diff options
| author | Michael Niedermayer <michaelni@gmx.at> | 2011-12-03 02:08:55 +0100 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2011-12-03 03:00:30 +0100 |
| commit | e4de71677f3adeac0f74b89ac8df5d417364df2c (patch) | |
| tree | 4792dd8d85d24f0f4eaddabb65f6044727907daa | |
| parent | 12804348f5babf56a315fa01751eea1ffdddf98a (diff) | |
| parent | d268b79e3436107c11ee8bcdf9f3645368bb3fcd (diff) | |
| download | ffmpeg-e4de71677f3adeac0f74b89ac8df5d417364df2c.tar.gz | |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
aac_latm: reconfigure decoder on audio specific config changes
latmdec: fix audio specific config parsing
Add avcodec_decode_audio4().
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Update developers documentation with coding conventions.
svq1dec: avoid undefined get_bits(0) call
ARM: h264dsp_neon cosmetics
ARM: make some NEON macros reusable
Do not memcpy raw video frames when using null muxer
fate: update asf seektest
vp8: flush buffers on size changes.
doc: improve general documentation for MacOSX
asf: use packet dts as approximation of pts
asf: do not call av_read_frame
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
Cleaned up alacenc.c
Conflicts:
doc/APIchanges
doc/developer.texi
libavcodec/8svx.c
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/nellymoserdec.c
libavcodec/tta.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/asfdec.c
tests/ref/seek/lavf_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
87 files changed, 2227 insertions, 1308 deletions
@@ -1267,7 +1267,8 @@ static void do_video_out(AVFormatContext *s, av_init_packet(&pkt); pkt.stream_index= ost->index; - if (s->oformat->flags & AVFMT_RAWPICTURE) { + if (s->oformat->flags & AVFMT_RAWPICTURE && + enc->codec->id == CODEC_ID_RAWVIDEO) { /* raw pictures are written as AVPicture structure to avoid any copies. We support temporarily the older method. */ @@ -1528,7 +1529,7 @@ static void flush_encoders(OutputStream *ost_table, int nb_ostreams) if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <=1) continue; - if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (os->oformat->flags & AVFMT_RAWPICTURE)) + if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (os->oformat->flags & AVFMT_RAWPICTURE) && enc->codec->id == CODEC_ID_RAWVIDEO) continue; for(;;) { diff --git a/doc/APIchanges b/doc/APIchanges index 97a5c0068d..23e568922c 100644 --- a/doc/APIchanges +++ b/doc/APIchanges @@ -22,6 +22,19 @@ API changes, most recent first: 2011-10-20 - b35e9e1 - lavu 51.22.0 Add av_strtok() to avstring.h. +2011-xx-xx - xxxxxxx - lavc 53.25.0 + Add nb_samples and extended_data fields to AVFrame. + Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE. + Deprecate avcodec_decode_audio3() in favor of avcodec_decode_audio4(). + avcodec_decode_audio4() writes output samples to an AVFrame, which allows + audio decoders to use get_buffer(). + +2011-xx-xx - xxxxxxx - lavc 53.24.0 + Change AVFrame.data[4]/base[4]/linesize[4]/error[4] to [8] at next major bump. + Change AVPicture.data[4]/linesize[4] to [8] at next major bump. + Change AVCodecContext.error[4] to [8] at next major bump. + Add AV_NUM_DATA_POINTERS to simplify the bump transition. + 2011-11-23 - bbb46f3 - lavu 51.18.0 Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and av_samples_alloc(), to samplefmt.h. diff --git a/doc/developer.texi b/doc/developer.texi index 2052854a40..800ca7d045 100644 --- a/doc/developer.texi +++ b/doc/developer.texi @@ -53,48 +53,26 @@ and should try to fix issues their commit causes. @anchor{Coding Rules} @section Coding Rules -FFmpeg is programmed in the ISO C90 language with a few additional -features from ISO C99, namely: -@itemize @bullet -@item -the @samp{inline} keyword; -@item -@samp{//} comments; -@item -designated struct initializers (@samp{struct s x = @{ .i = 17 @};}) -@item -compound literals (@samp{x = (struct s) @{ 17, 23 @};}) -@end itemize - -These features are supported by all compilers we care about, so we will not -accept patches to remove their use unless they absolutely do not impair -clarity and performance. +@subsection Code formatting conventions -All code must compile with recent versions of GCC and a number of other -currently supported compilers. To ensure compatibility, please do not use -additional C99 features or GCC extensions. Especially watch out for: +There are the following guidelines regarding the indentation in files: @itemize @bullet @item -mixing statements and declarations; -@item -@samp{long long} (use @samp{int64_t} instead); -@item -@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar; -@item -GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}). -@end itemize - Indent size is 4. -The presentation is one inspired by 'indent -i4 -kr -nut'. +@item The TAB character is forbidden outside of Makefiles as is any form of trailing whitespace. Commits containing either will be rejected by the git repository. +@item +You should try to limit your code lines to 80 characters; however, do so if and only if this improves readability. +@end itemize +The presentation is one inspired by 'indent -i4 -kr -nut'. The main priority in FFmpeg is simplicity and small code size in order to minimize the bug count. -Comments: Use the JavaDoc/Doxygen -format (see examples below) so that code documentation +@subsection Comments +Use the JavaDoc/Doxygen format (see examples below) so that code documentation can be generated automatically. All nontrivial functions should have a comment above them explaining what the function does, even if it is just one sentence. All structures and their member variables should be documented, too. @@ -128,11 +106,69 @@ int myfunc(int my_parameter) ... @end example +@subsection C language features + +FFmpeg is programmed in the ISO C90 language with a few additional +features from ISO C99, namely: +@itemize @bullet +@item +the @samp{inline} keyword; +@item +@samp{//} comments; +@item +designated struct initializers (@samp{struct s x = @{ .i = 17 @};}) +@item +compound literals (@samp{x = (struct s) @{ 17, 23 @};}) +@end itemize + +These features are supported by all compilers we care about, so we will not +accept patches to remove their use unless they absolutely do not impair +clarity and performance. + +All code must compile with recent versions of GCC and a number of other +currently supported compilers. To ensure compatibility, please do not use +additional C99 features or GCC extensions. Especially watch out for: +@itemize @bullet +@item +mixing statements and declarations; +@item +@samp{long long} (use @samp{int64_t} instead); +@item +@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar; +@item +GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}). +@end itemize + +@subsection Naming conventions +All names are using underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is +a valid function name and @samp{AVFilterGetVideo} is not. The only exception from this are structure names; +they should always be in the CamelCase + +There are following conventions for naming variables and functions: +@itemize @bullet +@item +For local variables no prefix is required. +@item +For variables and functions declared as @code{static} no prefixes are required. +@item +For variables and functions used internally by the library, @code{ff_} prefix should be used. +For example, @samp{ff_w64_demuxer}. +@item +For variables and functions used internally across multiple libraries, use @code{avpriv_}. For example, +@samp{avpriv_aac_parse_header}. +@item +For exported names, each library has its own prefixes. Just check the existing code and name accordingly. +@end itemize + +@subsection Miscellanous conventions +@itemize @bullet +@item fprintf and printf are forbidden in libavformat and libavcodec, please use av_log() instead. - +@item Casts should be used only when necessary. Unneeded parentheses should also be avoided if they don't make the code easier to understand. +@end itemize @section Development Policy diff --git a/doc/general.texi b/doc/general.texi index 120b7160c2..04ca71db91 100644 --- a/doc/general.texi +++ b/doc/general.texi @@ -840,13 +840,22 @@ bash directly to work around this: bash ./configure @end example -@subsection Darwin (MacOS X, iPhone) +@anchor{Darwin} +@subsection Darwin (OSX, iPhone) -MacOS X on PowerPC or ARM (iPhone) requires a preprocessor from +The toolchain provided with Xcode is sufficient to build the basic +unacelerated code. + +OSX on PowerPC or ARM (iPhone) requires a preprocessor from @url{http://github.com/yuvi/gas-preprocessor} to build the optimized assembler functions. Just download the Perl script and put it somewhere in your PATH, FFmpeg's configure will pick it up automatically. +OSX on amd64 and x86 requires @command{yasm} to build most of the +optimized assembler functions @url{http://mxcl.github.com/homebrew/, Homebrew}, +@url{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix} +or @url{http://www.macports.org, MacPorts} can easily provide it. + @section Windows To get help and instructions for building FFmpeg under Windows, check out @@ -1295,7 +1295,8 @@ static void do_video_out(AVFormatContext *s, av_init_packet(&pkt); pkt.stream_index= ost->index; - if (s->oformat->flags & AVFMT_RAWPICTURE) { + if (s->oformat->flags & AVFMT_RAWPICTURE && + enc->codec->id == CODEC_ID_RAWVIDEO) { /* raw pictures are written as AVPicture structure to avoid any copies. We support temporarily the older method. */ @@ -1560,7 +1561,7 @@ static void flush_encoders(OutputStream *ost_table, int nb_ostreams) if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <=1) continue; - if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (os->oformat->flags & AVFMT_RAWPICTURE)) + if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (os->oformat->flags & AVFMT_RAWPICTURE) && enc->codec->id == CODEC_ID_RAWVIDEO) continue; for(;;) { diff --git a/libavcodec/8svx.c b/libavcodec/8svx.c index efe554adc2..b4fc899a22 100644 --- a/libavcodec/8svx.c +++ b/libavcodec/8svx.c @@ -41,6 +41,7 @@ /** decoder context */ typedef struct EightSvxContext { + AVFrame frame; const int8_t *table; /* buffer used to store the whole audio decoded/interleaved chunk, @@ -99,11 +100,13 @@ static int delta_decode(int8_t *dst, const uint8_t *src, int src_size, return dst-dst0; } -static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size, - AVPacket *avpkt) +/** decode a frame */ +static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { EightSvxContext *esc = avctx->priv_data; - int out_data_size, n; + int n, out_data_size, ret; + uint8_t *out_date; uint8_t *src, *dst; /* decode and interleave the first packet */ @@ -145,19 +148,22 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_si memcpy(esc->samples, deinterleaved_samples, esc->samples_size); } - /* return single packed with fixed size */ - out_data_size = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx); - if (*data_size < out_data_size) { - av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n", *data_size); - return AVERROR(EINVAL); + /* get output buffer */ + esc->frame.nb_samples = (FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) +avctx->channels-1) / avctx->channels; + if ((ret = avctx->get_buffer(avctx, &esc->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } - *data_size = out_data_size; - dst = data; + *got_frame_ptr = 1; + *(AVFrame *)data = esc->frame; + + dst = esc->frame.data[0]; src = esc->samples + esc->samples_idx; + out_data_size = esc->frame.nb_samples * avctx->channels; for (n = out_data_size; n > 0; n--) *dst++ = *src++ + 128; - esc->samples_idx += *data_size; + esc->samples_idx += out_data_size; return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ? (avctx->frame_number == 0)*2 + out_data_size / 2 : @@ -184,6 +190,9 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx) } avctx->sample_fmt = AV_SAMPLE_FMT_U8; + avcodec_get_frame_defaults(&esc->frame); + avctx->coded_frame = &esc->frame; + return 0; } @@ -206,6 +215,7 @@ AVCodec ff_eightsvx_fib_decoder = { .init = eightsvx_decode_init, .decode = eightsvx_decode_frame, .close = eightsvx_decode_close, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"), }; @@ -217,6 +227,7 @@ AVCodec ff_eightsvx_exp_decoder = { .init = eightsvx_decode_init, .decode = eightsvx_decode_frame, .close = eightsvx_decode_close, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"), }; @@ -228,5 +239,6 @@ AVCodec ff_pcm_s8_planar_decoder = { .init = eightsvx_decode_init, .close = eightsvx_decode_close, .decode = eightsvx_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"), }; diff --git a/libavcodec/aac.h b/libavcodec/aac.h index d01534de3f..d31103684c 100644 --- a/libavcodec/aac.h +++ b/libavcodec/aac.h @@ -251,6 +251,7 @@ typedef struct { */ typedef struct { AVCodecContext *avctx; + AVFrame frame; MPEG4AudioConfig m4ac; diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c index 5a2b230d24..a046d991e6 100644 --- a/libavcodec/aacdec.c +++ b/libavcodec/aacdec.c @@ -471,15 +471,17 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx, * @param ac pointer to AACContext, may be null * @param avctx pointer to AVCCodecContext, used for logging * @param m4ac pointer to MPEG4AudioConfig, used for parsing - * @param data pointer to AVCodecContext extradata - * @param data_size size of AVCCodecContext extradata + * @param data pointer to buffer holding an audio specific config + * @param bit_size size of audio specific config or data in bits + * @param sync_extension look for an appended sync extension * * @return Returns error status or number of consumed bits. <0 - error */ static int decode_audio_specific_config(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, - const uint8_t *data, int data_size, int asclen) + const uint8_t *data, int bit_size, + int sync_extension) { GetBitContext gb; int i; @@ -489,9 +491,9 @@ static int decode_audio_specific_config(AACContext *ac, av_dlog(avctx, "%02x ", avctx->extradata[i]); av_dlog(avctx, "\n"); - init_get_bits(&gb, data, data_size * 8); + init_get_bits(&gb, data, bit_size); - if ((i = avpriv_mpeg4audio_get_config(m4ac, data, asclen/8)) < 0) + if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0) return -1; if (m4ac->sampling_index > 12) { av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index); @@ -591,7 +593,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx) if (avctx->extradata_size > 0) { if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac, avctx->extradata, - avctx->extradata_size, 8*avctx->extradata_size) < 0) + avctx->extradata_size*8, 1) < 0) return -1; } else { int sr, i; @@ -665,6 +667,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx) cbrt_tableinit(); + avcodec_get_frame_defaults(&ac->frame); + avctx->coded_frame = &ac->frame; + return 0; } @@ -2132,12 +2137,12 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb) } static int aac_decode_frame_int(AVCodecContext *avctx, void *data, - int *data_size, GetBitContext *gb) + int *got_frame_ptr, GetBitContext *gb) { AACContext *ac = avctx->priv_data; ChannelElement *che = NULL, *che_prev = NULL; enum RawDataBlockType elem_type, elem_type_prev = TYPE_END; - int err, elem_id, data_size_tmp; + int err, elem_id; int samples = 0, multiplier, audio_found = 0; if (show_bits(gb, 12) == 0xfff) { @@ -2250,24 +2255,26 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data, avctx->frame_size = samples; } - data_size_tmp = samples * avctx->channels * - av_get_bytes_per_sample(avctx->sample_fmt); - if (*data_size < data_size_tmp) { - av_log(avctx, AV_LOG_ERROR, - "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", - *data_size, data_size_tmp); - return -1; - } - *data_size = data_size_tmp; - if (samples) { + /* get output buffer */ + ac->frame.nb_samples = samples; + if ((err = avctx->get_buffer(avctx, &ac->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return err; + } + if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) - ac->fmt_conv.float_interleave(data, (const float **)ac->output_data, + ac->fmt_conv.float_interleave((float *)ac->frame.data[0], + (const float **)ac->output_data, samples, avctx->channels); else - ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, + ac->fmt_conv.float_to_int16_interleave((int16_t *)ac->frame.data[0], + (const float **)ac->output_data, samples, avctx->channels); + + *(AVFrame *)data = ac->frame; } + *got_frame_ptr = !!samples; if (ac->output_configured && audio_found) ac->output_configured = OC_LOCKED; @@ -2276,7 +2283,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data, } static int aac_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; @@ -2287,7 +2294,7 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data, init_get_bits(&gb, buf, buf_size * 8); - if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0) + if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb)) < 0) return err; buf_consumed = (get_bits_count(&gb) + 7) >> 3; @@ -2340,30 +2347,40 @@ static inline uint32_t latm_get_value(GetBitContext *b) static int latm_decode_audio_specific_config(struct LATMContext *latmctx, GetBitContext *gb, int asclen) { - AVCodecContext *avctx = latmctx->aac_ctx.avctx; - AACContext *ac= &latmctx->aac_ctx; - MPEG4AudioConfig m4ac=ac->m4ac; - int config_start_bit = get_bits_count(gb); - int bits_consumed, esize; + AACContext *ac = &latmctx->aac_ctx; + AVCodecContext *avctx = ac->avctx; + MPEG4AudioConfig m4ac = {0}; + int config_start_bit = get_bits_count(gb); + int sync_extension = 0; + int bits_consumed, esize; + + if (asclen) { + sync_extension = 1; + asclen = FFMIN(asclen, get_bits_left(gb)); + } else + asclen = get_bits_left(gb); if (config_start_bit % 8) { av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific " "config not byte aligned.\n", 1); return AVERROR_INVALIDDATA; - } else { - bits_consumed = - decode_audio_specific_config(ac, avctx, &m4ac, + } + bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac, gb->buffer + (config_start_bit / 8), - get_bits_left(gb) / 8, asclen); + asclen, sync_extension); - if (bits_consumed < 0) - return AVERROR_INVALIDDATA; - if(ac->m4ac.sample_rate != m4ac.sample_rate || m4ac.chan_config != ac->m4ac.chan_config) - ac->m4ac= m4ac; + if (bits_consumed < 0) + return AVERROR_INVALIDDATA; + + if (ac->m4ac.sample_rate != m4ac.sample_rate || + ac->m4ac.chan_config != m4ac.chan_config) { + + av_log(avctx, AV_LOG_INFO, "audio config changed\n"); + latmctx->initialized = 0; esize = (bits_consumed+7) / 8; - if (avctx->extradata_size <= esize) { + if (avctx->extradata_size < esize) { av_free(avctx->extradata); avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE); if (!avctx->extradata) @@ -2373,9 +2390,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, avctx->extradata_size = esize; memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize); memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE); - - skip_bits_long(gb, bits_consumed); } + skip_bits_long(gb, bits_consumed); return bits_consumed; } @@ -2512,8 +2528,8 @@ static int read_audio_mux_element(struct LATMContext *latmctx, } -static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size, - AVPacket *avpkt) +static int latm_decode_frame(AVCodecContext *avctx, void *out, + int *got_frame_ptr, AVPacket *avpkt) { struct LATMContext *latmctx = avctx->priv_data; int muxlength, err; @@ -2535,12 +2551,12 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size, if (!latmctx->initialized) { if (!avctx->extradata) { - *out_size = 0; + *got_frame_ptr = 0; return avpkt->size; } else { if ((err = decode_audio_specific_config( &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.m4ac, - avctx->extradata, avctx->extradata_size, 8*avctx->extradata_size)) < 0) + avctx->extradata, avctx->extradata_size*8, 1)) < 0) return err; latmctx->initialized = 1; } @@ -2553,7 +2569,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size, return AVERROR_INVALIDDATA; } - if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0) + if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb)) < 0) return err; return muxlength; @@ -2583,7 +2599,7 @@ AVCodec ff_aac_decoder = { .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, - .capabilities = CODEC_CAP_CHANNEL_CONF, + .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1, .channel_layouts = aac_channel_layout, }; @@ -2604,7 +2620,7 @@ AVCodec ff_aac_latm_decoder = { .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, - .capabilities = CODEC_CAP_CHANNEL_CONF, + .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1, .channel_layouts = aac_channel_layout, .flush = flush, }; diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c index 09b9a3102c..c650881430 100644 --- a/libavcodec/ac3dec.c +++ b/libavcodec/ac3dec.c @@ -208,6 +208,9 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx) } s->downmixed = 1; + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -1296,16 +1299,15 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) /** * Decode a single AC-3 frame. */ -static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, - AVPacket *avpkt) +static int ac3_decode_frame(AVCodecContext * avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; AC3DecodeContext *s = avctx->priv_data; - float *out_samples_flt = data; - int16_t *out_samples_s16 = data; - int blk, ch, err; - int data_size_orig, data_size_tmp; + float *out_samples_flt; + int16_t *out_samples_s16; + int blk, ch, err, ret; const uint8_t *channel_map; const float *output[AC3_MAX_CHANNELS]; @@ -1322,8 +1324,6 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, init_get_bits(&s->gbc, buf, buf_size * 8); /* parse the syncinfo */ - data_size_orig = *data_size; - *data_size = 0; err = parse_frame_header(s); if (err) { @@ -1345,6 +1345,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, /* TODO: add support for substreams and dependent frames */ if(s->frame_type == EAC3_FRAME_TYPE_DEPENDENT || s->substreamid) { av_log(avctx, AV_LOG_ERROR, "unsupported frame type : skipping frame\n"); + *got_frame_ptr = 0; return s->frame_size; } else { av_log(avctx, AV_LOG_ERROR, "invalid frame type\n"); @@ -1406,21 +1407,24 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, if (s->bitstream_mode == 0x7 && s->channels > 1) avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE; + /* get output buffer */ + s->frame.nb_samples = s->num_blocks * 256; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + out_samples_flt = (float *)s->frame.data[0]; + out_samples_s16 = (int16_t *)s->frame.data[0]; + /* decode the audio blocks */ channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on]; for (ch = 0; ch < s->out_channels; ch++) output[ch] = s->output[channel_map[ch]]; - data_size_tmp = s->num_blocks * 256 * avctx->channels; - data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples_s16); - if (data_size_orig < data_size_tmp) - return -1; - *data_size = data_size_tmp; for (blk = 0; blk < s->num_blocks; blk++) { if (!err && decode_audio_block(s, blk)) { av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n"); err = 1; } - if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) { s->fmt_conv.float_interleave(out_samples_flt, output, 256, s->out_channels); @@ -1431,8 +1435,10 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, out_samples_s16 += 256 * s->out_channels; } } - *data_size = s->num_blocks * 256 * avctx->channels * - av_get_bytes_per_sample(avctx->sample_fmt); + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return FFMIN(buf_size, s->frame_size); } @@ -1477,6 +1483,7 @@ AVCodec ff_ac3_decoder = { .init = ac3_decode_init, .close = ac3_decode_end, .decode = ac3_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"), .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE @@ -1499,6 +1506,7 @@ AVCodec ff_eac3_decoder = { .init = ac3_decode_init, .close = ac3_decode_end, .decode = ac3_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"), .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE diff --git a/libavcodec/ac3dec.h b/libavcodec/ac3dec.h index aa346cb020..0f5e86e6b3 100644 --- a/libavcodec/ac3dec.h +++ b/libavcodec/ac3dec.h @@ -68,6 +68,7 @@ typedef struct { AVClass *class; ///< class for AVOptions AVCodecContext *avctx; ///< parent context + AVFrame frame; ///< AVFrame for decoded output GetBitContext gbc; ///< bitstream reader ///@name Bit stream information diff --git a/libavcodec/adpcm.c b/libavcodec/adpcm.c index 1281acc33a..e305220af5 100644 --- a/libavcodec/adpcm.c +++ b/libavcodec/adpcm.c @@ -84,6 +84,7 @@ static const int swf_index_tables[4][16] = { /* end of tables */ typedef struct ADPCMDecodeContext { + AVFrame frame; ADPCMChannelStatus status[6]; } ADPCMDecodeContext; @@ -124,6 +125,10 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx) break; } avctx->sample_fmt = AV_SAMPLE_FMT_S16; + + avcodec_get_frame_defaults(&c->frame); + avctx->coded_frame = &c->frame; + return 0; } @@ -501,9 +506,8 @@ static int get_nb_samples(AVCodecContext *avctx, const uint8_t *buf, decode_top_nibble_next = 1; \ } -static int adpcm_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int adpcm_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; @@ -514,7 +518,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, const uint8_t *src; int st; /* stereo */ int count1, count2; - int nb_samples, coded_samples, out_bps, out_size; + int nb_samples, coded_samples, ret; nb_samples = get_nb_samples(avctx, buf, buf_size, &coded_samples); if (nb_samples <= 0) { @@ -522,22 +526,22 @@ static int adpcm_decode_frame(AVCodecContext *avctx, return AVERROR_INVALIDDATA; } - out_bps = av_get_bytes_per_sample(avctx->sample_fmt); - out_size = nb_samples * avctx->channels * out_bps; - if (*data_size < out_size) { - av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n"); - return AVERROR(EINVAL); + /* get output buffer */ + c->frame.nb_samples = nb_samples; + if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (short *)c->frame.data[0]; + /* use coded_samples when applicable */ /* it is always <= nb_samples, so the output buffer will be large enough */ if (coded_samples) { if (coded_samples != nb_samples) av_log(avctx, AV_LOG_WARNING, "mismatch in coded sample count\n"); - nb_samples = coded_samples; - out_size = nb_samples * avctx->channels * out_bps; + c->frame.nb_samples = nb_samples = coded_samples; } - samples = data; src = buf; st = avctx->channels == 2 ? 