);
q->out_samples = data;
init_get_bits(&q->gb, (const uint8_t*)buf16, IMC_BLOCK_SIZE * 8);
@ -672,13 +686,13 @@ static int imc_decode_frame(AVCodecContext * avctx,
if (imc_hdr != IMC_FRAME_ID) {
av_log(avctx, AV_LOG_ERROR, "imc frame header check failed!\n");
av_log(avctx, AV_LOG_ERROR, "got %x instead of 0x21.\n", imc_hdr);
return -1;
return AVERROR_INVALIDDATA;
}
stream_format_code = get_bits(&q->gb, 3);
if(stream_format_code & 1){
av_log(avctx, AV_LOG_ERROR, "Stream code format %X is not supported\n", stream_format_code);
return -1;
return AVERROR_INVALIDDATA;
}
// av_log(avctx, AV_LOG_DEBUG, "stream_format_code = %d\n", stream_format_code);
@ -738,10 +752,11 @@ static int imc_decode_frame(AVCodecContext * avctx,
}
}
if(bit_allocation (q, stream_format_code, 512 - bitscount - get_bits_count(&q->gb), flag) < 0) {
if((ret = bit_allocation (q, stream_format_code,
512 - bitscount - get_bits_count(&q->gb), flag)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Bit allocations failed\n");
q->decoder_reset = 1;
return -1;
return ret;
}
for(i = 0; i < BANDS; i++) {
@ -795,20 +810,20 @@ static int imc_decode_frame(AVCodecContext * avctx,
if(imc_get_coeffs(q) < 0) {
av_log(avctx, AV_LOG_ERROR, "Read coefficients failed\n");
q->decoder_reset = 1;
return 0;
return AVERROR_INVALIDDATA;
}
if(inverse_quant_coeff(q, stream_format_code) < 0) {
av_log(avctx, AV_LOG_ERROR, "Inverse quantization of coefficients failed\n");
q->decoder_reset = 1;
return 0;
return AVERROR_INVALIDDATA;
}
memset(q->skipFlags, 0, sizeof(q->skipFlags));
imc_imdct256(q);
*data_size = COEFFS * sizeof(float);
*data_size = out_size;
return IMC_BLOCK_SIZE;
}

1974
libavcodec/indeo3.c
View File

File diff suppressed because it is too large Load Diff

2640
libavcodec/indeo3data.h
View File

File diff suppressed because it is too large Load Diff

9
libavcodec/latm_parser.c
View File

@ -106,9 +106,8 @@ static int latm_parse(AVCodecParserContext *s1, AVCodecContext *avctx,
}
AVCodecParser ff_aac_latm_parser = {
{ CODEC_ID_AAC_LATM },
sizeof(LATMParseContext),
NULL,
latm_parse,
ff_parse_close
.codec_ids = { CODEC_ID_AAC_LATM },
.priv_data_size = sizeof(LATMParseContext),
.parser_parse = latm_parse,
.parser_close = ff_parse_close
};

40
libavcodec/libgsm.c
View File

@ -166,23 +166,39 @@ static av_cold int libgsm_decode_close(AVCodecContext *avctx) {
static int libgsm_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt) {
int i, ret;
struct gsm_state *s = avctx->priv_data;
uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
*data_size = 0; /* In case of error */
if(buf_size < avctx->block_align) return -1;
switch(avctx->codec_id) {
case CODEC_ID_GSM:
if(gsm_decode(avctx->priv_data,buf,data)) return -1;
*data_size = GSM_FRAME_SIZE*sizeof(int16_t);
break;
case CODEC_ID_GSM_MS:
if(gsm_decode(avctx->priv_data,buf,data) ||
gsm_decode(avctx->priv_data,buf+33,((int16_t*)data)+GSM_FRAME_SIZE)) return -1;
*data_size = GSM_FRAME_SIZE*sizeof(int16_t)*2;
int16_t *samples = data;
int out_size = avctx->frame_size * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
}
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
for (i = 0; i < avctx->frame_size / GSM_FRAME_SIZE; i++) {
if ((ret = gsm_decode(s, buf, samples)) < 0)
return -1;
buf += GSM_BLOCK_SIZE;
samples += GSM_FRAME_SIZE;
}
*data_size = out_size;
return avctx->block_align;
}
static void libgsm_flush(AVCodecContext *avctx) {
gsm_destroy(avctx->priv_data);
avctx->priv_data = gsm_create();
}
AVCodec ff_libgsm_decoder = {
.name = "libgsm",
.type = AVMEDIA_TYPE_AUDIO,
@ -190,6 +206,7 @@ AVCodec ff_libgsm_decoder = {
.init = libgsm_decode_init,
.close = libgsm_decode_close,
.decode = libgsm_decode_frame,
.flush = libgsm_flush,
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
};
@ -200,5 +217,6 @@ AVCodec ff_libgsm_ms_decoder = {
.init = libgsm_decode_init,
.close = libgsm_decode_close,
.decode = libgsm_decode_frame,
.flush = libgsm_flush,
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
};

