diff options
| author | Michael Niedermayer <michaelni@gmx.at> | 2011-10-29 02:08:54 +0200 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2011-10-29 02:08:54 +0200 |
| commit | 6faf0a21e18f314c48a886864145abe715be6572 (patch) | |
| tree | f67c3e543a8b2c3283875881536d0a69da515e5e /libavcodec | |
| parent | ed1aa8921749a1c70d4453326da7f7b5a6f6f6e7 (diff) | |
| parent | 61856d06eb30955290911140e6745bad93a25323 (diff) | |
| download | ffmpeg-6faf0a21e18f314c48a886864145abe715be6572.tar.gz | |
Merge remote-tracking branch 'qatar/master'
* qatar/master: (53 commits)
probe: Restore identification of files with very large id3 tags and no extension.
probe: Remove id3 tag presence as a criteria to do file extension checking.
mpegts: MP4 SL support
mpegts: MP4 OD support
mpegts: Add support for Sections in PMT
mpegts: Replace the MP4 descriptor parser with a recursive parser.
mpegts: Add support for multiple mp4 descriptors
mpegts: Parse mpeg2 SL descriptors.
isom: Add MPEG4SYSTEMS dummy object type indication.
aacdec: allow output reconfiguration on channel changes
nellymoserenc: take float input samples instead of int16
nellymoserdec: use dsp functions for overlap and windowing
nellymoserdec: do not fail if there is extra data in the packet
nellymoserdec: fail if output buffer is too small
nellymoserdec: remove pointless buffer size check.
lavf: add init_put_byte() to the list of visible symbols.
seek-test: free options dictionary after use
snow: do not draw_edge if emu_edge is set
tools/pktdumper: update to recent avformat api
seek-test: update to recent avformat api
...
Conflicts:
doc/APIchanges
libavcodec/mpegaudiodec.c
libavcodec/nellymoserdec.c
libavcodec/snow.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/avformat.h
libavformat/mpegts.c
libavformat/mxfdec.c
libavformat/utils.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec')
| -rw-r--r-- | libavcodec/apedec.c | 209 | ||||
| -rw-r--r-- | libavcodec/avcodec.h | 20 | ||||
| -rw-r--r-- | libavcodec/mpegaudiodec.c | 925 | ||||
| -rw-r--r-- | libavcodec/mpegaudiodec_float.c | 8 | ||||
| -rw-r--r-- | libavcodec/nellymoserdec.c | 42 | ||||
| -rw-r--r-- | libavcodec/nellymoserenc.c | 12 | ||||
| -rw-r--r-- | libavcodec/snow.c | 2 | ||||
| -rw-r--r-- | libavcodec/version.h | 5 | ||||
| -rw-r--r-- | libavcodec/wmadec.c | 25 | ||||
| -rw-r--r-- | libavcodec/wmaprodec.c | 35 | ||||
| -rw-r--r-- | libavcodec/wmavoice.c | 33 |
11 files changed, 692 insertions, 624 deletions
diff --git a/libavcodec/apedec.c b/libavcodec/apedec.c index 300a0097d8..9d2ce1dfaa 100644 --- a/libavcodec/apedec.c +++ b/libavcodec/apedec.c @@ -26,6 +26,7 @@ #include "get_bits.h" #include "bytestream.h" #include "libavutil/audioconvert.h" +#include "libavutil/avassert.h" /** * @file @@ -163,22 +164,34 @@ typedef struct APEContext { // TODO: dsputilize -static av_cold int ape_decode_init(AVCodecContext * avctx) +static av_cold int ape_decode_close(AVCodecContext *avctx) +{ + APEContext *s = avctx->priv_data; + int i; + + for (i = 0; i < APE_FILTER_LEVELS; i++) + av_freep(&s->filterbuf[i]); + + av_freep(&s->data); + return 0; +} + +static av_cold int ape_decode_init(AVCodecContext *avctx) { APEContext *s = avctx->priv_data; int i; if (avctx->extradata_size != 6) { av_log(avctx, AV_LOG_ERROR, "Incorrect extradata\n"); - return -1; + return AVERROR(EINVAL); } if (avctx->bits_per_coded_sample != 16) { av_log(avctx, AV_LOG_ERROR, "Only 16-bit samples are supported\n"); - return -1; + return AVERROR(EINVAL); } if (avctx->channels > 2) { av_log(avctx, AV_LOG_ERROR, "Only mono and stereo is supported\n"); - return -1; + return AVERROR(EINVAL); } s->avctx = avctx; s->channels = avctx->channels; @@ -186,34 +199,29 @@ static av_cold int ape_decode_init(AVCodecContext * avctx) s->compression_level = AV_RL16(avctx->extradata + 2); s->flags = AV_RL16(avctx->extradata + 4); - av_log(avctx, AV_LOG_DEBUG, "Compression Level: %d - Flags: %d\n", s->compression_level, s->flags); + av_log(avctx, AV_LOG_DEBUG, "Compression Level: %d - Flags: %d\n", + s->compression_level, s->flags); if (s->compression_level % 1000 || s->compression_level > COMPRESSION_LEVEL_INSANE) { - av_log(avctx, AV_LOG_ERROR, "Incorrect compression level %d\n", s->compression_level); - return -1; + av_log(avctx, AV_LOG_ERROR, "Incorrect compression level %d\n", + s->compression_level); + return AVERROR_INVALIDDATA; } s->fset = s->compression_level / 1000 - 1; for (i = 0; i < APE_FILTER_LEVELS; i++) { if (!ape_filter_orders[s->fset][i]) break; - s->filterbuf[i] = av_malloc((ape_filter_orders[s->fset][i] * 3 + HISTORY_SIZE) * 4); + FF_ALLOC_OR_GOTO(avctx, s->filterbuf[i], + (ape_filter_orders[s->fset][i] * 3 + HISTORY_SIZE) * 4, + filter_alloc_fail); } dsputil_init(&s->dsp, avctx); avctx->sample_fmt = AV_SAMPLE_FMT_S16; avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO; return 0; -} - -static av_cold int ape_decode_close(AVCodecContext * avctx) -{ - APEContext *s = avctx->priv_data; - int i; - - for (i = 0; i < APE_FILTER_LEVELS; i++) - av_freep(&s->filterbuf[i]); - - av_freep(&s->data); - return 0; +filter_alloc_fail: + ape_decode_close(avctx); + return AVERROR(ENOMEM); } /** @@ -228,7 +236,7 @@ static av_cold int ape_decode_close(AVCodecContext * avctx) #define BOTTOM_VALUE (TOP_VALUE >> 8) /** Start the decoder */ -static inline void range_start_decoding(APEContext * ctx) +static inline void range_start_decoding(APEContext *ctx) { ctx->rc.buffer = bytestream_get_byte(&ctx->ptr); ctx->rc.low = ctx->rc.buffer >> (8 - EXTRA_BITS); @@ -236,13 +244,16 @@ static inline void range_start_decoding(APEContext * ctx) } /** Perform normalization */ -static inline void range_dec_normalize(APEContext * ctx) +static inline void range_dec_normalize(APEContext *ctx) { while (ctx->rc.range <= BOTTOM_VALUE) { ctx->rc.buffer <<= 8; - if(ctx->ptr < ctx->data_end) + if(ctx->ptr < ctx->data_end) { ctx->rc.buffer += *ctx->ptr; - ctx->ptr++; + ctx->ptr++; + } else { + ctx->error = 1; + } ctx->rc.low = (ctx->rc.low << 8) | ((ctx->rc.buffer >> 1) & 0xFF); ctx->rc.range <<= 8; } @@ -254,7 +265,7 @@ static inline void range_dec_normalize(APEContext * ctx) * @param tot_f is the total frequency or (code_value)1<<shift * @return the culmulative frequency */ -static inline int range_decode_culfreq(APEContext * ctx, int tot_f) +static inline int range_decode_culfreq(APEContext *ctx, int tot_f) { range_dec_normalize(ctx); ctx->rc.help = ctx->rc.range / tot_f; @@ -266,7 +277,7 @@ static inline int range_decode_culfreq(APEContext * ctx, int tot_f) * @param ctx decoder context * @param shift number of bits to decode */ -static inline int range_decode_culshift(APEContext * ctx, int shift) +static inline int range_decode_culshift(APEContext *ctx, int shift) { range_dec_normalize(ctx); ctx->rc.help = ctx->rc.range >> shift; @@ -280,14 +291,14 @@ static inline int range_decode_culshift(APEContext * ctx, int shift) * @param sy_f the interval length (frequency of the symbol) * @param lt_f the lower end (frequency sum of < symbols) */ -static inline void range_decode_update(APEContext * ctx, int sy_f, int lt_f) +static inline void range_decode_update(APEContext *ctx, int sy_f, int lt_f) { ctx->rc.low -= ctx->rc.help * lt_f; ctx->rc.range = ctx->rc.help * sy_f; } /** Decode n bits (n <= 16) without modelling */ -static inline int range_decode_bits(APEContext * ctx, int n) +static inline int range_decode_bits(APEContext *ctx, int n) { int sym = range_decode_culshift(ctx, n); range_decode_update(ctx, 1, sym); @@ -339,7 +350,7 @@ static const uint16_t counts_diff_3980[21] = { * @param counts probability range start position * @param counts_diff probability range widths */ -static inline int range_get_symbol(APEContext * ctx, +static inline int range_get_symbol(APEContext *ctx, const uint16_t counts[], const uint16_t counts_diff[]) { @@ -374,7 +385,7 @@ static inline void update_rice(APERice *rice, int x) rice->k++; } -static inline int ape_decode_value(APEContext * ctx, APERice *rice) +static inline int ape_decode_value(APEContext *ctx, APERice *rice) { int x, overflow; @@ -441,7 +452,7 @@ static inline int ape_decode_value(APEContext * ctx, APERice *rice) return -(x >> 1); } -static void entropy_decode(APEContext * ctx, int blockstodecode, int stereo) +static void entropy_decode(APEContext *ctx, int blockstodecode, int stereo) { int32_t *decoded0 = ctx->decoded0; int32_t *decoded1 = ctx->decoded1; @@ -464,9 +475,11 @@ static void entropy_decode(APEContext * ctx, int blockstodecode, int stereo) range_dec_normalize(ctx); /* normalize to use up all bytes */ } -static void init_entropy_decoder(APEContext * ctx) +static int init_entropy_decoder(APEContext *ctx) { /* Read the CRC */ + if (ctx->data_end - ctx->ptr < 6) + return AVERROR_INVALIDDATA; ctx->CRC = bytestream_get_be32(&ctx->ptr); /* Read the frame flags if they exist */ @@ -474,6 +487,8 @@ static void init_entropy_decoder(APEContext * ctx) if ((ctx->fileversion > 3820) && (ctx->CRC & 0x80000000)) { ctx->CRC &= ~0x80000000; + if (ctx->data_end - ctx->ptr < 6) + return AVERROR_INVALIDDATA; ctx->frameflags = bytestream_get_be32(&ctx->ptr); } @@ -490,13 +505,15 @@ static void init_entropy_decoder(APEContext * ctx) ctx->ptr++; range_start_decoding(ctx); + + return 0; } static const int32_t initial_coeffs[4] = { 360, 317, -109, 98 }; -static void init_predictor_decoder(APEContext * ctx) +static void init_predictor_decoder(APEContext *ctx) { APEPredictor *p = &ctx->predictor; @@ -519,7 +536,10 @@ static inline int APESIGN(int32_t x) { return (x < 0) - (x > 0); } -static av_always_inline int predictor_update_filter(APEPredictor *p, const int decoded, const int filter, const int delayA, const int delayB, const int adaptA, const int adaptB) +static av_always_inline int predictor_update_filter(APEPredictor *p, + const int decoded, const int filter, + const int delayA, const int delayB, + const int adaptA, const int adaptB) { int32_t predictionA, predictionB, sign; @@ -563,7 +583,7 @@ static av_always_inline int predictor_update_filter(APEPredictor *p, const int d return p->filterA[filter]; } -static void predictor_decode_stereo(APEContext * ctx, int count) +static void predictor_decode_stereo(APEContext *ctx, int count) { APEPredictor *p = &ctx->predictor; int32_t *decoded0 = ctx->decoded0; @@ -571,9 +591,11 @@ static void predictor_decode_stereo(APEContext * ctx, int count) while (count--) { /* Predictor Y */ - *decoded0 = predictor_update_filter(p, *decoded0, 0, YDELAYA, YDELAYB, YADAPTCOEFFSA, YADAPTCOEFFSB); + *decoded0 = predictor_update_filter(p, *decoded0, 0, YDELAYA, YDELAYB, + YADAPTCOEFFSA, YADAPTCOEFFSB); decoded0++; - *decoded1 = predictor_update_filter(p, *decoded1, 1, XDELAYA, XDELAYB, XADAPTCOEFFSA, XADAPTCOEFFSB); + *decoded1 = predictor_update_filter(p, *decoded1, 1, XDELAYA, XDELAYB, + XADAPTCOEFFSA, XADAPTCOEFFSB); decoded1++; /* Combined */ @@ -587,7 +609,7 @@ static void predictor_decode_stereo(APEContext * ctx, int count) } } -static void predictor_decode_mono(APEContext * ctx, int count) +static void predictor_decode_mono(APEContext *ctx, int count) { APEPredictor *p = &ctx->predictor; int32_t *decoded0 = ctx->decoded0; @@ -632,7 +654,7 @@ static void predictor_decode_mono(APEContext * ctx, int count) p->lastA[0] = currentA; } -static void do_init_filter(APEFilter *f, int16_t * buf, int order) +static void do_init_filter(APEFilter *f, int16_t *buf, int order) { f->coeffs = buf; f->historybuffer = buf + order; @@ -644,20 +666,23 @@ static void do_init_filter(APEFilter *f, int16_t * buf, int order) f->avg = 0; } -static void init_filter(APEContext * ctx, APEFilter *f, int16_t * buf, int order) +static void init_filter(APEContext *ctx, APEFilter *f, int16_t *buf, int order) { do_init_filter(&f[0], buf, order); do_init_filter(&f[1], buf + order * 3 + HISTORY_SIZE, order); } -static void do_apply_filter(APEContext * ctx, int version, APEFilter *f, int32_t *data, int count, int order, int fracbits) +static void do_apply_filter(APEContext *ctx, int version, APEFilter *f, + int32_t *data, int count, int order, int fracbits) { int res; int absres; while (count--) { /* round fixedpoint scalar product */ - res = ctx->dsp.scalarproduct_and_madd_int16(f->coeffs, f->delay - order, f->adaptcoeffs - order, order, APESIGN(*data)); + res = ctx->dsp.scalarproduct_and_madd_int16(f->coeffs, f->delay - order, + f->adaptcoeffs - order, + order, APESIGN(*data)); res = (res + (1 << (fracbits - 1))) >> fracbits; res += *data; *data++ = res; @@ -676,7 +701,8 @@ static void do_apply_filter(APEContext * ctx, int version, APEFilter *f, int32_t /* Update the adaption coefficients */ absres = FFABS(res); if (absres) - *f->adaptcoeffs = ((res & (1<<31)) - (1<<30)) >> (25 + (absres <= f->avg*3) + (absres <= f->avg*4/3)); + *f->adaptcoeffs = ((res & (1<<31)) - (1<<30)) >> + (25 + (absres <= f->avg*3) + (absres <= f->avg*4/3)); else *f->adaptcoeffs = 0; @@ -699,8 +725,8 @@ static void do_apply_filter(APEContext * ctx, int version, APEFilter *f, int32_t } } -static void apply_filter(APEContext * ctx, APEFilter *f, - int32_t * data0, int32_t * data1, +static void apply_filter(APEContext *ctx, APEFilter *f, + int32_t *data0, int32_t *data1, int count, int order, int fracbits) { do_apply_filter(ctx, ctx->fileversion, &f[0], data0, count, order, fracbits); @@ -708,34 +734,38 @@ static void apply_filter(APEContext * ctx, APEFilter *f, do_apply_filter(ctx, ctx->fileversion, &f[1], data1, count, order, fracbits); } -static void ape_apply_filters(APEContext * ctx, int32_t * decoded0, - int32_t * decoded1, int count) +static void ape_apply_filters(APEContext *ctx, int32_t *decoded0, + int32_t *decoded1, int count) { int i; for (i = 0; i < APE_FILTER_LEVELS; i++) { if (!ape_filter_orders[ctx->fset][i]) break; - apply_filter(ctx, ctx->filters[i], decoded0, decoded1, count, ape_filter_orders[ctx->fset][i], ape_filter_fracbits[ctx->fset][i]); + apply_filter(ctx, ctx->filters[i], decoded0, decoded1, count, + ape_filter_orders[ctx->fset][i], + ape_filter_fracbits[ctx->fset][i]); } } -static void init_frame_decoder(APEContext * ctx) +static int init_frame_decoder(APEContext *ctx) { - int i; - init_entropy_decoder(ctx); + int i, ret; + if ((ret = init_entropy_decoder(ctx)) < 0) + return ret; init_predictor_decoder(ctx); for (i = 0; i < APE_FILTER_LEVELS; i++) { if (!ape_filter_orders[ctx->fset][i]) break; - init_filter(ctx, ctx->filters[i], ctx->filterbuf[i], ape_filter_orders[ctx->fset][i]); + init_filter(ctx, ctx->filters[i], ctx->filterbuf[i], + ape_filter_orders[ctx->fset][i]); } + return 0; } -static void ape_unpack_mono(APEContext * ctx, int count) +static void ape_unpack_mono(APEContext *ctx, int count) { - int32_t left; int32_t *decoded0 = ctx->decoded0; int32_t *decoded1 = ctx->decoded1; @@ -754,14 +784,11 @@ static void ape_unpack_mono(APEContext * ctx, int count) /* Pseudo-stereo - just copy left channel to right channel */ if (ctx->channels == 2) { - while (count--) { - left = *decoded0; - *(decoded1++) = *(decoded0++) = left; - } + memcpy(decoded1, decoded0, count * sizeof(*decoded1)); } } -static void ape_unpack_stereo(APEContext * ctx, int count) +static void ape_unpack_stereo(APEContext *ctx, int count) { int32_t left, right; int32_t *decoded0 = ctx->decoded0; @@ -789,7 +816,7 @@ static void ape_unpack_stereo(APEContext * ctx, int count) } } -static int ape_decode_frame(AVCodecContext * avctx, +static int ape_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) { @@ -797,49 +824,65 @@ static int ape_decode_frame(AVCodecContext * avctx, int buf_size = avpkt->size; APEContext *s = avctx->priv_data; int16_t *samples = data; - int nblocks; - int i, n; + uint32_t nblocks; + int i; int blockstodecode; int bytes_used; - if (buf_size == 0 && !s->samples) { - *data_size = 0; - return 0; - } - /* should not happen but who knows */ if (BLOCKS_PER_LOOP * 2 * avctx->channels > *data_size) { - av_log (avctx, AV_LOG_ERROR, "Packet size is too big to be handled in lavc! (max is %d where you have %d)\n", *data_size, s->samples * 2 * avctx->channels); - return -1; + av_log (avctx, AV_LOG_ERROR, "Output buffer is too small.\n"); + return AVERROR(EINVAL); } + /* this should never be negative, but bad things will happen if it is, so + check it just to make sure. */ + av_assert0(s->samples >= 0); + if(!s->samples){ - s->data = av_realloc(s->data, (buf_size + 3) & ~3); + uint32_t offset; + void *tmp_data; + + if (buf_size < 8) { + av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); + return AVERROR_INVALIDDATA; + } + + tmp_data = av_realloc(s->data, FFALIGN(buf_size, 4)); + if (!tmp_data) + return AVERROR(ENOMEM); + s->data = tmp_data; s->dsp.bswap_buf((uint32_t*)s->data, (const uint32_t*)buf, buf_size >> 2); s->ptr = s->last_ptr = s->data; s->data_end = s->data + buf_size; - nblocks = s->samples = bytestream_get_be32(&s->ptr); - n = bytestream_get_be32(&s->ptr); - if(n < 0 || n > 3){ + nblocks = bytestream_get_be32(&s->ptr); + offset = bytestream_get_be32(&s->ptr); + if (offset > 3) { av_log(avctx, AV_LOG_ERROR, "Incorrect offset passed\n"); s->data = NULL; - return -1; + return AVERROR_INVALIDDATA; } - s->ptr += n; + if (s->data_end - s->ptr < offset) { + av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); + return AVERROR_INVALIDDATA; + } + s->ptr += offset; - s->currentframeblocks = nblocks; - buf += 4; - if (s->samples <= 0) { - *data_size = 0; - return buf_size; + if (!nblocks || nblocks > INT_MAX) { + av_log(avctx, AV_LOG_ERROR, "Invalid sample count: %u.\n", nblocks); + return AVERROR_INVALIDDATA; } + s->currentframeblocks = s->samples = nblocks; memset(s->decoded0, 0, sizeof(s->decoded0)); memset(s->decoded1, 0, sizeof(s->decoded1)); /* Initialize the frame decoder */ - init_frame_decoder(s); + if (init_frame_decoder(s) < 0) { + av_log(avctx, AV_LOG_ERROR, "Error reading frame header\n"); + return AVERROR_INVALIDDATA; + } } if (!s->data) { @@ -858,10 +901,10 @@ static int ape_decode_frame(AVCodecContext * avctx, ape_unpack_stereo(s, blockstodecode); emms_c(); - if(s->error || s->ptr > s->data_end){ + if (s->error) { s->samples=0; av_log(avctx, AV_LOG_ERROR, "Error decoding frame\n"); - return -1; + return AVERROR_INVALIDDATA; } for (i = 0; i < blockstodecode; i++) { diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h index ad3dd01198..f5b06d3936 100644 --- a/libavcodec/avcodec.h +++ b/libavcodec/avcodec.h @@ -383,6 +383,8 @@ enum CodecID { CODEC_ID_MPEG2TS= 0x20000, /**< _FAKE_ codec to indicate a raw MPEG-2 TS * stream (only used by libavformat) */ + CODEC_ID_MPEG4SYSTEMS = 0x20001, /**< _FAKE_ codec to indicate a MPEG-4 Systems + * stream (only used by libavformat) */ CODEC_ID_FFMETADATA=0x21000, ///< Dummy codec for streams containing only metadata information. }; @@ -682,8 +684,10 @@ typedef struct RcOverride{ * assume the buffer was allocated by avcodec_default_get_buffer. */ #define CODEC_CAP_DR1 0x0002 +#if FF_API_PARSE_FRAME /* If 'parse_only' field is true, then avcodec_parse_frame() can be used. */ #define CODEC_CAP_PARSE_ONLY 0x0004 +#endif #define CODEC_CAP_TRUNCATED 0x0008 /* Codec can export data for HW decoding (XvMC). */ #define CODEC_CAP_HWACCEL 0x0010 @@ -1590,9 +1594,15 @@ typedef struct AVCodecContext { */ int block_align; - int parse_only; /* - decoding only: If true, only parsing is done - (function avcodec_parse_frame()). The frame - data is returned. Only MPEG codecs support this now. */ +#if FF_API_PARSE_FRAME + /** + * If true, only parsing is done. The frame data is returned. + * Only MPEG audio decoders support this now. + * - encoding: unused + * - decoding: Set by user + */ + attribute_deprecated int parse_only; +#endif /** * 0-> h263 quant 1-> mpeg quant @@ -4047,10 +4057,6 @@ int avcodec_decode_subtitle2(AVCodecContext *avctx, AVSubtitle *sub, */ void avsubtitle_free(AVSubtitle *sub); -int avcodec_parse_frame(AVCodecContext *avctx, uint8_t **pdata, - int *data_size_ptr, - uint8_t *buf, int buf_size); - /** * Encode an audio frame from samples into buf. * diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c index 286e08f27a..11d7f1fb93 100644 --- a/libavcodec/mpegaudiodec.c +++ b/libavcodec/mpegaudiodec.c @@ -21,7 +21,7 @@ /** * @file - * MPEG Audio decoder. + * MPEG Audio decoder */ #include "libavutil/audioconvert.h" @@ -63,7 +63,7 @@ typedef struct GranuleDef { typedef struct MPADecodeContext { MPA_DECODE_HEADER - uint8_t last_buf[2*BACKSTEP_SIZE + EXTRABYTES]; + uint8_t last_buf[2 * BACKSTEP_SIZE + EXTRABYTES]; int last_buf_size; /* next header (used in free format parsing) */ uint32_t free_format_next_header; @@ -74,9 +74,6 @@ typedef struct MPADecodeContext { DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT]; INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */ GranuleDef granules[2][2]; /* Used in Layer 3 */ -#ifdef DEBUG - int frame_count; -#endif int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3 int dither_state; int err_recognition; @@ -95,7 +92,7 @@ typedef struct MPADecodeContext { # define OUT_FMT AV_SAMPLE_FMT_FLT #else # define SHR(a,b) ((a)>>(b)) -/* WARNING: only correct for posititive numbers */ +/* WARNING: only correct for positive numbers */ # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5)) # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5)) # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5)) @@ -115,18 +112,16 @@ typedef struct MPADecodeContext { /* vlc structure for decoding layer 3 huffman tables */ static VLC huff_vlc[16]; static VLC_TYPE huff_vlc_tables[ - 0+128+128+128+130+128+154+166+ - 142+204+190+170+542+460+662+414 + 0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 + + 142 + 204 + 190 + 170 + 542 + 460 + 662 + 414 ][2]; static const int huff_vlc_tables_sizes[16] = { - 0, 128, 128, 128, 130, 128, 154, 166, - 142, 204, 190, 170, 542, 460, 662, 414 + 0, 128, 128, 128, 130, 128, 154, 166, + 142, 204, 190, 170, 542, 460, 662, 414 }; static VLC huff_quad_vlc[2]; -static VLC_TYPE huff_quad_vlc_tables[128+16][2]; -static const int huff_quad_vlc_tables_sizes[2] = { - 128, 16 -}; +static VLC_TYPE huff_quad_vlc_tables[128+16][2]; +static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 }; /* computed from band_size_long */ static uint16_t band_index_long[9][23]; #include "mpegaudio_tablegen.h" @@ -163,17 +158,19 @@ static const int32_t scale_factor_mult2[3][3] = { * Convert region