1 : 0; @@ -576,7 +580,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, cs->step_index = 88; } - samples = (short*)data + channel; + samples = (short *)c->frame.data[0] + channel; for (m = 0; m < 32; m++) { *samples = adpcm_ima_qt_expand_nibble(cs, src[0] & 0x0F, 3); @@ -628,7 +632,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } for (i = 0; i < avctx->channels; i++) { - samples = (short*)data + i; + samples = (short *)c->frame.data[0] + i; cs = &c->status[i]; for (n = nb_samples >> 1; n > 0; n--, src++) { uint8_t v = *src; @@ -965,7 +969,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } } - out_size = count * 28 * avctx->channels * out_bps; + c->frame.nb_samples = count * 28; src = src_end; break; } @@ -1144,7 +1148,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, prev[0][i] = (int16_t)bytestream_get_be16(&src); for (ch = 0; ch <= st; ch++) { - samples = (unsigned short *) data + ch; + samples = (short *)c->frame.data[0] + ch; /* Read in every sample for this channel. */ for (i = 0; i < nb_samples / 14; i++) { @@ -1177,7 +1181,10 @@ static int adpcm_decode_frame(AVCodecContext *avctx, default: return -1; } - *data_size = out_size; + + *got_frame_ptr = 1; + *(AVFrame *)data = c->frame; + return src - buf; } @@ -1190,6 +1197,7 @@ AVCodec ff_ ## name_ ## _decoder = { \ .priv_data_size = sizeof(ADPCMDecodeContext), \ .init = adpcm_decode_init, \ .decode = adpcm_decode_frame, \ + .capabilities = CODEC_CAP_DR1, \ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ } diff --git a/libavcodec/adx.h b/libavcodec/adx.h index 572483d28f..a14ddce499 100644 --- a/libavcodec/adx.h +++ b/libavcodec/adx.h @@ -40,6 +40,7 @@ typedef struct { } ADXChannelState; typedef struct { + AVFrame frame; int channels; ADXChannelState prev[2]; int header_parsed; diff --git a/libavcodec/adxdec.c b/libavcodec/adxdec.c index 0fed1220ef..f049def626 100644 --- a/libavcodec/adxdec.c +++ b/libavcodec/adxdec.c @@ -50,6 +50,10 @@ static av_cold int adx_decode_init(AVCodecContext *avctx) c->channels = avctx->channels; avctx->sample_fmt = AV_SAMPLE_FMT_S16; + + avcodec_get_frame_defaults(&c->frame); + avctx->coded_frame = &c->frame; + return 0; } @@ -89,36 +93,42 @@ static int adx_decode(ADXContext *c, int16_t *out, const uint8_t *in, int ch) return 0; } -static int adx_decode_frame(AVCodecContext *avctx, void *data, int *data_size, - AVPacket *avpkt) +static int adx_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { int buf_size = avpkt->size; ADXContext *c = avctx->priv_data; - int16_t *samples = data; + int16_t *samples; const uint8_t *buf = avpkt->data; - int num_blocks, ch; + int num_blocks, ch, ret; if (c->eof) { - *data_size = 0; + *got_frame_ptr = 0; return buf_size; } - /* 18 bytes of data are expanded into 32*2 bytes of audio, - so guard against buffer overflows */ + /* calculate number of blocks in the packet */ num_blocks = buf_size / (BLOCK_SIZE * c->channels); - if (num_blocks > *data_size / (BLOCK_SAMPLES * c->channels)) { - buf_size = (*data_size / (BLOCK_SAMPLES * c->channels)) * BLOCK_SIZE; - num_blocks = buf_size / (BLOCK_SIZE * c->channels); - } - if (!buf_size || buf_size % (BLOCK_SIZE * avctx->channels)) { + + /* if the packet is not an even multiple of BLOCK_SIZE, check for an EOF + packet */ + if (!num_blocks || buf_size % (BLOCK_SIZE * avctx->channels)) { if (buf_size >= 4 && (AV_RB16(buf) & 0x8000)) { c->eof = 1; - *data_size = 0; + *got_frame_ptr = 0; return avpkt->size; } return AVERROR_INVALIDDATA; } + /* get output buffer */ + c->frame.nb_samples = num_blocks * BLOCK_SAMPLES; + if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + samples = (int16_t *)c->frame.data[0]; + while (num_blocks--) { for (ch = 0; ch < c->channels; ch++) { if (adx_decode(c, samples + ch, buf, ch)) { @@ -132,7 +142,9 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data, int *data_size, samples += BLOCK_SAMPLES * c->channels; } - *data_size = (uint8_t*)samples - (uint8_t*)data; + *got_frame_ptr = 1; + *(AVFrame *)data = c->frame; + return buf - avpkt->data; } @@ -143,5 +155,6 @@ AVCodec ff_adpcm_adx_decoder = { .priv_data_size = sizeof(ADXContext), .init = adx_decode_init, .decode = adx_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"), }; diff --git a/libavcodec/alac.c b/libavcodec/alac.c index 4e143270a5..2788238c78 100644 --- a/libavcodec/alac.c +++ b/libavcodec/alac.c @@ -62,10 +62,10 @@ typedef struct { AVCodecContext *avctx; + AVFrame frame; GetBitContext gb; int numchannels; - int bytespersample; /* buffers */ int32_t *predicterror_buffer[MAX_CHANNELS]; @@ -351,9 +351,8 @@ static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS], } } -static int alac_decode_frame(AVCodecContext *avctx, - void *outbuffer, int *outputsize, - AVPacket *avpkt) +static int alac_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *inbuffer = avpkt->data; int input_buffer_size = avpkt->size; @@ -366,7 +365,7 @@ static int alac_decode_frame(AVCodecContext *avctx, int isnotcompressed; uint8_t interlacing_shift; uint8_t interlacing_leftweight; - int i, ch; + int i, ch, ret; init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8); @@ -401,14 +400,17 @@ static int alac_decode_frame(AVCodecContext *avctx, } else outputsamples = alac->setinfo_max_samples_per_frame; - alac->bytespersample = channels * av_get_bytes_per_sample(avctx->sample_fmt); - - if(outputsamples > *outputsize / alac->bytespersample){ - av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n"); - return -1; + /* get output buffer */ + if (outputsamples > INT32_MAX) { + av_log(avctx, AV_LOG_ERROR, "unsupported block size: %u\n", outputsamples); + return AVERROR_INVALIDDATA; + } + alac->frame.nb_samples = outputsamples; + if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } - *outputsize = outputsamples * alac->bytespersample; readsamplesize = alac->setinfo_sample_size - alac->extra_bits + channels - 1; if (readsamplesize > MIN_CACHE_BITS) { av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize); @@ -501,21 +503,23 @@ static int alac_decode_frame(AVCodecContext *avctx, switch(alac->setinfo_sample_size) { case 16: if (channels == 2) { - interleave_stereo_16(alac->outputsamples_buffer, outbuffer, - outputsamples); + interleave_stereo_16(alac->outputsamples_buffer, + (int16_t *)alac->frame.data[0], outputsamples); } else { + int16_t *outbuffer = (int16_t *)alac->frame.data[0]; for (i = 0; i < outputsamples; i++) { - ((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i]; + outbuffer[i] = alac->outputsamples_buffer[0][i]; } } break; case 24: if (channels == 2) { - interleave_stereo_24(alac->outputsamples_buffer, outbuffer, - outputsamples); + interleave_stereo_24(alac->outputsamples_buffer, + (int32_t *)alac->frame.data[0], outputsamples); } else { + int32_t *outbuffer = (int32_t *)alac->frame.data[0]; for (i = 0; i < outputsamples; i++) - ((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8; + outbuffer[i] = alac->outputsamples_buffer[0][i] << 8; } break; } @@ -523,6 +527,9 @@ static int alac_decode_frame(AVCodecContext *avctx, if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8) av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb)); + *got_frame_ptr = 1; + *(AVFrame *)data = alac->frame; + return input_buffer_size; } @@ -637,6 +644,9 @@ static av_cold int alac_decode_init(AVCodecContext * avctx) return ret; } + avcodec_get_frame_defaults(&alac->frame); + avctx->coded_frame = &alac->frame; + return 0; } @@ -648,5 +658,6 @@ AVCodec ff_alac_decoder = { .init = alac_decode_init, .close = alac_decode_close, .decode = alac_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), }; diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index c399471c1d..305a5b825b 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -75,20 +75,22 @@ typedef struct AlacEncodeContext { } AlacEncodeContext; -static void init_sample_buffers(AlacEncodeContext *s, const int16_t *input_samples) +static void init_sample_buffers(AlacEncodeContext *s, + const int16_t *input_samples) { int ch, i; - for(ch=0;ch<s->avctx->channels;ch++) { + for (ch = 0; ch < s->avctx->channels; ch++) { const int16_t *sptr = input_samples + ch; - for(i=0;i<s->avctx->frame_size;i++) { + for (i = 0; i < s->avctx->frame_size; i++) { s->sample_buf[ch][i] = *sptr; sptr += s->avctx->channels; } } } -static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size) +static void encode_scalar(AlacEncodeContext *s, int x, + int k, int write_sample_size) { int divisor, q, r; @@ -97,17 +99,17 @@ static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_s q = x / divisor; r = x % divisor; - if(q > 8) { + if (q > 8) { // write escape code and sample value directly put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE); put_bits(&s->pbctx, write_sample_size, x); } else { - if(q) + if (q) put_bits(&s->pbctx, q, (1<<q) - 1); put_bits(&s->pbctx, 1, 0); - if(k != 1) { - if(r > 0) + if (k != 1) { + if (r > 0) put_bits(&s->pbctx, k, r+1); else put_bits(&s->pbctx, k-1, 0); @@ -164,7 +166,7 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) /* calculate sum of 2nd order residual for each channel */ sum[0] = sum[1] = sum[2] = sum[3] = 0; - for(i=2; i<n; i++) { + for (i = 2; i < n; i++) { lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; sum[2] += FFABS((lt + rt) >> 1); @@ -181,8 +183,8 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) /* return mode with lowest score */ best = 0; - for(i=1; i<4; i++) { - if(score[i] < score[best]) { + for (i = 1; i < 4; i++) { + if (score[i] < score[best]) { best = i; } } @@ -205,7 +207,7 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s) break; case ALAC_CHMODE_LEFT_SIDE: - for(i=0; i<n; i++) { + for (i = 0; i < n; i++) { right[i] = left[i] - right[i]; } s->interlacing_leftweight = 1; @@ -213,7 +215,7 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s) break; case ALAC_CHMODE_RIGHT_SIDE: - for(i=0; i<n; i++) { + for (i = 0; i < n; i++) { tmp = right[i]; right[i] = left[i] - right[i]; left[i] = tmp + (right[i] >> 31); @@ -223,7 +225,7 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s) break; default: - for(i=0; i<n; i++) { + for (i = 0; i < n; i++) { tmp = left[i]; left[i] = (tmp + right[i]) >> 1; right[i] = tmp - right[i]; @@ -239,10 +241,10 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) int i; AlacLPCContext lpc = s->lpc[ch]; - if(lpc.lpc_order == 31) { + if (lpc.lpc_order == 31) { s->predictor_buf[0] = s->sample_buf[ch][0]; - for(i=1; i<s->avctx->frame_size; i++) + for (i = 1; i < s->avctx->frame_size; i++) s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1]; return; @@ -250,17 +252,17 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) // generalised linear predictor - if(lpc.lpc_order > 0) { + if (lpc.lpc_order > 0) { int32_t *samples = s->sample_buf[ch]; int32_t *residual = s->predictor_buf; // generate warm-up samples residual[0] = samples[0]; - for(i=1;i<=lpc.lpc_order;i++) + for (i = 1; i <= lpc.lpc_order; i++) residual[i] = samples[i] - samples[i-1]; // perform lpc on remaining samples - for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) { + for (i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) { int sum = 1 << (lpc.lpc_quant - 1), res_val, j; for (j = 0; j < lpc.lpc_order; j++) { @@ -303,7 +305,7 @@ static void alac_entropy_coder(AlacEncodeContext *s) int sign_modifier = 0, i, k; int32_t *samples = s->predictor_buf; - for(i=0;i < s->avctx->frame_size;) { + for (i = 0; i < s->avctx->frame_size;) { int x; k = av_log2((history >> 9) + 3); @@ -320,15 +322,15 @@ static void alac_entropy_coder(AlacEncodeContext *s) - ((history * s->rc.history_mult) >> 9); sign_modifier = 0; - if(x > 0xFFFF) + if (x > 0xFFFF) history = 0xFFFF; - if((history < 128) && (i < s->avctx->frame_size)) { + if (history < 128 && i < s->avctx->frame_size) { unsigned int block_size = 0; k = 7 - av_log2(history) + ((history + 16) >> 6); - while((*samples == 0) && (i < s->avctx->frame_size)) { + while (*samples == 0 && i < s->avctx->frame_size) { samples++; i++; block_size++; @@ -347,12 +349,12 @@ static void write_compressed_frame(AlacEncodeContext *s) { int i, j; - if(s->avctx->channels == 2) + if (s->avctx->channels == 2) alac_stereo_decorrelation(s); put_bits(&s->pbctx, 8, s->interlacing_shift); put_bits(&s->pbctx, 8, s->interlacing_leftweight); - for(i=0;i<s->avctx->channels;i++) { + for (i = 0; i < s->avctx->channels; i++) { calc_predictor_params(s, i); @@ -362,14 +364,14 @@ static void write_compressed_frame(AlacEncodeContext *s) put_bits(&s->pbctx, 3, s->rc.rice_modifier); put_bits(&s->pbctx, 5, s->lpc[i].lpc_order); // predictor coeff. table - for(j=0;j<s->lpc[i].lpc_order;j++) { + for (j = 0; j < s->lpc[i].lpc_order; j++) { put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); } } // apply lpc and entropy coding to audio samples - for(i=0;i<s->avctx->channels;i++) { + for (i = 0; i < s->avctx->channels; i++) { alac_linear_predictor(s, i); alac_entropy_coder(s); } @@ -384,7 +386,7 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) avctx->frame_size = DEFAULT_FRAME_SIZE; avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE; - if(avctx->sample_fmt != AV_SAMPLE_FMT_S16) { + if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) { av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n"); return -1; } @@ -395,7 +397,7 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) } // Set default compression level - if(avctx->compression_level == FF_COMPRESSION_DEFAULT) + if (avctx->compression_level == FF_COMPRESSION_DEFAULT) s->compression_level = 2; else s->compression_level = av_clip(avctx->compression_level, 0, 2); @@ -416,21 +418,23 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample); AV_WB8 (alac_extradata+21, avctx->channels); AV_WB32(alac_extradata+24, s->max_coded_frame_size); - AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate + AV_WB32(alac_extradata+28, + avctx->sample_rate * avctx->channels * avctx->bits_per_coded_sample); // average bitrate AV_WB32(alac_extradata+32, avctx->sample_rate); // Set relevant extradata fields - if(s->compression_level > 0) { + if (s->compression_level > 0) { AV_WB8(alac_extradata+18, s->rc.history_mult); AV_WB8(alac_extradata+19, s->rc.initial_history); AV_WB8(alac_extradata+20, s->rc.k_modifier); } s->min_prediction_order = DEFAULT_MIN_PRED_ORDER; - if(avctx->min_prediction_order >= 0) { - if(avctx->min_prediction_order < MIN_LPC_ORDER || + if (avctx->min_prediction_order >= 0) { + if (avctx->min_prediction_order < MIN_LPC_ORDER || avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) { - av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order); + av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", + avctx->min_prediction_order); return -1; } @@ -438,18 +442,20 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) } s->max_prediction_order = DEFAULT_MAX_PRED_ORDER; - if(avctx->max_prediction_order >= 0) { - if(avctx->max_prediction_order < MIN_LPC_ORDER || - avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) { - av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order); + if (avctx->max_prediction_order >= 0) { + if (avctx->max_prediction_order < MIN_LPC_ORDER || + avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) { + av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", + avctx->max_prediction_order); return -1; } s->max_prediction_order = avctx->max_prediction_order; } - if(s->max_prediction_order < s->min_prediction_order) { - av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n", + if (s->max_prediction_order < s->min_prediction_order) { + av_log(avctx, AV_LOG_ERROR, + "invalid prediction orders: min=%d max=%d\n", s->min_prediction_order, s->max_prediction_order); return -1; } @@ -474,12 +480,12 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, PutBitContext *pb = &s->pbctx; int i, out_bytes, verbatim_flag = 0; - if(avctx->frame_size > DEFAULT_FRAME_SIZE) { + if (avctx->frame_size > DEFAULT_FRAME_SIZE) { av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n"); return -1; } - if(buf_size < 2*s->max_coded_frame_size) { + if (buf_size < 2 * s->max_coded_frame_size) { av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); return -1; } @@ -487,11 +493,11 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, verbatim: init_put_bits(pb, frame, buf_size); - if((s->compression_level == 0) || verbatim_flag) { + if (s->compression_level == 0 || verbatim_flag) { // Verbatim mode const int16_t *samples = data; write_frame_header(s, 1); - for(i=0; i<avctx->frame_size*avctx->channels; i++) { + for (i = 0; i < avctx->frame_size * avctx->channels; i++) { put_sbits(pb, 16, *samples++); } } else { @@ -504,9 +510,9 @@ verbatim: flush_put_bits(pb); out_bytes = put_bits_count(pb) >> 3; - if(out_bytes > s->max_coded_frame_size) { + if (out_bytes > s->max_coded_frame_size) { /* frame too large. use verbatim mode */ - if(verbatim_flag || (s->compression_level == 0)) { + if (verbatim_flag || s->compression_level == 0) { /* still too large. must be an error. */ av_log(avctx, AV_LOG_ERROR, "error encoding frame\n"); return -1; @@ -537,6 +543,7 @@ AVCodec ff_alac_encoder = { .encode = alac_encode_frame, .close = alac_encode_close, .capabilities = CODEC_CAP_SMALL_LAST_FRAME, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE}, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), }; diff --git a/libavcodec/alsdec.c b/libavcodec/alsdec.c index fc98970bd7..0e5509c74a 100644 --- a/libavcodec/alsdec.c +++ b/libavcodec/alsdec.c @@ -191,6 +191,7 @@ typedef struct { typedef struct { AVCodecContext *avctx; + AVFrame frame; ALSSpecificConfig sconf; GetBitContext gb; DSPContext dsp; @@ -290,7 +291,7 @@ static av_cold int read_specific_config(ALSDecContext *ctx) init_get_bits(&gb, avctx->extradata, avctx->extradata_size * 8); config_offset = avpriv_mpeg4audio_get_config(&m4ac, avctx->extradata, - avctx->extradata_size); + avctx->extradata_size * 8, 1); if (config_offset < 0) return -1; @@ -1415,15 +1416,14 @@ static int read_frame_data(ALSDecContext *ctx, unsigned int ra_frame) /** Decode an ALS frame. */ -static int decode_frame(AVCodecContext *avctx, - void *data, int *data_size, +static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { ALSDecContext *ctx = avctx->priv_data; ALSSpecificConfig *sconf = &ctx->sconf; const uint8_t *buffer = avpkt->data; int buffer_size = avpkt->size; - int invalid_frame, size; + int invalid_frame, ret; unsigned int c, sample, ra_frame, bytes_read, shift; init_get_bits(&ctx->gb, buffer, buffer_size * 8); @@ -1448,21 +1448,17 @@ static int decode_frame(AVCodecContext *avctx, ctx->frame_id++; - // check for size of decoded data - size = ctx->cur_frame_length * avctx->channels * - av_get_bytes_per_sample(avctx->sample_fmt); - - if (size > *data_size) { - av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n"); - return -1; + /* get output buffer */ + ctx->frame.nb_samples = ctx->cur_frame_length; + if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } - *data_size = size; - // transform decoded frame into output format #define INTERLEAVE_OUTPUT(bps) \ { \ - int##bps##_t *dest = (int##bps##_t*) data; \ + int##bps##_t *dest = (int##bps##_t*)ctx->frame.data[0]; \ shift = bps - ctx->avctx->bits_per_raw_sample; \ for (sample = 0; sample < ctx->cur_frame_length; sample++) \ for (c = 0; c < avctx->channels; c++) \ @@ -1480,7 +1476,7 @@ static int decode_frame(AVCodecContext *avctx, int swap = HAVE_BIGENDIAN != sconf->msb_first; if (ctx->avctx->bits_per_raw_sample == 24) { - int32_t *src = data; + int32_t *src = (int32_t *)ctx->frame.data[0]; for (sample = 0; sample < ctx->cur_frame_length * avctx->channels; @@ -1501,22 +1497,25 @@ static int decode_frame(AVCodecContext *avctx, if (swap) { if (ctx->avctx->bits_per_raw_sample <= 16) { - int16_t *src = (int16_t*) data; + int16_t *src = (int16_t*) ctx->frame.data[0]; int16_t *dest = (int16_t*) ctx->crc_buffer; for (sample = 0; sample < ctx->cur_frame_length * avctx->channels; sample++) *dest++ = av_bswap16(src[sample]); } else { - ctx->dsp.bswap_buf((uint32_t*)ctx->crc_buffer, data, + ctx->dsp.bswap_buf((uint32_t*)ctx->crc_buffer, + (uint32_t *)ctx->frame.data[0], ctx->cur_frame_length * avctx->channels); } crc_source = ctx->crc_buffer; } else { - crc_source = data; + crc_source = ctx->frame.data[0]; } - ctx->crc = av_crc(ctx->crc_table, ctx->crc, crc_source, size); + ctx->crc = av_crc(ctx->crc_table, ctx->crc, crc_source, + ctx->cur_frame_length * avctx->channels * + av_get_bytes_per_sample(avctx->sample_fmt)); } @@ -1527,6 +1526,9 @@ static int decode_frame(AVCodecContext *avctx, } } + *got_frame_ptr = 1; + *(AVFrame *)data = ctx->frame; + bytes_read = invalid_frame ? buffer_size : (get_bits_count(&ctx->gb) + 7) >> 3; @@ -1724,6 +1726,9 @@ static av_cold int decode_init(AVCodecContext *avctx) dsputil_init(&ctx->dsp, avctx); + avcodec_get_frame_defaults(&ctx->frame); + avctx->coded_frame = &ctx->frame; + return 0; } @@ -1747,7 +1752,7 @@ AVCodec ff_als_decoder = { .close = decode_end, .decode = decode_frame, .flush = flush, - .capabilities = CODEC_CAP_SUBFRAMES, + .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("MPEG-4 Audio Lossless Coding (ALS)"), }; diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c index b8d826e139..0a4a7e6dda 100644 --- a/libavcodec/amrnbdec.c +++ b/libavcodec/amrnbdec.c @@ -95,6 +95,7 @@ #define AMR_AGC_ALPHA 0.9 typedef struct AMRContext { + AVFrame avframe; ///< AVFrame for decoded samples AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc) uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0 enum Mode cur_frame_mode; @@ -167,6 +168,9 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx) for (i = 0; i < 4; i++) p->prediction_error[i] = MIN_ENERGY; + avcodec_get_frame_defaults(&p->avframe); + avctx->coded_frame = &p->avframe; + return 0; } @@ -919,21 +923,29 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out) /// @} -static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, - AVPacket *avpkt) +static int amrnb_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { AMRContext *p = avctx->priv_data; // pointer to private data const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; - float *buf_out = data; // pointer to the output data buffer - int i, subframe; + float *buf_out; // pointer to the output data buffer + int i, subframe, ret; float fixed_gain_factor; AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing float synth_fixed_gain; // the fixed gain that synthesis should use const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use + /* get output buffer */ + p->avframe.nb_samples = AMR_BLOCK_SIZE; + if ((ret = avctx->get_buffer(avctx, &p->avframe)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + buf_out = (float *)p->avframe.data[0]; + p->cur_frame_mode = unpack_bitstream(p, buf, buf_size); if (p->cur_frame_mode == MODE_DTX) { av_log_missing_feature(avctx, "dtx mode", 0); @@ -1029,8 +1041,8 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3], 0.84, 0.16, LP_FILTER_ORDER); - /* report how many samples we got */ - *data_size = AMR_BLOCK_SIZE * sizeof(float); + *got_frame_ptr = 1; + *(AVFrame *)data = p->avframe; /* return the amount of bytes consumed if everything was OK */ return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC @@ -1044,6 +1056,7 @@ AVCodec ff_amrnb_decoder = { .priv_data_size = sizeof(AMRContext), .init = amrnb_decode_init, .decode = amrnb_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"), .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE}, }; diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c index fa3f8dd050..b0e54f9261 100644 --- a/libavcodec/amrwbdec.c +++ b/libavcodec/amrwbdec.c @@ -41,6 +41,7 @@ #include "amrwbdata.h" typedef struct { + AVFrame avframe; ///< AVFrame for decoded samples AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream enum Mode fr_cur_mode; ///< mode index of current frame uint8_t fr_quality; ///< frame quality index (FQI) @@ -102,6 +103,9 @@ static av_cold int amrwb_decode_init(AVCodecContext *avctx) for (i = 0; i < 4; i++) ctx->prediction_error[i] = MIN_ENERGY; + avcodec_get_frame_defaults(&ctx->avframe); + avctx->coded_frame = &ctx->avframe; + return 0; } @@ -1062,15 +1066,15 @@ static void update_sub_state(AMRWBContext *ctx) LP_ORDER_16k * sizeof(float)); } -static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, - AVPacket *avpkt) +static int amrwb_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { AMRWBContext *ctx = avctx->priv_data; AMRWBFrame *cf = &ctx->frame; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; int expected_fr_size, header_size; - float *buf_out = data; + float *buf_out; float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing float fixed_gain_factor; // fixed gain correction factor (gamma) float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use @@ -1080,7 +1084,15 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis float hb_gain; - int sub, i; + int sub, i, ret; + + /* get output buffer */ + ctx->avframe.nb_samples = 4 * AMRWB_SFR_SIZE_16k; + if ((ret = avctx->get_buffer(avctx, &ctx->avframe)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + buf_out = (float *)ctx->avframe.data[0]; header_size = decode_mime_header(ctx, buf); expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1; @@ -1088,7 +1100,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, if (buf_size < expected_fr_size) { av_log(avctx, AV_LOG_ERROR, "Frame too small (%d bytes). Truncated file?\n", buf_size); - *data_size = 0; + *got_frame_ptr = 0; return buf_size; } @@ -1219,8 +1231,8 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0])); memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float)); - /* report how many samples we got */ - *data_size = 4 * AMRWB_SFR_SIZE_16k * sizeof(float); + *got_frame_ptr = 1; + *(AVFrame *)data = ctx->avframe; return expected_fr_size; } @@ -1232,6 +1244,7 @@ AVCodec ff_amrwb_decoder = { .priv_data_size = sizeof(AMRWBContext), .init = amrwb_decode_init, .decode = amrwb_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate WideBand"), .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE}, }; diff --git a/libavcodec/apedec.c b/libavcodec/apedec.c index 10f75212ff..df7529835c 100644 --- a/libavcodec/apedec.c +++ b/libavcodec/apedec.c @@ -129,6 +129,7 @@ typedef struct APEPredictor { /** Decoder context */ typedef struct APEContext { AVCodecContext *avctx; + AVFrame frame; DSPContext dsp; int channels; int samples; ///< samples left to decode in current frame @@ -215,6 +216,10 @@ static av_cold int ape_decode_init(AVCodecContext *avctx) 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; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; filter_alloc_fail: ape_decode_close(avctx); @@ -805,16 +810,15 @@ static void ape_unpack_stereo(APEContext *ctx, int count) } } -static int ape_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int ape_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; APEContext *s = avctx->priv_data; - int16_t *samples = data; - int i; - int blockstodecode, out_size; + int16_t *samples; + int i, ret; + int blockstodecode; int bytes_used = 0; /* this should never be negative, but bad things will happen if it is, so @@ -826,7 +830,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *tmp_data; if (!buf_size) { - *data_size = 0; + *got_frame_ptr = 0; return 0; } if (buf_size < 8) { @@ -874,18 +878,19 @@ static int ape_decode_frame(AVCodecContext *avctx, } if (!s->data) { - *data_size = 0; + *got_frame_ptr = 0; return buf_size; } blockstodecode = FFMIN(BLOCKS_PER_LOOP, s->samples); - out_size = blockstodecode * avctx->channels * - 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); + /* get output buffer */ + s->frame.nb_samples = blockstodecode; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (int16_t *)s->frame.data[0]; s->error=0; @@ -909,7 +914,9 @@ static int ape_decode_frame(AVCodecContext *avctx, s->samples -= blockstodecode; - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return bytes_used; } @@ -927,7 +934,7 @@ AVCodec ff_ape_decoder = { .init = ape_decode_init, .close = ape_decode_close, .decode = ape_decode_frame, - .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY, + .