2
libavcodec/libtheoraenc.c
View File

@ -241,7 +241,7 @@ static av_cold int encode_init(AVCodecContext* avc_context)
header, comment, and tables.
Each one is prefixed with a 16bit size, then they
are concatenated together into ffmpeg's extradata.
are concatenated together into libavcodec's extradata.
*/
offset = 0;

2
libavcodec/mace.c
View File

@ -27,7 +27,7 @@
#include "avcodec.h"
/*
* Adapted to ffmpeg by Francois Revol <revol@free.fr>
* Adapted to libavcodec by Francois Revol <revol@free.fr>
* (removed 68k REG stuff, changed types, added some statics and consts,
* libavcodec api, context stuff, interlaced stereo out).
*/

9
libavcodec/mjpeg_parser.c
View File

@ -127,9 +127,8 @@ static int jpeg_parse(AVCodecParserContext *s,
AVCodecParser ff_mjpeg_parser = {
{ CODEC_ID_MJPEG },
sizeof(MJPEGParserContext),
NULL,
jpeg_parse,
ff_parse_close,
.codec_ids = { CODEC_ID_MJPEG },
.priv_data_size = sizeof(MJPEGParserContext),
.parser_parse = jpeg_parse,
.parser_close = ff_parse_close,
};

10
libavcodec/mlp_parser.c
View File

@ -345,9 +345,9 @@ lost_sync:
}
AVCodecParser ff_mlp_parser = {
{ CODEC_ID_MLP, CODEC_ID_TRUEHD },
sizeof(MLPParseContext),
mlp_init,
mlp_parse,
ff_parse_close,
.codec_ids = { CODEC_ID_MLP, CODEC_ID_TRUEHD },
.priv_data_size = sizeof(MLPParseContext),
.parser_init = mlp_init,
.parser_parse = mlp_parse,
.parser_close = ff_parse_close,
};

12
libavcodec/mpeg4video_parser.c
View File

@ -131,10 +131,10 @@ static int mpeg4video_parse(AVCodecParserContext *s,
AVCodecParser ff_mpeg4video_parser = {
{ CODEC_ID_MPEG4 },
sizeof(ParseContext1),
mpeg4video_parse_init,
mpeg4video_parse,
ff_parse1_close,
ff_mpeg4video_split,
.codec_ids = { CODEC_ID_MPEG4 },
.priv_data_size = sizeof(ParseContext1),
.parser_init = mpeg4video_parse_init,
.parser_parse = mpeg4video_parse,
.parser_close = ff_parse1_close,
.split = ff_mpeg4video_split,
};

2
libavcodec/mpeg4videodec.c
View File

@ -1878,7 +1878,7 @@ static int decode_user_data(MpegEncContext *s, GetBitContext *gb){
}
}
/* ffmpeg detection */
/* libavcodec detection */
e=sscanf(buf, "FFmpe%*[^b]b%d", &build)+3;
if(e!=4)
e=sscanf(buf, "FFmpeg v%d.%d.%d / libavcodec build: %d", &ver, &ver2, &ver3, &build);

9
libavcodec/mpegaudio_parser.c
View File

@ -101,9 +101,8 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
AVCodecParser ff_mpegaudio_parser = {
{ CODEC_ID_MP1, CODEC_ID_MP2, CODEC_ID_MP3 },
sizeof(MpegAudioParseContext),
NULL,
mpegaudio_parse,
ff_parse_close,
.codec_ids = { CODEC_ID_MP1, CODEC_ID_MP2, CODEC_ID_MP3 },
.priv_data_size = sizeof(MpegAudioParseContext),
.parser_parse = mpegaudio_parse,
.parser_close = ff_parse_close,
};