offsets to region sizes and truncate * size to big_values. */ -static void ff_region_offset2size(GranuleDef *g){ - int i, k, j=0; - g->region_size[2] = (576 / 2); - for(i=0;i<3;i++) { +static void ff_region_offset2size(GranuleDef *g) +{ + int i, k, j = 0; + g->region_size[2] = 576 / 2; + for (i = 0; i < 3; i++) { k = FFMIN(g->region_size[i], g->big_values); g->region_size[i] = k - j; j = k; } } -static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){ +static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g) +{ if (g->block_type == 2) g->region_size[0] = (36 / 2); else { @@ -187,17 +184,17 @@ static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){ g->region_size[1] = (576 / 2); } -static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2){ +static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2) +{ int l; - g->region_size[0] = - band_index_long[s->sample_rate_index][ra1 + 1] >> 1; + g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1; /* should not overflow */ l = FFMIN(ra1 + ra2 + 2, 22); - g->region_size[1] = - band_index_long[s->sample_rate_index][l] >> 1; + g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1; } -static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){ +static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g) +{ if (g->block_type == 2) { if (g->switch_point) { /* if switched mode, we handle the 36 first samples as @@ -212,12 +209,12 @@ static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){ g->short_start = 2 + (s->sample_rate_index != 8); } else { - g->long_end = 0; + g->long_end = 0; g->short_start = 0; } } else { g->short_start = 13; - g->long_end = 22; + g->long_end = 22; } } @@ -228,11 +225,11 @@ static inline int l1_unscale(int n, int mant, int scale_factor) int shift, mod; int64_t val; - shift = scale_factor_modshift[scale_factor]; - mod = shift & 3; + shift = scale_factor_modshift[scale_factor]; + mod = shift & 3; shift >>= 2; - val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]); - shift += n; + val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]); + shift += n; /* NOTE: at this point, 1 <= shift >= 21 + 15 */ return (int)((val + (1LL << (shift - 1))) >> shift); } @@ -241,8 +238,8 @@ static inline int l2_unscale_group(int steps, int mant, int scale_factor) { int shift, mod, val; - shift = scale_factor_modshift[scale_factor]; - mod = shift & 3; + shift = scale_factor_modshift[scale_factor]; + mod = shift & 3; shift >>= 2; val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod]; @@ -258,13 +255,13 @@ static inline int l3_unscale(int value, int exponent) unsigned int m; int e; - e = table_4_3_exp [4*value + (exponent&3)]; - m = table_4_3_value[4*value + (exponent&3)]; - e -= (exponent >> 2); - assert(e>=1); + e = table_4_3_exp [4 * value + (exponent & 3)]; + m = table_4_3_value[4 * value + (exponent & 3)]; + e -= exponent >> 2; + assert(e >= 1); if (e > 31) return 0; - m = (m + (1 << (e-1))) >> e; + m = (m + (1 << (e - 1))) >> e; return m; } @@ -272,7 +269,7 @@ static inline int l3_unscale(int value, int exponent) static av_cold int decode_init(AVCodecContext * avctx) { MPADecodeContext *s = avctx->priv_data; - static int init=0; + static int init = 0; int i, j, k; s->avctx = avctx; @@ -282,28 +279,31 @@ static av_cold int decode_init(AVCodecContext * avctx) avctx->sample_fmt= OUT_FMT; s->err_recognition = avctx->err_recognition; +#if FF_API_PARSE_FRAME if (!init && !avctx->parse_only) { +#else + if (!init) { +#endif int offset; /* scale factors table for layer 1/2 */ - for(i=0;i<64;i++) { + for (i = 0; i < 64; i++) { int shift, mod; /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */ - shift = (i / 3); - mod = i % 3; + shift = i / 3; + mod = i % 3; scale_factor_modshift[i] = mod | (shift << 2); } /* scale factor multiply for layer 1 */ - for(i=0;i<15;i++) { + for (i = 0; i < 15; i++) { int n, norm; n = i + 2; norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1); scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS); scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS); scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS); - av_dlog(avctx, "%d: norm=%x s=%x %x %x\n", - i, norm, + av_dlog(avctx, "%d: norm=%x s=%x %x %x\n", i, norm, scale_factor_mult[i][0], scale_factor_mult[i][1], scale_factor_mult[i][2]); @@ -313,7 +313,7 @@ static av_cold int decode_init(AVCodecContext * avctx) /* huffman decode tables */ offset = 0; - for(i=1;i<16;i++) { + for (i = 1; i < 16; i++) { const HuffTable *h = &mpa_huff_tables[i]; int xsize, x, y; uint8_t tmp_bits [512]; @@ -325,8 +325,8 @@ static av_cold int decode_init(AVCodecContext * avctx) xsize = h->xsize; j = 0; - for(x=0;x<xsize;x++) { - for(y=0;y<xsize;y++){ + for (x = 0; x < xsize; x++) { + for (y = 0; y < xsize; y++) { tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ]; tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++]; } @@ -343,7 +343,7 @@ static av_cold int decode_init(AVCodecContext * avctx) assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables)); offset = 0; - for(i=0;i<2;i++) { + for (i = 0; i < 2; i++) { huff_quad_vlc[i].table = huff_quad_vlc_tables+offset; huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i]; init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16, @@ -353,9 +353,9 @@ static av_cold int decode_init(AVCodecContext * avctx) } assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables)); - for(i=0;i<9;i++) { + for (i = 0; i < 9; i++) { k = 0; - for(j=0;j<22;j++) { + for (j = 0; j < 22; j++) { band_index_long[i][j] = k; k += band_size_long[i][j]; } @@ -366,21 +366,23 @@ static av_cold int decode_init(AVCodecContext * avctx) mpegaudio_tableinit(); - for (i = 0; i < 4; i++) - if (ff_mpa_quant_bits[i] < 0) - for (j = 0; j < (1<<(-ff_mpa_quant_bits[i]+1)); j++) { + for (i = 0; i < 4; i++) { + if (ff_mpa_quant_bits[i] < 0) { + for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) { int val1, val2, val3, steps; int val = j; - steps = ff_mpa_quant_steps[i]; - val1 = val % steps; - val /= steps; - val2 = val % steps; - val3 = val / steps; + steps = ff_mpa_quant_steps[i]; + val1 = val % steps; + val /= steps; + val2 = val % steps; + val3 = val / steps; division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8); } + } + } - for(i=0;i<7;i++) { + for (i = 0; i < 7; i++) { float f; INTFLOAT v; if (i != 6) { @@ -389,30 +391,30 @@ static av_cold int decode_init(AVCodecContext * avctx) } else { v = FIXR(1.0); } - is_table[0][i] = v; + is_table[0][ i] = v; is_table[1][6 - i] = v; } /* invalid values */ - for(i=7;i<16;i++) + for (i = 7; i < 16; i++) is_table[0][i] = is_table[1][i] = 0.0; - for(i=0;i<16;i++) { + for (i = 0; i < 16; i++) { double f; int e, k; - for(j=0;j<2;j++) { + for (j = 0; j < 2; j++) { e = -(j + 1) * ((i + 1) >> 1); f = pow(2.0, e / 4.0); k = i & 1; is_table_lsf[j][k ^ 1][i] = FIXR(f); - is_table_lsf[j][k][i] = FIXR(1.0); + is_table_lsf[j][k ][i] = FIXR(1.0); av_dlog(avctx, "is_table_lsf %d %d: %f %f\n", i, j, (float) is_table_lsf[j][0][i], (float) is_table_lsf[j][1][i]); } } - for(i=0;i<8;i++) { + for (i = 0; i < 8; i++) { float ci, cs, ca; ci = ci_table[i]; cs = 1.0 / sqrt(1.0 + ci * ci); @@ -431,27 +433,27 @@ static av_cold int decode_init(AVCodecContext * avctx) } /* compute mdct windows */ - for(i=0;i<36;i++) { - for(j=0; j<4; j++){ + for (i = 0; i < 36; i++) { + for (j = 0; j < 4; j++) { double d; - if(j==2 && i%3 != 1) + if (j == 2 && i % 3 != 1) continue; - d= sin(M_PI * (i + 0.5) / 36.0); - if(j==1){ - if (i>=30) d= 0; - else if(i>=24) d= sin(M_PI * (i - 18 + 0.5) / 12.0); - else if(i>=18) d= 1; - }else if(j==3){ - if (i< 6) d= 0; - else if(i< 12) d= sin(M_PI * (i - 6 + 0.5) / 12.0); - else if(i< 18) d= 1; + d = sin(M_PI * (i + 0.5) / 36.0); + if (j == 1) { + if (i >= 30) d = 0; + else if (i >= 24) d = sin(M_PI * (i - 18 + 0.5) / 12.0); + else if (i >= 18) d = 1; + } else if (j == 3) { + if (i < 6) d = 0; + else if (i < 12) d = sin(M_PI * (i - 6 + 0.5) / 12.0); + else if (i < 18) d = 1; } //merge last stage of imdct into the window coefficients - d*= 0.5 / cos(M_PI*(2*i + 19)/72); + d *= 0.5 / cos(M_PI * (2 * i + 19) / 72); - if(j==2) + if (j == 2) mdct_win[j][i/3] = FIXHR((d / (1<<5))); else mdct_win[j][i ] = FIXHR((d / (1<<5))); @@ -460,9 +462,9 @@ static av_cold int decode_init(AVCodecContext * avctx) /* NOTE: we do frequency inversion adter the MDCT by changing the sign of the right window coefs */ - for(j=0;j<4;j++) { - for(i=0;i<36;i+=2) { - mdct_win[j + 4][i] = mdct_win[j][i]; + for (j = 0; j < 4; j++) { + for (i = 0; i < 36; i += 2) { + mdct_win[j + 4][i ] = mdct_win[j][i ]; mdct_win[j + 4][i + 1] = -mdct_win[j][i + 1]; } } @@ -509,41 +511,41 @@ static void imdct12(INTFLOAT *out, INTFLOAT *in) { INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2; - in0= in[0*3]; - in1= in[1*3] + in[0*3]; - in2= in[2*3] + in[1*3]; - in3= in[3*3] + in[2*3]; - in4= in[4*3] + in[3*3]; - in5= in[5*3] + in[4*3]; + in0 = in[0*3]; + in1 = in[1*3] + in[0*3]; + in2 = in[2*3] + in[1*3]; + in3 = in[3*3] + in[2*3]; + in4 = in[4*3] + in[3*3]; + in5 = in[5*3] + in[4*3]; in5 += in3; in3 += in1; - in2= MULH3(in2, C3, 2); - in3= MULH3(in3, C3, 4); - - t1 = in0 - in4; - t2 = MULH3(in1 - in5, icos36h[4], 2); - - out[ 7]= - out[10]= t1 + t2; - out[ 1]= - out[ 4]= t1 - t2; - - in0 += SHR(in4, 1); - in4 = in0 + in2; - in5 += 2*in1; - in1 = MULH3(in5 + in3, icos36h[1], 1); - out[ 8]= - out[ 9]= in4 + in1; - out[ 2]= - out[ 3]= in4 - in1; - - in0 -= in2; - in5 = MULH3(in5 - in3, icos36h[7], 2); - out[ 0]= - out[ 5]= in0 - in5; - out[ 6]= - out[11]= in0 + in5; + in2 = MULH3(in2, C3, 2); + in3 = MULH3(in3, C3, 4); + + t1 = in0 - in4; + t2 = MULH3(in1 - in5, icos36h[4], 2); + + out[ 7] = + out[10] = t1 + t2; + out[ 1] = + out[ 4] = t1 - t2; + + in0 += SHR(in4, 1); + in4 = in0 + in2; + in5 += 2*in1; + in1 = MULH3(in5 + in3, icos36h[1], 1); + out[ 8] = + out[ 9] = in4 + in1; + out[ 2] = + out[ 3] = in4 - in1; + + in0 -= in2; + in5 = MULH3(in5 - in3, icos36h[7], 2); + out[ 0] = + out[ 5] = in0 - in5; + out[ 6] = + out[11] = in0 + in5; } /* cos(pi*i/18) */ @@ -564,12 +566,12 @@ static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win) INTFLOAT t0, t1, t2, t3, s0, s1, s2, s3; INTFLOAT tmp[18], *tmp1, *in1; - for(i=17;i>=1;i--) + for (i = 17; i >= 1; i--) in[i] += in[i-1]; - for(i=17;i>=3;i-=2) + for (i = 17; i >= 3; i -= 2) in[i] += in[i-2]; - for(j=0;j<2;j++) { + for (j = 0; j < 2; j++) { tmp1 = tmp + j; in1 = in + j; @@ -601,7 +603,7 @@ static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win) } i = 0; - for(j=0;j<4;j++) { + for (j = 0; j < 4; j++) { t0 = tmp[i]; t1 = tmp[i + 2]; s0 = t1 + t0; @@ -609,22 +611,22 @@ static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win) t2 = tmp[i + 1]; t3 = tmp[i + 3]; - s1 = MULH3(t3 + t2, icos36h[j], 2); - s3 = MULLx(t3 - t2, icos36[8 - j], FRAC_BITS); + s1 = MULH3(t3 + t2, icos36h[ j], 2); + s3 = MULLx(t3 - t2, icos36 [8 - j], FRAC_BITS); t0 = s0 + s1; t1 = s0 - s1; - out[(9 + j)*SBLIMIT] = MULH3(t1, win[9 + j], 1) + buf[9 + j]; - out[(8 - j)*SBLIMIT] = MULH3(t1, win[8 - j], 1) + buf[8 - j]; - buf[9 + j] = MULH3(t0, win[18 + 9 + j], 1); - buf[8 - j] = MULH3(t0, win[18 + 8 - j], 