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1, .flush = ape_flush, .long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"), }; diff --git a/libavcodec/arm/h264dsp_neon.S b/libavcodec/arm/h264dsp_neon.S index adc21f9500..9f4da2cb7b 100644 --- a/libavcodec/arm/h264dsp_neon.S +++ b/libavcodec/arm/h264dsp_neon.S @@ -19,55 +19,16 @@ */ #include "asm.S" - - .macro transpose_8x8 r0 r1 r2 r3 r4 r5 r6 r7 - vtrn.32 \r0, \r4 - vtrn.32 \r1, \r5 - vtrn.32 \r2, \r6 - vtrn.32 \r3, \r7 - vtrn.16 \r0, \r2 - vtrn.16 \r1, \r3 - vtrn.16 \r4, \r6 - vtrn.16 \r5, \r7 - vtrn.8 \r0, \r1 - vtrn.8 \r2, \r3 - vtrn.8 \r4, \r5 - vtrn.8 \r6, \r7 - .endm - - .macro transpose_4x4 r0 r1 r2 r3 - vtrn.16 \r0, \r2 - vtrn.16 \r1, \r3 - vtrn.8 \r0, \r1 - vtrn.8 \r2, \r3 - .endm - - .macro swap4 r0 r1 r2 r3 r4 r5 r6 r7 - vswp \r0, \r4 - vswp \r1, \r5 - vswp \r2, \r6 - vswp \r3, \r7 - .endm - - .macro transpose16_4x4 r0 r1 r2 r3 r4 r5 r6 r7 - vtrn.32 \r0, \r2 - vtrn.32 \r1, \r3 - vtrn.32 \r4, \r6 - vtrn.32 \r5, \r7 - vtrn.16 \r0, \r1 - vtrn.16 \r2, \r3 - vtrn.16 \r4, \r5 - vtrn.16 \r6, \r7 - .endm +#include "neon.S" /* chroma_mc8(uint8_t *dst, uint8_t *src, int stride, int h, int x, int y) */ - .macro h264_chroma_mc8 type +.macro h264_chroma_mc8 type function ff_\type\()_h264_chroma_mc8_neon, export=1 push {r4-r7, lr} ldrd r4, [sp, #20] -.ifc \type,avg + .ifc \type,avg mov lr, r0 -.endif + .endif pld [r1] pld [r1, r2] @@ -75,7 +36,7 @@ A muls r7, r4, r5 T mul r7, r4, r5 T cmp r7, #0 rsb r6, r7, r5, lsl #3 - rsb ip, r7, r4, lsl #3 + rsb r12, r7, r4, lsl #3 sub r4, r7, r4, lsl #3 sub r4, r4, r5, lsl #3 add r4, r4, #64 @@ -86,10 +47,10 @@ T cmp r7, #0 vdup.8 d0, r4 lsl r4, r2, #1 - vdup.8 d1, ip - vld1.64 {d4, d5}, [r1], r4 + vdup.8 d1, r12 + vld1.8 {d4, d5}, [r1], r4 vdup.8 d2, r6 - vld1.64 {d6, d7}, [r5], r4 + vld1.8 {d6, d7}, [r5], r4 vdup.8 d3, r7 vext.8 d5, d4, d5, #1 @@ -98,7 +59,7 @@ T cmp r7, #0 1: pld [r5] vmull.u8 q8, d4, d0 vmlal.u8 q8, d5, d1 - vld1.64 {d4, d5}, [r1], r4 + vld1.8 {d4, d5}, [r1], r4 vmlal.u8 q8, d6, d2 vext.8 d5, d4, d5, #1 vmlal.u8 q8, d7, d3 @@ -108,57 +69,57 @@ T cmp r7, #0 vmlal.u8 q9, d4, d2 vmlal.u8 q9, d5, d3 vrshrn.u16 d16, q8, #6 - vld1.64 {d6, d7}, [r5], r4 + vld1.8 {d6, d7}, [r5], r4 pld [r1] vrshrn.u16 d17, q9, #6 -.ifc \type,avg - vld1.64 {d20}, [lr,:64], r2 - vld1.64 {d21}, [lr,:64], r2 + .ifc \type,avg + vld1.8 {d20}, [lr,:64], r2 + vld1.8 {d21}, [lr,:64], r2 vrhadd.u8 q8, q8, q10 -.endif + .endif vext.8 d7, d6, d7, #1 - vst1.64 {d16}, [r0,:64], r2 - vst1.64 {d17}, [r0,:64], r2 + vst1.8 {d16}, [r0,:64], r2 + vst1.8 {d17}, [r0,:64], r2 bgt 1b pop {r4-r7, pc} 2: tst r6, r6 - add ip, ip, r6 + add r12, r12, r6 vdup.8 d0, r4 - vdup.8 d1, ip + vdup.8 d1, r12 beq 4f add r5, r1, r2 lsl r4, r2, #1 - vld1.64 {d4}, [r1], r4 - vld1.64 {d6}, [r5], r4 + vld1.8 {d4}, [r1], r4 + vld1.8 {d6}, [r5], r4 3: pld [r5] vmull.u8 q8, d4, d0 vmlal.u8 q8, d6, d1 - vld1.64 {d4}, [r1], r4 + vld1.8 {d4}, [r1], r4 vmull.u8 q9, d6, d0 vmlal.u8 q9, d4, d1 - vld1.64 {d6}, [r5], r4 + vld1.8 {d6}, [r5], r4 vrshrn.u16 d16, q8, #6 vrshrn.u16 d17, q9, #6 -.ifc \type,avg - vld1.64 {d20}, [lr,:64], r2 - vld1.64 {d21}, [lr,:64], r2 + .ifc \type,avg + vld1.8 {d20}, [lr,:64], r2 + vld1.8 {d21}, [lr,:64], r2 vrhadd.u8 q8, q8, q10 -.endif + .endif subs r3, r3, #2 pld [r1] - vst1.64 {d16}, [r0,:64], r2 - vst1.64 {d17}, [r0,:64], r2 + vst1.8 {d16}, [r0,:64], r2 + vst1.8 {d17}, [r0,:64], r2 bgt 3b pop {r4-r7, pc} -4: vld1.64 {d4, d5}, [r1], r2 - vld1.64 {d6, d7}, [r1], r2 +4: vld1.8 {d4, d5}, [r1], r2 + vld1.8 {d6, d7}, [r1], r2 vext.8 d5, d4, d5, #1 vext.8 d7, d6, d7, #1 @@ -166,36 +127,36 @@ T cmp r7, #0 subs r3, r3, #2 vmull.u8 q8, d4, d0 vmlal.u8 q8, d5, d1 - vld1.64 {d4, d5}, [r1], r2 + vld1.8 {d4, d5}, [r1], r2 vmull.u8 q9, d6, d0 vmlal.u8 q9, d7, d1 pld [r1] vext.8 d5, d4, d5, #1 vrshrn.u16 d16, q8, #6 vrshrn.u16 d17, q9, #6 -.ifc \type,avg - vld1.64 {d20}, [lr,:64], r2 - vld1.64 {d21}, [lr,:64], r2 + .ifc \type,avg + vld1.8 {d20}, [lr,:64], r2 + vld1.8 {d21}, [lr,:64], r2 vrhadd.u8 q8, q8, q10 -.endif - vld1.64 {d6, d7}, [r1], r2 + .endif + vld1.8 {d6, d7}, [r1], r2 vext.8 d7, d6, d7, #1 - vst1.64 {d16}, [r0,:64], r2 - vst1.64 {d17}, [r0,:64], r2 + vst1.8 {d16}, [r0,:64], r2 + vst1.8 {d17}, [r0,:64], r2 bgt 5b pop {r4-r7, pc} endfunc - .endm +.endm /* chroma_mc4(uint8_t *dst, uint8_t *src, int stride, int h, int x, int y) */ - .macro h264_chroma_mc4 type +.macro h264_chroma_mc4 type function ff_\type\()_h264_chroma_mc4_neon, export=1 push {r4-r7, lr} ldrd r4, [sp, #20] -.ifc \type,avg + .ifc \type,avg mov lr, r0 -.endif + .endif pld [r1] pld [r1, r2] @@ -203,7 +164,7 @@ A muls r7, r4, r5 T mul r7, r4, r5 T cmp r7, #0 rsb r6, r7, r5, lsl #3 - rsb ip, r7, r4, lsl #3 + rsb r12, r7, r4, lsl #3 sub r4, r7, r4, lsl #3 sub r4, r4, r5, lsl #3 add r4, r4, #64 @@ -214,10 +175,10 @@ T cmp r7, #0 vdup.8 d0, r4 lsl r4, r2, #1 - vdup.8 d1, ip - vld1.64 {d4}, [r1], r4 + vdup.8 d1, r12 + vld1.8 {d4}, [r1], r4 vdup.8 d2, r6 - vld1.64 {d6}, [r5], r4 + vld1.8 {d6}, [r5], r4 vdup.8 d3, r7 vext.8 d5, d4, d5, #1 @@ -231,22 +192,22 @@ T cmp r7, #0 1: pld [r5] vmull.u8 q8, d4, d0 vmlal.u8 q8, d6, d2 - vld1.64 {d4}, [r1], r4 + vld1.8 {d4}, [r1], r4 vext.8 d5, d4, d5, #1 vtrn.32 d4, d5 vmull.u8 q9, d6, d0 vmlal.u8 q9, d4, d2 - vld1.64 {d6}, [r5], r4 + vld1.8 {d6}, [r5], r4 vadd.i16 d16, d16, d17 vadd.i16 d17, d18, d19 vrshrn.u16 d16, q8, #6 subs r3, r3, #2 pld [r1] -.ifc \type,avg + .ifc \type,avg vld1.32 {d20[0]}, [lr,:32], r2 vld1.32 {d20[1]}, [lr,:32], r2 vrhadd.u8 d16, d16, d20 -.endif + .endif vext.8 d7, d6, d7, #1 vtrn.32 d6, d7 vst1.32 {d16[0]}, [r0,:32], r2 @@ -256,9 +217,9 @@ T cmp r7, #0 pop {r4-r7, pc} 2: tst r6, r6 - add ip, ip, r6 + add r12, r12, r6 vdup.8 d0, r4 - vdup.8 d1, ip + vdup.8 d1, r12 vtrn.32 d0, d1 beq 4f @@ -277,11 +238,11 @@ T cmp r7, #0 vadd.i16 d16, d16, d17 vadd.i16 d17, d18, d19 vrshrn.u16 d16, q8, #6 -.ifc \type,avg + .ifc \type,avg vld1.32 {d20[0]}, [lr,:32], r2 vld1.32 {d20[1]}, [lr,:32], r2 vrhadd.u8 d16, d16, d20 -.endif + .endif subs r3, r3, #2 pld [r1] vst1.32 {d16[0]}, [r0,:32], r2 @@ -290,8 +251,8 @@ T cmp r7, #0 pop {r4-r7, pc} -4: vld1.64 {d4}, [r1], r2 - vld1.64 {d6}, [r1], r2 +4: vld1.8 {d4}, [r1], r2 + vld1.8 {d6}, [r1], r2 vext.8 d5, d4, d5, #1 vext.8 d7, d6, d7, #1 vtrn.32 d4, d5 @@ -300,19 +261,19 @@ T cmp r7, #0 5: vmull.u8 q8, d4, d0 vmull.u8 q9, d6, d0 subs r3, r3, #2 - vld1.64 {d4}, [r1], r2 + vld1.8 {d4}, [r1], r2 vext.8 d5, d4, d5, #1 vtrn.32 d4, d5 vadd.i16 d16, d16, d17 vadd.i16 d17, d18, d19 pld [r1] vrshrn.u16 d16, q8, #6 -.ifc \type,avg + .ifc \type,avg vld1.32 {d20[0]}, [lr,:32], r2 vld1.32 {d20[1]}, [lr,:32], r2 vrhadd.u8 d16, d16, d20 -.endif - vld1.64 {d6}, [r1], r2 + .endif + vld1.8 {d6}, [r1], r2 vext.8 d7, d6, d7, #1 vtrn.32 d6, d7 pld [r1] @@ -322,9 +283,9 @@ T cmp r7, #0 pop {r4-r7, pc} endfunc - .endm +.endm - .macro h264_chroma_mc2 type +.macro h264_chroma_mc2 type function ff_\type\()_h264_chroma_mc2_neon, export=1 push {r4-r6, lr} ldr r4, [sp, #16] @@ -354,29 +315,29 @@ function ff_\type\()_h264_chroma_mc2_neon, export=1 vtrn.16 q2, q3 vmull.u8 q8, d4, d0 vmlal.u8 q8, d5, d1 -.ifc \type,avg + .ifc \type,avg vld1.16 {d18[0]}, [r0,:16], r2 vld1.16 {d18[1]}, [r0,:16] sub r0, r0, r2 -.endif + .endif vtrn.32 d16, d17 vadd.i16 d16, d16, d17 vrshrn.u16 d16, q8, #6 -.ifc \type,avg + .ifc \type,avg vrhadd.u8 d16, d16, d18 -.endif + .endif vst1.16 {d16[0]}, [r0,:16], r2 vst1.16 {d16[1]}, [r0,:16], r2 subs r3, r3, #2 bgt 1b pop {r4-r6, pc} 2: -.ifc \type,put + .ifc \type,put ldrh_post r5, r1, r2 strh_post r5, r0, r2 ldrh_post r6, r1, r2 strh_post r6, r0, r2 -.else + .else vld1.16 {d16[0]}, [r1], r2 vld1.16 {d16[1]}, [r1], r2 vld1.16 {d18[0]}, [r0,:16], r2 @@ -385,7 +346,7 @@ function ff_\type\()_h264_chroma_mc2_neon, export=1 vrhadd.u8 d16, d16, d18 vst1.16 {d16[0]}, [r0,:16], r2 vst1.16 {d16[1]}, [r0,:16], r2 -.endif + .endif subs r3, r3, #2 bgt 2b pop {r4-r6, pc} @@ -401,22 +362,22 @@ endfunc /* H.264 loop filter */ - .macro h264_loop_filter_start - ldr ip, [sp] +.macro h264_loop_filter_start + ldr r12, [sp] tst r2, r2 - ldr ip, [ip] + ldr r12, [r12] it ne tstne r3, r3 - vmov.32 d24[0], ip - and ip, ip, ip, lsl #16 + vmov.32 d24[0], r12 + and r12, r12, r12, lsl #16 it eq bxeq lr - ands ip, ip, ip, lsl #8 + ands r12, r12, r12, lsl #8 it lt bxlt lr - .endm +.endm - .macro h264_loop_filter_luma +.macro h264_loop_filter_luma vdup.8 q11, r2 @ alpha vmovl.u8 q12, d24 vabd.u8 q6, q8, q0 @ abs(p0 - q0) @@ -482,29 +443,29 @@ endfunc vqmovun.s16 d17, q6 vqmovun.s16 d0, q11 vqmovun.s16 d1, q12 - .endm +.endm function ff_h264_v_loop_filter_luma_neon, export=1 h264_loop_filter_start - vld1.64 {d0, d1}, [r0,:128], r1 - vld1.64 {d2, d3}, [r0,:128], r1 - vld1.64 {d4, d5}, [r0,:128], r1 + vld1.8 {d0, d1}, [r0,:128], r1 + vld1.8 {d2, d3}, [r0,:128], r1 + vld1.8 {d4, d5}, [r0,:128], r1 sub r0, r0, r1, lsl #2 sub r0, r0, r1, lsl #1 - vld1.64 {d20,d21}, [r0,:128], r1 - vld1.64 {d18,d19}, [r0,:128], r1 - vld1.64 {d16,d17}, [r0,:128], r1 + vld1.8 {d20,d21}, [r0,:128], r1 + vld1.8 {d18,d19}, [r0,:128], r1 + vld1.8 {d16,d17}, [r0,:128], r1 vpush {d8-d15} h264_loop_filter_luma sub r0, r0, r1, lsl #1 - vst1.64 {d8, d9}, [r0,:128], r1 - vst1.64 {d16,d17}, [r0,:128], r1 - vst1.64 {d0, d1}, [r0,:128], r1 - vst1.64 {d10,d11}, [r0,:128] + vst1.8 {d8, d9}, [r0,:128], r1 + vst1.8 {d16,d17}, [r0,:128], r1 + vst1.8 {d0, d1}, [r0,:128], r1 + vst1.8 {d10,d11}, [r0,:128] vpop {d8-d15} bx lr @@ -514,22 +475,22 @@ function ff_h264_h_loop_filter_luma_neon, export=1 h264_loop_filter_start sub r0, r0, #4 - vld1.64 {d6}, [r0], r1 - vld1.64 {d20}, [r0], r1 - vld1.64 {d18}, [r0], r1 - vld1.64 {d16}, [r0], r1 - vld1.64 {d0}, [r0], r1 - vld1.64 {d2}, [r0], r1 - vld1.64 {d4}, [r0], r1 - vld1.64 {d26}, [r0], r1 - vld1.64 {d7}, [r0], r1 - vld1.64 {d21}, [r0], r1 - vld1.64 {d19}, [r0], r1 - vld1.64 {d17}, [r0], r1 - vld1.64 {d1}, [r0], r1 - vld1.64 {d3}, [r0], r1 - vld1.64 {d5}, [r0], r1 - vld1.64 {d27}, [r0], r1 + vld1.8 {d6}, [r0], r1 + vld1.8 {d20}, [r0], r1 + vld1.8 {d18}, [r0], r1 + vld1.8 {d16}, [r0], r1 + vld1.8 {d0}, [r0], r1 + vld1.8 {d2}, [r0], r1 + vld1.8 {d4}, [r0], r1 + vld1.8 {d26}, [r0], r1 + vld1.8 {d7}, [r0], r1 + vld1.8 {d21}, [r0], r1 + vld1.8 {d19}, [r0], r1 + vld1.8 {d17}, [r0], r1 + vld1.8 {d1}, [r0], r1 + vld1.8 {d3}, [r0], r1 + vld1.8 {d5}, [r0], r1 + vld1.8 {d27}, [r0], r1 transpose_8x8 q3, q10, q9, q8, q0, q1, q2, q13 @@ -562,7 +523,7 @@ function ff_h264_h_loop_filter_luma_neon, export=1 bx lr endfunc - .macro h264_loop_filter_chroma +.macro h264_loop_filter_chroma vdup.8 d22, r2 @ alpha vmovl.u8 q12, d24 vabd.u8 d26, d16, d0 @ abs(p0 - q0) @@ -591,22 +552,22 @@ endfunc vsubw.s8 q11, q11, d4 vqmovun.s16 d16, q14 vqmovun.s16 d0, q11 - .endm +.endm function ff_h264_v_loop_filter_chroma_neon, export=1 h264_loop_filter_start sub r0, r0, r1, lsl #1 - vld1.64 {d18}, [r0,:64], r1 - vld1.64 {d16}, [r0,:64], r1 - vld1.64 {d0}, [r0,:64], r1 - vld1.64 {d2}, [r0,:64] + vld1.8 {d18}, [r0,:64], r1 + vld1.8 {d16}, [r0,:64], r1 + vld1.8 {d0}, [r0,:64], r1 + vld1.8 {d2}, [r0,:64] h264_loop_filter_chroma sub r0, r0, r1, lsl #1 - vst1.64 {d16}, [r0,:64], r1 - vst1.64 {d0}, [r0,:64], r1 + vst1.8 {d16}, [r0,:64], r1 + vst1.8 {d0}, [r0,:64], r1 bx lr endfunc @@ -651,20 +612,20 @@ endfunc /* H.264 qpel MC */ - .macro lowpass_const r +.macro lowpass_const r movw \r, #5 movt \r, #20 vmov.32 d6[0], \r - .endm +.endm - .macro lowpass_8 r0, r1, r2, r3, d0, d1, narrow=1 -.if \narrow +.macro lowpass_8 r0, r1, r2, r3, d0, d1, narrow=1 + .if \narrow t0 .req q0 t1 .req q8 -.else + .else t0 .req \d0 t1 .req \d1 -.endif + .endif vext.8 d2, \r0, \r1, #2 vext.8 d3, \r0, \r1, #3 vaddl.u8 q1, d2, d3 @@ -685,20 +646,20 @@ endfunc vaddl.u8 t1, \r2, d31 vmla.i16 t1, q9, d6[1] vmls.i16 t1, q10, d6[0] -.if \narrow + .if \narrow vqrshrun.s16 \d0, t0, #5 vqrshrun.s16 \d1, t1, #5 -.endif + .endif .unreq t0 .unreq t1 - .endm +.endm - .macro lowpass_8_1 r0, r1, d0, narrow=1 -.if \narrow +.macro lowpass_8_1 r0, r1, d0, narrow=1 + .if \narrow t0 .req q0 -.else + .else t0 .req \d0 -.endif + .endif vext.8 d2, \r0, \r1, #2 vext.8 d3, \r0, \r1, #3 vaddl.u8 q1, d2, d3 @@ -709,13 +670,13 @@ endfunc vaddl.u8 t0, \r0, d30 vmla.i16 t0, q1, d6[1] vmls.i16 t0, q2, d6[0] -.if \narrow + .if \narrow vqrshrun.s16 \d0, t0, #5 -.endif + .endif .unreq t0 - .endm +.endm - .macro lowpass_8.16 r0, r1, l0, h0, l1, h1, d +.macro lowpass_8.16 r0, r1, l0, h0, l1, h1, d vext.16 q1, \r0, \r1, #2 vext.16 q0, \r0, \r1, #3 vaddl.s16 q9, d2, d0 @@ -750,59 +711,59 @@ endfunc vrshrn.s32 d19, q1, #10 vqmovun.s16 \d, q9 - .endm +.endm function put_h264_qpel16_h_lowpass_neon_packed mov r4, lr - mov ip, #16 + mov r12, #16 mov r3, #8 bl put_h264_qpel8_h_lowpass_neon sub r1, r1, r2, lsl #4 add r1, r1, #8 - mov ip, #16 + mov r12, #16 mov lr, r4 b put_h264_qpel8_h_lowpass_neon endfunc - .macro h264_qpel_h_lowpass type +.macro h264_qpel_h_lowpass type function \type\()_h264_qpel16_h_lowpass_neon push {lr} - mov ip, #16 + mov r12, #16 bl \type\()_h264_qpel8_h_lowpass_neon sub r0, r0, r3, lsl #4 sub r1, r1, r2, lsl #4 add r0, r0, #8 add r1, r1, #8 - mov ip, #16 + mov r12, #16 pop {lr} endfunc function \type\()_h264_qpel8_h_lowpass_neon -1: vld1.64 {d0, d1}, [r1], r2 - vld1.64 {d16,d17}, [r1], r2 - subs ip, ip, #2 +1: vld1.8 {d0, d1}, [r1], r2 + vld1.8 {d16,d17}, [r1], r2 + subs r12, r12, #2 lowpass_8 d0, d1, d16, d17, d0, d16 -.ifc \type,avg + .ifc \type,avg vld1.8 {d2}, [r0,:64], r3 vrhadd.u8 d0, d0, d2 vld1.8 {d3}, [r0,:64] vrhadd.u8 d16, d16, d3 sub r0, r0, r3 -.endif - vst1.64 {d0}, [r0,:64], r3 - vst1.64 {d16}, [r0,:64], r3 + .endif + vst1.8 {d0}, [r0,:64], r3 + vst1.8 {d16}, [r0,:64], r3 bne 1b bx lr endfunc - .endm +.endm h264_qpel_h_lowpass put h264_qpel_h_lowpass avg - .macro h264_qpel_h_lowpass_l2 type +.macro h264_qpel_h_lowpass_l2 type function \type\()_h264_qpel16_h_lowpass_l2_neon push {lr} - mov ip, #16 + mov r12, #16 bl \type\()_h264_qpel8_h_lowpass_l2_neon sub r0, r0, r2, lsl #4 sub r1, r1, r2, lsl #4 @@ -810,31 +771,31 @@ function \type\()_h264_qpel16_h_lowpass_l2_neon add r0, r0, #8 add r1, r1, #8 add r3, r3, #8 - mov ip, #16 + mov r12, #16 pop {lr} endfunc function \type\()_h264_qpel8_h_lowpass_l2_neon -1: vld1.64 {d0, d1}, [r1], r2 - vld1.64 {d16,d17}, [r1], r2 - vld1.64 {d28}, [r3], r2 - vld1.64 {d29}, [r3], r2 - subs ip, ip, #2 +1: vld1.8 {d0, d1}, [r1], r2 + vld1.8 {d16,d17}, [r1], r2 + vld1.8 {d28}, [r3], r2 + vld1.8 {d29}, [r3], r2 + subs r12, r12, #2 lowpass_8 d0, d1, d16, d17, d0, d1 vrhadd.u8 q0, q0, q14 -.ifc \type,avg + .ifc \type,avg vld1.8 {d2}, [r0,:64], r2 vrhadd.u8 d0, d0, d2 vld1.8 {d3}, [r0,:64] vrhadd.u8 d1, d1, d3 sub r0, r0, r2 -.endif - vst1.64 {d0}, [r0,:64], r2 - vst1.64 {d1}, [r0,:64], r2 + .endif + vst1.8 {d0}, [r0,:64], r2 + vst1.8 {d1}, [r0,:64], r2 bne 1b bx lr endfunc - .endm +.endm h264_qpel_h_lowpass_l2 put h264_qpel_h_lowpass_l2 avg @@ -854,7 +815,7 @@ function put_h264_qpel16_v_lowpass_neon_packed b put_h264_qpel8_v_lowpass_neon endfunc - .macro h264_qpel_v_lowpass type +.macro h264_qpel_v_lowpass type function \type\()_h264_qpel16_v_lowpass_neon mov r4, lr bl \type\()_h264_qpel8_v_lowpass_neon @@ -871,19 +832,19 @@ function \type\()_h264_qpel16_v_lowpass_neon endfunc function \type\()_h264_qpel8_v_lowpass_neon - vld1.64 {d8}, [r1], r3 - vld1.64 {d10}, [r1], r3 - vld1.64 {d12}, [r1], r3 - vld1.64 {d14}, [r1], r3 - vld1.64 {d22}, [r1], r3 - vld1.64 {d24}, [r1], r3 - vld1.64 {d26}, [r1], r3 - vld1.64 {d28}, [r1], r3 - vld1.64 {d9}, [r1], r3 - vld1.64 {d11}, [r1], r3 - vld1.64 {d13}, [r1], r3 - vld1.64 {d15}, [r1], r3 - vld1.64 {d23}, [r1] + vld1.8 {d8}, [r1], r3 + vld1.8 {d10}, [r1], r3 + vld1.8 {d12}, [r1], r3 + vld1.8 {d14}, [r1], r3 + vld1.8 {d22}, [r1], r3 + vld1.8 {d24}, [r1], r3 + vld1.8 {d26}, [r1], r3 + vld1.8 {d28}, [r1], r3 + vld1.8 {d9}, [r1], r3 + vld1.8 {d11}, [r1], r3 + vld1.8 {d13}, [r1], r3 + vld1.8 {d15}, [r1], r3 + vld1.8 {d23}, [r1] transpose_8x8 q4, q5, q6, q7, q11, q12, q13, q14 lowpass_8 d8, d9, d10, d11, d8, d10 @@ -892,7 +853,7 @@ function \type\()_h264_qpel8_v_lowpass_neon lowpass_8 d26, d27, d28, d29, d26, d28 transpose_8x8 d8, d10, d12, d14, d22, d24, d26, d28 -.ifc \type,avg + .ifc \type,avg vld1.8 {d9}, [r0,:64], r2 vrhadd.u8 d8, d8, d9 vld1.8 {d11}, [r0,:64], r2 @@ -910,34 +871,34 @@ function \type\()_h264_qpel8_v_lowpass_neon vld1.8 {d29}, [r0,:64], r2 vrhadd.u8 d28, d28, d29 sub r0, r0, r2, lsl #3 -.endif + .endif - vst1.64 {d8}, [r0,:64], r2 - vst1.64 {d10}, [r0,:64], r2 - vst1.64 {d12}, [r0,:64], r2 - vst1.64 {d14}, [r0,:64], r2 - vst1.64 {d22}, [r0,:64], r2 - vst1.64 {d24}, [r0,:64], r2 - vst1.64 {d26}, [r0,:64], r2 - vst1.64 {d28}, [r0,:64], r2 + vst1.8 {d8}, [r0,:64], r2 + vst1.8 {d10}, [r0,:64], r2 + vst1.8 {d12}, [r0,:64], r2 + vst1.8 {d14}, [r0,:64], r2 + vst1.8 {d22}, [r0,:64], r2 + vst1.8 {d24}, [r0,:64], r2 + vst1.8 {d26}, [r0,:64], r2 + vst1.8 {d28}, [r0,:64], r2 bx lr endfunc - .endm +.endm h264_qpel_v_lowpass put h264_qpel_v_lowpass avg - .macro h264_qpel_v_lowpass_l2 type +.macro h264_qpel_v_lowpass_l2 type function \type\()_h264_qpel16_v_lowpass_l2_neon mov r4, lr bl \type\()_h264_qpel8_v_lowpass_l2_neon sub r1, r1, r3, lsl #2 bl \type\()_h264_qpel8_v_lowpass_l2_neon sub r0, r0, r3, lsl #4 - sub ip, ip, r2, lsl #4 + sub r12, r12, r2, lsl #4 add r0, r0, #8 - add ip, ip, #8 + add r12, r12, #8 sub r1, r1, r3, lsl #4 sub r1, r1, r3, lsl #2 add r1, r1, #8 @@ -947,19 +908,19 @@ function \type\()_h264_qpel16_v_lowpass_l2_neon endfunc function \type\()_h264_qpel8_v_lowpass_l2_neon - vld1.64 {d8}, [r1], r3 - vld1.64 {d10}, [r1], r3 - vld1.64 {d12}, [r1], r3 - vld1.64 {d14}, [r1], r3 - vld1.64 {d22}, [r1], r3 - vld1.64 {d24}, [r1], r3 - vld1.64 {d26}, [r1], r3 - vld1.64 {d28}, [r1], r3 - vld1.64 {d9}, [r1], r3 - vld1.64 {d11}, [r1], r3 - vld1.64 {d13}, [r1], r3 - vld1.64 {d15}, [r1], r3 - vld1.64 {d23}, [r1] + vld1.8 {d8}, [r1], r3 + vld1.8 {d10}, [r1], r3 + vld1.8 {d12}, [r1], r3 + vld1.8 {d14}, [r1], r3 + vld1.8 {d22}, [r1], r3 + vld1.8 {d24}, [r1], r3 + vld1.8 {d26}, [r1], r3 + vld1.8 {d28}, [r1], r3 + vld1.8 {d9}, [r1], r3 + vld1.8 {d11}, [r1], r3 + vld1.8 {d13}, [r1], r3 + vld1.8 {d15}, [r1], r3 + vld1.8 {d23}, [r1] transpose_8x8 q4, q5, q6, q7, q11, q12, q13, q14 lowpass_8 d8, d9, d10, d11, d8, d9 @@ -968,20 +929,20 @@ function \type\()_h264_qpel8_v_lowpass_l2_neon lowpass_8 d26, d27, d28, d29, d26, d27 transpose_8x8 d8, d9, d12, d13, d22, d23, d26, d27 - vld1.64 {d0}, [ip], r2 - vld1.64 {d1}, [ip], r2 - vld1.64 {d2}, [ip], r2 - vld1.64 {d3}, [ip], r2 - vld1.64 {d4}, [ip], r2 + vld1.8 {d0}, [r12], r2 + vld1.8 {d1}, [r12], r2 + vld1.8 {d2}, [r12], r2 + vld1.8 {d3}, [r12], r2 + vld1.8 {d4}, [r12], r2 vrhadd.u8 q0, q0, q4 - vld1.64 {d5}, [ip], r2 + vld1.8 {d5}, [r12], r2 vrhadd.u8 q1, q1, q6 - vld1.64 {d10}, [ip], r2 + vld1.8 {d10}, [r12], r2 vrhadd.u8 q2, q2, q11 - vld1.64 {d11}, [ip], r2 + vld1.8 {d11}, [r12], r2 vrhadd.u8 q5, q5, q13 -.ifc \type,avg + .ifc \type,avg vld1.8 {d16}, [r0,:64], r3 vrhadd.u8 d0, d0, d16 vld1.8 {d17}, [r0,:64], r3 @@ -999,51 +960,51 @@ function \type\()_h264_qpel8_v_lowpass_l2_neon vld1.8 {d17}, [r0,:64], r3 vrhadd.u8 d11, d11, d17 sub r0, r0, r3, lsl #3 -.endif + .endif - vst1.64 {d0}, [r0,:64], r3 - vst1.64 {d1}, [r0,:64], r3 - vst1.64 {d2}, [r0,:64], r3 - vst1.64 {d3}, [r0,:64], r3 - vst1.64 {d4}, [r0,:64], r3 - vst1.64 {d5}, [r0,:64], r3 - vst1.64 {d10}, [r0,:64], r3 - vst1.64 {d11}, [r0,:64], r3 + vst1.8 {d0}, [r0,:64], r3 + vst1.8 {d1}, [r0,:64], r3 + vst1.8 {d2}, [r0,:64], r3 + vst1.8 {d3}, [r0,:64], r3 + vst1.8 {d4}, [r0,:64], r3 + vst1.8 {d5}, [r0,:64], r3 + vst1.8 {d10}, [r0,:64], r3 + vst1.8 {d11}, [r0,:64], r3 bx lr endfunc - .endm +.endm h264_qpel_v_lowpass_l2 put h264_qpel_v_lowpass_l2 avg function put_h264_qpel8_hv_lowpass_neon_top - lowpass_const ip - mov ip, #12 -1: vld1.64 {d0, d1}, [r1], r3 - vld1.64 {d16,d17}, [r1], r3 - subs ip, ip, #2 + lowpass_const r12 + mov r12, #12 +1: vld1.8 {d0, d1}, [r1], r3 + vld1.8 {d16,d17}, [r1], r3 + subs r12, r12, #2 lowpass_8 d0, d1, d16, d17, q11, q12, narrow=0 - vst1.64 {d22-d25}, [r4,:128]! + vst1.8 {d22-d25}, [r4,:128]! bne 1b - vld1.64 {d0, d1}, [r1] + vld1.8 {d0, d1}, [r1] lowpass_8_1 d0, d1, q12, narrow=0 - mov ip, #-16 - add r4, r4, ip - vld1.64 {d30,d31}, [r4,:128], ip - vld1.64 {d20,d21}, [r4,:128], ip - vld1.64 {d18,d19}, [r4,:128], ip - vld1.64 {d16,d17}, [r4,:128], ip - vld1.64 {d14,d15}, [r4,:128], ip - vld1.64 {d12,d13}, [r4,:128], ip - vld1.64 {d10,d11}, [r4,:128], ip - vld1.64 {d8, d9}, [r4,:128], ip - vld1.64 {d6, d7}, [r4,:128], ip - vld1.64 {d4, d5}, [r4,:128], ip - vld1.64 {d2, d3}, [r4,:128], ip - vld1.64 {d0, d1}, [r4,:128] + mov r12, #-16 + add r4, r4, r12 + vld1.8 {d30,d31}, [r4,:128], r12 + vld1.8 {d20,d21}, [r4,:128], r12 + vld1.8 {d18,d19}, [r4,:128], r12 + vld1.8 {d16,d17}, [r4,:128], r12 + vld1.8 {d14,d15}, [r4,:128], r12 + vld1.8 {d12,d13}, [r4,:128], r12 + vld1.8 {d10,d11}, [r4,:128], r12 + vld1.8 {d8, d9}, [r4,:128], r12 + vld1.8 {d6, d7}, [r4,:128], r12 + vld1.8 {d4, d5}, [r4,:128], r12 + vld1.8 {d2, d3}, [r4,:128], r12 + vld1.8 {d0, d1}, [r4,:128] swap4 d1, d3, d5, d7, d8, d10, d12, d14 transpose16_4x4 q0, q1, q2, q3, q4, q5, q6, q7 @@ -1051,31 +1012,31 @@ function put_h264_qpel8_hv_lowpass_neon_top swap4 d17, d19, d21, d31, d24, d26, d28, d22 transpose16_4x4 q8, q9, q10, q15, q12, q13, q14, q11 - vst1.64 {d30,d31}, [r4,:128]! - vst1.64 {d6, d7}, [r4,:128]! - vst1.64 {d20,d21}, [r4,:128]! - vst1.64 {d4, d5}, [r4,:128]! - vst1.64 {d18,d19}, [r4,:128]! - vst1.64 {d2, d3}, [r4,:128]! - vst1.64 {d16,d17}, [r4,:128]! - vst1.64 {d0, d1}, [r4,:128] + vst1.8 {d30,d31}, [r4,:128]! + vst1.8 {d6, d7}, [r4,:128]! + vst1.8 {d20,d21}, [r4,:128]! + vst1.8 {d4, d5}, [r4,:128]! + vst1.8 {d18,d19}, [r4,:128]! + vst1.8 {d2, d3}, [r4,:128]! + vst1.8 {d16,d17}, [r4,:128]! + vst1.8 {d0, d1}, [r4,:128] lowpass_8.16 q4, q12, d8, d9, d24, d25, d8 lowpass_8.16 q5, q13, d10, d11, d26, d27, d9 lowpass_8.16 q6, q14, d12, d13, d28, d29, d10 lowpass_8.16 q7, q11, d14, d15, d22, d23, d11 - vld1.64 {d16,d17}, [r4,:128], ip - vld1.64 {d30,d31}, [r4,:128], ip + vld1.8 {d16,d17}, [r4,:128], r12 + vld1.8 {d30,d31}, [r4,:128], r12 lowpass_8.16 q8, q15, d16, d17, d30, d31, d12 - vld1.64 {d16,d17}, [r4,:128], ip - vld1.64 {d30,d31}, [r4,:128], ip + vld1.8 {d16,d17}, [r4,:128], r12 + vld1.8 {d30,d31}, [r4,:128], r12 lowpass_8.16 q8, q15, d16, d17, d30, d31, d13 - vld1.64 {d16,d17}, [r4,:128], ip - vld1.64 {d30,d31}, [r4,:128], ip + vld1.8 {d16,d17}, [r4,:128], r12 + vld1.8 {d30,d31}, [r4,:128], r12 lowpass_8.16 q8, q15, d16, d17, d30, d31, d14 - vld1.64 {d16,d17}, [r4,:128], ip - vld1.64 {d30,d31}, [r4,:128] + vld1.8 {d16,d17}, [r4,:128], r12 + vld1.8 {d30,d31}, [r4,:128] lowpass_8.16 q8, q15, d16, d17, d30, d31, d15 transpose_8x8 d12, d13, d14, d15, d8, d9, d10, d11 @@ -1083,11 +1044,11 @@ function put_h264_qpel8_hv_lowpass_neon_top bx lr endfunc - .macro h264_qpel8_hv_lowpass type +.macro h264_qpel8_hv_lowpass type function \type\()_h264_qpel8_hv_lowpass_neon mov r10, lr bl put_h264_qpel8_hv_lowpass_neon_top -.ifc \type,avg + .ifc \type,avg vld1.8 {d0}, [r0,:64], r2 vrhadd.u8 d12, d12, d0 vld1.8 {d1}, [r0,:64], r2 @@ -1105,39 +1066,39 @@ function \type\()_h264_qpel8_hv_lowpass_neon vld1.8 {d7}, [r0,:64], r2 vrhadd.u8 d11, d11, d7 sub r0, r0, r2, lsl #3 -.endif + .endif - vst1.64 {d12}, [r0,:64], r2 - vst1.64 {d13}, [r0,:64], r2 - vst1.64 {d14}, [r0,:64], r2 - vst1.64 {d15}, [r0,:64], r2 - vst1.64 {d8}, [r0,:64], r2 - vst1.64 {d9}, [r0,:64], r2 - vst1.64 {d10}, [r0,:64], r2 - vst1.64 {d11}, [r0,:64], r2 + vst1.8 {d12}, [r0,:64], r2 + vst1.8 {d13}, [r0,:64], r2 + vst1.8 {d14}, [r0,:64], r2 + vst1.8 {d15}, [r0,:64], r2 + vst1.8 {d8}, [r0,:64], r2 + vst1.8 {d9}, [r0,:64], r2 + vst1.8 {d10}, [r0,:64], r2 + vst1.8 {d11}, [r0,:64], r2 mov lr, r10 bx lr endfunc - .endm +.endm h264_qpel8_hv_lowpass put h264_qpel8_hv_lowpass avg - .macro h264_qpel8_hv_lowpass_l2 type +.macro h264_qpel8_hv_lowpass_l2 type function \type\()_h264_qpel8_hv_lowpass_l2_neon mov r10, lr bl put_h264_qpel8_hv_lowpass_neon_top - vld1.64 {d0, d1}, [r2,:128]! - vld1.64 {d2, d3}, [r2,:128]! + vld1.8 {d0, d1}, [r2,:128]! + vld1.8 {d2, d3}, [r2,:128]! vrhadd.u8 q0, q0, q6 - vld1.64 {d4, d5}, [r2,:128]! + vld1.8 {d4, d5}, [r2,:128]! vrhadd.u8 q1, q1, q7 - vld1.64 {d6, d7}, [r2,:128]! + vld1.8 {d6, d7}, [r2,:128]! vrhadd.u8 q2, q2, q4 vrhadd.u8 q3, q3, q5 -.ifc \type,avg + .ifc \type,avg vld1.8 {d16}, [r0,:64], r3 vrhadd.u8 d0, d0, d16 vld1.8 {d17}, [r0,:64], r3 @@ -1155,25 +1116,25 @@ function \type\()_h264_qpel8_hv_lowpass_l2_neon vld1.8 {d23}, [r0,:64], r3 vrhadd.u8 d7, d7, d23 sub r0, r0, r3, lsl #3 -.endif - vst1.64 {d0}, [r0,:64], r3 - vst1.64 {d1}, [r0,:64], r3 - vst1.64 {d2}, [r0,:64], r3 - vst1.64 {d3}, [r0,:64], r3 - vst1.64 {d4}, [r0,:64], r3 - vst1.64 {d5}, [r0,:64], r3 - vst1.64 {d6}, [r0,:64], r3 - vst1.64 {d7}, [r0,:64], r3 + .endif + vst1.8 {d0}, [r0,:64], r3 + vst1.8 {d1}, [r0,:64], r3 + vst1.8 {d2}, [r0,:64], r3 + vst1.8 {d3}, [r0,:64], r3 + vst1.8 {d4}, [r0,:64], r3 + vst1.8 {d5}, [r0,:64], r3 + vst1.8 {d6}, [r0,:64], r3 + vst1.8 {d7}, [r0,:64], r3 mov lr, r10 bx lr endfunc - .endm +.endm h264_qpel8_hv_lowpass_l2 put h264_qpel8_hv_lowpass_l2 avg - .macro h264_qpel16_hv type +.macro h264_qpel16_hv type function \type\()_h264_qpel16_hv_lowpass_neon mov r9, lr bl \type\()_h264_qpel8_hv_lowpass_neon @@ -1206,17 +1167,17 @@ function \type\()_h264_qpel16_hv_lowpass_l2_neon mov lr, r9 b \type\()_h264_qpel8_hv_lowpass_l2_neon endfunc - .endm +.endm h264_qpel16_hv put h264_qpel16_hv avg - .macro h264_qpel8 type +.macro h264_qpel8 type function ff_\type\()_h264_qpel8_mc10_neon, export=1 lowpass_const r3 mov r3, r1 sub r1, r1, #2 - mov ip, #8 + mov r12, #8 b \type\()_h264_qpel8_h_lowpass_l2_neon endfunc @@ -1224,7 +1185,7 @@ function ff_\type\()_h264_qpel8_mc20_neon, export=1 lowpass_const r3 sub r1, r1, #2 mov r3, r2 - mov ip, #8 + mov r12, #8 b \type\()_h264_qpel8_h_lowpass_neon endfunc @@ -1232,13 +1193,13 @@ function ff_\type\()_h264_qpel8_mc30_neon, export=1 lowpass_const r3 add r3, r1, #1 sub r1, r1, #2 - mov ip, #8 + mov r12, #8 b \type\()_h264_qpel8_h_lowpass_l2_neon endfunc function ff_\type\()_h264_qpel8_mc01_neon, export=1 push {lr} - mov ip, r1 + mov r12, r1 \type\()_h264_qpel8_mc01: lowpass_const r3 mov r3, r2 @@ -1261,12 +1222,12 @@ T mov sp, r0 mov r0, sp sub r1, r1, #2 mov r3, #8 - mov ip, #8 + mov r12, #8 vpush {d8-d15} bl put_h264_qpel8_h_lowpass_neon ldrd r0, [r11], #8 mov r3, r2 - add ip, sp, #64 + add r12, sp, #64 sub r1, r1, r2, lsl #1 mov r2, #8 bl \type\()_h264_qpel8_v_lowpass_l2_neon @@ -1287,7 +1248,7 @@ T mov sp, r0 sub r1, r1, #2 mov r3, #8 mov r0, sp - mov ip, #8 + mov r12, #8 vpush {d8-d15} bl put_h264_qpel8_h_lowpass_neon mov r4, r0 @@ -1372,7 +1333,7 @@ endfunc function ff_\type\()_h264_qpel8_mc03_neon, export=1 push {lr} - add ip, r1, r2 + add r12, r1, r2 b \type\()_h264_qpel8_mc01 endfunc @@ -1395,12 +1356,12 @@ function ff_\type\()_h264_qpel8_mc33_neon, export=1 sub r1, r1, #1 b \type\()_h264_qpel8_mc11 endfunc - .endm +.endm h264_qpel8 put h264_qpel8 avg - .macro h264_qpel16 type +.macro h264_qpel16 type function ff_\type\()_h264_qpel16_mc10_neon, export=1 lowpass_const r3 mov r3, r1 @@ -1424,7 +1385,7 @@ endfunc function ff_\type\()_h264_qpel16_mc01_neon, export=1 push {r4, lr} - mov ip, r1 + mov r12, r1 \type\()_h264_qpel16_mc01: lowpass_const r3 mov r3, r2 @@ -1451,7 +1412,7 @@ T mov sp, r0 bl put_h264_qpel16_h_lowpass_neon ldrd r0, [r11], #8 mov r3, r2 - add ip, sp, #64 + add r12, sp, #64 sub r1, r1, r2, lsl #1 mov r2, #16 bl \type\()_h264_qpel16_v_lowpass_l2_neon @@ -1554,7 +1515,7 @@ endfunc function ff_\type\()_h264_qpel16_mc03_neon, export=1 push {r4, lr} - add ip, r1, r2 + add r12, r1, r2 b \type\()_h264_qpel16_mc01 endfunc @@ -1577,14 +1538,14 @@ function ff_\type\()_h264_qpel16_mc33_neon, export=1 sub r1, r1, #1 b \type\()_h264_qpel16_mc11 endfunc - .endm +.endm h264_qpel16 put h264_qpel16 avg @ Biweighted prediction - .macro biweight_16 macs, macd +.macro biweight_16 macs, macd vdup.8 d0, r4 vdup.8 d1, r5 vmov q2, q8 @@ -1622,9 +1583,9 @@ endfunc vst1.8 {d24-d25},[r6,:128], r2 bne 1b pop {r4-r6, pc} - .endm +.endm - .macro biweight_8 macs, macd +.macro biweight_8 macs, macd vdup.8 d0, r4 vdup.8 d1, r5 vmov q1, q8 @@ -1652,9 +1613,9 @@ endfunc vst1.8 {d4},[r6,:64], r2 bne 1b pop {r4-r6, pc} - .endm +.endm - .macro biweight_4 macs, macd +.macro biweight_4 macs, macd vdup.8 d0, r4 vdup.8 d1, r5 vmov q1, q8 @@ -1694,9 +1655,9 @@ endfunc vst1.32 {d2[0]},[r6,:32], r2 vst1.32 {d2[1]},[r6,:32], r2 pop {r4-r6, pc} - .endm +.endm - .macro biweight_func w +.macro biweight_func w function ff_biweight_h264_pixels_\w\()_neon, export=1 push {r4-r6, lr} ldr r12, [sp, #16] @@ -1726,7 +1687,7 @@ function ff_biweight_h264_pixels_\w\()_neon, export=1 40: rsb r5, r5, #0 biweight_\w vmlsl.u8, vmlal.u8 endfunc - .endm +.endm biweight_func 16 biweight_func 8 @@ -1734,7 +1695,7 @@ endfunc @ Weighted prediction - .macro weight_16 add +.macro weight_16 add vdup.8 d0, r12 1: subs r2, r2, #2 vld1.8 {d20-d21},[r0,:128], r1 @@ -1761,9 +1722,9 @@ endfunc vst1.8 {d24-d25},[r4,:128], r1 bne 1b pop {r4, pc} - .endm +.endm - .macro weight_8 add +.macro weight_8 add vdup.8 d0, r12 1: subs r2, r2, #2 vld1.8 {d4},[r0,:64], r1 @@ -1782,9 +1743,9 @@ endfunc vst1.8 {d4},[r4,:64], r1 bne 1b pop {r4, pc} - .endm +.endm - .macro weight_4 add +.macro weight_4 add vdup.8 d0, r12 vmov q1, q8 vmov q10, q8 @@ -1818,9 +1779,9 @@ endfunc vst1.32 {d2[0]},[r4,:32], r1 vst1.32 {d2[1]},[r4,:32], r1 pop {r4, pc} - .endm +.endm - .macro weight_func w +.macro weight_func w function ff_weight_h264_pixels_\w\()_neon, export=1 push {r4, lr} ldr r12, [sp, #8] @@ -1845,7 +1806,7 @@ function ff_weight_h264_pixels_\w\()_neon, export=1 10: rsb r12, r12, #0 weight_\w vsub.s16 endfunc - .endm +.endm weight_func 16 weight_func 8 diff --git a/libavcodec/arm/neon.S b/libavcodec/arm/neon.S new file mode 100644 index 0000000000..716a607af7 --- /dev/null +++ b/libavcodec/arm/neon.S @@ -0,0 +1,59 @@ +/* + * Copyright (c) 2008 Mans Rullgard <mans@mansr.com> + * + * 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 + */ + +.macro transpose_8x8 r0, r1, r2, r3, r4, r5, r6, r7 + vtrn.32 \r0, \r4 + vtrn.32 \r1, \r5 + vtrn.32 \r2, \r6 + vtrn.32 \r3, \r7 + vtrn.16 \r0, \r2 + vtrn.16 \r1, \r3 + vtrn.16 \r4, \r6 + vtrn.16 \r5, \r7 + vtrn.8 \r0, \r1 + vtrn.8 \r2, \r3 + vtrn.8 \r4, \r5 + vtrn.8 \r6, \r7 +.endm + +.macro transpose_4x4 r0, r1, r2, r3 + vtrn.16 \r0, \r2 + vtrn.16 \r1, \r3 + vtrn.8 \r0, \r1 + vtrn.8 \r2, \r3 +.endm + +.macro swap4 r0, r1, r2, r3, r4, r5, r6, r7 + vswp \r0, \r4 + vswp \r1, \r5 + vswp \r2, \r6 + vswp \r3, \r7 +.endm + +.macro transpose16_4x4 r0, r1, r2, r3, r4, r5, r6, r7 + vtrn.32 \r0, \r2 + vtrn.32 \r1, \r3 + vtrn.32 \r4, \r6 + vtrn.32 \r5, \r7 + vtrn.16 \r0, \r1 + vtrn.16 \r2, \r3 + vtrn.16 \r4, \r5 + vtrn.16 \r6, \r7 +.endm diff --git a/libavcodec/arm/vp8dsp_neon.S b/libavcodec/arm/vp8dsp_neon.S index 1fb3753aab..b4ab1c7032 100644 --- a/libavcodec/arm/vp8dsp_neon.S +++ b/libavcodec/arm/vp8dsp_neon.S @@ -22,6 +22,7 @@ */ #include "asm.S" +#include "neon.S" function ff_vp8_luma_dc_wht_neon, export=1 vld1.16 {q0-q1}, [r1,:128] @@ -442,23 +443,6 @@ endfunc .endif .endm -.macro transpose8x16matrix - vtrn.32 q0, q4 - vtrn.32 q1, q5 - vtrn.32 q2, q6 - vtrn.32 q3, q7 - - vtrn.16 q0, q2 - vtrn.16 q1, q3 - vtrn.16 q4, q6 - vtrn.16 q5, q7 - - vtrn.8 q0, q1 - vtrn.8 q2, q3 - vtrn.8 q4, q5 - vtrn.8 q6, q7 -.endm - .macro vp8_v_loop_filter16 name, inner=0, simple=0 function ff_vp8_v_loop_filter16\name\()_neon, export=1 vpush {q4-q7} @@ -593,7 +577,7 @@ function ff_vp8_h_loop_filter16\name\()_neon, export=1 vld1.8 {d13}, [r0], r1 vld1.8 {d15}, [r0], r1 - transpose8x16matrix + transpose_8x8 q0, q1, q2, q3, q4, q5, q6, q7 vdup.8 q14, r2 @ flim_E .if !