11
libavcodec/mpegvideo_parser.c
View File

@ -177,10 +177,9 @@ static int mpegvideo_split(AVCodecContext *avctx,
}
AVCodecParser ff_mpegvideo_parser = {
{ CODEC_ID_MPEG1VIDEO, CODEC_ID_MPEG2VIDEO },
sizeof(ParseContext1),
NULL,
mpegvideo_parse,
ff_parse1_close,
mpegvideo_split,
.codec_ids = { CODEC_ID_MPEG1VIDEO, CODEC_ID_MPEG2VIDEO },
.priv_data_size = sizeof(ParseContext1),
.parser_parse = mpegvideo_parse,
.parser_close = ff_parse1_close,
.split = mpegvideo_split,
};

4
libavcodec/msmpeg4.c
View File

@ -1,5 +1,5 @@
/*
* MSMPEG4 backend for ffmpeg encoder and decoder
* MSMPEG4 backend for encoder and decoder
* Copyright (c) 2001 Fabrice Bellard
* Copyright (c) 2002-2004 Michael Niedermayer <michaelni@gmx.at>
*
@ -24,7 +24,7 @@
/**
* @file
* MSMPEG4 backend for ffmpeg encoder and decoder.
* MSMPEG4 backend for encoder and decoder
*/
#include "avcodec.h"

2
libavcodec/msmpeg4.h
View File

@ -1,5 +1,5 @@
/*
* MSMPEG4 backend for ffmpeg encoder and decoder
* MSMPEG4 backend for encoder and decoder
* copyright (c) 2007 Aurelien Jacobs <aurel@gnuage.org>
*
* This file is part of FFmpeg.

2
libavcodec/msmpeg4data.c
View File

@ -1,5 +1,5 @@
/*
* MSMPEG4 backend for ffmpeg encoder and decoder
* MSMPEG4 backend for encoder and decoder
* copyright (c) 2001 Fabrice Bellard
* copyright (c) 2002-2004 Michael Niedermayer <michaelni@gmx.at>
*

2
libavcodec/msmpeg4data.h
View File

@ -1,5 +1,5 @@
/*
* MSMPEG4 backend for ffmpeg encoder and decoder
* MSMPEG4 backend for encoder and decoder
* copyright (c) 2001 Fabrice Bellard
* copyright (c) 2002-2004 Michael Niedermayer <michaelni@gmx.at>
*

10
libavcodec/pnm_parser.c
View File

@ -84,9 +84,9 @@ retry:
}
AVCodecParser ff_pnm_parser = {
{ CODEC_ID_PGM, CODEC_ID_PGMYUV, CODEC_ID_PPM, CODEC_ID_PBM, CODEC_ID_PAM},
sizeof(ParseContext),
NULL,
pnm_parse,
ff_parse_close,
.codec_ids = { CODEC_ID_PGM, CODEC_ID_PGMYUV, CODEC_ID_PPM,
CODEC_ID_PBM, CODEC_ID_PAM },
.priv_data_size = sizeof(ParseContext),
.parser_parse = pnm_parse,
.parser_close = ff_parse_close,
};

4
libavcodec/rpza.c
View File

@ -30,8 +30,8 @@
* Note that this decoder reads big endian RGB555 pixel values from the
* bytestream, arranges them in the host's endian order, and outputs
* them to the final rendered map in the same host endian order. This is
* intended behavior as the ffmpeg documentation states that RGB555 pixels
* shall be stored in native CPU endianness.
* intended behavior as the libavcodec documentation states that RGB555
* pixels shall be stored in native CPU endianness.
*/
#include <stdio.h>

14
libavcodec/rv34_parser.c
View File

@ -78,18 +78,16 @@ static int rv34_parse(AVCodecParserContext *s,
#ifdef CONFIG_RV30_PARSER
AVCodecParser ff_rv30_parser = {
{ CODEC_ID_RV30 },
sizeof(RV34ParseContext),
NULL,
rv34_parse,
.codec_ids = { CODEC_ID_RV30 },
.priv_data_size = sizeof(RV34ParseContext),
.parser_parse = rv34_parse,
};
#endif
#ifdef CONFIG_RV40_PARSER
AVCodecParser ff_rv40_parser = {
{ CODEC_ID_RV40 },
sizeof(RV34ParseContext),
NULL,
rv34_parse,
.codec_ids = { CODEC_ID_RV40 },
.priv_data_size = sizeof(RV34ParseContext),
.parser_parse = rv34_parse,
};
#endif