1); + out[(9 + j) * SBLIMIT] = MULH3(t1, win[ 9 + j], 1) + buf[9 + j]; + out[(8 - j) * SBLIMIT] = MULH3(t1, win[ 8 - j], 1) + buf[8 - j]; + buf[ 9 + j ] = MULH3(t0, win[18 + 9 + j], 1); + buf[ 8 - j ] = MULH3(t0, win[18 + 8 - j], 1); t0 = s2 + s3; t1 = s2 - s3; - out[(9 + 8 - j)*SBLIMIT] = MULH3(t1, win[9 + 8 - j], 1) + buf[9 + 8 - j]; - out[( j)*SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j]; - buf[9 + 8 - j] = MULH3(t0, win[18 + 9 + 8 - j], 1); - buf[ + j] = MULH3(t0, win[18 + j], 1); + out[(9 + 8 - j) * SBLIMIT] = MULH3(t1, win[ 9 + 8 - j], 1) + buf[9 + 8 - j]; + out[ j * SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j]; + buf[ 9 + 8 - j ] = MULH3(t0, win[18 + 9 + 8 - j], 1); + buf[ j ] = MULH3(t0, win[18 + j], 1); i += 4; } @@ -632,10 +634,10 @@ static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win) s1 = MULH3(tmp[17], icos36h[4], 2); t0 = s0 + s1; t1 = s0 - s1; - out[(9 + 4)*SBLIMIT] = MULH3(t1, win[9 + 4], 1) + buf[9 + 4]; - out[(8 - 4)*SBLIMIT] = MULH3(t1, win[8 - 4], 1) + buf[8 - 4]; - buf[9 + 4] = MULH3(t0, win[18 + 9 + 4], 1); - buf[8 - 4] = MULH3(t0, win[18 + 8 - 4], 1); + out[(9 + 4) * SBLIMIT] = MULH3(t1, win[ 9 + 4], 1) + buf[9 + 4]; + out[(8 - 4) * SBLIMIT] = MULH3(t1, win[ 8 - 4], 1) + buf[8 - 4]; + buf[ 9 + 4 ] = MULH3(t0, win[18 + 9 + 4], 1); + buf[ 8 - 4 ] = MULH3(t0, win[18 + 8 - 4], 1); } /* return the number of decoded frames */ @@ -651,23 +653,22 @@ static int mp_decode_layer1(MPADecodeContext *s) bound = SBLIMIT; /* allocation bits */ - for(i=0;i<bound;i++) { - for(ch=0;ch<s->nb_channels;ch++) { + for (i = 0; i < bound; i++) { + for (ch = 0; ch < s->nb_channels; ch++) { allocation[ch][i] = get_bits(&s->gb, 4); } } - for(i=bound;i<SBLIMIT;i++) { + for (i = bound; i < SBLIMIT; i++) allocation[0][i] = get_bits(&s->gb, 4); - } /* scale factors */ - for(i=0;i<bound;i++) { - for(ch=0;ch<s->nb_channels;ch++) { + for (i = 0; i < bound; i++) { + for (ch = 0; ch < s->nb_channels; ch++) { if (allocation[ch][i]) scale_factors[ch][i] = get_bits(&s->gb, 6); } } - for(i=bound;i<SBLIMIT;i++) { + for (i = bound; i < SBLIMIT; i++) { if (allocation[0][i]) { scale_factors[0][i] = get_bits(&s->gb, 6); scale_factors[1][i] = get_bits(&s->gb, 6); @@ -675,9 +676,9 @@ static int mp_decode_layer1(MPADecodeContext *s) } /* compute samples */ - for(j=0;j<12;j++) { - for(i=0;i<bound;i++) { - for(ch=0;ch<s->nb_channels;ch++) { + for (j = 0; j < 12; j++) { + for (i = 0; i < bound; i++) { + for (ch = 0; ch < s->nb_channels; ch++) { n = allocation[ch][i]; if (n) { mant = get_bits(&s->gb, n + 1); @@ -688,7 +689,7 @@ static int mp_decode_layer1(MPADecodeContext *s) s->sb_samples[ch][j][i] = v; } } - for(i=bound;i<SBLIMIT;i++) { + for (i = bound; i < SBLIMIT; i++) { n = allocation[0][i]; if (n) { mant = get_bits(&s->gb, n + 1); @@ -717,8 +718,8 @@ static int mp_decode_layer2(MPADecodeContext *s) /* select decoding table */ table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels, - s->sample_rate, s->lsf); - sblimit = ff_mpa_sblimit_table[table]; + s->sample_rate, s->lsf); + sblimit = ff_mpa_sblimit_table[table]; alloc_table = ff_mpa_alloc_tables[table]; if (s->mode == MPA_JSTEREO) @@ -729,18 +730,18 @@ static int mp_decode_layer2(MPADecodeContext *s) av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit); /* sanity check */ - if( bound > sblimit ) bound = sblimit; + if (bound > sblimit) + bound = sblimit; /* parse bit allocation */ j = 0; - for(i=0;i<bound;i++) { + for (i = 0; i < bound; i++) { bit_alloc_bits = alloc_table[j]; - for(ch=0;ch<s->nb_channels;ch++) { + for (ch = 0; ch < s->nb_channels; ch++) bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits); - } j += 1 << bit_alloc_bits; } - for(i=bound;i<sblimit;i++) { + for (i = bound; i < sblimit; i++) { bit_alloc_bits = alloc_table[j]; v = get_bits(&s->gb, bit_alloc_bits); bit_alloc[0][i] = v; @@ -749,19 +750,19 @@ static int mp_decode_layer2(MPADecodeContext *s) } /* scale codes */ - for(i=0;i<sblimit;i++) { - for(ch=0;ch<s->nb_channels;ch++) { + for (i = 0; i < sblimit; i++) { + for (ch = 0; ch < s->nb_channels; ch++) { if (bit_alloc[ch][i]) scale_code[ch][i] = get_bits(&s->gb, 2); } } /* scale factors */ - for(i=0;i<sblimit;i++) { - for(ch=0;ch<s->nb_channels;ch++) { + for (i = 0; i < sblimit; i++) { + for (ch = 0; ch < s->nb_channels; ch++) { if (bit_alloc[ch][i]) { sf = scale_factors[ch][i]; - switch(scale_code[ch][i]) { + switch (scale_code[ch][i]) { default: case 0: sf[0] = get_bits(&s->gb, 6); @@ -789,12 +790,12 @@ static int mp_decode_layer2(MPADecodeContext *s) } /* samples */ - for(k=0;k<3;k++) { - for(l=0;l<12;l+=3) { + for (k = 0; k < 3; k++) { + for (l = 0; l < 12; l += 3) { j = 0; - for(i=0;i<bound;i++) { + for (i = 0; i < bound; i++) { bit_alloc_bits = alloc_table[j]; - for(ch=0;ch<s->nb_channels;ch++) { + for (ch = 0; ch < s->nb_channels; ch++) { b = bit_alloc[ch][i]; if (b) { scale = scale_factors[ch][i][k]; @@ -808,13 +809,13 @@ static int mp_decode_layer2(MPADecodeContext *s) steps = ff_mpa_quant_steps[qindex]; s->sb_samples[ch][k * 12 + l + 0][i] = - l2_unscale_group(steps, v2 & 15, scale); + l2_unscale_group(steps, v2 & 15, scale); s->sb_samples[ch][k * 12 + l + 1][i] = l2_unscale_group(steps, (v2 >> 4) & 15, scale); s->sb_samples[ch][k * 12 + l + 2][i] = l2_unscale_group(steps, v2 >> 8 , scale); } else { - for(m=0;m<3;m++) { + for (m = 0; m < 3; m++) { v = get_bits(&s->gb, bits); v = l1_unscale(bits - 1, v, scale); s->sb_samples[ch][k * 12 + l + m][i] = v; @@ -830,7 +831,7 @@ static int mp_decode_layer2(MPADecodeContext *s) j += 1 << bit_alloc_bits; } /* XXX: find a way to avoid this duplication of code */ - for(i=bound;i<sblimit;i++) { + for (i = bound; i < sblimit; i++) { bit_alloc_bits = alloc_table[j]; b = bit_alloc[0][i]; if (b) { @@ -860,7 +861,7 @@ static int mp_decode_layer2(MPADecodeContext *s) s->sb_samples[1][k * 12 + l + 2][i] = l2_unscale_group(steps, v, scale1); } else { - for(m=0;m<3;m++) { + for (m = 0; m < 3; m++) { mant = get_bits(&s->gb, bits); s->sb_samples[0][k * 12 + l + m][i] = l1_unscale(bits - 1, mant, scale0); @@ -880,8 +881,8 @@ static int mp_decode_layer2(MPADecodeContext *s) j += 1 << bit_alloc_bits; } /* fill remaining samples to zero */ - for(i=sblimit;i<SBLIMIT;i++) { - for(ch=0;ch<s->nb_channels;ch++) { + for (i = sblimit; i < SBLIMIT; i++) { + for (ch = 0; ch < s->nb_channels; ch++) { s->sb_samples[ch][k * 12 + l + 0][i] = 0; s->sb_samples[ch][k * 12 + l + 1][i] = 0; s->sb_samples[ch][k * 12 + l + 2][i] = 0; @@ -892,28 +893,28 @@ static int mp_decode_layer2(MPADecodeContext *s) return 3 * 12; } -#define SPLIT(dst,sf,n)\ - if(n==3){\ - int m= (sf*171)>>9;\ - dst= sf - 3*m;\ - sf=m;\ - }else if(n==4){\ - dst= sf&3;\ - sf>>=2;\ - }else if(n==5){\ - int m= (sf*205)>>10;\ - dst= sf - 5*m;\ - sf=m;\ - }else if(n==6){\ - int m= (sf*171)>>10;\ - dst= sf - 6*m;\ - sf=m;\ - }else{\ - dst=0;\ +#define SPLIT(dst,sf,n) \ + if (n == 3) { \ + int m = (sf * 171) >> 9; \ + dst = sf - 3 * m; \ + sf = m; \ + } else if (n == 4) { \ + dst = sf & 3; \ + sf >>= 2; \ + } else if (n == 5) { \ + int m = (sf * 205) >> 10; \ + dst = sf - 5 * m; \ + sf = m; \ + } else if (n == 6) { \ + int m = (sf * 171) >> 10; \ + dst = sf - 6 * m; \ + sf = m; \ + } else { \ + dst = 0; \ } -static av_always_inline void lsf_sf_expand(int *slen, - int sf, int n1, int n2, int n3) +static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2, + int n3) { SPLIT(slen[3], sf, n3) SPLIT(slen[2], sf, n2) @@ -921,8 +922,7 @@ static av_always_inline void lsf_sf_expand(int *slen, slen[0] = sf; } -static void exponents_from_scale_factors(MPADecodeContext *s, - GranuleDef *g, +static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g, int16_t *exponents) { const uint8_t *bstab, *pretab; @@ -930,30 +930,30 @@ static void exponents_from_scale_factors(MPADecodeContext *s, int16_t *exp_ptr; exp_ptr = exponents; - gain = g->global_gain - 210; - shift = g->scalefac_scale + 1; + gain = g->global_gain - 210; + shift = g->scalefac_scale + 1; - bstab = band_size_long[s->sample_rate_index]; + bstab = band_size_long[s->sample_rate_index]; pretab = mpa_pretab[g->preflag]; - for(i=0;i<g->long_end;i++) { + for (i = 0; i < g->long_end; i++) { v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400; len = bstab[i]; - for(j=len;j>0;j--) + for (j = len; j > 0; j--) *exp_ptr++ = v0; } if (g->short_start < 13) { - bstab = band_size_short[s->sample_rate_index]; + bstab = band_size_short[s->sample_rate_index]; gains[0] = gain - (g->subblock_gain[0] << 3); gains[1] = gain - (g->subblock_gain[1] << 3); gains[2] = gain - (g->subblock_gain[2] << 3); - k = g->long_end; - for(i=g->short_start;i<13;i++) { + k = g->long_end; + for (i = g->short_start; i < 13; i++) { len = bstab[i]; - for(l=0;l<3;l++) { + for (l = 0; l < 3; l++) { v0 = gains[l] - (g->scale_factors[k++] << shift) + 400; - for(j=len;j>0;j--) - *exp_ptr++ = v0; + for (j = len; j > 0; j--) + *exp_ptr++ = v0; } } } @@ -962,22 +962,21 @@ static void exponents_from_scale_factors(MPADecodeContext *s, /* handle n = 0 too */ static inline int get_bitsz(GetBitContext *s, int n) { - if (n == 0) - return 0; - else - return get_bits(s, n); + return n ? get_bits(s, n) : 0; } -static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2){ - if(s->in_gb.buffer && *pos >= s->gb.size_in_bits){ - s->gb= s->in_gb; - s->in_gb.buffer=NULL; +static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, + int *end_pos2) +{ + if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) { + s->gb = s->in_gb; + s->in_gb.buffer = NULL; assert((get_bits_count(&s->gb) & 7) == 0); skip_bits_long(&s->gb, *pos - *end_pos); - *end_pos2= - *end_pos= *end_pos2 + get_bits_count(&s->gb) - *pos; - *pos= get_bits_count(&s->gb); + *end_pos2 = + *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos; + *pos = get_bits_count(&s->gb); } } @@ -988,13 +987,13 @@ static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_ *dst = v; */ #if CONFIG_FLOAT -#define READ_FLIP_SIGN(dst,src)\ - v = AV_RN32A(src) ^ (get_bits1(&s->gb)<<31);\ - AV_WN32A(dst, v); +#define READ_FLIP_SIGN(dst,src) \ + v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \ + AV_WN32A(dst, v); #else -#define READ_FLIP_SIGN(dst,src)\ - v= -get_bits1(&s->gb);\ - *(dst) = (*(src) ^ v) - v; +#define READ_FLIP_SIGN(dst,src) \ + v = -get_bits1(&s->gb); \ + *(dst) = (*(src) ^ v) - v; #endif static int huffman_decode(MPADecodeContext *s, GranuleDef *g, @@ -1004,43 +1003,43 @@ static int huffman_decode(MPADecodeContext *s, GranuleDef *g, int i; int last_pos, bits_left; VLC *vlc; - int end_pos= FFMIN(end_pos2, s->gb.size_in_bits); + int end_pos = FFMIN(end_pos2, s->gb.size_in_bits); /* low frequencies (called big values) */ s_index = 0; - for(i=0;i<3;i++) { + for (i = 0; i < 3; i++) { int j, k, l, linbits; j = g->region_size[i]; if (j == 0) continue; /* select vlc table */ - k = g->table_select[i]; - l = mpa_huff_data[k][0]; + k = g->table_select[i]; + l = mpa_huff_data[k][0]; linbits = mpa_huff_data[k][1]; - vlc = &huff_vlc[l]; + vlc = &huff_vlc[l]; - if(!l){ - memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*2*j); - s_index += 2*j; + if (!l) { + memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j); + s_index += 2 * j; continue; } /* read huffcode and compute each couple */ - for(;j>0;j--) { + for (; j > 0; j--) { int exponent, x, y; int v; - int pos= get_bits_count(&s->gb); + int pos = get_bits_count(&s->gb); if (pos >= end_pos){ // av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index); switch_buffer(s, &pos, &end_pos, &end_pos2); // av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos); - if(pos >= end_pos) + if (pos >= end_pos) break; } y = get_vlc2(&s->gb, vlc->table, 7, 3); - if(!y){ + if (!y) { g->sb_hybrid[s_index ] = g->sb_hybrid[s_index+1] = 0; s_index += 2; @@ -1051,54 +1050,54 @@ static int huffman_decode(MPADecodeContext *s, GranuleDef *g, av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n", i, g->region_size[i] - j, x, y, exponent); - if(y&16){ + if (y & 16) { x = y >> 5; y = y & 0x0f; - if (x < 15){ - READ_FLIP_SIGN(g->sb_hybrid+s_index, RENAME(expval_table)[ exponent ]+x) - }else{ + if (x < 15) { + READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x) + } else { x += get_bitsz(&s->gb, linbits); - v = l3_unscale(x, exponent); + v = l3_unscale(x, exponent); if (get_bits1(&s->gb)) v = -v; g->sb_hybrid[s_index] = v; } - if (y < 15){ - READ_FLIP_SIGN(g->sb_hybrid+s_index+1, RENAME(expval_table)[ exponent ]+y) - }else{ + if (y < 15) { + READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y) + } else { y += get_bitsz(&s->gb, linbits); - v = l3_unscale(y, exponent); + v = l3_unscale(y, exponent); if (get_bits1(&s->gb)) v = -v; g->sb_hybrid[s_index+1] = v; } - }else{ + } else { x = y >> 5; y = y & 0x0f; x += y; - if (x < 15){ - READ_FLIP_SIGN(g->sb_hybrid+s_index+!!y, RENAME(expval_table)[ exponent ]+x) - }else{ + if (x < 15) { + READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x) + } else { x += get_bitsz(&s->gb, linbits); - v = l3_unscale(x, exponent); + v = l3_unscale(x, exponent); if (get_bits1(&s->gb)) v = -v; g->sb_hybrid[s_index+!!y] = v; } - g->sb_hybrid[s_index+ !y] = 0; + g->sb_hybrid[s_index + !y] = 0; } - s_index+=2; + s_index += 2; } } /* high frequencies */ vlc = &huff_quad_vlc[g->count1table_select]; - last_pos=0; + last_pos = 0; while (s_index <= 572) { int pos, code; pos = get_bits_count(&s->gb); if (pos >= end_pos) { - if (pos > end_pos2 && last_pos){ + if (pos > end_pos2 && last_pos) { /* some encoders generate an incorrect size for this part. We must go back into the data */ s_index -= 4; @@ -1111,25 +1110,25 @@ static int huffman_decode(MPADecodeContext *s, GranuleDef *g, // av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index); switch_buffer(s, &pos, &end_pos, &end_pos2); // av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index); - if(pos >= end_pos) + if (pos >= end_pos) break; } - last_pos= pos; + last_pos = pos; code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1); av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code); - g->sb_hybrid[s_index+0]= - g->sb_hybrid[s_index+1]= - g->sb_hybrid[s_index+2]= - g->sb_hybrid[s_index+3]= 0; - while(code){ - static const int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0}; + g->sb_hybrid[s_index+0] = + g->sb_hybrid[s_index+1] = + g->sb_hybrid[s_index+2] = + g->sb_hybrid[s_index+3] = 0; + while (code) { + static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 }; int v; - int pos= s_index+idxtab[code]; - code ^= 8>>idxtab[code]; - READ_FLIP_SIGN(g->sb_hybrid+pos, RENAME(exp_table)+exponents[pos]) + int pos = s_index + idxtab[code]; + code ^= 8 >> idxtab[code]; + READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos]) } - s_index+=4; + s_index += 4; } /* skip extension bits */ bits_left = end_pos2 - get_bits_count(&s->gb); @@ -1137,14 +1136,14 @@ static int huffman_decode(MPADecodeContext *s, GranuleDef *g, if (bits_left < 0 && (s->err_recognition & AV_EF_BITSTREAM)) { av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left); s_index=0; - }else if(bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)){ + } else if (bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)) { av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left); - s_index=0; + s_index = 0; } - memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index)); + memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index)); skip_bits_long(&s->gb, bits_left); - i= get_bits_count(&s->gb); + i = get_bits_count(&s->gb); switch_buffer(s, &i, &end_pos, &end_pos2); return 0; @@ -1163,34 +1162,32 @@ static void reorder_block(MPADecodeContext *s, GranuleDef *g) return; if (g->switch_point) { - if (s->sample_rate_index != 8) { + if (s->sample_rate_index != 8) ptr = g->sb_hybrid + 36; - } else { + else ptr = g->sb_hybrid + 48; - } } else { ptr = g->sb_hybrid; } - for(i=g->short_start;i<13;i++) { - len = band_size_short[s->sample_rate_index][i]; + for (i = g->short_start; i < 13; i++) { + len = band_size_short[s->sample_rate_index][i]; ptr1 = ptr; - dst = tmp; - for(j=len;j>0;j--) { + dst = tmp; + for (j = len; j > 0; j--) { *dst++ = ptr[0*len]; *dst++ = ptr[1*len]; *dst++ = ptr[2*len]; ptr++; } - ptr+=2*len; + ptr += 2 * len; memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1)); } } #define ISQRT2 FIXR(0.70710678118654752440) -static void compute_stereo(MPADecodeContext *s, - GranuleDef *g0, GranuleDef *g1) +static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1) { int i, j, k, l; int sf_max, sf, len, non_zero_found; @@ -1214,17 +1211,17 @@ static void compute_stereo(MPADecodeContext *s, non_zero_found_short[1] = 0; non_zero_found_short[2] = 0; k = (13 - g1->short_start) * 3 + g1->long_end - 3; - for(i = 12;i >= g1->short_start;i--) { + for (i = 12; i >= g1->short_start; i--) { /* for last band, use previous scale factor */ if (i != 11) k -= 3; len = band_size_short[s->sample_rate_index][i]; - for(l=2;l>=0;l--) { + for (l = 2; l >= 0; l--) { tab0 -= len; tab1 -= len; if (!non_zero_found_short[l]) { /* test if non zero band. if so, stop doing i-stereo */ - for(j=0;j<len;j++) { + for (j = 0; j < len; j++) { if (tab1[j] != 0) { non_zero_found_short[l] = 1; goto found1; @@ -1236,19 +1233,19 @@ static void compute_stereo(MPADecodeContext *s, v1 = is_tab[0][sf]; v2 = is_tab[1][sf]; - for(j=0;j<len;j++) { - tmp0 = tab0[j]; + for (j = 0; j < len; j++) { + tmp0 = tab0[j]; tab0[j] = MULLx(tmp0, v1, FRAC_BITS); tab1[j] = MULLx(tmp0, v2, FRAC_BITS); } } else { - found1: +found1: if (s->mode_ext & MODE_EXT_MS_STEREO) { /* lower part of the spectrum : do ms stereo if enabled */ - for(j=0;j<len;j++) { - tmp0 = tab0[j]; - tmp1 = tab1[j]; + for (j = 0; j < len; j++) { + tmp0 = tab0[j]; + tmp1 = tab1[j]; tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS); tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS); } @@ -1258,41 +1255,41 @@ static void compute_stereo(MPADecodeContext *s, } non_zero_found = non_zero_found_short[0] | - non_zero_found_short[1] | - non_zero_found_short[2]; + non_zero_found_short[1] | + non_zero_found_short[2]; - for(i = g1->long_end - 1;i >= 0;i--) { - len = band_size_long[s->sample_rate_index][i]; + for (i = g1->long_end - 1;i >= 0;i--) { + len = band_size_long[s->sample_rate_index][i]; tab0 -= len; tab1 -= len; /* test if non zero band. if so, stop doing i-stereo */ if (!non_zero_found) { - for(j=0;j<len;j++) { + for (j = 0; j < len; j++) { if (tab1[j] != 0) { non_zero_found = 1; goto found2; } } /* for last band, use previous scale factor */ - k = (i == 21) ? 20 : i; + k = (i == 21) ? 20 : i; sf = g1->scale_factors[k]; if (sf >= sf_max) goto found2; v1 = is_tab[0][sf]; v2 = is_tab[1][sf]; - for(j=0;j<len;j++) { - tmp0 = tab0[j]; + for (j = 0; j < len; j++) { + tmp0 = tab0[j]; tab0[j] = MULLx(tmp0, v1, FRAC_BITS); tab1[j] = MULLx(tmp0, v2, FRAC_BITS); } } else { - found2: +found2: if (s->mode_ext & MODE_EXT_MS_STEREO) { /* lower part of the spectrum : do ms stereo if enabled */ - for(j=0;j<len;j++) { - tmp0 = tab0[j]; - tmp1 = tab1[j]; + for (j = 0; j < len; j++) { + tmp0 = tab0[j]; + tmp1 = tab1[j]; tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS); tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS); } @@ -1305,9 +1302,9 @@ static void compute_stereo(MPADecodeContext *s, global gain */ tab0 = g0->sb_hybrid; tab1 = g1->sb_hybrid; - for(i=0;i<576;i++) { - tmp0 = tab0[i]; - tmp1 = tab1[i]; + for (i = 0; i < 576; i++) { + tmp0 = tab0[i]; + tmp1 = tab1[i]; tab0[i] = tmp0 + tmp1; tab1[i] = tmp0 - tmp1; } @@ -1326,8 +1323,8 @@ static void compute_stereo(MPADecodeContext *s, int tmp0 = ptr[-1-j]; \ int tmp1 = ptr[ j]; \ int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \ - ptr[-1-j] = 4*(tmp2 - MULH(tmp1, csa_table[j][2])); \ - ptr[ j] = 4*(tmp2 + MULH(tmp0, csa_table[j][3])); \ + ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \ + ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \ } while (0) #endif @@ -1347,7 +1344,7 @@ static void compute_antialias(MPADecodeContext *s, GranuleDef *g) } ptr = g->sb_hybrid + 18; - for(i = n;i > 0;i--) { + for (i = n; i > 0; i--) { AA(0); AA(1); AA(2); @@ -1361,23 +1358,21 @@ static void compute_antialias(MPADecodeContext *s, GranuleDef *g) } } -static void compute_imdct(MPADecodeContext *s, - GranuleDef *g, - INTFLOAT *sb_samples, - INTFLOAT *mdct_buf) +static void compute_imdct(MPADecodeContext *s, GranuleDef *g, + INTFLOAT *sb_samples, INTFLOAT *mdct_buf) { INTFLOAT *win, *win1, *out_ptr, *ptr, *buf, *ptr1; INTFLOAT out2[12]; int i, j, mdct_long_end, sblimit; /* find last non zero block */ - ptr = g->sb_hybrid + 576; + ptr = g->sb_hybrid + 576; ptr1 = g->sb_hybrid + 2 * 18; while (ptr >= ptr1) { int32_t *p; ptr -= 6; - p= (int32_t*)ptr; - if(p[0] | p[1] | p[2] | p[3] | p[4] | p[5]) + p = (int32_t*)ptr; + if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5]) break; } sblimit = ((ptr - g->sb_hybrid) / 18) + 1; @@ -1394,7 +1389,7 @@ static void compute_imdct(MPADecodeContext *s, buf = mdct_buf; ptr = g->sb_hybrid; - for(j=0;j<mdct_long_end;j++) { + for (j = 0; j < mdct_long_end; j++) { /* apply window & overlap with previous buffer */ out_ptr = sb_samples + j; /* select window */ @@ -1405,33 +1400,33 @@ static void compute_imdct(MPADecodeContext *s, /* select frequency inversion */ win = win1 + ((4 * 36) & -(j & 1)); imdct36(out_ptr, buf, ptr, win); - out_ptr += 18*SBLIMIT; - ptr += 18; - buf += 18; + out_ptr += 18 * SBLIMIT; + ptr += 18; + buf += 18; } - for(j=mdct_long_end;j<sblimit;j++) { + for (j = mdct_long_end; j < sblimit; j++) { /* select frequency inversion */ - win = mdct_win[2] + ((4 * 36) & -(j & 1)); + win = mdct_win[2] + ((4 * 36) & -(j & 1)); out_ptr = sb_samples + j; - for(i=0; i<6; i++){ + for (i = 0; i < 6; i++) { *out_ptr = buf[i]; out_ptr += SBLIMIT; } imdct12(out2, ptr + 0); - for(i=0;i<6;i++) { + for (i = 0; i < 6; i++) { *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*1]; buf[i + 6*2] = MULH3(out2[i + 6], win[i + 6], 1); out_ptr += SBLIMIT; } imdct12(out2, ptr + 1); - for(i=0;i<6;i++) { + for (i = 0; i < 6; i++) { *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*2]; buf[i + 6*0] = MULH3(out2[i + 6], win[i + 6], 1); out_ptr += SBLIMIT; } imdct12(out2, ptr + 2); - for(i=0;i<6;i++) { + for (i = 0; i < 6; i++) { buf[i + 6*0] = MULH3(out2[i ], win[i ], 1) + buf[i + 6*0]; buf[i + 6*1] = MULH3(out2[i + 6], win[i + 6], 1); buf[i + 6*2] = 0; @@ -1440,12 +1435,12 @@ static void compute_imdct(MPADecodeContext *s, buf += 18; } /* zero bands */ - for(j=sblimit;j<SBLIMIT;j++) { + for (j = sblimit; j < SBLIMIT; j++) { /* overlap */ out_ptr = sb_samples + j; - for(i=0;i<18;i++) { + for (i = 0; i < 18; i++) { *out_ptr = buf[i]; - buf[i] = 0; + buf[i] = 0; out_ptr += SBLIMIT; } buf += 18; @@ -1472,21 +1467,21 @@ static int mp_decode_layer3(MPADecodeContext *s) else skip_bits(&s->gb, 5); nb_granules = 2; - for(ch=0;ch<s->nb_channels;ch++) { + for (ch = 0; ch < s->nb_channels; ch++) { s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */ s->granules[ch][1].scfsi = get_bits(&s->gb, 4); } } - for(gr=0;gr<nb_granules;gr++) { - for(ch=0;ch<s->nb_channels;ch++) { + for (gr = 0; gr < nb_granules; gr++) { + for (ch = 0; ch < s->nb_channels; ch++) { av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch); g = &s->granules[ch][gr]; g->part2_3_length = get_bits(&s->gb, 12); - g->big_values = get_bits(&s->gb, 9); - if(g->big_values > 288){ + g->big_values = get_bits(&s->gb, 9); + if (g->big_values > 288) { av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n"); - return -1; + return AVERROR_INVALIDDATA; } g->global_gain = get_bits(&s->gb, 8); @@ -1502,21 +1497,21 @@ static int mp_decode_layer3(MPADecodeContext *s) blocksplit_flag = get_bits1(&s->gb); if (blocksplit_flag) { g->block_type = get_bits(&s->gb, 2); - if (g->block_type == 0){ + if (g->block_type == 0) { av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n"); - return -1; + return AVERROR_INVALIDDATA; } g->switch_point = get_bits1(&s->gb); - for(i=0;i<2;i++) + for (i = 0; i < 2; i++) g->table_select[i] = get_bits(&s->gb, 5); - for(i=0;i<3;i++) + for (i = 0; i < 3; i++) g->subblock_gain[i] = get_bits(&s->gb, 3); ff_init_short_region(s, g); } else { int region_address1, region_address2; g->block_type = 0; g->switch_point = 0; - for(i=0;i<3;i++) + for (i = 0; i < 3; i++) g->table_select[i] = get_bits(&s->gb, 5); /* compute huffman coded region sizes */ region_address1 = get_bits(&s->gb, 4); @@ -1531,38 +1526,38 @@ static int mp_decode_layer3(MPADecodeContext *s) g->preflag = 0; if (!s->lsf) g->preflag = get_bits1(&s->gb); - g->scalefac_scale = get_bits1(&s->gb); + g->scalefac_scale = get_bits1(&s->gb); g->count1table_select = get_bits1(&s->gb); av_dlog(s->avctx, "block_type=%d switch_point=%d\n", g->block_type, g->switch_point); } } - if (!s->adu_mode) { - const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3); - assert((get_bits_count(&s->gb) & 7) == 0); - /* now we get bits from the main_data_begin offset */ - av_dlog(s->avctx, "seekback: %d\n", main_data_begin); -//av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size); + if (!s->adu_mode) { + const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3); + assert((get_bits_count(&s->gb) & 7) == 0); + /* now we get bits from the main_data_begin offset */ + av_dlog(s->avctx, "seekback: %d\n", main_data_begin); + //av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size); - memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES); - s->in_gb= s->gb; + memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES); + s->in_gb = s->gb; init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8); skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin)); - } + } - for(gr=0;gr<nb_granules;gr++) { - for(ch=0;ch<s->nb_channels;ch++) { + for (gr = 0; gr < nb_granules; gr++) { + for (ch = 0; ch < s->nb_channels; ch++) { g = &s->granules[ch][gr]; - if(get_bits_count(&s->gb)<0){ + if (get_bits_count(&s->gb) < 0) { av_log(s->avctx, AV_LOG_DEBUG, "mdb:%d, lastbuf:%d skipping granule %d\n", - main_data_begin, s->last_buf_size, gr); + main_data_begin, s->last_buf_size, gr); skip_bits_long(&s->gb, g->part2_3_length); memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid)); - if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer){ + if (get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer) { skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits); - s->gb= s->in_gb; - s->in_gb.buffer=NULL; + s->gb = s->in_gb; + s->in_gb.buffer = NULL; } continue; } @@ -1580,39 +1575,39 @@ static int mp_decode_layer3(MPADecodeContext *s) if (g->block_type == 2) { n = g->switch_point ? 17 : 18; j = 0; - if(slen1){ - for(i=0;i<n;i++) + if (slen1) { + for (i = 0; i < n; i++) g->scale_factors[j++] = get_bits(&s->gb, slen1); - }else{ - for(i=0;i<n;i++) + } else { + for (i = 0; i < n; i++) g->scale_factors[j++] = 0; } - if(slen2){ - for(i=0;i<18;i++) + if (slen2) { + for (i = 0; i < 18; i++) g->scale_factors[j++] = get_bits(&s->gb, slen2); - for(i=0;i<3;i++) + for (i = 0; i < 3; i++) g->scale_factors[j++] = 0; - }else{ - for(i=0;i<21;i++) + } else { + for (i = 0; i < 21; i++) g->scale_factors[j++] = 0; } } else { sc = s->granules[ch][0].scale_factors; j = 0; - for(k=0;k<4;k++) { - n = (k == 0 ? 6 : 5); + for (k = 0; k < 4; k++) { + n = k == 0 ? 6 : 5; if ((g->scfsi & (0x8 >> k)) == 0) { slen = (k < 2) ? slen1 : slen2; - if(slen){ - for(i=0;i<n;i++) + if (slen) { + for (i = 0; i < n; i++) g->scale_factors[j++] = get_bits(&s->gb, slen); - }else{ - for(i=0;i<n;i++) + } else { + for (i = 0; i < n; i++) g->scale_factors[j++] = 0; } } else { /* simply copy from last granule */ - for(i=0;i<n;i++) { + for (i = 0; i < n; i++) { g->scale_factors[j] = sc[j]; j++; } @@ -1624,11 +1619,11 @@ static int mp_decode_layer3(MPADecodeContext *s) int tindex, tindex2, slen[4], sl, sf; /* LSF scale factors */ - if (g->block_type == 2) { + if (g->block_type == 2) tindex = g->switch_point ? 2 : 1; - } else { + else tindex = 0; - } + sf = g->scalefac_compress; if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) { /* intensity stereo case */ @@ -1659,19 +1654,19 @@ static int mp_decode_layer3(MPADecodeContext *s) } j = 0; - for(k=0;k<4;k++) { - n = lsf_nsf_table[tindex2][tindex][k]; + for (k = 0; k < 4; k++) { + n = lsf_nsf_table[tindex2][tindex][k]; sl = slen[k]; - if(sl){ - for(i=0;i<n;i++) + if (sl) { + for (i = 0; i < n; i++) g->scale_factors[j++] = get_bits(&s->gb, sl); - }else{ - for(i=0;i<n;i++) + } else { + for (i = 0; i < n; i++) g->scale_factors[j++] = 0; } } /* XXX: should compute exact size */ - for(;j<40;j++) + for (; j < 40; j++) g->scale_factors[j] = 0; } @@ -1684,7 +1679,7 @@ static int mp_decode_layer3(MPADecodeContext *s) if (s->nb_channels == 2) compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]); - for(ch=0;ch<s->nb_channels;ch++) { + for (ch = 0; ch < s->nb_channels; ch++) { g = &s->granules[ch][gr]; reorder_block(s, g); @@ -1692,18 +1687,18 @@ static int mp_decode_layer3(MPADecodeContext *s) compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]); } } /* gr */ - if(get_bits_count(&s->gb)<0) + if (get_bits_count(&s->gb) < 0) skip_bits_long(&s->gb, -get_bits_count(&s->gb)); return nb_granules * 18; } -static int mp_decode_frame(MPADecodeContext *s, - OUT_INT *samples, const uint8_t *buf, int buf_size) +static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples, + const uint8_t *buf, int buf_size) { int i, nb_frames, ch; OUT_INT *samples_ptr; - init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8); + init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8); /* skip error protection field */ if (s->error_protection) @@ -1724,28 +1719,28 @@ static int mp_decode_frame(MPADecodeContext *s, nb_frames = mp_decode_layer3(s); s->last_buf_size=0; - if(s->in_gb.buffer){ + if (s->in_gb.buffer) { align_get_bits(&s->gb); - i= get_bits_left(&s->gb)>>3; - if(i >= 0 && i <= BACKSTEP_SIZE){ + i = get_bits_left(&s->gb)>>3; + if (i >= 0 && i <= BACKSTEP_SIZE) { memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i); s->last_buf_size=i; - }else + } else av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i); - s->gb= s->in_gb; - s->in_gb.buffer= NULL; + s->gb = s->in_gb; + s->in_gb.buffer = NULL; } align_get_bits(&s->gb); assert((get_bits_count(&s->gb) & 7) == 0); - i= get_bits_left(&s->gb)>>3; + i = get_bits_left(&s->gb) >> 3; - if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){ - if(i<0) + if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) { + if (i < 0) av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i); - i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE); + i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE); } - assert(i <= buf_size - HEADER_SIZE && i>= 0); + assert(i <= buf_size - HEADER_SIZE && i >= 0); memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i); s->last_buf_size += i; @@ -1753,9 +1748,9 @@ static int mp_decode_frame(MPADecodeContext *s, } /* apply the synthesis filter */ - for(ch=0;ch<s->nb_channels;ch++) { + for (ch = 0; ch < s->nb_channels; ch++) { samples_ptr = samples + ch; - for(i=0;i<nb_frames;i++) { + for (i = 0; i < nb_frames; i++) { RENAME(ff_mpa_synth_filter)( &s->mpadsp, s->synth_buf[ch], &(s->synth_buf_offset[ch]), @@ -1769,74 +1764,80 @@ static int mp_decode_frame(MPADecodeContext *s, return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels; } -static int decode_frame(AVCodecContext * avctx, - void *data, int *data_size, +static int decode_frame(AVCodecContext * avctx, void *data, int *data_size, AVPacket *avpkt) { - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; MPADecodeContext *s = avctx->priv_data; uint32_t header; int out_size; OUT_INT *out_samples = data; - if(buf_size < HEADER_SIZE) - return -1; + if (buf_size < HEADER_SIZE) + return AVERROR_INVALIDDATA; header = AV_RB32(buf); - if(ff_mpa_check_header(header) < 0){ + if (ff_mpa_check_header(header) < 0) { av_log(avctx, AV_LOG_ERROR, "Header missing\n"); - return -1; + return AVERROR_INVALIDDATA; } if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) { /* free format: prepare to compute frame size */ s->frame_size = -1; - return -1; + return AVERROR_INVALIDDATA; } /* update codec info */ - avctx->channels = s->nb_channels; + avctx->channels = s->nb_channels; avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO; if (!avctx->bit_rate) avctx->bit_rate = s->bit_rate; avctx->sub_id = s->layer; - if(*data_size < 1152*avctx->channels*sizeof(OUT_INT)) - return -1; + if (*data_size < avctx->frame_size * avctx->channels * sizeof(OUT_INT)) + return AVERROR(EINVAL); *data_size = 0; - if(s->frame_size<=0 || s->frame_size > buf_size){ + if (s->frame_size <= 0 || s->frame_size > buf_size) { av_log(avctx, AV_LOG_ERROR, "incomplete frame\n"); - return -1; + return AVERROR_INVALIDDATA; }else if(s->frame_size < buf_size){ av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n"); buf_size= s->frame_size; } out_size = mp_decode_frame(s, out_samples, buf, buf_size); - if(out_size>=0){ - *data_size = out_size; + if (out_size >= 0) { + *data_size = out_size; avctx->sample_rate = s->sample_rate; //FIXME maybe move the other codec info stuff from above here too - }else - av_log(avctx, AV_LOG_DEBUG, "Error while decoding MPEG audio frame.