\simple @@ -604,7 +588,7 @@ function ff_vp8_h_loop_filter16\name\()_neon, export=1 sub r0, r0, r1, lsl #4 @ backup 16 rows - transpose8x16matrix + transpose_8x8 q0, q1, q2, q3, q4, q5, q6, q7 @ Store pixels: vst1.8 {d0}, [r0], r1 @@ -658,7 +642,7 @@ function ff_vp8_h_loop_filter8uv\name\()_neon, export=1 vld1.8 {d14}, [r0], r2 vld1.8 {d15}, [r1], r2 - transpose8x16matrix + transpose_8x8 q0, q1, q2, q3, q4, q5, q6, q7 vdup.8 q14, r3 @ flim_E vdup.8 q15, r12 @ flim_I @@ -669,7 +653,7 @@ function ff_vp8_h_loop_filter8uv\name\()_neon, export=1 sub r0, r0, r2, lsl #3 @ backup u 8 rows sub r1, r1, r2, lsl #3 @ backup v 8 rows - transpose8x16matrix + transpose_8x8 q0, q1, q2, q3, q4, q5, q6, q7 @ Store pixels: vst1.8 {d0}, [r0], r2 diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c index f341b1c554..c796235c80 100644 --- a/libavcodec/atrac1.c +++ b/libavcodec/atrac1.c @@ -72,6 +72,7 @@ typedef struct { * The atrac1 context, holds all needed parameters for decoding */ typedef struct { + AVFrame frame; AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer @@ -273,14 +274,14 @@ static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) static int atrac1_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; AT1Ctx *q = avctx->priv_data; - int ch, ret, out_size; + int ch, ret; GetBitContext gb; - float* samples = data; + float *samples; if (buf_size < 212 * q->channels) { @@ -288,12 +289,13 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data, return AVERROR_INVALIDDATA; } - out_size = q->channels * AT1_SU_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); + /* get output buffer */ + q->frame.nb_samples = AT1_SU_SAMPLES; + if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (float *)q->frame.data[0]; for (ch = 0; ch < q->channels; ch++) { AT1SUCtx* su = &q->SUs[ch]; @@ -321,7 +323,9 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data, AT1_SU_SAMPLES, 2); } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = q->frame; + return avctx->block_align; } @@ -389,6 +393,9 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx) q->SUs[1].spectrum[0] = q->SUs[1].spec1; q->SUs[1].spectrum[1] = q->SUs[1].spec2; + avcodec_get_frame_defaults(&q->frame); + avctx->coded_frame = &q->frame; + return 0; } @@ -401,5 +408,6 @@ AVCodec ff_atrac1_decoder = { .init = atrac1_decode_init, .close = atrac1_decode_end, .decode = atrac1_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"), }; diff --git a/libavcodec/atrac3.c b/libavcodec/atrac3.c index 25beeeeb6c..ccbb718e00 100644 --- a/libavcodec/atrac3.c +++ b/libavcodec/atrac3.c @@ -86,6 +86,7 @@ typedef struct { } channel_unit; typedef struct { + AVFrame frame; GetBitContext gb; //@{ /** stream data */ @@ -823,16 +824,16 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf, * @param avctx pointer to the AVCodecContext */ -static int atrac3_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) { +static int atrac3_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; ATRAC3Context *q = avctx->priv_data; - int result = 0, out_size; + int result; const uint8_t* databuf; - float *samples_flt = data; - int16_t *samples_s16 = data; + float *samples_flt; + int16_t *samples_s16; if (buf_size < avctx->block_align) { av_log(avctx, AV_LOG_ERROR, @@ -840,12 +841,14 @@ static int atrac3_decode_frame(AVCodecContext *avctx, return AVERROR_INVALIDDATA; } - out_size = SAMPLES_PER_FRAME * q->channels * - 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); + /* get output buffer */ + q->frame.nb_samples = SAMPLES_PER_FRAME; + if ((result = avctx->get_buffer(avctx, &q->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return result; } + samples_flt = (float *)q->frame.data[0]; + samples_s16 = (int16_t *)q->frame.data[0]; /* Check if we need to descramble and what buffer to pass on. */ if (q->scrambled_stream) { @@ -875,7 +878,9 @@ static int atrac3_decode_frame(AVCodecContext *avctx, (const float **)q->outSamples, SAMPLES_PER_FRAME, q->channels); } - *data_size = out_size; + + *got_frame_ptr = 1; + *(AVFrame *)data = q->frame; return avctx->block_align; } @@ -1047,6 +1052,9 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx) } } + avcodec_get_frame_defaults(&q->frame); + avctx->coded_frame = &q->frame; + return 0; } @@ -1060,6 +1068,6 @@ AVCodec ff_atrac3_decoder = .init = atrac3_decode_init, .close = atrac3_decode_close, .decode = atrac3_decode_frame, - .capabilities = CODEC_CAP_SUBFRAMES, + .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"), }; diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h index 1381a3e018..dfd15e5683 100644 --- a/libavcodec/avcodec.h +++ b/libavcodec/avcodec.h @@ -491,8 +491,10 @@ enum CodecID { #define CH_LAYOUT_STEREO_DOWNMIX AV_CH_LAYOUT_STEREO_DOWNMIX #endif +#if FF_API_OLD_DECODE_AUDIO /* in bytes */ #define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio +#endif /** * Required number of additionally allocated bytes at the end of the input bitstream for decoding. @@ -947,21 +949,37 @@ typedef struct AVPacket { * sizeof(AVFrame) must not be used outside libav*. */ typedef struct AVFrame { +#if FF_API_DATA_POINTERS +#define AV_NUM_DATA_POINTERS 4 +#else +#define AV_NUM_DATA_POINTERS 8 +#endif /** - * pointer to the picture planes. + * pointer to the picture/channel planes. * This might be different from the first allocated byte - * - encoding: - * - decoding: + * - encoding: Set by user + * - decoding: set by AVCodecContext.get_buffer() + */ + uint8_t *data[AV_NUM_DATA_POINTERS]; + + /** + * Size, in bytes, of the data for each picture/channel plane. + * + * For audio, only linesize[0] may be set. For planar audio, each channel + * plane must be the same size. + * + * - encoding: Set by user (video only) + * - decoding: set by AVCodecContext.get_buffer() */ - uint8_t *data[4]; - int linesize[4]; + int linesize[AV_NUM_DATA_POINTERS]; + /** * pointer to the first allocated byte of the picture. Can be used in get_buffer/release_buffer. * This isn't used by libavcodec unless the default get/release_buffer() is used. * - encoding: * - decoding: */ - uint8_t *base[4]; + uint8_t *base[AV_NUM_DATA_POINTERS]; /** * 1 -> keyframe, 0-> not * - encoding: Set by libavcodec. @@ -1008,7 +1026,7 @@ typedef struct AVFrame { * buffer age (1->was last buffer and dint change, 2->..., ...). * Set to INT_MAX if the buffer has not been used yet. * - encoding: unused - * - decoding: MUST be set by get_buffer(). + * - decoding: MUST be set by get_buffer() for video. */ int age; @@ -1085,7 +1103,7 @@ typedef struct AVFrame { * - encoding: Set by libavcodec. if flags&CODEC_FLAG_PSNR. * - decoding: unused */ - uint64_t error[4]; + uint64_t error[AV_NUM_DATA_POINTERS]; /** * type of the buffer (to keep track of who has to deallocate data[*]) @@ -1207,6 +1225,33 @@ typedef struct AVFrame { void *thread_opaque; /** + * number of audio samples (per channel) described by this frame + * - encoding: unused + * - decoding: Set by libavcodec + */ + int nb_samples; + + /** + * pointers to the data planes/channels. + * + * For video, this should simply point to data[]. + * + * For planar audio, each channel has a separate data pointer, and + * linesize[0] contains the size of each channel buffer. + * For packed audio, there is just one data pointer, and linesize[0] + * contains the total size of the buffer for all channels. + * + * Note: Both data and extended_data will always be set by get_buffer(), + * but for planar audio with more channels that can fit in data, + * extended_data must be used by the decoder in order to access all + * channels. + * + * encoding: unused + * decoding: set by AVCodecContext.get_buffer() + */ + uint8_t **extended_data; + + /** * frame timestamp estimated using various heuristics, in stream time base * - encoding: unused * - decoding: set by libavcodec, read by user. @@ -1379,7 +1424,7 @@ typedef struct AVCodecContext { * @param offset offset into the AVFrame.data from which the slice should be read */ void (*draw_horiz_band)(struct AVCodecContext *s, - const AVFrame *src, int offset[4], + const AVFrame *src, int offset[AV_NUM_DATA_POINTERS], int y, int type, int height); /* audio only */ @@ -1602,15 +1647,56 @@ typedef struct AVCodecContext { /** * Called at the beginning of each frame to get a buffer for it. - * If pic.reference is set then the frame will be read later by libavcodec. - * avcodec_align_dimensions2() should be used to find the required width and - * height, as they normally need to be rounded up to the next multiple of 16. + * + * The function will set AVFrame.data[], AVFrame.linesize[]. + * AVFrame.extended_data[] must also be set, but it should be the same as + * AVFrame.data[] except for planar audio with more channels than can fit + * in AVFrame.data[]. In that case, AVFrame.data[] shall still contain as + * many data pointers as it can hold. + * * if CODEC_CAP_DR1 is not set then get_buffer() must call * avcodec_default_get_buffer() instead of providing buffers allocated by * some other means. + * + * AVFrame.data[] should be 32- or 16-byte-aligned unless the CPU doesn't + * need it. avcodec_default_get_buffer() aligns the output buffer properly, + * but if get_buffer() is overridden then alignment considerations should + * be taken into account. + * + * @see avcodec_default_get_buffer() + * + * Video: + * + * If pic.reference is set then the frame will be read later by libavcodec. + * avcodec_align_dimensions2() should be used to find the required width and + * height, as they normally need to be rounded up to the next multiple of 16. + * * If frame multithreading is used and thread_safe_callbacks is set, - * it may be called from a different thread, but not from more than one at once. - * Does not need to be reentrant. + * it may be called from a different thread, but not from more than one at + * once. Does not need to be reentrant. + * + * @see release_buffer(), reget_buffer() + * @see avcodec_align_dimensions2() + * + * Audio: + * + * Decoders request a buffer of a particular size by setting + * AVFrame.nb_samples prior to calling get_buffer(). The decoder may, + * however, utilize only part of the buffer by setting AVFrame.nb_samples + * to a smaller value in the output frame. + * + * Decoders cannot use the buffer after returning from + * avcodec_decode_audio4(), so they will not call release_buffer(), as it + * is assumed to be released immediately upon return. + * + * As a convenience, av_samples_get_buffer_size() and + * av_samples_fill_arrays() in libavutil may be used by custom get_buffer() + * functions to find the required data size and to fill data pointers and + * linesize. In AVFrame.linesize, only linesize[0] may be set for audio + * since all planes must be the same size. + * + * @see av_samples_get_buffer_size(), av_samples_fill_arrays() + * * - encoding: unused * - decoding: Set by libavcodec, user can override. */ @@ -1929,7 +2015,7 @@ typedef struct AVCodecContext { * - encoding: Set by libavcodec if flags&CODEC_FLAG_PSNR. * - decoding: unused */ - uint64_t error[4]; + uint64_t error[AV_NUM_DATA_POINTERS]; /** * motion estimation comparison function @@ -3253,8 +3339,8 @@ typedef struct AVHWAccel { * the last component is alpha */ typedef struct AVPicture { - uint8_t *data[4]; - int linesize[4]; ///< number of bytes per line + uint8_t *data[AV_NUM_DATA_POINTERS]; + int linesize[AV_NUM_DATA_POINTERS]; ///< number of bytes per line } AVPicture; #define AVPALETTE_SIZE 1024 @@ -3922,7 +4008,7 @@ void avcodec_align_dimensions(AVCodecContext *s, int *width, int *height); * according to avcodec_get_edge_width() before. */ void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height, - int linesize_align[4]); + int linesize_align[AV_NUM_DATA_POINTERS]); enum PixelFormat avcodec_default_get_format(struct AVCodecContext *s, const enum PixelFormat * fmt); @@ -4005,7 +4091,12 @@ int avcodec_open(AVCodecContext *avctx, AVCodec *codec); */ int avcodec_open2(AVCodecContext *avctx, AVCodec *codec, AVDictionary **options); +#if FF_API_OLD_DECODE_AUDIO /** + * Wrapper function which calls avcodec_decode_audio4. + * + * @deprecated Use avcodec_decode_audio4 instead. + * * Decode the audio frame of size avpkt->size from avpkt->data into samples. * Some decoders may support multiple frames in a single AVPacket, such * decoders would then just decode the first frame. In this case, @@ -4040,6 +4131,8 @@ int avcodec_open2(AVCodecContext *avctx, AVCodec *codec, AVDictionary **options) * * @param avctx the codec context * @param[out] samples the output buffer, sample type in avctx->sample_fmt + * If the sample format is planar, each channel plane will + * be the same size, with no padding between channels. * @param[in,out] frame_size_ptr the output buffer size in bytes * @param[in] avpkt The input AVPacket containing the input buffer. * You can create such packet with av_init_packet() and by then setting @@ -4048,9 +4141,46 @@ int avcodec_open2(AVCodecContext *avctx, AVCodec *codec, AVDictionary **options) * @return On error a negative value is returned, otherwise the number of bytes * used or zero if no frame data was decompressed (used) from the input AVPacket. */ -int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples, +attribute_deprecated int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples, int *frame_size_ptr, AVPacket *avpkt); +#endif + +/** + * Decode the audio frame of size avpkt->size from avpkt->data into frame. + * + * Some decoders may support multiple frames in a single AVPacket. Such + * decoders would then just decode the first frame. In this case, + * avcodec_decode_audio4 has to be called again with an AVPacket containing + * the remaining data in order to decode the second frame, etc... + * Even if no frames are returned, the packet needs to be fed to the decoder + * with remaining data until it is completely consumed or an error occurs. + * + * @warning The input buffer, avpkt->data must be FF_INPUT_BUFFER_PADDING_SIZE + * larger than the actual read bytes because some optimized bitstream + * readers read 32 or 64 bits at once and could read over the end. + * + * @note You might have to align the input buffer. The alignment requirements + * depend on the CPU and the decoder. + * + * @param avctx the codec context + * @param[out] frame The AVFrame in which to store decoded audio samples. + * Decoders request a buffer of a particular size by setting + * AVFrame.nb_samples prior to calling get_buffer(). The + * decoder may, however, only utilize part of the buffer by + * setting AVFrame.nb_samples to a smaller value in the + * output frame. + * @param[out] got_frame_ptr Zero if no frame could be decoded, otherwise it is + * non-zero. + * @param[in] avpkt The input AVPacket containing the input buffer. + * At least avpkt->data and avpkt->size should be set. Some + * decoders might also require additional fields to be set. + * @return A negative error code is returned if an error occurred during + * decoding, otherwise the number of bytes consumed from the input + * AVPacket is returned. + */ +int avcodec_decode_audio4(AVCodecContext *avctx, AVFrame *frame, + int *got_frame_ptr, AVPacket *avpkt); /** * Decode the video frame of size avpkt->size from avpkt->data into picture. diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c index 4f716824c7..94be26ed49 100644 --- a/libavcodec/binkaudio.c +++ b/libavcodec/binkaudio.c @@ -45,6 +45,7 @@ static float quant_table[96]; #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11) typedef struct { + AVFrame frame; GetBitContext gb; DSPContext dsp; FmtConvertContext fmt_conv; @@ -147,6 +148,9 @@ static av_cold int decode_init(AVCodecContext *avctx) else return -1; + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -293,6 +297,7 @@ static av_cold int decode_end(AVCodecContext *avctx) ff_rdft_end(&s->trans.rdft); else if (CONFIG_BINKAUDIO_DCT_DECODER) ff_dct_end(&s->trans.dct); + return 0; } @@ -302,20 +307,19 @@ static void get_bits_align32(GetBitContext *s) if (n) skip_bits(s, n); } -static int decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { BinkAudioContext *s = avctx->priv_data; - int16_t *samples = data; + int16_t *samples; GetBitContext *gb = &s->gb; - int out_size, consumed = 0; + int ret, consumed = 0; if (!get_bits_left(gb)) { uint8_t *buf; /* handle end-of-stream */ if (!avpkt->size) { - *data_size = 0; + *got_frame_ptr = 0; return 0; } if (avpkt->size < 4) { @@ -334,11 +338,13 @@ static int decode_frame(AVCodecContext *avctx, skip_bits_long(gb, 32); } - out_size = s->block_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); + /* get output buffer */ + s->frame.nb_samples = s->block_size / avctx->channels; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (int16_t *)s->frame.data[0]; if (decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT)) { av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n"); @@ -346,7 +352,9 @@ static int decode_frame(AVCodecContext *avctx, } get_bits_align32(gb); - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return consumed; } @@ -358,7 +366,7 @@ AVCodec ff_binkaudio_rdft_decoder = { .init = decode_init, .close = decode_end, .decode = decode_frame, - .capabilities = CODEC_CAP_DELAY, + .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)") }; @@ -370,6 +378,6 @@ AVCodec ff_binkaudio_dct_decoder = { .init = decode_init, .close = decode_end, .decode = decode_frame, - .capabilities = CODEC_CAP_DELAY, + .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)") }; diff --git a/libavcodec/cook.c b/libavcodec/cook.c index 6a076d9310..b285f88a15 100644 --- a/libavcodec/cook.c +++ b/libavcodec/cook.c @@ -122,6 +122,7 @@ typedef struct cook { void (* saturate_output) (struct cook *q, int chan, float *out); AVCodecContext* avctx; + AVFrame frame; GetBitContext gb; /* stream data */ int nb_channels; @@ -131,6 +132,7 @@ typedef struct cook { int samples_per_channel; /* states */ AVLFG random_state; + int discarded_packets; /* transform data */ FFTContext mdct_ctx; @@ -896,7 +898,8 @@ mlt_compensate_output(COOKContext *q, float *decode_buffer, float *out, int chan) { imlt_gain(q, decode_buffer, gains_ptr, previous_buffer); - q->saturate_output (q, chan, out); + if (out) + q->saturate_output(q, chan, out); } @@ -953,24 +956,28 @@ static void decode_subpacket(COOKContext *q, COOKSubpacket *p, * @param avctx pointer to the AVCodecContext */ -static int cook_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) { +static int cook_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; COOKContext *q = avctx->priv_data; - int i, out_size; + float *samples = NULL; + int i, ret; int offset = 0; int chidx = 0; if (buf_size < avctx->block_align) return buf_size; - out_size = q->nb_channels * q->samples_per_channel * - 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); + /* get output buffer */ + if (q->discarded_packets >= 2) { + q->frame.nb_samples = q->samples_per_channel; + if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + samples = (float *)q->frame.data[0]; } /* estimate subpacket sizes */ @@ -990,15 +997,21 @@ static int cook_decode_frame(AVCodecContext *avctx, q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv; q->subpacket[i].ch_idx = chidx; av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align); - decode_subpacket(q, &q->subpacket[i], buf + offset, data); + decode_subpacket(q, &q->subpacket[i], buf + offset, samples); offset += q->subpacket[i].size; chidx += q->subpacket[i].num_channels; av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb)); } - *data_size = out_size; /* Discard the first two frames: no valid audio. */ - if (avctx->frame_number < 2) *data_size = 0; + if (q->discarded_packets < 2) { + q->discarded_packets++; + *got_frame_ptr = 0; + return avctx->block_align; + } + + *got_frame_ptr = 1; + *(AVFrame *)data = q->frame; return avctx->block_align; } @@ -1246,6 +1259,9 @@ static av_cold int cook_decode_init(AVCodecContext *avctx) else avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO; + avcodec_get_frame_defaults(&q->frame); + avctx->coded_frame = &q->frame; + #ifdef DEBUG dump_cook_context(q); #endif @@ -1262,5 +1278,6 @@ AVCodec ff_cook_decoder = .init = cook_decode_init, .close = cook_decode_close, .decode = cook_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("COOK"), }; diff --git a/libavcodec/dca.c b/libavcodec/dca.c index edafb967b4..7c43b922b1 100644 --- a/libavcodec/dca.c +++ b/libavcodec/dca.c @@ -261,6 +261,7 @@ static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int id typedef struct { AVCodecContext *avctx; + AVFrame frame; /* Frame header */ int frame_type; ///< type of the current frame int samples_deficit; ///< deficit sample count @@ -1634,9 +1635,8 @@ static void dca_exss_parse_header(DCAContext *s) * Main frame decoding function * FIXME add arguments */ -static int dca_decode_frame(AVCodecContext * avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int dca_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; @@ -1644,9 +1644,8 @@ static int dca_decode_frame(AVCodecContext * avctx, int lfe_samples; int num_core_channels = 0; int i, ret; - float *samples_flt = data; - int16_t *samples_s16 = data; - int out_size; + float *samples_flt; + int16_t *samples_s16; DCAContext *s = avctx->priv_data; int channels; int core_ss_end; @@ -1832,11 +1831,14 @@ static int dca_decode_frame(AVCodecContext * avctx, avctx->channels = channels; } - out_size = 256 / 8 * s->sample_blocks * channels * - av_get_bytes_per_sample(avctx->sample_fmt); - if (*data_size < out_size) - return AVERROR(EINVAL); - *data_size = out_size; + /* get output buffer */ + s->frame.nb_samples = 256 * (s->sample_blocks / 8); + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + samples_flt = (float *)s->frame.data[0]; + samples_s16 = (int16_t *)s->frame.data[0]; /* filter to get final output */ for (i = 0; i < (s->sample_blocks / 8); i++) { @@ -1870,6 +1872,9 @@ static int dca_decode_frame(AVCodecContext * avctx, s->lfe_data[i] = s->lfe_data[i + lfe_samples]; } + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return buf_size; } @@ -1912,6 +1917,9 @@ static av_cold int dca_decode_init(AVCodecContext * avctx) avctx->channels = avctx->request_channels; } + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -1940,7 +1948,7 @@ AVCodec ff_dca_decoder = { .decode = dca_decode_frame, .close = dca_decode_end, .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"), - .capabilities = CODEC_CAP_CHANNEL_CONF, + .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, diff --git a/libavcodec/dpcm.c b/libavcodec/dpcm.c index 2f87f51774..b96461984b 100644 --- a/libavcodec/dpcm.c +++ b/libavcodec/dpcm.c @@ -42,6 +42,7 @@ #include "bytestream.h" typedef struct DPCMContext { + AVFrame frame; int channels; int16_t roq_square_array[256]; int sample[2]; ///< previous sample (for SOL_DPCM) @@ -162,22 +163,25 @@ static av_cold int dpcm_decode_init(AVCodecContext *avctx) else avctx->sample_fmt = AV_SAMPLE_FMT_S16; + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } -static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size, - AVPacket *avpkt) +static int dpcm_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; const uint8_t *buf_end = buf + buf_size; DPCMContext *s = avctx->priv_data; - int out = 0; + int out = 0, ret; int predictor[2]; int ch = 0; int stereo = s->channels - 1; - int16_t *output_samples = data; + int16_t *output_samples; /* calculate output size */ switch(avctx->codec->id) { @@ -197,15 +201,18 @@ static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size, out = buf_size; break; } - out *= av_get_bytes_per_sample(avctx->sample_fmt); if (out <= 0) { av_log(avctx, AV_LOG_ERROR, "packet is too small\n"); return AVERROR(EINVAL); } - if (*data_size < out) { - av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n"); - return AVERROR(EINVAL); + + /* get output buffer */ + s->frame.nb_samples = out / s->channels; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + output_samples = (int16_t *)s->frame.data[0]; switch(avctx->codec->id) { @@ -307,7 +314,9 @@ static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size, break; } - *data_size = out; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return buf_size; } @@ -319,6 +328,7 @@ AVCodec ff_ ## name_ ## _decoder = { \ .priv_data_size = sizeof(DPCMContext), \ .init = dpcm_decode_init, \ .decode = dpcm_decode_frame, \ + .capabilities = CODEC_CAP_DR1, \ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ } diff --git a/libavcodec/dsicinav.c b/libavcodec/dsicinav.c index 3cf32ca32b..4c6c41c2fe 100644 --- a/libavcodec/dsicinav.c +++ b/libavcodec/dsicinav.c @@ -44,6 +44,7 @@ typedef struct CinVideoContext { } CinVideoContext; typedef struct CinAudioContext { + AVFrame frame; int initial_decode_frame; int delta; } CinAudioContext; @@ -318,25 +319,28 @@ static av_cold int cinaudio_decode_init(AVCodecContext *avctx) cin->delta = 0; avctx->sample_fmt = AV_SAMPLE_FMT_S16; + avcodec_get_frame_defaults(&cin->frame); + avctx->coded_frame = &cin->frame; + return 0; } -static int cinaudio_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int cinaudio_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; CinAudioContext *cin = avctx->priv_data; const uint8_t *buf_end = buf + avpkt->size; - int16_t *samples = data; - int delta, out_size; - - out_size = (avpkt->size - cin->initial_decode_frame) * - 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); + int16_t *samples; + int delta, ret; + + /* get output buffer */ + cin->frame.nb_samples = avpkt->size - cin->initial_decode_frame; + if ((ret = avctx->get_buffer(avctx, &cin->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (int16_t *)cin->frame.data[0]; delta = cin->delta; if (cin->initial_decode_frame) { @@ -352,7 +356,8 @@ static int cinaudio_decode_frame(AVCodecContext *avctx, } cin->delta = delta; - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = cin->frame; return avpkt->size; } @@ -377,5 +382,6 @@ AVCodec ff_dsicinaudio_decoder = { .priv_data_size = sizeof(CinAudioContext), .init = cinaudio_decode_init, .decode = cinaudio_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Delphine Software International CIN audio"), }; diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c index c140440436..ca0863107a 100644 --- a/libavcodec/flacdec.c +++ b/libavcodec/flacdec.c @@ -49,6 +49,7 @@ typedef struct FLACContext { FLACSTREAMINFO AVCodecContext *avctx; ///< parent AVCodecContext + AVFrame frame; GetBitContext gb; ///< GetBitContext initialized to start at the current frame int blocksize; ///< number of samples in the current frame @@ -116,6 +117,9 @@ static av_cold int flac_decode_init(AVCodecContext *avctx) allocate_buffers(s); s->got_streaminfo = 1; + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -542,20 +546,18 @@ static int decode_frame(FLACContext *s) return 0; } -static int flac_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int flac_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; FLACContext *s = avctx->priv_data; int i, j = 0, bytes_read = 0; - int16_t *samples_16 = data; - int32_t *samples_32 = data; - int alloc_data_size= *data_size; - int output_size; + int16_t *samples_16; + int32_t *samples_32; + int ret; - *data_size=0; + *got_frame_ptr = 0; if (s->max_framesize == 0) { s->max_framesize = @@ -586,15 +588,14 @@ static int flac_decode_frame(AVCodecContext *avctx, } bytes_read = (get_bits_count(&s->gb)+7)/8; - /* check if allocated data size is large enough for output */ - output_size = s->blocksize * s->channels * - av_get_bytes_per_sample(avctx->sample_fmt); - if (output_size > alloc_data_size) { - av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than " - "allocated data size\n"); - return -1; + /* get output buffer */ + s->frame.nb_samples = s->blocksize; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } - *data_size = output_size; + samples_16 = (int16_t *)s->frame.data[0]; + samples_32 = (int32_t *)s->frame.data[0]; #define DECORRELATE(left, right)\ assert(s->channels == 2);\ @@ -639,6 +640,9 @@ static int flac_decode_frame(AVCodecContext *avctx, buf_size - bytes_read, buf_size); } + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return bytes_read; } @@ -662,5 +666,6 @@ AVCodec ff_flac_decoder = { .init = flac_decode_init, .close = flac_decode_close, .decode = flac_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"), }; diff --git a/libavcodec/g722.h b/libavcodec/g722.h index 5edb6c8119..69e7a86e25 100644 --- a/libavcodec/g722.h +++ b/libavcodec/g722.h @@ -26,10 +26,12 @@ #define AVCODEC_G722_H #include <stdint.h> +#include "avcodec.h" #define PREV_SAMPLES_BUF_SIZE 1024 typedef struct { + AVFrame frame; int16_t prev_samples[PREV_SAMPLES_BUF_SIZE]; ///< memory of past decoded samples int prev_samples_pos; ///< the number of values in prev_samples diff --git a/libavcodec/g722dec.c b/libavcodec/g722dec.c index 2be47159a4..652a1aa4ae 100644 --- a/libavcodec/g722dec.c +++ b/libavcodec/g722dec.c @@ -66,6 +66,9 @@ static av_cold int g722_decode_init(AVCodecContext * avctx) c->band[1].scale_factor = 2; c->prev_samples_pos = 22; + avcodec_get_frame_defaults(&c->frame); + avctx->coded_frame = &c->frame; + return 0; } @@ -81,20 +84,22 @@ static const int16_t *low_inv_quants[3] = { ff_g722_low_inv_quant6, ff_g722_low_inv_quant4 }; static int g722_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { G722Context *c = avctx->priv_data; - int16_t *out_buf = data; - int j, out_len; + int16_t *out_buf; + int j, ret; const int skip = 8 - avctx->bits_per_coded_sample; const int16_t *quantizer_table = low_inv_quants[skip]; GetBitContext gb; - out_len = avpkt->size * 2 * av_get_bytes_per_sample(avctx->sample_fmt); - if (*data_size < out_len) { - av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); - return AVERROR(EINVAL); + /* get output buffer */ + c->frame.nb_samples = avpkt->size * 2; + if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + out_buf = (int16_t *)c->frame.data[0]; init_get_bits(&gb, avpkt->data, avpkt->size * 8); @@ -128,7 +133,10 @@ static int g722_decode_frame(AVCodecContext *avctx, void *data, c->prev_samples_pos = 22; } } - *data_size = out_len; + + *got_frame_ptr = 1; + *(AVFrame *)data = c->frame; + return avpkt->size; } @@ -139,5 +147,6 @@ AVCodec ff_adpcm_g722_decoder = { .priv_data_size = sizeof(G722Context), .init = g722_decode_init, .decode = g722_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"), }; diff --git a/libavcodec/g726.c b/libavcodec/g726.c index ae1b5a3001..8c02a392cc 100644 --- a/libavcodec/g726.c +++ b/libavcodec/g726.c @@ -75,6 +75,7 @@ typedef struct G726Tables { typedef struct G726Context { AVClass *class; + AVFrame frame; G726Tables tbls; /**< static tables needed for computation */ Float11 sr[2]; /**< prev. reconstructed samples */ @@ -427,26 +428,31 @@ static av_cold int g726_decode_init(AVCodecContext *avctx) avctx->sample_fmt = AV_SAMPLE_FMT_S16; + avcodec_get_frame_defaults(&c->frame); + avctx->coded_frame = &c->frame; + return 0; } -static int g726_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int g726_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; G726Context *c = avctx->priv_data; - int16_t *samples = data; + int16_t *samples; GetBitContext gb; - int out_samples, out_size; + int out_samples, ret; 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); + + /* get output buffer */ + c->frame.nb_samples = out_samples; + if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (int16_t *)c->frame.data[0]; init_get_bits(&gb, buf, buf_size * 8); @@ -456,7 +462,9 @@ static int g726_decode_frame(AVCodecContext *avctx, if (get_bits_left(&gb) > 0) av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n"); - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = c->frame; + return buf_size; } @@ -474,6 +482,7 @@ AVCodec ff_adpcm_g726_decoder = { .init = g726_decode_init, .decode = g726_decode_frame, .flush = g726_decode_flush, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"), }; #endif diff --git a/libavcodec/gsmdec.c b/libavcodec/gsmdec.c index 2d1dee8384..eec515b41f 100644 --- a/libavcodec/gsmdec.c +++ b/libavcodec/gsmdec.c @@ -32,6 +32,8 @@ static av_cold int gsm_init(AVCodecContext *avctx) { + GSMContext *s = avctx->priv_data; + avctx->channels = 1; if (!avctx->sample_rate) avctx->sample_rate = 8000; @@ -47,30 +49,35 @@ static av_cold int gsm_init(AVCodecContext *avctx) avctx->block_align = GSM_MS_BLOCK_SIZE; } + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } static int gsm_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { + GSMContext *s = avctx->priv_data; int res; GetBitContext gb; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; - int16_t *samples = data; - int frame_bytes = avctx->frame_size * - av_get_bytes_per_sample(avctx->sample_fmt); - - if (*data_size < frame_bytes) { - av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); - return AVERROR(EINVAL); - } + int16_t *samples; if (buf_size < avctx->block_align) { av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); return AVERROR_INVALIDDATA; } + /* get output buffer */ + s->frame.nb_samples = avctx->frame_size; + if ((res = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return res; + } + samples = (int16_t *)s->frame.data[0]; + switch (avctx->codec_id) { case CODEC_ID_GSM: init_get_bits(&gb, buf, buf_size * 8); @@ -85,7 +92,10 @@ static int gsm_decode_frame(AVCodecContext *avctx, void *data, if (res < 0) return res; } - *data_size = frame_bytes; + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return avctx->block_align; } @@ -103,6 +113,7 @@ AVCodec ff_gsm_decoder = { .init = gsm_init, .decode = gsm_decode_frame, .flush = gsm_flush, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("GSM"), }; @@ -114,5 +125,6 @@ AVCodec ff_gsm_ms_decoder = { .init = gsm_init, .decode = gsm_decode_frame, .flush = gsm_flush, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("GSM Microsoft variant"), }; diff --git a/libavcodec/gsmdec_data.h b/libavcodec/gsmdec_data.h index bb4f159c60..4e2c7e6846 100644 --- a/libavcodec/gsmdec_data.h +++ b/libavcodec/gsmdec_data.h @@ -23,6 +23,7 @@ #define AVCODEC_GSMDEC_DATA #include <stdint.h> +#include "avcodec.h" // input and output sizes in byte #define GSM_BLOCK_SIZE 33 @@ -30,6 +31,7 @@ #define GSM_FRAME_SIZE 160 typedef struct { + AVFrame frame; // Contains first 120 elements from the previous frame // (used by long_term_synth according to the "lag"), // then in the following 160 elements the current diff --git a/libavcodec/huffyuv.c b/libavcodec/huffyuv.c index 1ac1913745..574daacc0b 100644 --- a/libavcodec/huffyuv.c +++ b/libavcodec/huffyuv.c @@ -956,8 +956,8 @@ static inline int encode_bgra_bitstream(HYuvContext *s, int count, int planes){ #if CONFIG_HUFFYUV_DECODER || CONFIG_FFVHUFF_DECODER static void draw_slice(HYuvContext *s, int y){ - int h, cy; - int offset[4]; + int h, cy, i; + int offset[AV_NUM_DATA_POINTERS]; if(s->avctx->draw_horiz_band==NULL) return; @@ -974,7 +974,8 @@ static void draw_slice(HYuvContext *s, int y){ offset[0] = s->picture.linesize[0]*y; offset[1] = s->picture.linesize[1]*cy; offset[2] = s->picture.linesize[2]*cy; - offset[3] = 0; + for (i = 3; i < AV_NUM_DATA_POINTERS; i++) + offset[i] = 0; emms_c(); s->avctx->draw_horiz_band(s->avctx, &s->picture, offset, y, 3, h); diff --git a/libavcodec/imc.c b/libavcodec/imc.c index bb6f095875..3919797038 100644 --- a/libavcodec/imc.c +++ b/libavcodec/imc.c @@ -51,6 +51,8 @@ #define COEFFS 256 typedef struct { + AVFrame frame; + float old_floor[BANDS]; float flcoeffs1[BANDS]; float flcoeffs2[BANDS]; @@ -168,6 +170,10 @@ static av_cold int imc_decode_init(AVCodecContext * avctx) dsputil_init(&q->dsp, avctx); avctx->sample_fmt = AV_SAMPLE_FMT_FLT; avctx->channel_layout = AV_CH_LAYOUT_MONO; + + avcodec_get_frame_defaults(&q->frame); + avctx->coded_frame = &q->frame; + return 0; } @@ -649,9 +655,8 @@ static int imc_get_coeffs (IMCContext* q) { return 0; } -static int imc_decode_frame(AVCodecContext * avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int imc_decode_frame(AVCodecContext * avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; @@ -659,7 +664,7 @@ static int imc_decode_frame(AVCodecContext * avctx, IMCContext *q = avctx->priv_data; int stream_format_code; - int imc_hdr, i, j, out_size, ret; + int imc_hdr, i, j, ret; int flag; int bits, summer; int counter, bitscount; @@ -670,15 +675,16 @@ static int imc_decode_frame(AVCodecContext * avctx, return AVERROR_INVALIDDATA; } - 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); + /* get output buffer */ + q->frame.nb_samples = COEFFS; + if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + q->out_samples = (float *)q->frame.data[0]; 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); /* Check the frame header */ @@ -823,7 +829,8 @@ static int imc_decode_frame(AVCodecContext * avctx, imc_imdct256(q); - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = q->frame; return IMC_BLOCK_SIZE; } @@ -834,6 +841,7 @@ static av_cold int imc_decode_close(AVCodecContext * avctx) IMCContext *q = avctx->priv_data; ff_fft_end(&q->fft); + return 0; } @@ -846,5 +854,6 @@ AVCodec ff_imc_decoder = { .init = imc_decode_init, .close = imc_decode_close, .decode = imc_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("IMC (Intel Music Coder)"), }; diff --git a/libavcodec/internal.h b/libavcodec/internal.h index 694d344da1..352be67854 100644 --- a/libavcodec/internal.h +++ b/libavcodec/internal.h @@ -31,12 +31,15 @@ typedef struct InternalBuffer { int last_pic_num; - uint8_t *base[4]; - uint8_t *data[4]; - int linesize[4]; + uint8_t *base[AV_NUM_DATA_POINTERS]; + uint8_t *data[AV_NUM_DATA_POINTERS]; + int linesize[AV_NUM_DATA_POINTERS]; int width; int height; enum PixelFormat pix_fmt; + uint8_t **extended_data; + int audio_data_size; + int nb_channels; } InternalBuffer; typedef struct AVCodecInternal { diff --git a/libavcodec/libgsm.c b/libavcodec/libgsm.c index 95919a4c43..9ba9da9e3f 100644 --- a/libavcodec/libgsm.c +++ b/libavcodec/libgsm.c @@ -124,7 +124,14 @@ AVCodec ff_libgsm_ms_encoder = { .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"), }; +typedef struct LibGSMDecodeContext { + AVFrame frame; + struct gsm_state *state; +} LibGSMDecodeContext; + static av_cold int libgsm_decode_init(AVCodecContext *avctx) { + LibGSMDecodeContext *s = avctx->priv_data; + if (avctx->channels > 1) { av_log(avctx, AV_LOG_ERROR, "Mono required for GSM, got %d channels\n", avctx->channels); @@ -139,7 +146,7 @@ static av_cold int libgsm_decode_init(AVCodecContext *avctx) { avctx->sample_fmt = AV_SAMPLE_FMT_S16; - avctx->priv_data = gsm_create(); + s->state = gsm_create(); switch(avctx->codec_id) { case CODEC_ID_GSM: @@ -154,59 +161,72 @@ static av_cold int libgsm_decode_init(AVCodecContext *avctx) { } } + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } static av_cold int libgsm_decode_close(AVCodecContext *avctx) { - gsm_destroy(avctx->priv_data); - avctx->priv_data = NULL; + LibGSMDecodeContext *s = avctx->priv_data; + + gsm_destroy(s->state); + s->state = NULL; return 0; } -static int libgsm_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) { +static int libgsm_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ int i, ret; - struct gsm_state *s = avctx->priv_data; + LibGSMDecodeContext *s = avctx->priv_data; uint8_t *buf = avpkt->data; int buf_size = avpkt->size; - 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); - } + int16_t *samples; if (buf_size < avctx->block_align) { av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); return AVERROR_INVALIDDATA; } + /* get output buffer */ + s->frame.nb_samples = avctx->frame_size; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + samples = (int16_t *)s->frame.data[0]; + for (i = 0; i < avctx->frame_size / GSM_FRAME_SIZE; i++) { - if ((ret = gsm_decode(s, buf, samples)) < 0) + if ((ret = gsm_decode(s->state, buf, samples)) < 0) return -1; buf += GSM_BLOCK_SIZE; samples += GSM_FRAME_SIZE; } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return avctx->block_align; } static void libgsm_flush(AVCodecContext *avctx) { - gsm_destroy(avctx->priv_data); - avctx->priv_data = gsm_create(); + LibGSMDecodeContext *s = avctx->priv_data; + + gsm_destroy(s->state); + s->state = gsm_create(); } AVCodec ff_libgsm_decoder = { .name = "libgsm", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_GSM, + .priv_data_size = sizeof(LibGSMDecodeContext), .init = libgsm_decode_init, .close = libgsm_decode_close, .decode = libgsm_decode_frame, .flush = libgsm_flush, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"), }; @@ -214,9 +234,11 @@ AVCodec ff_libgsm_ms_decoder = { .name = "libgsm_ms", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_GSM_MS, + .priv_data_size = sizeof(LibGSMDecodeContext), .init = libgsm_decode_init, .close = libgsm_decode_close, .decode = libgsm_decode_frame, .flush = libgsm_flush, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"), }; diff --git a/libavcodec/libopencore-amr.c b/libavcodec/libopencore-amr.c index e456c407a4..0de7130f19 100644 --- a/libavcodec/libopencore-amr.c +++ b/libavcodec/libopencore-amr.c @@ -79,6 +79,7 @@ static int get_bitrate_mode(int bitrate, void *log_ctx) typedef struct AMRContext { AVClass *av_class; + AVFrame frame; void *dec_state; void *enc_state; int enc_bitrate; @@ -112,6 +113,9 @@ static av_cold int amr_nb_decode_init(AVCodecContext *avctx) return AVERROR(ENOSYS); } + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -120,26 +124,28 @@ static av_cold int amr_nb_decode_close(AVCodecContext *avctx) AMRContext *s = avctx->priv_data; Decoder_Interface_exit(s->dec_state); + return 0; } static int amr_nb_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; AMRContext *s = avctx->priv_data; static const uint8_t block_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 }; enum Mode dec_mode; - int packet_size, out_size; + int packet_size, ret; av_dlog(avctx, "amr_decode_frame buf=%p buf_size=%d frame_count=%d!!\n", buf, buf_size, avctx->frame_number); - out_size = 160 * 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); + /* get output buffer */ + s->frame.nb_samples = 160; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } dec_mode = (buf[0] >> 3) & 0x000F; @@ -154,8 +160,10 @@ static int amr_nb_decode_frame(AVCodecContext *avctx, void *data, av_dlog(avctx, "packet_size=%d buf= 0x%X %X %X %X\n", packet_size, buf[0], buf[1], buf[2], buf[3]); /* call decoder */ - Decoder_Interface_Decode(s->dec_state, buf, data, 0); - *data_size = out_size; + Decoder_Interface_Decode(s->dec_state, buf, (short *)s->frame.data[0], 0); + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; return packet_size; } @@ -168,6 +176,7 @@ AVCodec ff_libopencore_amrnb_decoder = { .init = amr_nb_decode_init, .close = amr_nb_decode_close, .decode = amr_nb_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("OpenCORE Adaptive Multi-Rate (AMR) Narrow-Band"), }; @@ -251,6 +260,7 @@ AVCodec ff_libopencore_amrnb_encoder = { #include <opencore-amrwb/if_rom.h> typedef struct AMRWBContext { + AVFrame frame; void *state; } AMRWBContext; @@ -267,23 +277,27 @@ static av_cold int amr_wb_decode_init(AVCodecContext *avctx) return AVERROR(ENOSYS); } + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } static int amr_wb_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; AMRWBContext *s = avctx->priv_data; - int mode; - int packet_size, out_size; + int mode, ret; + int packet_size; static const uint8_t block_size[16] = {18, 24, 33, 37, 41, 47, 51, 59, 61, 6, 6, 0, 0, 0, 1, 1}; - out_size = 320 * 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); + /* get output buffer */ + s->frame.nb_samples = 320; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } mode = (buf[0] >> 3) & 0x000F; @@ -295,8 +309,11 @@ static int amr_wb_decode_frame(AVCodecContext *avctx, void *data, return AVERROR_INVALIDDATA; } - D_IF_decode(s->state, buf, data, _good_frame); - *data_size = out_size; + D_IF_decode(s->state, buf, (short *)s->frame.data[0], _good_frame); + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return packet_size; } @@ -316,6 +333,7 @@ AVCodec ff_libopencore_amrwb_decoder = { .init = amr_wb_decode_init, .close = amr_wb_decode_close, .decode = amr_wb_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("OpenCORE Adaptive Multi-Rate (AMR) Wide-Band"), }; diff --git a/libavcodec/libspeexdec.c b/libavcodec/libspeexdec.c index 91f190525d..fdc39532b7 100644 --- a/libavcodec/libspeexdec.c +++ b/libavcodec/libspeexdec.c @@ -25,6 +25,7 @@ #include "avcodec.h" typedef struct { + AVFrame frame; SpeexBits bits; SpeexStereoState stereo; void *dec_state; @@ -89,26 +90,29 @@ static av_cold int libspeex_decode_init(AVCodecContext *avctx) s->stereo = (SpeexStereoState)SPEEX_STEREO_STATE_INIT; speex_decoder_ctl(s->dec_state, SPEEX_SET_HANDLER, &callback); } + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } -static int libspeex_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int libspeex_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { uint8_t *buf = avpkt->data; int buf_size = avpkt->size; LibSpeexContext *s = avctx->priv_data; - int16_t *output = data; - int out_size, ret, consumed = 0; - - /* check output buffer size */ - out_size = s->frame_size * avctx->channels * - 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); + int16_t *output; + int ret, consumed = 0; + + /* get output buffer */ + s->frame.nb_samples = s->frame_size; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + output = (int16_t *)s->frame.data[0]; /* if there is not enough data left for the smallest possible frame, reset the libspeex buffer using the current packet, otherwise ignore @@ -116,7 +120,7 @@ static int libspeex_decode_frame(AVCodecContext *avctx, if (speex_bits_remaining(&s->bits) < 43) { /* check for flush packet */ if (!buf || !buf_size) { - *data_size = 0; + *got_frame_ptr = 0; return buf_size; } /* set new buffer */ @@ -133,7 +137,9 @@ static int libspeex_decode_frame(AVCodecContext *avctx, if (avctx->channels == 2) speex_decode_stereo_int(output, s->frame_size, &s->stereo); - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return consumed; } @@ -163,6 +169,6 @@ AVCodec ff_libspeex_decoder = { .close = libspeex_decode_close, .decode = libspeex_decode_frame, .flush = libspeex_decode_flush, - .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY, + .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("libspeex Speex"), }; diff --git a/libavcodec/mace.c b/libavcodec/mace.c index 980ebb9027..ffa11ad80d 100644 --- a/libavcodec/mace.c +++ b/libavcodec/mace.c @@ -153,6 +153,7 @@ typedef struct ChannelData { } ChannelData; typedef struct MACEContext { + AVFrame frame; ChannelData chd[2]; } MACEContext; @@ -228,30 +229,35 @@ static void chomp6(ChannelData *chd, int16_t *output, uint8_t val, static av_cold int mace_decode_init(AVCodecContext * avctx) { + MACEContext *ctx = avctx->priv_data; + if (avctx->channels > 2) return -1; avctx->sample_fmt = AV_SAMPLE_FMT_S16; + + avcodec_get_frame_defaults(&ctx->frame); + avctx->coded_frame = &ctx->frame; + return 0; } -static int mace_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int mace_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; - int16_t *samples = data; + int16_t *samples; MACEContext *ctx = avctx->priv_data; - int i, j, k, l; - int out_size; + int i, j, k, l, ret; int is_mace3 = (avctx->codec_id == CODEC_ID_MACE3); - out_size = 3 * (buf_size << (1 - is_mace3)) * - 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); + /* get output buffer */ + ctx->frame.nb_samples = 3 * (buf_size << (1 - is_mace3)) / avctx->channels; + if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (int16_t *)ctx->frame.data[0]; for(i = 0; i < avctx->channels; i++) { int16_t *output = samples + i; @@ -277,7 +283,8 @@ static int mace_decode_frame(AVCodecContext *avctx, } } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = ctx->frame; return buf_size; } @@ -289,6 +296,7 @@ AVCodec ff_mace3_decoder = { .priv_data_size = sizeof(MACEContext), .init = mace_decode_init, .decode = mace_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("MACE (Macintosh Audio Compression/Expansion) 3:1"), }; @@ -299,6 +307,7 @@ AVCodec ff_mace6_decoder = { .priv_data_size = sizeof(MACEContext), .init = mace_decode_init, .decode = mace_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("MACE (Macintosh Audio Compression/Expansion) 6:1"), }; diff --git a/libavcodec/mlpdec.c b/libavcodec/mlpdec.c index 0396423b92..4b439ddb67 100644 --- a/libavcodec/mlpdec.c +++ b/libavcodec/mlpdec.c @@ -120,6 +120,7 @@ typedef struct SubStream { typedef struct MLPDecodeContext { AVCodecContext *avctx; + AVFrame frame; //! Current access unit being read has a major sync. int is_major_sync_unit; @@ -242,6 +243,9 @@ static av_cold int mlp_decode_init(AVCodecContext *avctx) m->substream[substr].lossless_check_data = 0xffffffff; dsputil_init(&m->dsp, avctx); + avcodec_get_frame_defaults(&m->frame); + avctx->coded_frame = &m->frame; + return 0; } @@ -946,13 +950,14 @@ static void rematrix_channels(MLPDecodeContext *m, unsigned int substr) /** Write the audio data into the output buffer. */ static int output_data(MLPDecodeContext *m, unsigned int substr, - uint8_t *data, unsigned int *data_size) + void *data, int *got_frame_ptr) { + AVCodecContext *avctx = m->avctx; SubStream *s = &m->substream[substr]; unsigned int i, out_ch = 0; - int out_size; - int32_t *data_32 = (int32_t*) data; - int16_t *data_16 = (int16_t*) data; + int32_t *data_32; + int16_t *data_16; + int ret; int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32); if (m->avctx->channels != s->max_matrix_channel + 1) { @@ -960,11 +965,14 @@ static int output_data(MLPDecodeContext *m, unsigned int substr, return AVERROR_INVALIDDATA; } - out_size = s->blockpos * m->avctx->channels * - av_get_bytes_per_sample(m->avctx->sample_fmt); - - if (*data_size < out_size) - return AVERROR(EINVAL); + /* get output buffer */ + m->frame.nb_samples = s->blockpos; + if ((ret = avctx->get_buffer(avctx, &m->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + data_32 = (int32_t *)m->frame.data[0]; + data_16 = (int16_t *)m->frame.data[0]; for (i = 0; i < s->blockpos; i++) { for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) { @@ -977,7 +985,8 @@ static int output_data(MLPDecodeContext *m, unsigned int substr, } } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = m->frame; return 0; } @@ -986,8 +995,8 @@ static int output_data(MLPDecodeContext *m, unsigned int substr, * @return negative on error, 0 if not enough data is present in the input stream, * otherwise the number of bytes consumed. */ -static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size, - AVPacket *avpkt) +static int read_access_unit(AVCodecContext *avctx, void* data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; @@ -1023,7 +1032,7 @@ static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size, if (!m->params_valid) { av_log(m->avctx, AV_LOG_WARNING, "Stream parameters not seen; skipping frame.\n"); - *data_size = 0; + *got_frame_ptr = 0; return length; } @@ -1168,7 +1177,7 @@ next_substr: rematrix_channels(m, m->max_decoded_substream); - if ((ret = output_data(m, m->max_decoded_substream, data, data_size)) < 0) + if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0) return ret; return length; @@ -1189,6 +1198,7 @@ AVCodec ff_mlp_decoder = { .priv_data_size = sizeof(MLPDecodeContext), .init = mlp_decode_init, .decode = read_access_unit, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"), }; @@ -1200,6 +1210,7 @@ AVCodec ff_truehd_decoder = { .priv_data_size = sizeof(MLPDecodeContext), .init = mlp_decode_init, .decode = read_access_unit, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("TrueHD"), }; #endif /* CONFIG_TRUEHD_DECODER */ diff --git a/libavcodec/mpc.h b/libavcodec/mpc.h index cd5769234c..8b4deef689 100644 --- a/libavcodec/mpc.h +++ b/libavcodec/mpc.h @@ -50,6 +50,7 @@ typedef struct { }Band; typedef struct { + AVFrame frame; DSPContext dsp; MPADSPContext mpadsp; GetBitContext gb; diff --git a/libavcodec/mpc7.c b/libavcodec/mpc7.c index 1e76ddf56a..7393250971 100644 --- a/libavcodec/mpc7.c +++ b/libavcodec/mpc7.c @@ -136,6 +136,10 @@ static av_cold int mpc7_decode_init(AVCodecContext * avctx) } } vlc_initialized = 1; + + avcodec_get_frame_defaults(&c->frame); + avctx->coded_frame = &c->frame; + return 0; } @@ -192,9 +196,8 @@ static int get_scale_idx(GetBitContext *gb, int ref) return ref + t; } -static int mpc7_decode_frame(AVCodecContext * avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int mpc7_decode_frame(AVCodecContext * avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; @@ -204,7 +207,7 @@ static int mpc7_decode_frame(AVCodecContext * avctx, int i, ch; int mb = -1; Band *bands = c->bands; - int off, out_size; + int off, ret; int bits_used, bits_avail; memset(bands, 0, sizeof(*bands) * (c->maxbands + 1)); @@ -213,10 +216,11 @@ static int mpc7_decode_frame(AVCodecContext * avctx, return AVERROR(EINVAL); } - out_size = (buf[1] ? c->lastframelen : MPC_FRAME_SIZE) * 4; - if (*data_size < out_size) { - av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); - return AVERROR(EINVAL); + /* get output buffer */ + c->frame.nb_samples = buf[1] ? c->lastframelen : MPC_FRAME_SIZE; + if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } bits = av_malloc(((buf_size - 1) & ~3) + FF_INPUT_BUFFER_PADDING_SIZE); @@ -276,7 +280,7 @@ static int mpc7_decode_frame(AVCodecContext * avctx, for(ch = 0; ch < 2; ch++) idx_to_quant(c, &gb, bands[i].res[ch], c->Q[ch] + off); - ff_mpc_dequantize_and_synth(c, mb, data, 2); + ff_mpc_dequantize_and_synth(c, mb, c->frame.data[0], 2); av_free(bits); @@ -288,10 +292,12 @@ static int mpc7_decode_frame(AVCodecContext * avctx, } if(c->frames_to_skip){ c->frames_to_skip--; - *data_size = 0; + *got_frame_ptr = 0; return buf_size; } - *data_size = out_size; + + *got_frame_ptr = 1; + *(AVFrame *)data = c->frame; return buf_size; } @@ -312,5 +318,6 @@ AVCodec ff_mpc7_decoder = { .init = mpc7_decode_init, .decode = mpc7_decode_frame, .flush = mpc7_decode_flush, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"), }; diff --git a/libavcodec/mpc8.c b/libavcodec/mpc8.c index 2f6bde3231..a4750ad961 100644 --- a/libavcodec/mpc8.c +++ b/libavcodec/mpc8.c @@ -230,12 +230,15 @@ static av_cold int mpc8_decode_init(AVCodecContext * avctx) &mpc8_q8_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC); } vlc_initialized = 1; + + avcodec_get_frame_defaults(&c->frame); + avctx->coded_frame = &c->frame; + return 0; } -static int mpc8_decode_frame(AVCodecContext * avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int mpc8_decode_frame(AVCodecContext * avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; @@ -243,14 +246,15 @@ static int mpc8_decode_frame(AVCodecContext * avctx, GetBitContext gb2, *gb = &gb2; int i, j, k, ch, cnt, res, t; Band *bands = c->bands; - int off, out_size; + int off; int maxband, keyframe; int last[2]; - out_size = MPC_FRAME_SIZE * 2 * avctx->channels; - if (*data_size < out_size) { - av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); - return AVERROR(EINVAL); + /* get output buffer */ + c->frame.nb_samples = MPC_FRAME_SIZE; + if ((res = avctx->get_buffer(avctx, &c->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return res; } keyframe = c->cur_frame == 0; @@ -403,14 +407,16 @@ static int