2
libavcodec/sparc/dsputil_vis.c
View File

@ -19,7 +19,7 @@
*/
/* The *no_round* functions have been added by James A. Morrison, 2003,2004.
The vis code from libmpeg2 was adapted for ffmpeg by James A. Morrison.
The vis code from libmpeg2 was adapted for libavcodec by James A. Morrison.
*/
#include "config.h"

6
libavcodec/truemotion1data.h
View File

@ -2,9 +2,9 @@
* Duck Truemotion v1 Decoding Tables
*
* Data in this file was originally part of VpVision from On2 which is
* distributed under the GNU GPL. It is redistributed with ffmpeg under the
* GNU LGPL using the common understanding that data tables necessary for
* decoding algorithms are not necessarily licensable.
* distributed under the GNU GPL. It is redistributed with libavcodec under
* the GNU LGPL using the common understanding that data tables necessary
* for decoding algorithms are not necessarily copyrightable.
*
* This file is part of FFmpeg.
*

11
libavcodec/vc1_parser.c
View File

@ -185,10 +185,9 @@ static int vc1_split(AVCodecContext *avctx,
}
AVCodecParser ff_vc1_parser = {
{ CODEC_ID_VC1 },
sizeof(VC1ParseContext),
NULL,
vc1_parse,
ff_parse1_close,
vc1_split,
.codec_ids = { CODEC_ID_VC1 },
.priv_data_size = sizeof(VC1ParseContext),
.parser_parse = vc1_parse,
.parser_close = ff_parse1_close,
.split = vc1_split,
};

9
libavcodec/vp3.c
View File

@ -1588,9 +1588,6 @@ static av_cold int allocate_tables(AVCodecContext *avctx)
return 0;
}
/*
* This is the ffmpeg/libavcodec API init function.
*/
static av_cold int vp3_decode_init(AVCodecContext *avctx)
{
Vp3DecodeContext *s = avctx->priv_data;
@ -1842,9 +1839,6 @@ static int vp3_update_thread_context(AVCodecContext *dst, const AVCodecContext *
return 0;
}
/*
* This is the ffmpeg/libavcodec API frame decode function.
*/
static int vp3_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
@ -2002,9 +1996,6 @@ error:
static void vp3_decode_flush(AVCodecContext *avctx);
/*
* This is the ffmpeg/libavcodec API module cleanup function.
*/
static av_cold int vp3_decode_end(AVCodecContext *avctx)
{
Vp3DecodeContext *s = avctx->priv_data;

8
libavcodec/vp3_parser.c
View File

@ -36,9 +36,7 @@ static int parse(AVCodecParserContext *s,
}
AVCodecParser ff_vp3_parser = {
{ CODEC_ID_THEORA, CODEC_ID_VP3,
CODEC_ID_VP6, CODEC_ID_VP6F, CODEC_ID_VP6A },
0,
NULL,
parse,
.codec_ids = { CODEC_ID_THEORA, CODEC_ID_VP3, CODEC_ID_VP6,
CODEC_ID_VP6F, CODEC_ID_VP6A },
.parser_parse = parse,
};

6
libavcodec/vp8_parser.c
View File

@ -33,8 +33,6 @@ static int parse(AVCodecParserContext *s,
}
AVCodecParser ff_vp8_parser = {
{ CODEC_ID_VP8 },
0,
NULL,
parse,
.codec_ids = { CODEC_ID_VP8 },
.parser_parse = parse,
};

2
libavfilter/vf_drawtext.c
View File

@ -22,7 +22,7 @@
/**
* @file
* drawtext filter, based on the original FFmpeg vhook/drawtext.c
* drawtext filter, based on the original vhook/drawtext.c
* filter by Gustavo Sverzut Barbieri
*/

2
libavfilter/vsrc_testsrc.c
View File

@ -24,7 +24,7 @@
* Misc test sources.
*
* testsrc is based on the test pattern generator demuxer by Nicolas George:
* http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/2007-October/037845.html
* http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2007-October/037845.html
*
* rgbtestsrc is ported from MPlayer libmpcodecs/vf_rgbtest.c by
* Michael Niedermayer.