\n"); //FIXME return -1 / but also return the number of bytes consumed + } else { + av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n"); + /* Only return an error if the bad frame makes up the whole packet. + If there is more data in the packet, just consume the bad frame + instead of returning an error, which would discard the whole + packet. */ + if (buf_size == avpkt->size) + return out_size; + } s->frame_size = 0; return buf_size; } -static void flush(AVCodecContext *avctx){ +static void flush(AVCodecContext *avctx) +{ MPADecodeContext *s = avctx->priv_data; memset(s->synth_buf, 0, sizeof(s->synth_buf)); - s->last_buf_size= 0; + s->last_buf_size = 0; } #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER -static int decode_frame_adu(AVCodecContext * avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size, + AVPacket *avpkt) { - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; MPADecodeContext *s = avctx->priv_data; uint32_t header; int len, out_size; @@ -1846,8 +1847,8 @@ static int decode_frame_adu(AVCodecContext * avctx, // Discard too short frames if (buf_size < HEADER_SIZE) { - *data_size = 0; - return buf_size; + av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); + return AVERROR_INVALIDDATA; } @@ -1858,25 +1859,29 @@ static int decode_frame_adu(AVCodecContext * avctx, header = AV_RB32(buf) | 0xffe00000; if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame - *data_size = 0; - return buf_size; + av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n"); + return AVERROR_INVALIDDATA; } avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header); /* update codec info */ avctx->sample_rate = s->sample_rate; - avctx->channels = s->nb_channels; + avctx->channels = s->nb_channels; if (!avctx->bit_rate) avctx->bit_rate = s->bit_rate; avctx->sub_id = s->layer; + if (*data_size < avctx->frame_size * avctx->channels * sizeof(OUT_INT)) + return AVERROR(EINVAL); + s->frame_size = len; - if (avctx->parse_only) { +#if FF_API_PARSE_FRAME + if (avctx->parse_only) out_size = buf_size; - } else { - out_size = mp_decode_frame(s, out_samples, buf, buf_size); - } + else +#endif + out_size = mp_decode_frame(s, out_samples, buf, buf_size); *data_size = out_size; return buf_size; @@ -1889,9 +1894,9 @@ static int decode_frame_adu(AVCodecContext * avctx, * Context for MP3On4 decoder */ typedef struct MP3On4DecodeContext { - int frames; ///< number of mp3 frames per block (number of mp3 decoder instances) - int syncword; ///< syncword patch - const uint8_t *coff; ///< channels offsets in output buffer + int frames; ///< number of mp3 frames per block (number of mp3 decoder instances) + int syncword; ///< syncword patch + const uint8_t *coff; ///< channel offsets in output buffer MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance OUT_INT *decoded_buf; ///< output buffer for decoded samples } MP3On4DecodeContext; @@ -1899,17 +1904,20 @@ typedef struct MP3On4DecodeContext { #include "mpeg4audio.h" /* Next 3 arrays are indexed by channel config number (passed via codecdata) */ -static const uint8_t mp3Frames[8] = {0,1,1,2,3,3,4,5}; /* number of mp3 decoder instances */ + +/* number of mp3 decoder instances */ +static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 }; + /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */ static const uint8_t chan_offset[8][5] = { - {0}, - {0}, // C - {0}, // FLR - {2,0}, // C FLR - {2,0,3}, // C FLR BS - {2,0,3}, // C FLR BLRS - {2,0,4,3}, // C FLR BLRS LFE - {2,0,6,4,3}, // C FLR BLRS BLR LFE + { 0 }, + { 0 }, // C + { 0 }, // FLR + { 2, 0 }, // C FLR + { 2, 0, 3 }, // C FLR BS + { 2, 0, 3 }, // C FLR BLRS + { 2, 0, 4, 3 }, // C FLR BLRS LFE + { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE }; /* mp3on4 channel layouts */ @@ -1946,17 +1954,17 @@ static int decode_init_mp3on4(AVCodecContext * avctx) if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) { av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n"); - return -1; + return AVERROR_INVALIDDATA; } avpriv_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size); if (!cfg.chan_config || cfg.chan_config > 7) { av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n"); - return -1; + return AVERROR_INVALIDDATA; } - s->frames = mp3Frames[cfg.chan_config]; - s->coff = chan_offset[cfg.chan_config]; - avctx->channels = ff_mpeg4audio_channels[cfg.chan_config]; + s->frames = mp3Frames[cfg.chan_config]; + s->coff = chan_offset[cfg.chan_config]; + avctx->channels = ff_mpeg4audio_channels[cfg.chan_config]; avctx->channel_layout = chan_layout[cfg.chan_config]; if (cfg.sample_rate < 16000) @@ -2024,8 +2032,8 @@ static int decode_frame_mp3on4(AVCodecContext * avctx, void *data, int *data_size, AVPacket *avpkt) { - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; MP3On4DecodeContext *s = avctx->priv_data; MPADecodeContext *m; int fsize, len = buf_size, out_size = 0; @@ -2039,10 +2047,9 @@ static int decode_frame_mp3on4(AVCodecContext * avctx, return AVERROR(EINVAL); } - *data_size = 0; // Discard too short frames if (buf_size < HEADER_SIZE) - return -1; + return AVERROR_INVALIDDATA; // If only one decoder interleave is not needed outptr = s->frames == 1 ? out_samples : s->decoded_buf; @@ -2053,8 +2060,8 @@ static int decode_frame_mp3on4(AVCodecContext * avctx, for (fr = 0; fr < s->frames; fr++) { fsize = AV_RB16(buf) >> 4; fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE); - m = s->mp3decctx[fr]; - assert (m != NULL); + m = s->mp3decctx[fr]; + assert(m != NULL); header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header @@ -2071,23 +2078,23 @@ static int decode_frame_mp3on4(AVCodecContext * avctx, ch += m->nb_channels; out_size += mp_decode_frame(m, outptr, buf, fsize); - buf += fsize; - len -= fsize; + buf += fsize; + len -= fsize; - if(s->frames > 1) { + if (s->frames > 1) { n = m->avctx->frame_size*m->nb_channels; /* interleave output data */ bp = out_samples + s->coff[fr]; - if(m->nb_channels == 1) { - for(j = 0; j < n; j++) { + if (m->nb_channels == 1) { + for (j = 0; j < n; j++) { *bp = s->decoded_buf[j]; bp += avctx->channels; } } else { - for(j = 0; j < n; j++) { + for (j = 0; j < n; j++) { bp[0] = s->decoded_buf[j++]; bp[1] = s->decoded_buf[j]; - bp += avctx->channels; + bp += avctx->channels; } } } @@ -2111,7 +2118,9 @@ AVCodec ff_mp1_decoder = { .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame, +#if FF_API_PARSE_FRAME .capabilities = CODEC_CAP_PARSE_ONLY, +#endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"), }; @@ -2124,7 +2133,9 @@ AVCodec ff_mp2_decoder = { .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame, +#if FF_API_PARSE_FRAME .capabilities = CODEC_CAP_PARSE_ONLY, +#endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), }; @@ -2137,7 +2148,9 @@ AVCodec ff_mp3_decoder = { .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame, +#if FF_API_PARSE_FRAME .capabilities = CODEC_CAP_PARSE_ONLY, +#endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"), }; @@ -2150,7 +2163,9 @@ AVCodec ff_mp3adu_decoder = { .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame_adu, +#if FF_API_PARSE_FRAME .capabilities = CODEC_CAP_PARSE_ONLY, +#endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"), }; diff --git a/libavcodec/mpegaudiodec_float.c b/libavcodec/mpegaudiodec_float.c index 312b84278f..4482168a3e 100644 --- a/libavcodec/mpegaudiodec_float.c +++ b/libavcodec/mpegaudiodec_float.c @@ -30,7 +30,9 @@ AVCodec ff_mp1float_decoder = { .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame, +#if FF_API_PARSE_FRAME .capabilities = CODEC_CAP_PARSE_ONLY, +#endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"), }; @@ -43,7 +45,9 @@ AVCodec ff_mp2float_decoder = { .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame, +#if FF_API_PARSE_FRAME .capabilities = CODEC_CAP_PARSE_ONLY, +#endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), }; @@ -56,7 +60,9 @@ AVCodec ff_mp3float_decoder = { .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame, +#if FF_API_PARSE_FRAME .capabilities = CODEC_CAP_PARSE_ONLY, +#endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"), }; @@ -69,7 +75,9 @@ AVCodec ff_mp3adufloat_decoder = { .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame_adu, +#if FF_API_PARSE_FRAME .capabilities = CODEC_CAP_PARSE_ONLY, +#endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"), }; diff --git a/libavcodec/nellymoserdec.c b/libavcodec/nellymoserdec.c index 04d966173a..cd054826f1 100644 --- a/libavcodec/nellymoserdec.c +++ b/libavcodec/nellymoserdec.c @@ -48,7 +48,7 @@ typedef struct NellyMoserDecodeContext { AVCodecContext* avctx; float *float_buf; - float state[NELLY_BUF_LEN]; + DECLARE_ALIGNED(16, float, state)[NELLY_BUF_LEN]; AVLFG random_state; GetBitContext gb; float scale_bias; @@ -58,23 +58,6 @@ typedef struct NellyMoserDecodeContext { DECLARE_ALIGNED(32, float, imdct_out)[NELLY_BUF_LEN * 2]; } NellyMoserDecodeContext; -static void overlap_and_window(NellyMoserDecodeContext *s, float *state, float *audio, float *a_in) -{ - int bot, top; - - bot = 0; - top = NELLY_BUF_LEN-1; - - while (bot < NELLY_BUF_LEN) { - audio[bot] = a_in [bot]*ff_sine_128[bot] - +state[bot]*ff_sine_128[top]; - - bot++; - top--; - } - memcpy(state, a_in + NELLY_BUF_LEN, sizeof(float)*NELLY_BUF_LEN); -} - static void nelly_decode_block(NellyMoserDecodeContext *s, const unsigned char block[NELLY_BLOCK_LEN], float audio[NELLY_SAMPLES]) @@ -125,7 +108,9 @@ static void nelly_decode_block(NellyMoserDecodeContext *s, s->imdct_ctx.imdct_calc(&s->imdct_ctx, s->imdct_out, aptr); /* XXX: overlapping and windowing should be part of a more generic imdct function */ - overlap_and_window(s, s->state, aptr, s->imdct_out); + s->dsp.vector_fmul_reverse(s->state, s->state, ff_sine_128, NELLY_BUF_LEN); + s->dsp.vector_fmul_add(aptr, s->imdct_out, ff_sine_128, s->state, NELLY_BUF_LEN); + memcpy(s->state, s->imdct_out + NELLY_BUF_LEN, sizeof(float)*NELLY_BUF_LEN); } } @@ -172,20 +157,21 @@ static int decode_tag(AVCodecContext * avctx, float *samples_flt = data; *data_size = 0; - if (buf_size < avctx->block_align) { - return buf_size; - } - - if (buf_size % NELLY_BLOCK_LEN) { - av_log(avctx, AV_LOG_ERROR, "Tag size %d.\n", buf_size); - return buf_size; - } block_size = NELLY_SAMPLES * av_get_bytes_per_sample(avctx->sample_fmt); - blocks = FFMIN(buf_size / NELLY_BLOCK_LEN, data_max / block_size); + blocks = buf_size / NELLY_BLOCK_LEN; + if (blocks <= 0) { + av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); + return AVERROR_INVALIDDATA; + } + if (data_max < blocks * block_size) { av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); return AVERROR(EINVAL); } + if (buf_size % NELLY_BLOCK_LEN) { + av_log(avctx, AV_LOG_WARNING, "Leftover bytes: %d.\n", + buf_size % NELLY_BLOCK_LEN); + } /* Normal numbers of blocks for sample rates: * 8000 Hz - 1 * 11025 Hz - 2 diff --git a/libavcodec/nellymoserenc.c b/libavcodec/nellymoserenc.c index 1d35cda9a1..5af1c5c6ca 100644 --- a/libavcodec/nellymoserenc.c +++ b/libavcodec/nellymoserenc.c @@ -146,7 +146,7 @@ static av_cold int encode_init(AVCodecContext *avctx) avctx->frame_size = NELLY_SAMPLES; s->avctx = avctx; - ff_mdct_init(&s->mdct_ctx, 8, 0, 1.0); + ff_mdct_init(&s->mdct_ctx, 8, 0, 32768.0); dsputil_init(&s->dsp, avctx); /* Generate overlap window */ @@ -352,17 +352,15 @@ static void encode_block(NellyMoserEncodeContext *s, unsigned char *output, int static int encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size, void *data) { NellyMoserEncodeContext *s = avctx->priv_data; - const int16_t *samples = data; + const float *samples = data; int i; if (s->last_frame) return 0; if (data) { - for (i = 0; i < avctx->frame_size; i++) { - s->buf[s->bufsel][i] = samples[i]; - } - for (; i < NELLY_SAMPLES; i++) { + memcpy(s->buf[s->bufsel], samples, avctx->frame_size * sizeof(*samples)); + for (i = avctx->frame_size; i < NELLY_SAMPLES; i++) { s->buf[s->bufsel][i] = 0; } s->bufsel = 1 - s->bufsel; @@ -393,5 +391,5 @@ AVCodec ff_nellymoser_encoder = { .close = encode_end, .