Merge remote-tracking branch 'qatar/master'

* 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>

Changelog

@@ -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:
 

libavcodec/aac_parser.c

@@ -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,
 };

libavcodec/ac3_parser.c

@@ -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,
 };

libavcodec/ac3enc.c

@@ -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;

libavcodec/ac3enc_fixed.c

@@ -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)

libavcodec/ac3enc_float.c

@@ -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)

libavcodec/ac3enc_template.c

@@ -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)
 {

libavcodec/cavs_parser.c

@@ -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,
 };

libavcodec/dca_parser.c

@@ -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,
 };

libavcodec/dirac_parser.c

@@ -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,
 };

libavcodec/dnxhd_parser.c

@@ -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,
 };

libavcodec/dvbsub.c

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

libavcodec/dvbsub_parser.c

@@ -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,
 };

libavcodec/dvbsubdec.c

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

libavcodec/dvdsub_parser.c

@@ -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,
 };

libavcodec/dvdsubdec.c

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

libavcodec/dvdsubenc.c

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

libavcodec/flac_parser.c

@@ -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,
 };

libavcodec/g726.c

@@ -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;
-    }
-    if(index>3){
-        av_log(avctx, AV_LOG_ERROR, "Unsupported number of bits %d\n", index+2);
-        return -1;
+        return AVERROR(EINVAL);
     }
-    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

libavcodec/gsmdec.c

@@ -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"),
 };

libavcodec/h261_parser.c

@@ -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,
 };

libavcodec/h263.c

@@ -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

libavcodec/h263_parser.c

@@ -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,
 };

libavcodec/h264_parser.c

@@ -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,
 };

libavcodec/idcinvideo.c

@@ -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,

libavcodec/imc.c

@@ -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;
 }

libavcodec/latm_parser.c

@@ -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
 };

libavcodec/libgsm.c

@@ -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"),
 };

libavcodec/libtheoraenc.c

@@ -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;
 

libavcodec/mace.c

@@ -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).
  */

libavcodec/mjpeg_parser.c

@@ -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,
 };

libavcodec/mlp_parser.c

@@ -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,
 };

libavcodec/mpeg4video_parser.c

@@ -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,
 };

libavcodec/mpeg4videodec.c

@@ -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);

libavcodec/mpegaudio_parser.c

@@ -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,
 };

libavcodec/mpegvideo_parser.c

@@ -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,
 };

libavcodec/msmpeg4.c

@@ -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"

libavcodec/msmpeg4.h

@@ -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.

libavcodec/msmpeg4data.c

@@ -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>
  *

libavcodec/msmpeg4data.h

@@ -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>
  *

libavcodec/pnm_parser.c

@@ -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,
 };

libavcodec/rpza.c

@@ -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>

libavcodec/rv34_parser.c

@@ -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

libavcodec/sparc/dsputil_vis.c

@@ -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"

libavcodec/truemotion1data.h

@@ -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.
  *

libavcodec/vc1_parser.c

@@ -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,
 };

libavcodec/vp3.c

@@ -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;

libavcodec/vp3_parser.c

@@ -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,
 };

libavcodec/vp8_parser.c

@@ -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,
 };

libavfilter/vf_drawtext.c

@@ -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
  */
 

libavfilter/vsrc_testsrc.c

@@ -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.

libavformat/Makefile

@@ -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

libavformat/asfdec.c

@@ -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

libavformat/au.c

@@ -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 },

libavformat/avio.c

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

libavformat/aviobuf.c

@@ -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.

libavformat/avisynth.c

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

libavformat/cutils.c

@@ -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.

libavformat/dv.c

@@ -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.

libavformat/file.c

@@ -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.

libavformat/flvenc.c

@@ -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);
 

libavformat/gsmdec.c

@@ -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,
+};

libavformat/id3v2.c

@@ -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[] = {

libavformat/mpegts.c

@@ -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) {

libavformat/os_support.c

@@ -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
  *

libavformat/os_support.h

@@ -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.

libavformat/rawdec.c

@@ -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",

libavformat/riff.c

@@ -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;
 }

libavformat/rso.h

@@ -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 */

libavformat/rtsp.c

@@ -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;
         }

libavformat/rtsp.h

@@ -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 -

libavutil/log.c

@@ -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
     }
 

libavutil/parseutils.c

@@ -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;