2
libavformat/Makefile
View File

@ -93,7 +93,7 @@ OBJS-$(CONFIG_FOURXM_DEMUXER) += 4xm.o
OBJS-$(CONFIG_FRAMECRC_MUXER) += framecrcenc.o
OBJS-$(CONFIG_FRAMEMD5_MUXER) += md5enc.o
OBJS-$(CONFIG_GIF_MUXER) += gif.o
OBJS-$(CONFIG_GSM_DEMUXER) += rawdec.o
OBJS-$(CONFIG_GSM_DEMUXER) += gsmdec.o
OBJS-$(CONFIG_GXF_DEMUXER) += gxf.o
OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o audiointerleave.o
OBJS-$(CONFIG_G722_DEMUXER) += rawdec.o

2
libavformat/asfdec.c
View File

@ -358,7 +358,7 @@ static int asf_read_stream_properties(AVFormatContext *s, int64_t size)
/* Extract palette from extradata if bpp <= 8 */
/* This code assumes that extradata contains only palette */
/* This is true for all paletted codecs implemented in ffmpeg */
/* This is true for all paletted codecs implemented in libavcodec */
if (st->codec->extradata_size && (st->codec->bits_per_coded_sample <= 8)) {
int av_unused i;
#if HAVE_BIGENDIAN

2
libavformat/au.c
View File

@ -35,7 +35,7 @@
/* if we don't know the size in advance */
#define AU_UNKNOWN_SIZE ((uint32_t)(~0))
/* The ffmpeg codecs we support, and the IDs they have in the file */
/* The libavcodec codecs we support, and the IDs they have in the file */
static const AVCodecTag codec_au_tags[] = {
{ CODEC_ID_PCM_MULAW, 1 },
{ CODEC_ID_PCM_S8, 2 },

2
libavformat/avio.c
View File

@ -1,5 +1,5 @@
/*
* Unbuffered io for ffmpeg system
* unbuffered I/O
* Copyright (c) 2001 Fabrice Bellard
*
* This file is part of FFmpeg.

2
libavformat/aviobuf.c
View File

@ -1,5 +1,5 @@
/*
* Buffered I/O for ffmpeg system
* buffered I/O
* Copyright (c) 2000,2001 Fabrice Bellard
*
* This file is part of FFmpeg.

2
libavformat/avisynth.c
View File

@ -1,5 +1,5 @@
/*
* AVISynth support for ffmpeg system
* AVISynth support
* Copyright (c) 2006 DivX, Inc.
*
* This file is part of FFmpeg.

2
libavformat/cutils.c
View File

@ -1,5 +1,5 @@
/*
* Various simple utilities for ffmpeg system
* various simple utilities for libavformat
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.

2
libavformat/dv.c
View File

@ -96,7 +96,7 @@ static const uint8_t* dv_extract_pack(uint8_t* frame, enum dv_pack_type t)
/*
* There's a couple of assumptions being made here:
* 1. By default we silence erroneous (0x8000/16bit 0x800/12bit) audio samples.
* We can pass them upwards when ffmpeg will be ready to deal with them.
* We can pass them upwards when libavcodec will be ready to deal with them.
* 2. We don't do software emphasis.
* 3. Audio is always returned as 16bit linear samples: 12bit nonlinear samples
* are converted into 16bit linear ones.

2
libavformat/file.c
View File

@ -1,5 +1,5 @@
/*
* Buffered file io for ffmpeg system
* buffered file I/O
* Copyright (c) 2001 Fabrice Bellard
*
* This file is part of FFmpeg.