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY, .long_name = NULL_IF_CONFIG_SMALL("Nellymoser Asao"), - .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, + .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE}, }; diff --git a/libavcodec/snow.c b/libavcodec/snow.c index a4ebf803e0..ca01f9c684 100644 --- a/libavcodec/snow.c +++ b/libavcodec/snow.c @@ -1665,7 +1665,7 @@ static int frame_start(SnowContext *s){ int w= s->avctx->width; //FIXME round up to x16 ? int h= s->avctx->height; - if(s->current_picture.data[0] && !(s->avctx->flags&CODEC_FLAG_EMU_EDGE)){ + if (s->current_picture.data[0] && !(s->avctx->flags&CODEC_FLAG_EMU_EDGE)) { s->dsp.draw_edges(s->current_picture.data[0], s->current_picture.linesize[0], w , h , EDGE_WIDTH , EDGE_WIDTH , EDGE_TOP | EDGE_BOTTOM); diff --git a/libavcodec/version.h b/libavcodec/version.h index 0fd5e72874..f8b9920b19 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -21,7 +21,7 @@ #define AVCODEC_VERSION_H #define LIBAVCODEC_VERSION_MAJOR 53 -#define LIBAVCODEC_VERSION_MINOR 23 +#define LIBAVCODEC_VERSION_MINOR 24 #define LIBAVCODEC_VERSION_MICRO 0 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ @@ -101,5 +101,8 @@ #ifndef FF_API_GET_ALPHA_INFO #define FF_API_GET_ALPHA_INFO (LIBAVCODEC_VERSION_MAJOR < 54) #endif +#ifndef FF_API_PARSE_FRAME +#define FF_API_PARSE_FRAME (LIBAVCODEC_VERSION_MAJOR < 54) +#endif #endif /* AVCODEC_VERSION_H */ diff --git a/libavcodec/wmadec.c b/libavcodec/wmadec.c index b4a4c800af..bf6e7ee71e 100644 --- a/libavcodec/wmadec.c +++ b/libavcodec/wmadec.c @@ -816,7 +816,7 @@ static int wma_decode_superframe(AVCodecContext *avctx, const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; WMACodecContext *s = avctx->priv_data; - int nb_frames, bit_offset, i, pos, len; + int nb_frames, bit_offset, i, pos, len, out_size; uint8_t *q; int16_t *samples; @@ -838,13 +838,19 @@ static int wma_decode_superframe(AVCodecContext *avctx, if (s->use_bit_reservoir) { /* read super frame header */ skip_bits(&s->gb, 4); /* super frame index */ - nb_frames = get_bits(&s->gb, 4) - 1; + nb_frames = get_bits(&s->gb, 4) - (s->last_superframe_len <= 0); + } else { + nb_frames = 1; + } - if((nb_frames+1) * s->nb_channels * s->frame_len * sizeof(int16_t) > *data_size){ - av_log(s->avctx, AV_LOG_ERROR, "Insufficient output space\n"); - goto fail; - } + out_size = nb_frames * s->frame_len * s->nb_channels * + av_get_bytes_per_sample(avctx->sample_fmt); + if (*data_size < out_size) { + av_log(s->avctx, AV_LOG_ERROR, "Insufficient output space\n"); + goto fail; + } + if (s->use_bit_reservoir) { bit_offset = get_bits(&s->gb, s->byte_offset_bits + 3); if (s->last_superframe_len > 0) { @@ -873,6 +879,7 @@ static int wma_decode_superframe(AVCodecContext *avctx, if (wma_decode_frame(s, samples) < 0) goto fail; samples += s->nb_channels * s->frame_len; + nb_frames--; } /* read each frame starting from bit_offset */ @@ -901,10 +908,6 @@ static int wma_decode_superframe(AVCodecContext *avctx, s->last_superframe_len = len; memcpy(s->last_superframe, buf + pos, len); } else { - if(s->nb_channels * s->frame_len * sizeof(int16_t) > *data_size){ - av_log(s->avctx, AV_LOG_ERROR, "Insufficient output space\n"); - goto fail; - } /* single frame decode */ if (wma_decode_frame(s, samples) < 0) goto fail; @@ -912,7 +915,7 @@ static int wma_decode_superframe(AVCodecContext *avctx, } //av_log(NULL, AV_LOG_ERROR, "%d %d %d %d outbytes:%d eaten:%d\n", s->frame_len_bits, s->block_len_bits, s->frame_len, s->block_len, (int8_t *)samples - (int8_t *)data, s->block_align); - *data_size = (int8_t *)samples - (int8_t *)data; + *data_size = out_size; return buf_size; fail: /* when error, we reset the bit reservoir */ diff --git a/libavcodec/wmaprodec.c b/libavcodec/wmaprodec.c index 119027b7b7..868a28393d 100644 --- a/libavcodec/wmaprodec.c +++ b/libavcodec/wmaprodec.c @@ -86,12 +86,14 @@ * subframe in order to reconstruct the output samples. */ +#include "libavutil/intreadwrite.h" #include "avcodec.h" #include "internal.h" #include "get_bits.h" #include "put_bits.h" #include "wmaprodata.h" #include "dsputil.h" +#include "fmtconvert.h" #include "sinewin.h" #include "wma.h" @@ -166,6 +168,7 @@ typedef struct WMAProDecodeCtx { /* generic decoder variables */ AVCodecContext* avctx; ///< codec context for av_log DSPContext dsp; ///< accelerated DSP functions + FmtConvertContext fmt_conv; uint8_t frame_data[MAX_FRAMESIZE + FF_INPUT_BUFFER_PADDING_SIZE];///< compressed frame data PutBitContext pb; ///< context for filling the frame_data buffer @@ -279,6 +282,7 @@ static av_cold int decode_init(AVCodecContext *avctx) s->avctx = avctx; dsputil_init(&s->dsp, avctx); + ff_fmt_convert_init(&s->fmt_conv, avctx); init_put_bits(&s->pb, s->frame_data, MAX_FRAMESIZE); avctx->sample_fmt = AV_SAMPLE_FMT_FLT; @@ -767,7 +771,7 @@ static int decode_coeffs(WMAProDecodeCtx *s, int c) /* Integers 0..15 as single-precision floats. The table saves a costly int to float conversion, and storing the values as integers allows fast sign-flipping. */ - static const int fval_tab[16] = { + static const uint32_t fval_tab[16] = { 0x00000000, 0x3f800000, 0x40000000, 0x40400000, 0x40800000, 0x40a00000, 0x40c00000, 0x40e00000, 0x41000000, 0x41100000, 0x41200000, 0x41300000, @@ -799,7 +803,7 @@ static int decode_coeffs(WMAProDecodeCtx *s, int c) 4 vector coded large values) */ while ((s->transmit_num_vec_coeffs || !rl_mode) && (cur_coeff + 3 < ci->num_vec_coeffs)) { - int vals[4]; + uint32_t vals[4]; int i; unsigned int idx; @@ -809,15 +813,15 @@ static int decode_coeffs(WMAProDecodeCtx *s, int c) for (i = 0; i < 4; i += 2) { idx = get_vlc2(&s->gb, vec2_vlc.table, VLCBITS, VEC2MAXDEPTH); if (idx == HUFF_VEC2_SIZE - 1) { - int v0, v1; + uint32_t v0, v1; v0 = get_vlc2(&s->gb, vec1_vlc.table, VLCBITS, VEC1MAXDEPTH); if (v0 == HUFF_VEC1_SIZE - 1) v0 += ff_wma_get_large_val(&s->gb); v1 = get_vlc2(&s->gb, vec1_vlc.table, VLCBITS, VEC1MAXDEPTH); if (v1 == HUFF_VEC1_SIZE - 1) v1 += ff_wma_get_large_val(&s->gb); - ((float*)vals)[i ] = v0; - ((float*)vals)[i+1] = v1; + vals[i ] = ((av_alias32){ .f32 = v0 }).u32; + vals[i+1] = ((av_alias32){ .f32 = v1 }).u32; } else { vals[i] = fval_tab[symbol_to_vec2[idx] >> 4 ]; vals[i+1] = fval_tab[symbol_to_vec2[idx] & 0xF]; @@ -833,8 +837,8 @@ static int decode_coeffs(WMAProDecodeCtx *s, int c) /** decode sign */ for (i = 0; i < 4; i++) { if (vals[i]) { - int sign = get_bits1(&s->gb) - 1; - *(uint32_t*)&ci->coeffs[cur_coeff] = vals[i] ^ sign<<31; + uint32_t sign = get_bits1(&s->gb) - 1; + AV_WN32A(&ci->coeffs[cur_coeff], vals[i] ^ sign << 31); num_zeros = 0; } else { ci->coeffs[cur_coeff] = 0; @@ -1281,6 +1285,7 @@ static int decode_frame(WMAProDecodeCtx *s) int more_frames = 0; int len = 0; int i; + const float *out_ptr[WMAPRO_MAX_CHANNELS]; /** check for potential output buffer overflow */ if (s->num_channels * s->samples_per_frame > s->samples_end - s->samples) { @@ -1356,18 +1361,12 @@ static int decode_frame(WMAProDecodeCtx *s) } /** interleave samples and write them to the output buffer */ - for (i = 0; i < s->num_channels; i++) { - float* ptr = s->samples + i; - int incr = s->num_channels; - float* iptr = s->channel[i].out; - float* iend = iptr + s->samples_per_frame; - - // FIXME should create/use a DSP function here - while (iptr < iend) { - *ptr = *iptr++; - ptr += incr; - } + for (i = 0; i < s->num_channels; i++) + out_ptr[i] = s->channel[i].out; + s->fmt_conv.float_interleave(s->samples, out_ptr, s->samples_per_frame, + s->num_channels); + for (i = 0; i < s->num_channels; i++) { /** reuse second half of the IMDCT output for the next frame */ memcpy(&s->channel[i].out[0], &s->channel[i].out[s->samples_per_frame], diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index b7a6f88a5b..ca7b368f63 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -1730,7 +1730,7 @@ static int synth_superframe(AVCodecContext *ctx, { WMAVoiceContext *s = ctx->priv_data; GetBitContext *gb = &s->gb, s_gb; - int n, res, n_samples = 480; + int n, res, out_size, n_samples = 480; double lsps[MAX_FRAMES][MAX_LSPS]; const double *mean_lsf = s->lsps == 16 ? wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode]; @@ -1748,7 +1748,10 @@ static int synth_superframe(AVCodecContext *ctx, s->sframe_cache_size = 0; } - if ((res = check_bits_for_superframe(gb, s)) == 1) return 1; + if ((res = check_bits_for_superframe(gb, s)) == 1) { + *data_size = 0; + return 1; + } /* First bit is speech/music bit, it differentiates between WMAVoice * speech samples (the actual codec) and WMAVoice music samples, which @@ -1789,6 +1792,14 @@ static int synth_superframe(AVCodecContext *ctx, stabilize_lsps(lsps[n], s->lsps); } + out_size = n_samples * av_get_bytes_per_sample(ctx->sample_fmt); + if (*data_size < out_size) { + av_log(ctx, AV_LOG_ERROR, + "Output buffer too small (%d given - %zu needed)\n", + *data_size, out_size); + return -1; + } + /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */ for (n = 0; n < 3; n++) { if (!s->has_residual_lsps) { @@ -1808,8 +1819,10 @@ static int synth_superframe(AVCodecContext *ctx, &samples[n * MAX_FRAMESIZE], lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], &excitation[s->history_nsamples + n * MAX_FRAMESIZE], - &synth[s->lsps + n * MAX_FRAMESIZE]))) + &synth[s->lsps + n * MAX_FRAMESIZE]))) { + *data_size = 0; return res; + } } /* Statistics? FIXME - we don't check for length, a slight overrun @@ -1821,7 +1834,7 @@ static int synth_superframe(AVCodecContext *ctx, } /* Specify nr. of output samples */ - *data_size = n_samples * sizeof(float); + *data_size = out_size; /* Update history */ memcpy(s->prev_lsps, lsps[2], @@ -1915,22 +1928,16 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, GetBitContext *gb = &s->gb; int size, res, pos; - if (*data_size < 480 * sizeof(float)) { - av_log(ctx, AV_LOG_ERROR, - "Output buffer too small (%d given - %zu needed)\n", - *data_size, 480 * sizeof(float)); - return -1; - } - *data_size = 0; - /* Packets are sometimes a multiple of ctx->block_align, with a packet * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer * feeds us ASF packets, which may concatenate multiple "codec" packets * in a single "muxer" packet, so we artificially emulate that by * capping the packet size at ctx->block_align. */ for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); - if (!size) + if (!size) { + *data_size = 0; return 0; + } init_get_bits(&s->gb, avpkt->data, size << 3); /* size == ctx->block_align is used to indicate whether we are dealing with |