17
libavformat/flvenc.c
View File

@ -60,10 +60,10 @@ typedef struct FLVContext {
int64_t duration_offset;
int64_t filesize_offset;
int64_t duration;
int64_t delay; ///< first dts delay (needed for AVC & Speex)
} FLVContext;
typedef struct FLVStreamContext {
int delay; ///< first dts delay for each stream (needed for AVC & Speex)
int64_t last_ts; ///< last timestamp for each stream
} FLVStreamContext;
@ -210,6 +210,8 @@ static int flv_write_header(AVFormatContext *s)
s->streams[i]->priv_data = sc;
sc->last_ts = -1;
}
flv->delay = AV_NOPTS_VALUE;
avio_write(pb, "FLV", 3);
avio_w8(pb,1);
avio_w8(pb, FLV_HEADER_FLAG_HASAUDIO * !!audio_enc
@ -444,10 +446,15 @@ static int flv_write_packet(AVFormatContext *s, AVPacket *pkt)
av_log(s, AV_LOG_ERROR, "malformated aac bitstream, use -absf aac_adtstoasc\n");
return -1;
}
if (!sc->delay && pkt->dts < 0)
sc->delay = -pkt->dts;
if (flv->delay == AV_NOPTS_VALUE)
flv->delay = -pkt->dts;
if (pkt->dts < -flv->delay) {
av_log(s, AV_LOG_WARNING, "Packets are not in the proper order with "
"respect to DTS\n");
return AVERROR(EINVAL);
}
ts = pkt->dts + sc->delay; // add delay to force positive dts
ts = pkt->dts + flv->delay; // add delay to force positive dts
/* check Speex packet duration */
if (enc->codec_id == CODEC_ID_SPEEX && ts - sc->last_ts > 160) {
@ -481,7 +488,7 @@ static int flv_write_packet(AVFormatContext *s, AVPacket *pkt)
avio_write(pb, data ? data : pkt->data, size);
avio_wb32(pb,size+flags_size+11); // previous tag size
flv->duration = FFMAX(flv->duration, pkt->pts + sc->delay + pkt->duration);
flv->duration = FFMAX(flv->duration, pkt->pts + flv->delay + pkt->duration);
avio_flush(pb);

132
libavformat/gsmdec.c Normal file
View File

@ -0,0 +1,132 @@
/*
* RAW GSM demuxer
* Copyright (c) 2011 Justin Ruggles
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "avformat.h"
#define GSM_BLOCK_SIZE 33
#define GSM_BLOCK_SAMPLES 160
#define GSM_SAMPLE_RATE 8000
typedef struct {
AVClass *class;
int sample_rate;
} GSMDemuxerContext;
static int gsm_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret, size;
size = GSM_BLOCK_SIZE * 32;
pkt->pos = avio_tell(s->pb);
pkt->stream_index = 0;
ret = av_get_packet(s->pb, pkt, size);
if (ret < GSM_BLOCK_SIZE) {
av_free_packet(pkt);
return ret < 0 ? ret : AVERROR(EIO);
}
pkt->size = ret;
pkt->duration = ret / GSM_BLOCK_SIZE;
pkt->pts = pkt->pos / GSM_BLOCK_SIZE;
return 0;
}
static int gsm_read_header(AVFormatContext *s, AVFormatParameters *ap)
{
GSMDemuxerContext *c = s->priv_data;
AVStream *st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = s->iformat->value;
st->codec->channels = 1;
st->codec->sample_rate = c->sample_rate;
st->codec->block_align = GSM_BLOCK_SIZE;
st->codec->bit_rate = GSM_BLOCK_SIZE * 8 * c->sample_rate / GSM_BLOCK_SAMPLES;
av_set_pts_info(st, 64, GSM_BLOCK_SAMPLES, GSM_SAMPLE_RATE);
return 0;
}
static int gsm_read_seek2(AVFormatContext *s, int stream_index, int64_t min_ts,
int64_t ts, int64_t max_ts, int flags)
{
GSMDemuxerContext *c = s->priv_data;
/* convert timestamps to file positions */
if (!(flags & AVSEEK_FLAG_BYTE)) {
if (stream_index < 0) {
AVRational bitrate_q = { GSM_BLOCK_SAMPLES, c->sample_rate * GSM_BLOCK_SIZE };
ts = av_rescale_q(ts, AV_TIME_BASE_Q, bitrate_q);
min_ts = av_rescale_q(min_ts, AV_TIME_BASE_Q, bitrate_q);
max_ts = av_rescale_q(max_ts, AV_TIME_BASE_Q, bitrate_q);
} else {
ts *= GSM_BLOCK_SIZE;
min_ts *= GSM_BLOCK_SIZE;
max_ts *= GSM_BLOCK_SIZE;
}
}
/* round to nearest block boundary */
ts = (ts + GSM_BLOCK_SIZE / 2) / GSM_BLOCK_SIZE * GSM_BLOCK_SIZE;
ts = FFMAX(0, ts);
/* handle min/max */
while (ts < min_ts)
ts += GSM_BLOCK_SIZE;
while (ts > max_ts)
ts -= GSM_BLOCK_SIZE;
if (ts < min_ts || ts > max_ts)
return -1;
return avio_seek(s->pb, ts, SEEK_SET);
}
static const AVOption options[] = {
{ "sample_rate", "", offsetof(GSMDemuxerContext, sample_rate),
AV_OPT_TYPE_INT, {.dbl = GSM_SAMPLE_RATE}, 1, INT_MAX / GSM_BLOCK_SIZE,
AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass class = {
.class_name = "gsm demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_gsm_demuxer = {
.name = "gsm",
.long_name = NULL_IF_CONFIG_SMALL("raw GSM"),
.priv_data_size = sizeof(GSMDemuxerContext),
.read_header = gsm_read_header,
.read_packet = gsm_read_packet,
.read_seek2 = gsm_read_seek2,
.extensions = "gsm",
.value = CODEC_ID_GSM,
.priv_class = &class,
};

2
libavformat/id3v2.c
View File

@ -392,7 +392,7 @@ typedef struct ID3v2EMFunc {
const char *tag3;
const char *tag4;
void (*read)(AVFormatContext*, AVIOContext*, int, char*, ID3v2ExtraMeta **);
void (*free)(void *);
void (*free)(void *obj);
} ID3v2EMFunc;
static const ID3v2EMFunc id3v2_extra_meta_funcs[] = {

2
libavformat/mpegts.c
View File

@ -1475,7 +1475,7 @@ static void pmt_cb(MpegTSFilter *filter, const uint8_t *section, int section_len
if (pid < 0)
break;
/* now create ffmpeg stream */
/* now create stream */
if (ts->pids[pid] && ts->pids[pid]->type == MPEGTS_PES) {
pes = ts->pids[pid]->u.pes_filter.opaque;
if (!pes->st) {

2
libavformat/os_support.c
View File

@ -1,5 +1,5 @@
/*
* Various utilities for ffmpeg system
* various OS-feature replacement utilities
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
* copyright (c) 2002 Francois Revol
*

2
libavformat/os_support.h
View File

@ -1,5 +1,5 @@
/*
* various utilities for ffmpeg system
* various OS-feature replacement utilities
* copyright (c) 2000, 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.

12
libavformat/rawdec.c
View File

@ -193,18 +193,6 @@ AVInputFormat ff_g722_demuxer = {
};
#endif
#if CONFIG_GSM_DEMUXER
AVInputFormat ff_gsm_demuxer = {
.name = "gsm",
.long_name = NULL_IF_CONFIG_SMALL("raw GSM"),
.read_header = ff_raw_audio_read_header,
.read_packet = ff_raw_read_partial_packet,
.flags= AVFMT_GENERIC_INDEX,
.extensions = "gsm",
.value = CODEC_ID_GSM,
};
#endif
#if CONFIG_LATM_DEMUXER
AVInputFormat ff_latm_demuxer = {
.name = "latm",

17
libavformat/riff.c
View File

@ -400,11 +400,13 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
avio_wl32(pb, enc->sample_rate);
if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3 || enc->codec_id == CODEC_ID_GSM_MS) {
bps = 0;
} else if (enc->codec_id == CODEC_ID_ADPCM_G726) {
bps = 4;
} else {
if (!(bps = av_get_bits_per_sample(enc->codec_id)))
bps = 16; // default to 16
if (!(bps = av_get_bits_per_sample(enc->codec_id))) {
if (enc->bits_per_coded_sample)
bps = enc->bits_per_coded_sample;
else
bps = 16; // default to 16
}
}
if(bps != enc->bits_per_coded_sample && enc->bits_per_coded_sample){
av_log(enc, AV_LOG_WARNING, "requested bits_per_coded_sample (%d) and actually stored (%d) differ\n", enc->bits_per_coded_sample, bps);
@ -415,12 +417,10 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
//blkalign = 144 * enc->bit_rate/enc->sample_rate;
} else if (enc->codec_id == CODEC_ID_AC3) {
blkalign = 3840; //maximum bytes per frame
} else if (enc->codec_id == CODEC_ID_ADPCM_G726) { //
blkalign = 1;
} else if (enc->block_align != 0) { /* specified by the codec */
blkalign = enc->block_align;
} else
blkalign = enc->channels*bps >> 3;
blkalign = bps * enc->channels / av_gcd(8, bps);
if (enc->codec_id == CODEC_ID_PCM_U8 ||
enc->codec_id == CODEC_ID_PCM_S24LE ||
enc->codec_id == CODEC_ID_PCM_S32LE ||
@ -572,6 +572,9 @@ int ff_get_wav_header(AVIOContext *pb, AVCodecContext *codec, int size)
codec->channels = 0;
codec->sample_rate = 0;
}
/* override bits_per_coded_sample for G.726 */
if (codec->codec_id == CODEC_ID_ADPCM_G726)
codec->bits_per_coded_sample = codec->bit_rate / codec->sample_rate;
return 0;
}

2
libavformat/rso.h
View File

@ -26,7 +26,7 @@
#define RSO_HEADER_SIZE 8
/* The ffmpeg codecs we support, and the IDs they have in the file */
/* The libavcodec codecs we support, and the IDs they have in the file */
extern const AVCodecTag ff_codec_rso_tags[];
#endif /* AVFORMAT_RSO_H */

12
libavformat/rtsp.c
View File

@ -65,6 +65,12 @@
{ name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
{ "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
#define RTSP_MEDIATYPE_OPTS(name, longname) \
{ name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
{ "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
{ "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
{ "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
const AVOption ff_rtsp_options[] = {
{ "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
@ -74,11 +80,13 @@ const AVOption ff_rtsp_options[] = {
{ "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
{ "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
{ NULL },
};
static const AVOption sdp_options[] = {
RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
{ NULL },
};
@ -325,6 +333,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
case 'm':
/* new stream */
s1->skip_media = 0;
codec_type = AVMEDIA_TYPE_UNKNOWN;
get_word(st_type, sizeof(st_type), &p);
if (!strcmp(st_type, "audio")) {
codec_type = AVMEDIA_TYPE_AUDIO;
@ -332,7 +341,8 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
codec_type = AVMEDIA_TYPE_VIDEO;
} else if (!strcmp(st_type, "application")) {
codec_type = AVMEDIA_TYPE_DATA;
} else {
}
if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
s1->skip_media = 1;
return;
}

5
libavformat/rtsp.h
View File

@ -354,6 +354,11 @@ typedef struct RTSPState {
* Various option flags for the RTSP muxer/demuxer.
*/
int rtsp_flags;
/**
* Mask of all requested media types
*/
int media_type_mask;
} RTSPState;
#define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -

8
libavutil/log.c
View File

@ -52,17 +52,17 @@ static void colored_fputs(int level, const char *str){
#if defined(_WIN32) && !defined(__MINGW32CE__)
CONSOLE_SCREEN_BUFFER_INFO con_info;
con = GetStdHandle(STD_ERROR_HANDLE);
use_color = (con != INVALID_HANDLE_VALUE) && !getenv("NO_COLOR") && !getenv("FFMPEG_FORCE_NOCOLOR");
use_color = (con != INVALID_HANDLE_VALUE) && !getenv("NO_COLOR") && !getenv("AV_LOG_FORCE_NOCOLOR");
if (use_color) {
GetConsoleScreenBufferInfo(con, &con_info);
attr_orig = con_info.wAttributes;
background = attr_orig & 0xF0;
}
#elif HAVE_ISATTY
use_color= !getenv("NO_COLOR") && !getenv("FFMPEG_FORCE_NOCOLOR") &&
(getenv("TERM") && isatty(2) || getenv("FFMPEG_FORCE_COLOR"));
use_color= !getenv("NO_COLOR") && !getenv("AV_LOG_FORCE_NOCOLOR") &&
(getenv("TERM") && isatty(2) || getenv("AV_LOG_FORCE_COLOR"));
#else
use_color= getenv("FFMPEG_FORCE_COLOR") && !getenv("NO_COLOR") && !getenv("FFMPEG_FORCE_NOCOLOR");
use_color= getenv("AV_LOG_FORCE_COLOR") && !getenv("NO_COLOR") && !getenv("AV_LOG_FORCE_NOCOLOR");
#endif
}

4
libavutil/parseutils.c
View File

@ -411,9 +411,7 @@ static int date_get_num(const char **pp,
* function call, or NULL in case the function fails to match all of
* the fmt string and therefore an error occurred
*/
static
const char *small_strptime(const char *p, const char *fmt,
struct tm *dt)
static const char *small_strptime(const char *p, const char *fmt, struct tm *dt)
{
int c, val;