mpc8_decode_frame(AVCodecContext * avctx, } } - ff_mpc_dequantize_and_synth(c, maxband, data, avctx->channels); + ff_mpc_dequantize_and_synth(c, maxband, c->frame.data[0], avctx->channels); c->cur_frame++; c->last_bits_used = get_bits_count(gb); if(c->cur_frame >= c->frames) c->cur_frame = 0; - *data_size = out_size; + + *got_frame_ptr = 1; + *(AVFrame *)data = c->frame; return c->cur_frame ? c->last_bits_used >> 3 : buf_size; } @@ -422,5 +428,6 @@ AVCodec ff_mpc8_decoder = { .priv_data_size = sizeof(MPCContext), .init = mpc8_decode_init, .decode = mpc8_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Musepack SV8"), }; diff --git a/libavcodec/mpeg4audio.c b/libavcodec/mpeg4audio.c index ac546ba137..67445935c4 100644 --- a/libavcodec/mpeg4audio.c +++ b/libavcodec/mpeg4audio.c @@ -76,12 +76,13 @@ static inline int get_sample_rate(GetBitContext *gb, int *index) avpriv_mpeg4audio_sample_rates[*index]; } -int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf, int buf_size) +int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf, + int bit_size, int sync_extension) { GetBitContext gb; int specific_config_bitindex; - init_get_bits(&gb, buf, buf_size*8); + init_get_bits(&gb, buf, bit_size); c->object_type = get_object_type(&gb); c->sample_rate = get_sample_rate(&gb, &c->sampling_index); c->chan_config = get_bits(&gb, 4); @@ -117,7 +118,7 @@ int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf, int bu return -1; } - if (c->ext_object_type != AOT_SBR) { + if (c->ext_object_type != AOT_SBR && sync_extension) { while (get_bits_left(&gb) > 15) { if (show_bits(&gb, 11) == 0x2b7) { // sync extension get_bits(&gb, 11); diff --git a/libavcodec/mpeg4audio.h b/libavcodec/mpeg4audio.h index 3d1f7e0c4c..21000a9acc 100644 --- a/libavcodec/mpeg4audio.h +++ b/libavcodec/mpeg4audio.h @@ -42,14 +42,17 @@ typedef struct { extern const int avpriv_mpeg4audio_sample_rates[16]; extern const uint8_t ff_mpeg4audio_channels[8]; + /** * Parse MPEG-4 systems extradata to retrieve audio configuration. * @param[in] c MPEG4AudioConfig structure to fill. * @param[in] buf Extradata from container. - * @param[in] buf_size Extradata size. + * @param[in] bit_size Extradata size in bits. + * @param[in] sync_extension look for a sync extension after config if true. * @return On error -1 is returned, on success AudioSpecificConfig bit index in extradata. */ -int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf, int buf_size); +int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf, + int bit_size, int sync_extension); enum AudioObjectType { AOT_NULL, diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c index 7dded97cf4..5cc3c74339 100644 --- a/libavcodec/mpegaudiodec.c +++ b/libavcodec/mpegaudiodec.c @@ -79,6 +79,7 @@ typedef struct MPADecodeContext { int err_recognition; AVCodecContext* avctx; MPADSPContext mpadsp; + AVFrame frame; } MPADecodeContext; #if CONFIG_FLOAT @@ -479,6 +480,10 @@ static av_cold int decode_init(AVCodecContext * avctx) if (avctx->codec_id == CODEC_ID_MP3ADU) s->adu_mode = 1; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -1581,7 +1586,7 @@ static int mp_decode_layer3(MPADecodeContext *s) static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples, const uint8_t *buf, int buf_size) { - int i, nb_frames, ch; + int i, nb_frames, ch, ret; OUT_INT *samples_ptr; init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8); @@ -1629,8 +1634,16 @@ static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples, 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; + } - break; + /* get output buffer */ + if (!samples) { + s->frame.nb_samples = s->avctx->frame_size; + if ((ret = s->avctx->get_buffer(s->avctx, &s->frame)) < 0) { + av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + samples = (OUT_INT *)s->frame.data[0]; } /* apply the synthesis filter */ @@ -1650,7 +1663,7 @@ static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples, 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 *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; @@ -1658,7 +1671,6 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size, MPADecodeContext *s = avctx->priv_data; uint32_t header; int out_size; - OUT_INT *out_samples = data; if (buf_size < HEADER_SIZE) return AVERROR_INVALIDDATA; @@ -1681,10 +1693,6 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size, 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); - *data_size = 0; - if (s->frame_size <= 0 || s->frame_size > buf_size) { av_log(avctx, AV_LOG_ERROR, "incomplete frame\n"); return AVERROR_INVALIDDATA; @@ -1693,9 +1701,10 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size, buf_size= s->frame_size; } - out_size = mp_decode_frame(s, out_samples, buf, buf_size); + out_size = mp_decode_frame(s, NULL, buf, buf_size); if (out_size >= 0) { - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; avctx->sample_rate = s->sample_rate; //FIXME maybe move the other codec info stuff from above here too } else { @@ -1704,6 +1713,7 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size, If there is more data in the packet, just consume the bad frame instead of returning an error, which would discard the whole packet. */ + *got_frame_ptr = 0; if (buf_size == avpkt->size) return out_size; } @@ -1719,15 +1729,14 @@ static void flush(AVCodecContext *avctx) } #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 *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; MPADecodeContext *s = avctx->priv_data; uint32_t header; int len, out_size; - OUT_INT *out_samples = data; len = buf_size; @@ -1757,9 +1766,6 @@ static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size, 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 FF_API_PARSE_FRAME @@ -1767,9 +1773,11 @@ static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size, out_size = buf_size; else #endif - out_size = mp_decode_frame(s, out_samples, buf, buf_size); + out_size = mp_decode_frame(s, NULL, buf, buf_size); + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; - *data_size = out_size; return buf_size; } #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */ @@ -1780,6 +1788,7 @@ static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size, * Context for MP3On4 decoder */ typedef struct MP3On4DecodeContext { + AVFrame *frame; 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 @@ -1843,7 +1852,8 @@ static int decode_init_mp3on4(AVCodecContext * avctx) return AVERROR_INVALIDDATA; } - avpriv_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size); + avpriv_mpeg4audio_get_config(&cfg, avctx->extradata, + avctx->extradata_size * 8, 1); if (!cfg.chan_config || cfg.chan_config > 7) { av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n"); return AVERROR_INVALIDDATA; @@ -1870,6 +1880,7 @@ static int decode_init_mp3on4(AVCodecContext * avctx) // Put decoder context in place to make init_decode() happy avctx->priv_data = s->mp3decctx[0]; decode_init(avctx); + s->frame = avctx->coded_frame; // Restore mp3on4 context pointer avctx->priv_data = s; s->mp3decctx[0]->adu_mode = 1; // Set adu mode @@ -1914,9 +1925,8 @@ static void flush_mp3on4(AVCodecContext *avctx) } -static int decode_frame_mp3on4(AVCodecContext * avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int decode_frame_mp3on4(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; @@ -1924,14 +1934,17 @@ static int decode_frame_mp3on4(AVCodecContext * avctx, MPADecodeContext *m; int fsize, len = buf_size, out_size = 0; uint32_t header; - OUT_INT *out_samples = data; + OUT_INT *out_samples; OUT_INT *outptr, *bp; - int fr, j, n, ch; + int fr, j, n, ch, ret; - if (*data_size < MPA_FRAME_SIZE * avctx->channels * sizeof(OUT_INT)) { - av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n"); - return AVERROR(EINVAL); + /* get output buffer */ + s->frame->nb_samples = MPA_FRAME_SIZE; + if ((ret = avctx->get_buffer(avctx, s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + out_samples = (OUT_INT *)s->frame->data[0]; // Discard too short frames if (buf_size < HEADER_SIZE) @@ -1990,7 +2003,10 @@ static int decode_frame_mp3on4(AVCodecContext * avctx, /* update codec info */ avctx->sample_rate = s->mp3decctx[0]->sample_rate; - *data_size = out_size; + s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT)); + *got_frame_ptr = 1; + *(AVFrame *)data = *s->frame; + return buf_size; } #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */ @@ -2005,7 +2021,9 @@ AVCodec ff_mp1_decoder = { .init = decode_init, .decode = decode_frame, #if FF_API_PARSE_FRAME - .capabilities = CODEC_CAP_PARSE_ONLY, + .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1, +#else + .capabilities = CODEC_CAP_DR1, #endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"), @@ -2020,7 +2038,9 @@ AVCodec ff_mp2_decoder = { .init = decode_init, .decode = decode_frame, #if FF_API_PARSE_FRAME - .capabilities = CODEC_CAP_PARSE_ONLY, + .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1, +#else + .capabilities = CODEC_CAP_DR1, #endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), @@ -2035,7 +2055,9 @@ AVCodec ff_mp3_decoder = { .init = decode_init, .decode = decode_frame, #if FF_API_PARSE_FRAME - .capabilities = CODEC_CAP_PARSE_ONLY, + .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1, +#else + .capabilities = CODEC_CAP_DR1, #endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"), @@ -2050,7 +2072,9 @@ AVCodec ff_mp3adu_decoder = { .init = decode_init, .decode = decode_frame_adu, #if FF_API_PARSE_FRAME - .capabilities = CODEC_CAP_PARSE_ONLY, + .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1, +#else + .capabilities = CODEC_CAP_DR1, #endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"), @@ -2065,6 +2089,7 @@ AVCodec ff_mp3on4_decoder = { .init = decode_init_mp3on4, .close = decode_close_mp3on4, .decode = decode_frame_mp3on4, + .capabilities = CODEC_CAP_DR1, .flush = flush_mp3on4, .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"), }; diff --git a/libavcodec/mpegaudiodec_float.c b/libavcodec/mpegaudiodec_float.c index 4482168a3e..7b6dcd96a1 100644 --- a/libavcodec/mpegaudiodec_float.c +++ b/libavcodec/mpegaudiodec_float.c @@ -31,7 +31,9 @@ AVCodec ff_mp1float_decoder = { .init = decode_init, .decode = decode_frame, #if FF_API_PARSE_FRAME - .capabilities = CODEC_CAP_PARSE_ONLY, + .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1, +#else + .capabilities = CODEC_CAP_DR1, #endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"), @@ -46,7 +48,9 @@ AVCodec ff_mp2float_decoder = { .init = decode_init, .decode = decode_frame, #if FF_API_PARSE_FRAME - .capabilities = CODEC_CAP_PARSE_ONLY, + .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1, +#else + .capabilities = CODEC_CAP_DR1, #endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), @@ -61,7 +65,9 @@ AVCodec ff_mp3float_decoder = { .init = decode_init, .decode = decode_frame, #if FF_API_PARSE_FRAME - .capabilities = CODEC_CAP_PARSE_ONLY, + .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1, +#else + .capabilities = CODEC_CAP_DR1, #endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"), @@ -76,7 +82,9 @@ AVCodec ff_mp3adufloat_decoder = { .init = decode_init, .decode = decode_frame_adu, #if FF_API_PARSE_FRAME - .capabilities = CODEC_CAP_PARSE_ONLY, + .capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1, +#else + .capabilities = CODEC_CAP_DR1, #endif .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"), @@ -91,6 +99,7 @@ AVCodec ff_mp3on4float_decoder = { .init = decode_init_mp3on4, .close = decode_close_mp3on4, .decode = decode_frame_mp3on4, + .capabilities = CODEC_CAP_DR1, .flush = flush_mp3on4, .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"), }; diff --git a/libavcodec/mpegvideo.c b/libavcodec/mpegvideo.c index 6893a3a8cd..1d3bb0cdf2 100644 --- a/libavcodec/mpegvideo.c +++ b/libavcodec/mpegvideo.c @@ -2351,7 +2351,8 @@ void ff_draw_horiz_band(MpegEncContext *s, int y, int h){ if (s->avctx->draw_horiz_band) { AVFrame *src; - int offset[4]; + int offset[AV_NUM_DATA_POINTERS]; + int i; if(s->pict_type==AV_PICTURE_TYPE_B || s->low_delay || (s->avctx->slice_flags&SLICE_FLAG_CODED_ORDER)) src= (AVFrame*)s->current_picture_ptr; @@ -2361,15 +2362,14 @@ void ff_draw_horiz_band(MpegEncContext *s, int y, int h){ return; if(s->pict_type==AV_PICTURE_TYPE_B && s->picture_structure == PICT_FRAME && s->out_format != FMT_H264){ - offset[0]= - offset[1]= - offset[2]= - offset[3]= 0; + for (i = 0; i < AV_NUM_DATA_POINTERS; i++) + offset[i] = 0; }else{ offset[0]= y * s->linesize; offset[1]= offset[2]= (y >> s->chroma_y_shift) * s->uvlinesize; - offset[3]= 0; + for (i = 3; i < AV_NUM_DATA_POINTERS; i++) + offset[i] = 0; } emms_c(); diff --git a/libavcodec/nellymoserdec.c b/libavcodec/nellymoserdec.c index cf73121ef8..3386b8241e 100644 --- a/libavcodec/nellymoserdec.c +++ b/libavcodec/nellymoserdec.c @@ -47,6 +47,7 @@ typedef struct NellyMoserDecodeContext { AVCodecContext* avctx; + AVFrame frame; float *float_buf; DECLARE_ALIGNED(16, float, state)[NELLY_BUF_LEN]; AVLFG random_state; @@ -142,33 +143,31 @@ static av_cold int decode_init(AVCodecContext * avctx) { ff_init_ff_sine_windows(7); avctx->channel_layout = AV_CH_LAYOUT_MONO; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } -static int decode_tag(AVCodecContext * avctx, - void *data, int *data_size, - AVPacket *avpkt) { +static int decode_tag(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ const uint8_t *buf = avpkt->data; const uint8_t *side=av_packet_get_side_data(avpkt, 'F', NULL); int buf_size = avpkt->size; NellyMoserDecodeContext *s = avctx->priv_data; - int data_max = *data_size; - int blocks, i, block_size; - int16_t *samples_s16 = data; - float *samples_flt = data; - *data_size = 0; + int blocks, i, ret; + int16_t *samples_s16; + float *samples_flt; - block_size = NELLY_SAMPLES * av_get_bytes_per_sample(avctx->sample_fmt); 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); @@ -183,6 +182,15 @@ static int decode_tag(AVCodecContext * avctx, if(side && blocks>1 && avctx->sample_rate%11025==0 && (1<<((side[0]>>2)&3)) == blocks) avctx->sample_rate= 11025*(blocks/2); + /* get output buffer */ + s->frame.nb_samples = NELLY_SAMPLES * blocks; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + samples_s16 = (int16_t *)s->frame.data[0]; + samples_flt = (float *)s->frame.data[0]; + for (i=0 ; i<blocks ; i++) { if (avctx->sample_fmt == SAMPLE_FMT_FLT) { nelly_decode_block(s, buf, samples_flt); @@ -194,7 +202,9 @@ static int decode_tag(AVCodecContext * avctx, } buf += NELLY_BLOCK_LEN; } - *data_size = blocks * block_size; + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; return buf_size; } @@ -204,6 +214,7 @@ static av_cold int decode_end(AVCodecContext * avctx) { av_freep(&s->float_buf); ff_mdct_end(&s->imdct_ctx); + return 0; } @@ -215,6 +226,7 @@ AVCodec ff_nellymoser_decoder = { .init = decode_init, .close = decode_end, .decode = decode_tag, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Nellymoser Asao"), .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, diff --git a/libavcodec/pcm.c b/libavcodec/pcm.c index 92519c0534..66b095370c 100644 --- a/libavcodec/pcm.c +++ b/libavcodec/pcm.c @@ -192,6 +192,7 @@ static int pcm_encode_frame(AVCodecContext *avctx, } typedef struct PCMDecode { + AVFrame frame; short table[256]; } PCMDecode; @@ -223,6 +224,9 @@ static av_cold int pcm_decode_init(AVCodecContext * avctx) if (avctx->sample_fmt == AV_SAMPLE_FMT_S32) avctx->bits_per_raw_sample = av_get_bits_per_sample(avctx->codec->id); + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -243,22 +247,20 @@ static av_cold int pcm_decode_init(AVCodecContext * avctx) dst += size / 8; \ } -static int pcm_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int pcm_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *src = avpkt->data; int buf_size = avpkt->size; PCMDecode *s = avctx->priv_data; - int sample_size, c, n, out_size; + int sample_size, c, n, ret, samples_per_block; uint8_t *samples; int32_t *dst_int32_t; - samples = data; - sample_size = av_get_bits_per_sample(avctx->codec_id)/8; /* av_get_bits_per_sample returns 0 for CODEC_ID_PCM_DVD */ + samples_per_block = 1; if (CODEC_ID_PCM_DVD == avctx->codec_id) { if (avctx->bits_per_coded_sample != 20 && avctx->bits_per_coded_sample != 24) { @@ -268,10 +270,13 @@ static int pcm_decode_frame(AVCodecContext *avctx, return AVERROR(EINVAL); } /* 2 samples are interleaved per block in PCM_DVD */ + samples_per_block = 2; sample_size = avctx->bits_per_coded_sample * 2 / 8; - } else if (avctx->codec_id == CODEC_ID_PCM_LXF) + } else if (avctx->codec_id == CODEC_ID_PCM_LXF) { /* we process 40-bit blocks per channel for LXF */ + samples_per_block = 2; sample_size = 5; + } if (sample_size == 0) { av_log(avctx, AV_LOG_ERROR, "Invalid sample_size\n"); @@ -290,14 +295,13 @@ static int pcm_decode_frame(AVCodecContext *avctx, n = buf_size/sample_size; - out_size = n * av_get_bytes_per_sample(avctx->sample_fmt); - if (avctx->codec_id == CODEC_ID_PCM_DVD || - avctx->codec_id == CODEC_ID_PCM_LXF) - out_size *= 2; - if (*data_size < out_size) { - av_log(avctx, AV_LOG_ERROR, "output buffer too small\n"); - return AVERROR(EINVAL); + /* get output buffer */ + s->frame.nb_samples = n * samples_per_block / avctx->channels; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = s->frame.data[0]; switch(avctx->codec->id) { case CODEC_ID_PCM_U32LE: @@ -403,7 +407,7 @@ static int pcm_decode_frame(AVCodecContext *avctx, case CODEC_ID_PCM_DVD: { const uint8_t *src8; - dst_int32_t = data; + dst_int32_t = (int32_t *)s->frame.data[0]; n /= avctx->channels; switch (avctx->bits_per_coded_sample) { case 20: @@ -435,7 +439,7 @@ static int pcm_decode_frame(AVCodecContext *avctx, { int i; const uint8_t *src8; - dst_int32_t = data; + dst_int32_t = (int32_t *)s->frame.data[0]; n /= avctx->channels; //unpack and de-planerize for (i = 0; i < n; i++) { @@ -456,7 +460,10 @@ static int pcm_decode_frame(AVCodecContext *avctx, default: return -1; } - *data_size = out_size; + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return buf_size; } @@ -485,6 +492,7 @@ AVCodec ff_ ## name_ ## _decoder = { \ .priv_data_size = sizeof(PCMDecode), \ .init = pcm_decode_init, \ .decode = pcm_decode_frame, \ + .capabilities = CODEC_CAP_DR1, \ .sample_fmts = (const enum AVSampleFormat[]){sample_fmt_,AV_SAMPLE_FMT_NONE}, \ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ } diff --git a/libavcodec/qcelpdec.c b/libavcodec/qcelpdec.c index 76480f0d85..47b1f68fe5 100644 --- a/libavcodec/qcelpdec.c +++ b/libavcodec/qcelpdec.c @@ -56,6 +56,7 @@ typedef enum typedef struct { + AVFrame avframe; GetBitContext gb; qcelp_packet_rate bitrate; QCELPFrame frame; /**< unpacked data frame */ @@ -97,6 +98,9 @@ static av_cold int qcelp_decode_init(AVCodecContext *avctx) for(i=0; i<10; i++) q->prev_lspf[i] = (i+1)/11.; + avcodec_get_frame_defaults(&q->avframe); + avctx->coded_frame = &q->avframe; + return 0; } @@ -682,23 +686,25 @@ static void postfilter(QCELPContext *q, float *samples, float *lpc) 160, 0.9375, &q->postfilter_agc_mem); } -static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size, - AVPacket *avpkt) +static int qcelp_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; QCELPContext *q = avctx->priv_data; - float *outbuffer = data; - int i, out_size; + float *outbuffer; + int i, ret; float quantized_lspf[10], lpc[10]; float gain[16]; float *formant_mem; - out_size = 160 * 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); + /* get output buffer */ + q->avframe.nb_samples = 160; + if ((ret = avctx->get_buffer(avctx, &q->avframe)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + outbuffer = (float *)q->avframe.data[0]; if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) { warn_insufficient_frame_quality(avctx, "bitrate cannot be determined."); @@ -783,7 +789,8 @@ erasure: memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf)); q->prev_bitrate = q->bitrate; - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = q->avframe; return buf_size; } @@ -795,6 +802,7 @@ AVCodec ff_qcelp_decoder = .id = CODEC_ID_QCELP, .init = qcelp_decode_init, .decode = qcelp_decode_frame, + .capabilities = CODEC_CAP_DR1, .priv_data_size = sizeof(QCELPContext), .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"), }; diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index e2eb0eacb8..cc71825838 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -130,6 +130,8 @@ typedef struct { * QDM2 decoder context */ typedef struct { + AVFrame frame; + /// Parameters from codec header, do not change during playback int nb_channels; ///< number of channels int channels; ///< number of channels @@ -1876,6 +1878,9 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) avctx->sample_fmt = AV_SAMPLE_FMT_S16; + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + // dump_context(s); return 0; } @@ -1956,30 +1961,27 @@ static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) } -static int qdm2_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int qdm2_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; QDM2Context *s = avctx->priv_data; - int16_t *out = data; - int i, out_size; + int16_t *out; + int i, ret; if(!buf) return 0; if(buf_size < s->checksum_size) return -1; - out_size = 16 * s->channels * s->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); + /* get output buffer */ + s->frame.nb_samples = 16 * s->frame_size; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } - - av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", - buf_size, buf, s->checksum_size, data, *data_size); + out = (int16_t *)s->frame.data[0]; for (i = 0; i < 16; i++) { if (qdm2_decode(s, buf, out) < 0) @@ -1987,7 +1989,8 @@ static int qdm2_decode_frame(AVCodecContext *avctx, out += s->channels * s->frame_size; } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; return s->checksum_size; } @@ -2001,5 +2004,6 @@ AVCodec ff_qdm2_decoder = .init = qdm2_decode_init, .close = qdm2_decode_close, .decode = qdm2_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), }; diff --git a/libavcodec/ra144.h b/libavcodec/ra144.h index 722b42e68a..9665534f7b 100644 --- a/libavcodec/ra144.h +++ b/libavcodec/ra144.h @@ -34,6 +34,7 @@ typedef struct { AVCodecContext *avctx; + AVFrame frame; LPCContext lpc_ctx; unsigned int old_energy; ///< previous frame energy diff --git a/libavcodec/ra144dec.c b/libavcodec/ra144dec.c index 2e57e5054c..428f788733 100644 --- a/libavcodec/ra144dec.c +++ b/libavcodec/ra144dec.c @@ -38,6 +38,10 @@ static av_cold int ra144_decode_init(AVCodecContext * avctx) ractx->lpc_coef[1] = ractx->lpc_tables[1]; avctx->sample_fmt = AV_SAMPLE_FMT_S16; + + avcodec_get_frame_defaults(&ractx->frame); + avctx->coded_frame = &ractx->frame; + return 0; } @@ -54,8 +58,8 @@ static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs, } /** Uncompress one block (20 bytes -> 160*2 bytes). */ -static int ra144_decode_frame(AVCodecContext * avctx, void *vdata, - int *data_size, AVPacket *avpkt) +static int ra144_decode_frame(AVCodecContext * avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; @@ -64,23 +68,25 @@ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata, uint16_t block_coefs[NBLOCKS][LPC_ORDER]; // LPC coefficients of each sub-block unsigned int lpc_refl[LPC_ORDER]; // LPC reflection coefficients of the frame int i, j; - int out_size; - int16_t *data = vdata; + int ret; + int16_t *samples; unsigned int energy; RA144Context *ractx = avctx->priv_data; GetBitContext gb; - out_size = NBLOCKS * BLOCKSIZE * 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); + /* get output buffer */ + ractx->frame.nb_samples = NBLOCKS * BLOCKSIZE; + if ((ret = avctx->get_buffer(avctx, &ractx->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (int16_t *)ractx->frame.data[0]; if(buf_size < FRAMESIZE) { av_log(avctx, AV_LOG_ERROR, "Frame too small (%d bytes). Truncated file?\n", buf_size); - *data_size = 0; + *got_frame_ptr = 0; return buf_size; } init_get_bits(&gb, buf, FRAMESIZE * 8); @@ -106,7 +112,7 @@ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata, do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb); for (j=0; j < BLOCKSIZE; j++) - *data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2); + *samples++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2); } ractx->old_energy = energy; @@ -114,7 +120,9 @@ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata, FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]); - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = ractx->frame; + return FRAMESIZE; } @@ -125,5 +133,6 @@ AVCodec ff_ra_144_decoder = { .priv_data_size = sizeof(RA144Context), .init = ra144_decode_init, .decode = ra144_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"), }; diff --git a/libavcodec/ra288.c b/libavcodec/ra288.c index a91a06cfa1..3a9f409f4a 100644 --- a/libavcodec/ra288.c +++ b/libavcodec/ra288.c @@ -36,6 +36,7 @@ #define RA288_BLOCKS_PER_FRAME 32 typedef struct { + AVFrame frame; DSPContext dsp; DECLARE_ALIGNED(16, float, sp_lpc)[FFALIGN(36, 8)]; ///< LPC coefficients for speech data (spec: A) DECLARE_ALIGNED(16, float, gain_lpc)[FFALIGN(10, 8)]; ///< LPC coefficients for gain (spec: GB) @@ -62,6 +63,10 @@ static av_cold int ra288_decode_init(AVCodecContext *avctx) RA288Context *ractx = avctx->priv_data; avctx->sample_fmt = AV_SAMPLE_FMT_FLT; dsputil_init(&ractx->dsp, avctx); + + avcodec_get_frame_defaults(&ractx->frame); + avctx->coded_frame = &ractx->frame; + return 0; } @@ -165,12 +170,12 @@ static void backward_filter(RA288Context *ractx, } static int ra288_decode_frame(AVCodecContext * avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; - float *out = data; - int i, out_size; + float *out; + int i, ret; RA288Context *ractx = avctx->priv_data; GetBitContext gb; @@ -181,12 +186,13 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data, return AVERROR_INVALIDDATA; } - out_size = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME * - 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); + /* get output buffer */ + ractx->frame.nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME; + if ((ret = avctx->get_buffer(avctx, &ractx->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + out = (float *)ractx->frame.data[0]; init_get_bits(&gb, buf, avctx->block_align * 8); @@ -208,7 +214,9 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data, } } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = ractx->frame; + return avctx->block_align; } @@ -219,5 +227,6 @@ AVCodec ff_ra_288_decoder = { .priv_data_size = sizeof(RA288Context), .init = ra288_decode_init, .decode = ra288_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), }; diff --git a/libavcodec/s302m.c b/libavcodec/s302m.c index 713d80f5d9..4a17fa102e 100644 --- a/libavcodec/s302m.c +++ b/libavcodec/s302m.c @@ -25,6 +25,10 @@ #define AES3_HEADER_LEN 4 +typedef struct S302MDecodeContext { + AVFrame frame; +} S302MDecodeContext; + static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf, int buf_size) { @@ -83,10 +87,12 @@ static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf, } static int s302m_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { + S302MDecodeContext *s = avctx->priv_data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; + int block_size, ret; int frame_size = s302m_parse_frame_header(avctx, buf, buf_size); if (frame_size < 0) @@ -95,11 +101,18 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data, buf_size -= AES3_HEADER_LEN; buf += AES3_HEADER_LEN; - if (*data_size < 4 * buf_size * 8 / (avctx->bits_per_coded_sample + 4)) - return -1; + /* get output buffer */ + block_size = (avctx->bits_per_coded_sample + 4) / 4; + s->frame.nb_samples = 2 * (buf_size / block_size) / avctx->channels; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + + buf_size = (s->frame.nb_samples * avctx->channels / 2) * block_size; if (avctx->bits_per_coded_sample == 24) { - uint32_t *o = data; + uint32_t *o = (uint32_t *)s->frame.data[0]; for (; buf_size > 6; buf_size -= 7) { *o++ = (av_reverse[buf[2]] << 24) | (av_reverse[buf[1]] << 16) | @@ -110,9 +123,8 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data, (av_reverse[buf[3] & 0x0f] << 4); buf += 7; } - *data_size = (uint8_t*) o - (uint8_t*) data; } else if (avctx->bits_per_coded_sample == 20) { - uint32_t *o = data; + uint32_t *o = (uint32_t *)s->frame.data[0]; for (; buf_size > 5; buf_size -= 6) { *o++ = (av_reverse[buf[2] & 0xf0] << 28) | (av_reverse[buf[1]] << 20) | @@ -122,9 +134,8 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data, (av_reverse[buf[3]] << 12); buf += 6; } - *data_size = (uint8_t*) o - (uint8_t*) data; } else { - uint16_t *o = data; + uint16_t *o = (uint16_t *)s->frame.data[0]; for (; buf_size > 4; buf_size -= 5) { *o++ = (av_reverse[buf[1]] << 8) | av_reverse[buf[0]]; @@ -133,10 +144,22 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data, (av_reverse[buf[2]] >> 4); buf += 5; } - *data_size = (uint8_t*) o - (uint8_t*) data; } - return buf - avpkt->data; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + + return avpkt->size; +} + +static int s302m_decode_init(AVCodecContext *avctx) +{ + S302MDecodeContext *s = avctx->priv_data; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + + return 0; } @@ -144,6 +167,9 @@ AVCodec ff_s302m_decoder = { .name = "s302m", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_S302M, + .priv_data_size = sizeof(S302MDecodeContext), + .init = s302m_decode_init, .decode = s302m_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"), }; diff --git a/libavcodec/shorten.c b/libavcodec/shorten.c index ee01e886a9..26ce6fe885 100644 --- a/libavcodec/shorten.c +++ b/libavcodec/shorten.c @@ -79,6 +79,7 @@ static const uint8_t is_audio_command[10] = { 1, 1, 1, 1, 0, 0, 0, 1, 1, 0 }; typedef struct ShortenContext { AVCodecContext *avctx; + AVFrame frame; GetBitContext gb; int min_framesize, max_framesize; @@ -112,6 +113,9 @@ static av_cold int shorten_decode_init(AVCodecContext * avctx) s->avctx = avctx; avctx->sample_fmt = AV_SAMPLE_FMT_S16; + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -394,15 +398,13 @@ static int read_header(ShortenContext *s) return 0; } -static int shorten_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int shorten_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; ShortenContext *s = avctx->priv_data; int i, input_buf_size = 0; - int16_t *samples = data; int ret; /* allocate internal bitstream buffer */ @@ -436,7 +438,7 @@ static int shorten_decode_frame(AVCodecContext *avctx, /* do not decode until buffer has at least max_framesize bytes or the end of the file has been reached */ if (buf_size < s->max_framesize && avpkt->data) { - *data_size = 0; + *got_frame_ptr = 0; return input_buf_size; } } @@ -448,13 +450,13 @@ static int shorten_decode_frame(AVCodecContext *avctx, if (!s->got_header) { if ((ret = read_header(s)) < 0) return ret; - *data_size = 0; + *got_frame_ptr = 0; goto finish_frame; } /* if quit command was read previously, don't decode anything */ if (s->got_quit_command) { - *data_size = 0; + *got_frame_ptr = 0; return avpkt->size; } @@ -464,7 +466,7 @@ static int shorten_decode_frame(AVCodecContext *avctx, int len; if (get_bits_left(&s->gb) < 3+FNSIZE) { - *data_size = 0; + *got_frame_ptr = 0; break; } @@ -472,7 +474,7 @@ static int shorten_decode_frame(AVCodecContext *avctx, if (cmd > FN_VERBATIM) { av_log(avctx, AV_LOG_ERROR, "unknown shorten function %d\n", cmd); - *data_size = 0; + *got_frame_ptr = 0; break; } @@ -507,7 +509,7 @@ static int shorten_decode_frame(AVCodecContext *avctx, break; } if (cmd == FN_BLOCKSIZE || cmd == FN_QUIT) { - *data_size = 0; + *got_frame_ptr = 0; break; } } else { @@ -571,19 +573,23 @@ static int shorten_decode_frame(AVCodecContext *avctx, /* if this is the last channel in the block, output the samples */ s->cur_chan++; if (s->cur_chan == s->channels) { - int out_size = s->blocksize * s->channels * - 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); + /* get output buffer */ + s->frame.nb_samples = s->blocksize; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } - interleave_buffer(samples, s->channels, s->blocksize, s->decoded); - *data_size = out_size; + /* interleave output */ + interleave_buffer((int16_t *)s->frame.data[0], s->channels, + s->blocksize, s->decoded); + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; } } } if (s->cur_chan < s->channels) - *data_size = 0; + *got_frame_ptr = 0; finish_frame: s->bitindex = get_bits_count(&s->gb) - 8*((get_bits_count(&s->gb))/8); @@ -614,6 +620,7 @@ static av_cold int shorten_decode_close(AVCodecContext *avctx) } av_freep(&s->bitstream); av_freep(&s->coeffs); + return 0; } @@ -625,6 +632,6 @@ AVCodec ff_shorten_decoder = { .init = shorten_decode_init, .close = shorten_decode_close, .decode = shorten_decode_frame, - .capabilities = CODEC_CAP_DELAY, + .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1, .long_name= NULL_IF_CONFIG_SMALL("Shorten"), }; diff --git a/libavcodec/sipr.c b/libavcodec/sipr.c index 20d9da15e6..d66c14fb12 100644 --- a/libavcodec/sipr.c +++ b/libavcodec/sipr.c @@ -507,20 +507,23 @@ static av_cold int sipr_decoder_init(AVCodecContext * avctx) avctx->sample_fmt = AV_SAMPLE_FMT_FLT; + avcodec_get_frame_defaults(&ctx->frame); + avctx->coded_frame = &ctx->frame; + return 0; } -static int sipr_decode_frame(AVCodecContext *avctx, void *datap, - int *data_size, AVPacket *avpkt) +static int sipr_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { SiprContext *ctx = avctx->priv_data; const uint8_t *buf=avpkt->data; SiprParameters parm; const SiprModeParam *mode_par = &modes[ctx->mode]; GetBitContext gb; - float *data = datap; + float *samples; int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE; - int i, out_size; + int i, ret; ctx->avctx = avctx; if (avpkt->size < (mode_par->bits_per_frame >> 3)) { @@ -530,27 +533,27 @@ static int sipr_decode_frame(AVCodecContext *avctx, void *datap, return -1; } - out_size = mode_par->frames_per_packet * subframe_size * - mode_par->subframe_count * - av_get_bytes_per_sample(avctx->sample_fmt); - if (*data_size < out_size) { - av_log(avctx, AV_LOG_ERROR, - "Error processing packet: output buffer (%d) too small\n", - *data_size); - return -1; + /* get output buffer */ + ctx->frame.nb_samples = mode_par->frames_per_packet * subframe_size * + mode_par->subframe_count; + if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (float *)ctx->frame.data[0]; init_get_bits(&gb, buf, mode_par->bits_per_frame); for (i = 0; i < mode_par->frames_per_packet; i++) { decode_parameters(&parm, &gb, mode_par); - ctx->decode_frame(ctx, &parm, data); + ctx->decode_frame(ctx, &parm, samples); - data += subframe_size * mode_par->subframe_count; + samples += subframe_size * mode_par->subframe_count; } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = ctx->frame; return mode_par->bits_per_frame >> 3; } @@ -562,5 +565,6 @@ AVCodec ff_sipr_decoder = { .priv_data_size = sizeof(SiprContext), .init = sipr_decoder_init, .decode = sipr_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"), }; diff --git a/libavcodec/smacker.c b/libavcodec/smacker.c index 19bfab4ffd..a983922fc7 100644 --- a/libavcodec/smacker.c +++ b/libavcodec/smacker.c @@ -559,31 +559,43 @@ static av_cold int decode_end(AVCodecContext *avctx) } +typedef struct SmackerAudioContext { + AVFrame frame; +} SmackerAudioContext; + static av_cold int smka_decode_init(AVCodecContext *avctx) { + SmackerAudioContext *s = avctx->priv_data; + if (avctx->channels < 1 || avctx->channels > 2) { av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n"); return AVERROR(EINVAL); } avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO; avctx->sample_fmt = avctx->bits_per_coded_sample == 8 ? AV_SAMPLE_FMT_U8 : AV_SAMPLE_FMT_S16; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } /** * Decode Smacker audio data */ -static int smka_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) +static int smka_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { + SmackerAudioContext *s = avctx->priv_data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; GetBitContext gb; HuffContext h[4]; VLC vlc[4]; - int16_t *samples = data; - uint8_t *samples8 = data; + int16_t *samples; + uint8_t *samples8; int val; - int i, res; + int i, res, ret; int unp_size; int bits, stereo; int pred[2] = {0, 0}; @@ -599,15 +611,11 @@ static int smka_decode_frame(AVCodecContext *avctx, void *data, int *data_size, if(!get_bits1(&gb)){ av_log(avctx, AV_LOG_INFO, "Sound: no data\n"); - *data_size = 0; + *got_frame_ptr = 0; return 1; } stereo = get_bits1(&gb); bits = get_bits1(&gb); - if (unp_size & 0xC0000000 || unp_size > *data_size) { - av_log(avctx, AV_LOG_ERROR, "Frame is too large to fit in buffer\n"); - return -1; - } if (stereo ^ (avctx->channels != 1)) { av_log(avctx, AV_LOG_ERROR, "channels mismatch\n"); return AVERROR(EINVAL); @@ -617,6 +625,15 @@ static int smka_decode_frame(AVCodecContext *avctx, void *data, int *data_size, return AVERROR(EINVAL); } + /* get output buffer */ + s->frame.nb_samples = unp_size / (avctx->channels * (bits + 1)); + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + samples = (int16_t *)s->frame.data[0]; + samples8 = s->frame.data[0]; + memset(vlc, 0, sizeof(VLC) * 4); memset(h, 0, sizeof(HuffContext) * 4); // Initialize @@ -706,7 +723,9 @@ static int smka_decode_frame(AVCodecContext *avctx, void *data, int *data_size, av_free(h[i].values); } - *data_size = unp_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return buf_size; } @@ -726,8 +745,10 @@ AVCodec ff_smackaud_decoder = { .name = "smackaud", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_SMACKAUDIO, + .priv_data_size = sizeof(SmackerAudioContext), .init = smka_decode_init, .decode = smka_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Smacker audio"), }; diff --git a/libavcodec/svq1dec.c b/libavcodec/svq1dec.c index 431913c57d..83fe07e4c7 100644 --- a/libavcodec/svq1dec.c +++ b/libavcodec/svq1dec.c @@ -195,7 +195,8 @@ static const uint8_t string_table[256] = { #define SVQ1_CALC_CODEBOOK_ENTRIES(cbook)\ codebook = (const uint32_t *) cbook[level];\ - bit_cache = get_bits (bitbuf, 4*stages);\ + if (stages > 0)\ + bit_cache = get_bits (bitbuf, 4*stages);\ /* calculate codebook entries for this vector */\ for (j=0; j < stages; j++) {\ entries[j] = (((bit_cache >> (4*(stages - j - 1))) & 0xF) + 16*j) << (level + 1);\ diff --git a/libavcodec/truespeech.c b/libavcodec/truespeech.c index b0a8e44150..bb4ce6f219 100644 --- a/libavcodec/truespeech.c +++ b/libavcodec/truespeech.c @@ -34,6 +34,7 @@ * TrueSpeech decoder context */ typedef struct { + AVFrame frame; DSPContext dsp; /* input data */ uint8_t buffer[32]; @@ -69,6 +70,9 @@ static av_cold int truespeech_decode_init(AVCodecContext * avctx) dsputil_init(&c->dsp, avctx); + avcodec_get_frame_defaults(&c->frame); + avctx->coded_frame = &c->frame; + return 0; } @@ -299,17 +303,16 @@ static void truespeech_save_prevvec(TSContext *c) c->prevfilt[i] = c->cvector[i]; } -static int truespeech_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int truespeech_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; TSContext *c = avctx->priv_data; int i, j; - short *samples = data; - int iterations, out_size; + int16_t *samples; + int iterations, ret; iterations = buf_size / 32; @@ -319,13 +322,15 @@ static int truespeech_decode_frame(AVCodecContext *avctx, return -1; } - out_size = iterations * 240 * 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); + /* get output buffer */ + c->frame.nb_samples = iterations * 240; + if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (int16_t *)c->frame.data[0]; - memset(samples, 0, out_size); + memset(samples, 0, iterations * 240 * sizeof(*samples)); for(j = 0; j < iterations; j++) { truespeech_read_frame(c, buf); @@ -345,7 +350,8 @@ static int truespeech_decode_frame(AVCodecContext *avctx, truespeech_save_prevvec(c); } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = c->frame; return buf_size; } @@ -357,5 +363,6 @@ AVCodec ff_truespeech_decoder = { .priv_data_size = sizeof(TSContext), .init = truespeech_decode_init, .decode = truespeech_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"), }; diff --git a/libavcodec/tta.c b/libavcodec/tta.c index 7ff5116ffb..3c4c152213 100644 --- a/libavcodec/tta.c +++ b/libavcodec/tta.c @@ -56,6 +56,7 @@ typedef struct TTAChannel { typedef struct TTAContext { AVCodecContext *avctx; + AVFrame frame; GetBitContext gb; int format, channels, bps, data_length; @@ -288,17 +289,19 @@ static av_cold int tta_decode_init(AVCodecContext * avctx) return -1; } + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } -static int tta_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int tta_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; TTAContext *s = avctx->priv_data; - int i, out_size; + int i, ret; int cur_chan = 0, framelen = s->frame_length; int32_t *p; @@ -309,10 +312,11 @@ static int tta_decode_frame(AVCodecContext *avctx, if (!s->total_frames && s->last_frame_length) framelen = s->last_frame_length; - out_size = framelen * s->channels * av_get_bytes_per_sample(avctx->sample_fmt); - if (*data_size < out_size) { - av_log(avctx, AV_LOG_ERROR, "Output buffer size is too small.\n"); - return -1; + /* get output buffer */ + s->frame.nb_samples = framelen; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } // decode directly to output buffer for 24-bit sample format @@ -409,20 +413,20 @@ static int tta_decode_frame(AVCodecContext *avctx, // convert to output buffer switch(s->bps) { case 1: { - uint8_t *samples = data; + uint8_t *samples = (int16_t *)s->frame.data[0]; for (p = s->decode_buffer; p < s->decode_buffer + (framelen * s->channels); p++) *samples++ = *p + 0x80; break; } case 2: { - uint16_t *samples = data; + uint16_t *samples = (int16_t *)s->frame.data[0]; for (p = s->decode_buffer; p < s->decode_buffer + (framelen * s->channels); p++) *samples++ = *p; break; } case 3: { // shift samples for 24-bit sample format - int32_t *samples = data; + int32_t *samples = (int16_t *)s->frame.data[0]; for (p = s->decode_buffer; p < s->decode_buffer + (framelen * s->channels); p++) *samples++ <<= 8; // reset decode buffer @@ -433,7 +437,8 @@ static int tta_decode_frame(AVCodecContext *avctx, av_log(s->avctx, AV_LOG_ERROR, "Error, only 16bit samples supported!\n"); } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; return buf_size; } @@ -455,5 +460,6 @@ AVCodec ff_tta_decoder = { .init = tta_decode_init, .close = tta_decode_close, .decode = tta_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("True Audio (TTA)"), }; diff --git a/libavcodec/twinvq.c b/libavcodec/twinvq.c index bf8b48b15e..1810ce8c88 100644 --- a/libavcodec/twinvq.c +++ b/libavcodec/twinvq.c @@ -174,6 +174,7 @@ static const ModeTab mode_44_48 = { typedef struct TwinContext { AVCodecContext *avctx; + AVFrame frame; DSPContext dsp; FFTContext mdct_ctx[3]; @@ -195,6 +196,7 @@ typedef struct TwinContext { float *curr_frame; ///< non-interleaved output float *prev_frame; ///< non-interleaved previous frame int last_block_pos[2]; + int discarded_packets; float *cos_tabs[3]; @@ -676,6 +678,9 @@ static void imdct_output(TwinContext *tctx, enum FrameType ftype, int wtype, i); } + if (!out) + return; + size2 = tctx->last_block_pos[0]; size1 = mtab->size - size2; if (tctx->avctx->channels == 2) { @@ -811,16 +816,16 @@ static void read_and_decode_spectrum(TwinContext *tctx, GetBitContext *gb, } static int twin_decode_frame(AVCodecContext * avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; TwinContext *tctx = avctx->priv_data; GetBitContext gb; const ModeTab *mtab = tctx->mtab; - float *out = data; + float *out = NULL; enum FrameType ftype; - int window_type, out_size; + int window_type, ret; static const enum FrameType wtype_to_ftype_table[] = { FT_LONG, FT_LONG, FT_SHORT, FT_LONG, FT_MEDIUM, FT_LONG, FT_LONG, FT_MEDIUM, FT_MEDIUM @@ -832,11 +837,14 @@ static int twin_decode_frame(AVCodecContext * avctx, void *data, return AVERROR(EINVAL); } - out_size = mtab->size * avctx->channels * - 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); + /* get output buffer */ + if (tctx->discarded_packets >= 2) { + tctx->frame.nb_samples = mtab->size; + if ((ret = avctx->get_buffer(avctx, &tctx->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + out = (float *)tctx->frame.data[0]; } init_get_bits(&gb, buf, buf_size * 8); @@ -856,12 +864,14 @@ static int twin_decode_frame(AVCodecContext * avctx, void *data, FFSWAP(float*, tctx->curr_frame, tctx->prev_frame); - if (tctx->avctx->frame_number < 2) { - *data_size=0; + if (tctx->discarded_packets < 2) { + tctx->discarded_packets++; + *got_frame_ptr = 0; return buf_size; } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = tctx->frame;; return buf_size; } @@ -1153,6 +1163,9 @@ static av_cold int twin_decode_init(AVCodecContext *avctx) memset_float(tctx->bark_hist[0][0], 0.1, FF_ARRAY_ELEMS(tctx->bark_hist)); + avcodec_get_frame_defaults(&tctx->frame); + avctx->coded_frame = &tctx->frame; + return 0; } @@ -1164,5 +1177,6 @@ AVCodec ff_twinvq_decoder = { .init = twin_decode_init, .close = twin_decode_close, .decode = twin_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("VQF TwinVQ"), }; diff --git a/libavcodec/utils.c b/libavcodec/utils.c index f023a8967c..fa94b7c124 100644 --- a/libavcodec/utils.c +++ b/libavcodec/utils.c @@ -127,7 +127,10 @@ void avcodec_set_dimensions(AVCodecContext *s, int width, int height){ #define INTERNAL_BUFFER_SIZE (32+1) -void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height, int linesize_align[4]){ +void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height, + int linesize_align[AV_NUM_DATA_POINTERS]) +{ + int i; int w_align= 1; int h_align= 1; @@ -213,10 +216,8 @@ void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height, int l *height+=2; // some of the optimized chroma MC reads one line too much // which is also done in mpeg decoders with lowres > 0 - linesize_align[0] = - linesize_align[1] = - linesize_align[2] = - linesize_align[3] = STRIDE_ALIGN; + for (i = 0; i < AV_NUM_DATA_POINTERS; i++) + linesize_align[i] = STRIDE_ALIGN; //STRIDE_ALIGN is 8 for SSE* but this does not work for SVQ1 chroma planes //we could change STRIDE_ALIGN to 16 for x86/sse but it would increase the //picture size unneccessarily in some cases. The solution here is not @@ -225,16 +226,15 @@ void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height, int l if(s->codec_id == CODEC_ID_SVQ1 || s->codec_id == CODEC_ID_VP5 || s->codec_id == CODEC_ID_VP6 || s->codec_id == CODEC_ID_VP6F || s->codec_id == CODEC_ID_VP6A || s->codec_id == CODEC_ID_DIRAC) { - linesize_align[0] = - linesize_align[1] = - linesize_align[2] = 16; + for (i = 0; i < AV_NUM_DATA_POINTERS; i++) + linesize_align[i] = 16; } #endif } void avcodec_align_dimensions(AVCodecContext *s, int *width, int *height){ int chroma_shift = av_pix_fmt_descriptors[s->pix_fmt].log2_chroma_w; - int linesize_align[4]; + int linesize_align[AV_NUM_DATA_POINTERS]; int align; avcodec_align_dimensions2(s, width, height, linesize_align); align = FFMAX(linesize_align[0], linesize_align[3]); @@ -260,7 +260,108 @@ void ff_init_buffer_info(AVCodecContext *s, AVFrame *pic) pic->format = s->pix_fmt; } -int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){ +static int audio_get_buffer(AVCodecContext *avctx, AVFrame *frame) +{ + AVCodecInternal *avci = avctx->internal; + InternalBuffer *buf; + int buf_size, ret, i, needs_extended_data; + + buf_size = av_samples_get_buffer_size(NULL, avctx->channels, + frame->nb_samples, avctx->sample_fmt, + 32); + if (buf_size < 0) + return AVERROR(EINVAL); + + needs_extended_data = av_sample_fmt_is_planar(avctx->sample_fmt) && + avctx->channels > AV_NUM_DATA_POINTERS; + + /* allocate InternalBuffer if needed */ + if (!avci->buffer) { + avci->buffer = av_mallocz(sizeof(InternalBuffer)); + if (!avci->buffer) + return AVERROR(ENOMEM); + } + buf = avci->buffer; + + /* if there is a previously-used internal buffer, check its size and + channel count to see if we can reuse it */ + if (buf->extended_data) { + /* if current buffer is too small, free it */ + if (buf->extended_data[0] && buf_size > buf->audio_data_size) { + av_free(buf->extended_data[0]); + if (buf->extended_data != buf->data) + av_free(&buf->extended_data); + buf->extended_data = NULL; + buf->data[0] = NULL; + } + /* if number of channels has changed, reset and/or free extended data + pointers but leave data buffer in buf->data[0] for reuse */ + if (buf->nb_channels != avctx->channels) { + if (buf->extended_data != buf->data) + av_free(buf->extended_data); + buf->extended_data = NULL; + } + } + + /* if there is no previous buffer or the previous buffer cannot be used + as-is, allocate a new buffer and/or rearrange the channel pointers */ + if (!buf->extended_data) { + /* if the channel pointers will fit, just set extended_data to data, + otherwise allocate the extended_data channel pointers */ + if (needs_extended_data) { + buf->extended_data = av_mallocz(avctx->channels * + sizeof(*buf->extended_data)); + if (!buf->extended_data) + return AVERROR(ENOMEM); + } else { + buf->extended_data = buf->data; + } + + /* if there is a previous buffer and it is large enough, reuse it and + just fill-in new channel pointers and linesize, otherwise allocate + a new buffer */ + if (buf->extended_data[0]) { + ret = av_samples_fill_arrays(buf->extended_data, &buf->linesize[0], + buf->extended_data[0], avctx->channels, + frame->nb_samples, avctx->sample_fmt, + 32); + } else { + ret = av_samples_alloc(buf->extended_data, &buf->linesize[0], + avctx->channels, frame->nb_samples, + avctx->sample_fmt, 32); + } + if (ret) + return ret; + + /* if data was not used for extended_data, we need to copy as many of + the extended_data channel pointers as will fit */ + if (needs_extended_data) { + for (i = 0; i < AV_NUM_DATA_POINTERS; i++) + buf->data[i] = buf->extended_data[i]; + } + buf->audio_data_size = buf_size; + buf->nb_channels = avctx->channels; + } + + /* copy InternalBuffer info to the AVFrame */ + frame->type = FF_BUFFER_TYPE_INTERNAL; + frame->extended_data = buf->extended_data; + frame->linesize[0] = buf->linesize[0]; + memcpy(frame->data, buf->data, sizeof(frame->data)); + + if (avctx->pkt) frame->pkt_pts = avctx->pkt->pts; + else frame->pkt_pts = AV_NOPTS_VALUE; + frame->reordered_opaque = avctx->reordered_opaque; + + if (avctx->debug & FF_DEBUG_BUFFERS) + av_log(avctx, AV_LOG_DEBUG, "default_get_buffer called on frame %p, " + "internal audio buffer used\n", frame); + + return 0; +} + +static int video_get_buffer(AVCodecContext *s, AVFrame *pic) +{ int i; int w= s->width; int h= s->height; @@ -295,7 +396,7 @@ int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){ return -1; } - for(i=0; i<4; i++){ + for (i = 0; i < AV_NUM_DATA_POINTERS; i++) { av_freep(&buf->base[i]); buf->data[i]= NULL; } @@ -310,7 +411,7 @@ int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){ int tmpsize; int unaligned; AVPicture picture; - int stride_align[4]; + int stride_align[AV_NUM_DATA_POINTERS]; const int pixel_size = av_pix_fmt_descriptors[s->pix_fmt].comp[0].step_minus1+1; avcodec_get_chroma_sub_sample(s->pix_fmt, &h_chroma_shift, &v_chroma_shift); @@ -363,6 +464,10 @@ int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){ else buf->data[i] = buf->base[i] + FFALIGN((buf->linesize[i]*EDGE_WIDTH>>v_shift) + (pixel_size*EDGE_WIDTH>>h_shift), stride_align[i]); } + for (; i < AV_NUM_DATA_POINTERS; i++) { + buf->base[i] = buf->data[i] = NULL; + buf->linesize[i] = 0; + } if(size[1] && !size[2]) ff_set_systematic_pal2((uint32_t*)buf->data[1], s->pix_fmt); buf->width = s->width; @@ -372,11 +477,12 @@ int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){ } pic->type= FF_BUFFER_TYPE_INTERNAL; - for(i=0; i<4; i++){ + for (i = 0; i < AV_NUM_DATA_POINTERS; i++) { pic->base[i]= buf->base[i]; pic->data[i]= buf->data[i]; pic->linesize[i]= buf->linesize[i]; } + pic->extended_data = pic->data; avci->buffer_count++; if (s->pkt) { @@ -399,11 +505,25 @@ int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){ return 0; } +int avcodec_default_get_buffer(AVCodecContext *avctx, AVFrame *frame) +{ + switch (avctx->codec_type) { + case AVMEDIA_TYPE_VIDEO: + return video_get_buffer(avctx, frame); + case AVMEDIA_TYPE_AUDIO: + return audio_get_buffer(avctx, frame); + default: + return -1; + } +} + void avcodec_default_release_buffer(AVCodecContext *s, AVFrame *pic){ int i; InternalBuffer *buf, *last; AVCodecInternal *avci = s->internal; + assert(s->codec_type == AVMEDIA_TYPE_VIDEO); + assert(pic->type==FF_BUFFER_TYPE_INTERNAL); assert(avci->buffer_count); @@ -421,7 +541,7 @@ void avcodec_default_release_buffer(AVCodecContext *s, AVFrame *pic){ FFSWAP(InternalBuffer, *buf, *last); } - for(i=0; i<4; i++){ + for (i = 0; i < AV_NUM_DATA_POINTERS; i++) { pic->data[i]=NULL; // pic->base[i]=NULL; } @@ -436,6 +556,8 @@ int avcodec_default_reget_buffer(AVCodecContext *s, AVFrame *pic){ AVFrame temp_pic; int i; + assert(s->codec_type == AVMEDIA_TYPE_VIDEO); + /* If no picture return a new buffer */ if(pic->data[0] == NULL) { /* We will copy from buffer, so must be readable */ @@ -455,7 +577,7 @@ int avcodec_default_reget_buffer(AVCodecContext *s, AVFrame *pic){ * Not internal type and reget_buffer not overridden, emulate cr buffer */ temp_pic = *pic; - for(i = 0; i < 4; i++) + for(i = 0; i < AV_NUM_DATA_POINTERS; i++) pic->data[i] = pic->base[i] = NULL; pic->opaque = NULL; /* Allocate new frame */ @@ -862,36 +984,73 @@ int attribute_align_arg avcodec_decode_video2(AVCodecContext *avctx, AVFrame *pi return ret; } +#if FF_API_OLD_DECODE_AUDIO int attribute_align_arg avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples, int *frame_size_ptr, AVPacket *avpkt) { - int ret; + AVFrame frame; + int ret, got_frame = 0; + + if (avctx->get_buffer != avcodec_default_get_buffer) { + av_log(avctx, AV_LOG_ERROR, "A custom get_buffer() cannot be used with " + "avcodec_decode_audio3()\n"); + return AVERROR(EINVAL); + } + + ret = avcodec_decode_audio4(avctx, &frame, &got_frame, avpkt); + + if (ret >= 0 && got_frame) { + int ch, plane_size; + int planar = av_sample_fmt_is_planar(avctx->sample_fmt); + int data_size = av_samples_get_buffer_size(&plane_size, avctx->channels, + frame.nb_samples, + avctx->sample_fmt, 1); + if (*frame_size_ptr < data_size) { + av_log(avctx, AV_LOG_ERROR, "output buffer size is too small for " + "the current frame (%d < %d)\n", *frame_size_ptr, data_size); + return AVERROR(EINVAL); + } + + memcpy(samples, frame.extended_data[0], plane_size); + + if (planar && avctx->channels > 1) { + uint8_t *out = ((uint8_t *)samples) + plane_size; + for (ch = 1; ch < avctx->channels; ch++) { + memcpy(out, frame.extended_data[ch], plane_size); + out += plane_size; + } + } + *frame_size_ptr = data_size; + } else { + *frame_size_ptr = 0; + } + return ret; +} +#endif + +int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx, + AVFrame *frame, + int *got_frame_ptr, + AVPacket *avpkt) +{ + int ret = 0; + + *got_frame_ptr = 0; if (!avpkt->data && avpkt->size) { av_log(avctx, AV_LOG_ERROR, "invalid packet: NULL data, size != 0\n"); return AVERROR(EINVAL); } - if((avctx->codec->capabilities & CODEC_CAP_DELAY) || avpkt->size){ + if ((avctx->codec->capabilities & CODEC_CAP_DELAY) || avpkt->size) { av_packet_split_side_data(avpkt); avctx->pkt = avpkt; - //FIXME remove the check below _after_ ensuring that all audio check that the available space is enough - if(*frame_size_ptr < AVCODEC_MAX_AUDIO_FRAME_SIZE){ - av_log(avctx, AV_LOG_ERROR, "buffer smaller than AVCODEC_MAX_AUDIO_FRAME_SIZE\n"); - return -1; - } - if(*frame_size_ptr < FF_MIN_BUFFER_SIZE || - *frame_size_ptr < avctx->channels * avctx->frame_size * sizeof(int16_t)){ - av_log(avctx, AV_LOG_ERROR, "buffer %d too small\n", *frame_size_ptr); - return -1; + ret = avctx->codec->decode(avctx, frame, got_frame_ptr, avpkt); + if (ret >= 0 && *got_frame_ptr) { + avctx->frame_number++; + frame->pkt_dts = avpkt->dts; } - - ret = avctx->codec->decode(avctx, samples, frame_size_ptr, avpkt); - avctx->frame_number++; - }else{ - ret= 0; - *frame_size_ptr=0; } return ret; } @@ -1230,7 +1389,8 @@ void avcodec_flush_buffers(AVCodecContext *avctx) avctx->codec->flush(avctx); } -void avcodec_default_free_buffers(AVCodecContext *s){ +static void video_free_buffers(AVCodecContext *s) +{ AVCodecInternal *avci = s->internal; int i, j; @@ -1252,6 +1412,37 @@ void avcodec_default_free_buffers(AVCodecContext *s){ avci->buffer_count=0; } +static void audio_free_buffers(AVCodecContext *avctx) +{ + AVCodecInternal *avci = avctx->internal; + InternalBuffer *buf; + + if (!avci->buffer) + return; + buf = avci->buffer; + + if (buf->extended_data) { + av_free(buf->extended_data[0]); + if (buf->extended_data != buf->data) + av_free(buf->extended_data); + } + av_freep(&avci->buffer); +} + +void avcodec_default_free_buffers(AVCodecContext *avctx) +{ + switch (avctx->codec_type) { + case AVMEDIA_TYPE_VIDEO: + video_free_buffers(avctx); + break; + case AVMEDIA_TYPE_AUDIO: + audio_free_buffers(avctx); + break; + default: + break; + } +} + #if FF_API_OLD_FF_PICT_TYPES char av_get_pict_type_char(int pict_type){ return av_get_picture_type_char(pict_type); diff --git a/libavcodec/version.h b/libavcodec/version.h index b955116c6a..70dbd0001e 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -21,8 +21,8 @@ #define AVCODEC_VERSION_H #define LIBAVCODEC_VERSION_MAJOR 53 -#define LIBAVCODEC_VERSION_MINOR 39 -#define LIBAVCODEC_VERSION_MICRO 1 +#define LIBAVCODEC_VERSION_MINOR 40 +#define LIBAVCODEC_VERSION_MICRO 0 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ LIBAVCODEC_VERSION_MINOR, \ @@ -110,6 +110,11 @@ #ifndef FF_API_TIFFENC_COMPLEVEL #define FF_API_TIFFENC_COMPLEVEL (LIBAVCODEC_VERSION_MAJOR < 54) #endif - +#ifndef FF_API_DATA_POINTERS +#define FF_API_DATA_POINTERS (LIBAVCODEC_VERSION_MAJOR < 54) +#endif +#ifndef FF_API_OLD_DECODE_AUDIO +#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 54) +#endif #endif /* AVCODEC_VERSION_H */ diff --git a/libavcodec/vmdav.c b/libavcodec/vmdav.c index 06c6255cc5..ec7c967f74 100644 --- a/libavcodec/vmdav.c +++ b/libavcodec/vmdav.c @@ -466,6 +466,7 @@ static av_cold int vmdvideo_decode_end(AVCodecContext *avctx) #define BLOCK_TYPE_SILENCE 3 typedef struct VmdAudioContext { + AVFrame frame; int out_bps; int chunk_size; } VmdAudioContext; @@ -507,6 +508,9 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx) s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2); + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, " "block align = %d, sample rate = %d\n", avctx->channels, avctx->bits_per_coded_sample, avctx->block_align, @@ -544,22 +548,21 @@ static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size, } } -static int vmdaudio_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; const uint8_t *buf_end; int buf_size = avpkt->size; VmdAudioContext *s = avctx->priv_data; int block_type, silent_chunks, audio_chunks; - int nb_samples, out_size; - uint8_t *output_samples_u8 = data; - int16_t *output_samples_s16 = data; + int ret; + uint8_t *output_samples_u8; + int16_t *output_samples_s16; if (buf_size < 16) { av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n"); - *data_size = 0; + *got_frame_ptr = 0; return buf_size; } @@ -590,10 +593,15 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx, /* ensure output buffer is large enough */ audio_chunks = buf_size / s->chunk_size; - nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) / avctx->channels; - out_size = nb_samples * avctx->channels * s->out_bps; - if (*data_size < out_size) - return -1; + + /* get output buffer */ + s->frame.nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) / avctx->channels; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + output_samples_u8 = s->frame.data[0]; + output_samples_s16 = (int16_t *)s->frame.data[0]; /* decode silent chunks */ if (silent_chunks > 0) { @@ -623,7 +631,9 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx, } } - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return avpkt->size; } @@ -651,5 +661,6 @@ AVCodec ff_vmdaudio_decoder = { .priv_data_size = sizeof(VmdAudioContext), .init = vmdaudio_decode_init, .decode = vmdaudio_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"), }; diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c index 0457d8b454..03ecc38ed4 100644 --- a/libavcodec/vorbisdec.c +++ b/libavcodec/vorbisdec.c @@ -125,6 +125,7 @@ typedef struct { typedef struct vorbis_context_s { AVCodecContext *avccontext; + AVFrame frame; GetBitContext gb; DSPContext dsp; FmtConvertContext fmt_conv; @@ -1037,6 +1038,9 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext) avccontext->sample_rate = vc->audio_samplerate; avccontext->frame_size = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2; + avcodec_get_frame_defaults(&vc->frame); + avccontext->coded_frame = &vc->frame; + return 0; } @@ -1609,16 +1613,15 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) // Return the decoded audio packet through the standard api -static int vorbis_decode_frame(AVCodecContext *avccontext, - void *data, int *data_size, - AVPacket *avpkt) +static int vorbis_decode_frame(AVCodecContext *avccontext, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; vorbis_context *vc = avccontext->priv_data; GetBitContext *gb = &(vc->gb); const float *channel_ptrs[255]; - int i, len, out_size; + int i, len, ret; av_dlog(NULL, "packet length %d \n", buf_size); @@ -1629,18 +1632,18 @@ static int vorbis_decode_frame(AVCodecContext *avccontext, if (!vc->first_frame) { vc->first_frame = 1; - *data_size = 0; + *got_frame_ptr = 0; return buf_size; } av_dlog(NULL, "parsed %d bytes %d bits, returned %d samples (*ch*bits) \n", get_bits_count(gb) / 8, get_bits_count(gb) % 8, len); - out_size = len * vc->audio_channels * - av_get_bytes_per_sample(avccontext->sample_fmt); - if (*data_size < out_size) { - av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n"); - return AVERROR(EINVAL); + /* get output buffer */ + vc->frame.nb_samples = len; + if ((ret = avccontext->get_buffer(avccontext, &vc->frame)) < 0) { + av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } if (vc->audio_channels > 8) { @@ -1653,12 +1656,15 @@ static int vorbis_decode_frame(AVCodecContext *avccontext, } if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT) - vc->fmt_conv.float_interleave(data, channel_ptrs, len, vc->audio_channels); + vc->fmt_conv.float_interleave((float *)vc->frame.data[0], channel_ptrs, + len, vc->audio_channels); else - vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len, + vc->fmt_conv.float_to_int16_interleave((int16_t *)vc->frame.data[0], + channel_ptrs, len, vc->audio_channels); - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = vc->frame; return buf_size; } @@ -1682,6 +1688,7 @@ AVCodec ff_vorbis_decoder = { .init = vorbis_decode_init, .close = vorbis_decode_close, .decode = vorbis_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Vorbis"), .channel_layouts = ff_vorbis_channel_layouts, .sample_fmts = (const enum AVSampleFormat[]) { diff --git a/libavcodec/vp3.c b/libavcodec/vp3.c index 83ec4d80f0..a6a3109dba 100644 --- a/libavcodec/vp3.c +++ b/libavcodec/vp3.c @@ -1335,8 +1335,8 @@ end: */ static void vp3_draw_horiz_band(Vp3DecodeContext *s, int y) { - int h, cy; - int offset[4]; + int h, cy, i; + int offset[AV_NUM_DATA_POINTERS]; if (HAVE_THREADS && s->avctx->active_thread_type&FF_THREAD_FRAME) { int y_flipped = s->flipped_image ? s->avctx->height-y : y; @@ -1362,7 +1362,8 @@ static void vp3_draw_horiz_band(Vp3DecodeContext *s, int y) offset[0] = s->current_frame.linesize[0]*y; offset[1] = s->current_frame.linesize[1]*cy; offset[2] = s->current_frame.linesize[2]*cy; - offset[3] = 0; + for (i = 3; i < AV_NUM_DATA_POINTERS; i++) + offset[i] = 0; emms_c(); s->avctx->draw_horiz_band(s->avctx, &s->current_frame, offset, y, 3, h); diff --git a/libavcodec/vp8.c b/libavcodec/vp8.c index b374fa4222..8cf67ba491 100644 --- a/libavcodec/vp8.c +++ b/libavcodec/vp8.c @@ -51,8 +51,7 @@ static int vp8_alloc_frame(VP8Context *s, AVFrame *f) int ret; if ((ret = ff_thread_get_buffer(s->avctx, f)) < 0) return ret; - if (s->num_maps_to_be_freed) { - assert(!s->maps_are_invalid); + if (s->num_maps_to_be_freed && !s->maps_are_invalid) { f->ref_index[0] = s->segmentation_maps[--s->num_maps_to_be_freed]; } else if (!(f->ref_index[0] = av_mallocz(s->mb_width * s->mb_height))) { ff_thread_release_buffer(s->avctx, f); @@ -1568,13 +1567,15 @@ static int vp8_decode_frame(AVCodecContext *avctx, void *data, int *data_size, VP8Context *s = avctx->priv_data; int ret, mb_x, mb_y, i, y, referenced; enum AVDiscard skip_thresh; - AVFrame *av_uninit(curframe), *prev_frame = s->framep[VP56_FRAME_CURRENT]; + AVFrame *av_uninit(curframe), *prev_frame; release_queued_segmaps(s, 0); if ((ret = decode_frame_header(s, avpkt->data, avpkt->size)) < 0) return ret; + prev_frame = s->framep[VP56_FRAME_CURRENT]; + referenced = s->update_last || s->update_golden == VP56_FRAME_CURRENT || s->update_altref == VP56_FRAME_CURRENT; @@ -1815,6 +1816,7 @@ static int vp8_decode_update_thread_context(AVCodecContext *dst, const AVCodecCo if (s->macroblocks_base && (s_src->mb_width != s->mb_width || s_src->mb_height != s->mb_height)) { free_buffers(s); + s->maps_are_invalid = 1; } s->prob[0] = s_src->prob[!s_src->update_probabilities]; diff --git a/libavcodec/wavpack.c b/libavcodec/wavpack.c index 5ff9795ef8..1ccaec665d 100644 --- a/libavcodec/wavpack.c +++ b/libavcodec/wavpack.c @@ -115,8 +115,6 @@ typedef struct WavpackFrameContext { int float_shift; int float_max_exp; WvChannel ch[2]; - int samples_left; - int max_samples; int pos; SavedContext sc, extra_sc; } WavpackFrameContext; @@ -125,6 +123,7 @@ typedef struct WavpackFrameContext { typedef struct WavpackContext { AVCodecContext *avctx; + AVFrame frame; WavpackFrameContext *fdec[WV_MAX_FRAME_DECODERS]; int fdec_num; @@ -133,7 +132,6 @@ typedef struct WavpackContext { int mkv_mode; int block; int samples; - int samples_left; int ch_offset; } WavpackContext; @@ -485,7 +483,6 @@ static float wv_get_value_float(WavpackFrameContext *s, uint32_t *crc, int S) static void wv_reset_saved_context(WavpackFrameContext *s) { s->pos = 0; - s->samples_left = 0; s->sc.crc = s->extra_sc.crc = 0xFFFFFFFF; } @@ -502,8 +499,7 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb, vo float *dstfl = dst; const int channel_pad = s->avctx->channels - 2; - if(s->samples_left == s->samples) - s->one = s->zero = s->zeroes = 0; + s->one = s->zero = s->zeroes = 0; do{ L = wv_get_value(s, gb, 0, &last); if(last) break; @@ -594,13 +590,8 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb, vo dst16 += channel_pad; } count++; - }while(!last && count < s->max_samples); + } while (!last && count < s->samples); - if (last) - s->samples_left = 0; - else - s->samples_left -= count; - if(!s->samples_left){ wv_reset_saved_context(s); if(crc != s->CRC){ av_log(s->avctx, AV_LOG_ERROR, "CRC error\n"); @@ -610,15 +601,7 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb, vo av_log(s->avctx, AV_LOG_ERROR, "Extra bits CRC error\n"); return -1; } - }else{ - s->pos = pos; - s->sc.crc = crc; - s->sc.bits_used = get_bits_count(&s->gb); - if(s->got_extra_bits){ - s->extra_sc.crc = crc_extra_bits; - s->extra_sc.bits_used = get_bits_count(&s->gb_extra_bits); - } - } + return count * 2; } @@ -635,8 +618,7 @@ static inline int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb, void float *dstfl = dst; const int channel_stride = s->avctx->channels; - if(s->samples_left == s->samples) - s->one = s->zero = s->zeroes = 0; + s->one = s->zero = s->zeroes = 0; do{ T = wv_get_value(s, gb, 0, &last); S = 0; @@ -675,13 +657,8 @@ static inline int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb, void dst16 += channel_stride; } count++; - }while(!last && count < s->max_samples); + } while (!last && count < s->samples); - if (last) - s->samples_left = 0; - else - s->samples_left -= count; - if(!s->samples_left){ wv_reset_saved_context(s); if(crc != s->CRC){ av_log(s->avctx, AV_LOG_ERROR, "CRC error\n"); @@ -691,15 +668,7 @@ static inline int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb, void av_log(s->avctx, AV_LOG_ERROR, "Extra bits CRC error\n"); return -1; } - }else{ - s->pos = pos; - s->sc.crc = crc; - s->sc.bits_used = get_bits_count(&s->gb); - if(s->got_extra_bits){ - s->extra_sc.crc = crc_extra_bits; - s->extra_sc.bits_used = get_bits_count(&s->gb_extra_bits); - } - } + return count; } @@ -743,6 +712,9 @@ static av_cold int wavpack_decode_init(AVCodecContext *avctx) s->fdec_num = 0; + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -759,7 +731,7 @@ static av_cold int wavpack_decode_end(AVCodecContext *avctx) } static int wavpack_decode_block(AVCodecContext *avctx, int block_no, - void *data, int *data_size, + void *data, int *got_frame_ptr, const uint8_t *buf, int buf_size) { WavpackContext *wc = avctx->priv_data; @@ -774,7 +746,7 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no, int bpp, chan, chmask; if (buf_size == 0){ - *data_size = 0; + *got_frame_ptr = 0; return 0; } @@ -789,18 +761,16 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no, return -1; } - if(!s->samples_left){ memset(s->decorr, 0, MAX_TERMS * sizeof(Decorr)); memset(s->ch, 0, sizeof(s->ch)); s->extra_bits = 0; s->and = s->or = s->shift = 0; s->got_extra_bits = 0; - } if(!wc->mkv_mode){ s->samples = AV_RL32(buf); buf += 4; if(!s->samples){ - *data_size = 0; + *got_frame_ptr = 0; return 0; } }else{ @@ -829,13 +799,6 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no, wc->ch_offset += 1 + s->stereo; - s->max_samples = *data_size / (bpp * avctx->channels); - s->max_samples = FFMIN(s->max_samples, s->samples); - if(s->samples_left > 0){ - s->max_samples = FFMIN(s->max_samples, s->samples_left); - buf = buf_end; - } - // parse metadata blocks while(buf < buf_end){ id = *buf++; @@ -1064,7 +1027,7 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no, } if(id & WP_IDF_ODD) buf++; } - if(!s->samples_left){ + if(!got_terms){ av_log(avctx, AV_LOG_ERROR, "No block with decorrelation terms\n"); return -1; @@ -1101,16 +1064,6 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no, s->got_extra_bits = 0; } } - s->samples_left = s->samples; - }else{ - init_get_bits(&s->gb, orig_buf + s->sc.offset, s->sc.size); - skip_bits_long(&s->gb, s->sc.bits_used); - if(s->got_extra_bits){ - init_get_bits(&s->gb_extra_bits, orig_buf + s->extra_sc.offset, - s->extra_sc.size); - skip_bits_long(&s->gb_extra_bits, s->extra_sc.bits_used); - } - } if(s->stereo_in){ if(avctx->sample_fmt == AV_SAMPLE_FMT_S16) @@ -1167,7 +1120,7 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no, } } - wc->samples_left = s->samples_left; + *got_frame_ptr = 1; return samplecount * bpp; } @@ -1181,23 +1134,40 @@ static void wavpack_decode_flush(AVCodecContext *avctx) wv_reset_saved_context(s->fdec[i]); } -static int wavpack_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int wavpack_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { WavpackContext *s = avctx->priv_data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; - int frame_size; + int frame_size, ret; int samplecount = 0; s->block = 0; - s->samples_left = 0; s->ch_offset = 0; + /* determine number of samples */ if(s->mkv_mode){ s->samples = AV_RL32(buf); buf += 4; + } else { + if (s->multichannel) + s->samples = AV_RL32(buf + 4); + else + s->samples = AV_RL32(buf); + } + if (s->samples <= 0) { + av_log(avctx, AV_LOG_ERROR, "Invalid number of samples: %d\n", + s->samples); + return AVERROR(EINVAL); + } + + /* get output buffer */ + s->frame.nb_samples = s->samples; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + while(buf_size > 0){ if(!s->multichannel){ frame_size = buf_size; @@ -1216,17 +1186,19 @@ static int wavpack_decode_frame(AVCodecContext *avctx, wavpack_decode_flush(avctx); return -1; } - if((samplecount = wavpack_decode_block(avctx, s->block, data, - data_size, buf, frame_size)) < 0) { + if((samplecount = wavpack_decode_block(avctx, s->block, s->frame.data[0], + got_frame_ptr, buf, frame_size)) < 0) { wavpack_decode_flush(avctx); return -1; } s->block++; buf += frame_size; buf_size -= frame_size; } - *data_size = samplecount * avctx->channels; - return s->samples_left > 0 ? 0 : avpkt->size; + if (*got_frame_ptr) + *(AVFrame *)data = s->frame; + + return avpkt->size; } AVCodec ff_wavpack_decoder = { @@ -1238,6 +1210,6 @@ AVCodec ff_wavpack_decoder = { .close = wavpack_decode_end, .decode = wavpack_decode_frame, .flush = wavpack_decode_flush, - .capabilities = CODEC_CAP_SUBFRAMES, + .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("WavPack"), }; diff --git a/libavcodec/wma.h b/libavcodec/wma.h index fd9d5d8d3b..6c8e944b79 100644 --- a/libavcodec/wma.h +++ b/libavcodec/wma.h @@ -65,6 +65,7 @@ typedef struct CoefVLCTable { typedef struct WMACodecContext { AVCodecContext* avctx; + AVFrame frame; GetBitContext gb; PutBitContext pb; int sample_rate; diff --git a/libavcodec/wmadec.c b/libavcodec/wmadec.c index 40315d4291..0b2e49981d 100644 --- a/libavcodec/wmadec.c +++ b/libavcodec/wmadec.c @@ -136,6 +136,10 @@ static int wma_decode_init(AVCodecContext * avctx) } avctx->sample_fmt = AV_SAMPLE_FMT_S16; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -814,14 +818,13 @@ static int wma_decode_frame(WMACodecContext *s, int16_t *samples) return 0; } -static int wma_decode_superframe(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int wma_decode_superframe(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; WMACodecContext *s = avctx->priv_data; - int nb_frames, bit_offset, i, pos, len, out_size; + int nb_frames, bit_offset, i, pos, len, ret; uint8_t *q; int16_t *samples; @@ -836,8 +839,6 @@ static int wma_decode_superframe(AVCodecContext *avctx, if(s->block_align) buf_size = s->block_align; - samples = data; - init_get_bits(&s->gb, buf, buf_size*8); if (s->use_bit_reservoir) { @@ -848,12 +849,13 @@ static int wma_decode_superframe(AVCodecContext *avctx, nb_frames = 1; } - 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; + /* get output buffer */ + s->frame.nb_samples = nb_frames * s->frame_len; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; } + samples = (int16_t *)s->frame.data[0]; if (s->use_bit_reservoir) { bit_offset = get_bits(&s->gb, s->byte_offset_bits + 3); @@ -920,7 +922,10 @@ 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 = out_size; + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + return buf_size; fail: /* when error, we reset the bit reservoir */ @@ -945,6 +950,7 @@ AVCodec ff_wmav1_decoder = { .close = ff_wma_end, .decode = wma_decode_superframe, .flush = flush, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"), }; @@ -957,5 +963,6 @@ AVCodec ff_wmav2_decoder = { .close = ff_wma_end, .decode = wma_decode_superframe, .flush = flush, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"), }; diff --git a/libavcodec/wmaprodec.c b/libavcodec/wmaprodec.c index 868a28393d..3f022c7208 100644 --- a/libavcodec/wmaprodec.c +++ b/libavcodec/wmaprodec.c @@ -167,6 +167,7 @@ typedef struct { typedef struct WMAProDecodeCtx { /* generic decoder variables */ AVCodecContext* avctx; ///< codec context for av_log + AVFrame frame; ///< AVFrame for decoded output DSPContext dsp; ///< accelerated DSP functions FmtConvertContext fmt_conv; uint8_t frame_data[MAX_FRAMESIZE + @@ -209,8 +210,6 @@ typedef struct WMAProDecodeCtx { uint32_t frame_num; ///< current frame number (not used for decoding) GetBitContext gb; ///< bitstream reader context int buf_bit_size; ///< buffer size in bits - float* samples; ///< current samplebuffer pointer - float* samples_end; ///< maximum samplebuffer pointer uint8_t drc_gain; ///< gain for the DRC tool int8_t skip_frame; ///< skip output step int8_t parsed_all_subframes; ///< all subframes decoded? @@ -453,6 +452,10 @@ static av_cold int decode_init(AVCodecContext *avctx) dump_context(s); avctx->channel_layout = channel_mask; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } @@ -1279,22 +1282,15 @@ static int decode_subframe(WMAProDecodeCtx *s) *@return 0 if the trailer bit indicates that this is the last frame, * 1 if there are additional frames */ -static int decode_frame(WMAProDecodeCtx *s) +static int decode_frame(WMAProDecodeCtx *s, int *got_frame_ptr) { + AVCodecContext *avctx = s->avctx; GetBitContext* gb = &s->gb; int more_frames = 0; int len = 0; - int i; + int i, ret; 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) { - /** return an error if no frame could be decoded at all */ - av_log(s->avctx, AV_LOG_ERROR, - "not enough space for the output samples\n"); - s->packet_loss = 1; - return 0; - } + float *samples; /** get frame length */ if (s->len_prefix) @@ -1360,10 +1356,19 @@ static int decode_frame(WMAProDecodeCtx *s) } } + /* get output buffer */ + s->frame.nb_samples = s->samples_per_frame; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + s->packet_loss = 1; + return 0; + } + samples = (float *)s->frame.data[0]; + /** interleave samples and write them to the output buffer */ 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->fmt_conv.float_interleave(samples, out_ptr, s->samples_per_frame, s->num_channels); for (i = 0; i < s->num_channels; i++) { @@ -1375,8 +1380,10 @@ static int decode_frame(WMAProDecodeCtx *s) if (s->skip_frame) { s->skip_frame = 0; - } else - s->samples += s->num_channels * s->samples_per_frame; + *got_frame_ptr = 0; + } else { + *got_frame_ptr = 1; + } if (s->len_prefix) { if (len != (get_bits_count(gb) - s->frame_offset) + 2) { @@ -1473,8 +1480,8 @@ static void save_bits(WMAProDecodeCtx *s, GetBitContext* gb, int len, *@param avpkt input packet *@return number of bytes that were read from the input buffer */ -static int decode_packet(AVCodecContext *avctx, - void *data, int *data_size, AVPacket* avpkt) +static int decode_packet(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket* avpkt) { WMAProDecodeCtx *s = avctx->priv_data; GetBitContext* gb = &s->pgb; @@ -1483,9 +1490,7 @@ static int decode_packet(AVCodecContext *avctx, int num_bits_prev_frame; int packet_sequence_number; - s->samples = data; - s->samples_end = (float*)((int8_t*)data + *data_size); - *data_size = 0; + *got_frame_ptr = 0; if (s->packet_done || s->packet_loss) { s->packet_done = 0; @@ -1532,7 +1537,7 @@ static int decode_packet(AVCodecContext *avctx, /** decode the cross packet frame if it is valid */ if (!s->packet_loss) - decode_frame(s); + decode_frame(s, got_frame_ptr); } else if (s->num_saved_bits - s->frame_offset) { av_dlog(avctx, "ignoring %x previously saved bits\n", s->num_saved_bits - s->frame_offset); @@ -1555,7 +1560,7 @@ static int decode_packet(AVCodecContext *avctx, (frame_size = show_bits(gb, s->log2_frame_size)) && frame_size <= remaining_bits(s, gb)) { save_bits(s, gb, frame_size, 0); - s->packet_done = !decode_frame(s); + s->packet_done = !decode_frame(s, got_frame_ptr); } else if (!s->len_prefix && s->num_saved_bits > get_bits_count(&s->gb)) { /** when the frames do not have a length prefix, we don't know @@ -1565,7 +1570,7 @@ static int decode_packet(AVCodecContext *avctx, therefore we save the incoming packet first, then we append the "previous frame" data from the next packet so that we get a buffer that only contains full frames */ - s->packet_done = !decode_frame(s); + s->packet_done = !decode_frame(s, got_frame_ptr); } else s->packet_done = 1; } @@ -1577,10 +1582,14 @@ static int decode_packet(AVCodecContext *avctx, save_bits(s, gb, remaining_bits(s, gb), 0); } - *data_size = (int8_t *)s->samples - (int8_t *)data; s->packet_offset = get_bits_count(gb) & 7; + if (s->packet_loss) + return AVERROR_INVALIDDATA; + + if (*got_frame_ptr) + *(AVFrame *)data = s->frame; - return (s->packet_loss) ? AVERROR_INVALIDDATA : get_bits_count(gb) >> 3; + return get_bits_count(gb) >> 3; } /** @@ -1611,7 +1620,7 @@ AVCodec ff_wmapro_decoder = { .init = decode_init, .close = decode_end, .decode = decode_packet, - .capabilities = CODEC_CAP_SUBFRAMES, + .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, .flush= flush, .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 9 Professional"), }; diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index 244b630922..45383b033b 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -131,6 +131,7 @@ typedef struct { * @name Global values specified in the stream header / extradata or used all over. * @{ */ + AVFrame frame; GetBitContext gb; ///< packet bitreader. During decoder init, ///< it contains the extradata from the ///< demuxer. During decoding, it contains @@ -438,6 +439,9 @@ static av_cold int wmavoice_decode_init(AVCodecContext *ctx) ctx->sample_fmt = AV_SAMPLE_FMT_FLT; + avcodec_get_frame_defaults(&s->frame); + ctx->coded_frame = &s->frame; + return 0; } @@ -1725,17 +1729,17 @@ static int check_bits_for_superframe(GetBitContext *orig_gb, * @return 0 on success, <0 on error or 1 if there was not enough data to * fully parse the superframe */ -static int synth_superframe(AVCodecContext *ctx, - float *samples, int *data_size) +static int synth_superframe(AVCodecContext *ctx, int *got_frame_ptr) { WMAVoiceContext *s = ctx->priv_data; GetBitContext *gb = &s->gb, s_gb; - int n, res, out_size, n_samples = 480; + int n, res, 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]; float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; float synth[MAX_LSPS + MAX_SFRAMESIZE]; + float *samples; memcpy(synth, s->synth_history, s->lsps * sizeof(*synth)); @@ -1749,7 +1753,7 @@ static int synth_superframe(AVCodecContext *ctx, } if ((res = check_bits_for_superframe(gb, s)) == 1) { - *data_size = 0; + *got_frame_ptr = 0; return 1; } @@ -1792,13 +1796,14 @@ 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 - %d needed)\n", - *data_size, out_size); - return -1; + /* get output buffer */ + s->frame.nb_samples = 480; + if ((res = ctx->get_buffer(ctx, &s->frame)) < 0) { + av_log(ctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return res; } + s->frame.nb_samples = n_samples; + samples = (float *)s->frame.data[0]; /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */ for (n = 0; n < 3; n++) { @@ -1820,7 +1825,7 @@ static int synth_superframe(AVCodecContext *ctx, lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], &excitation[s->history_nsamples + n * MAX_FRAMESIZE], &synth[s->lsps + n * MAX_FRAMESIZE]))) { - *data_size = 0; + *got_frame_ptr = 0; return res; } } @@ -1833,8 +1838,7 @@ static int synth_superframe(AVCodecContext *ctx, skip_bits(gb, 10 * (res + 1)); } - /* Specify nr. of output samples */ - *data_size = out_size; + *got_frame_ptr = 1; /* Update history */ memcpy(s->prev_lsps, lsps[2], @@ -1922,7 +1926,7 @@ static void copy_bits(PutBitContext *pb, * For more information about frames, see #synth_superframe(). */ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { WMAVoiceContext *s = ctx->priv_data; GetBitContext *gb = &s->gb; @@ -1935,7 +1939,7 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, * capping the packet size at ctx->block_align. */ for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); if (!size) { - *data_size = 0; + *got_frame_ptr = 0; return 0; } init_get_bits(&s->gb, avpkt->data, size << 3); @@ -1956,10 +1960,11 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); flush_put_bits(&s->pb); s->sframe_cache_size += s->spillover_nbits; - if ((res = synth_superframe(ctx, data, data_size)) == 0 && - *data_size > 0) { + if ((res = synth_superframe(ctx, got_frame_ptr)) == 0 && + *got_frame_ptr) { cnt += s->spillover_nbits; s->skip_bits_next = cnt & 7; + *(AVFrame *)data = s->frame; return cnt >> 3; } else skip_bits_long (gb, s->spillover_nbits - cnt + @@ -1974,11 +1979,12 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, s->sframe_cache_size = 0; s->skip_bits_next = 0; pos = get_bits_left(gb); - if ((res = synth_superframe(ctx, data, data_size)) < 0) { + if ((res = synth_superframe(ctx, got_frame_ptr)) < 0) { return res; - } else if (*data_size > 0) { + } else if (*got_frame_ptr) { int cnt = get_bits_count(gb); s->skip_bits_next = cnt & 7; + *(AVFrame *)data = s->frame; return cnt >> 3; } else if ((s->sframe_cache_size = pos) > 0) { /* rewind bit reader to start of last (incomplete) superframe... */ @@ -2046,7 +2052,7 @@ AVCodec ff_wmavoice_decoder = { .init = wmavoice_decode_init, .close = wmavoice_decode_end, .decode = wmavoice_decode_packet, - .capabilities = CODEC_CAP_SUBFRAMES, + .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, .flush = wmavoice_flush, .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), }; diff --git a/libavcodec/ws-snd1.c b/libavcodec/ws-snd1.c index 17b8cbfa3a..b7e28f5e56 100644 --- a/libavcodec/ws-snd1.c +++ b/libavcodec/ws-snd1.c @@ -37,26 +37,37 @@ static const int8_t ws_adpcm_4bit[] = { 0, 1, 2, 3, 4, 5, 6, 8 }; +typedef struct WSSndContext { + AVFrame frame; +} WSSndContext; + static av_cold int ws_snd_decode_init(AVCodecContext *avctx) { + WSSndContext *s = avctx->priv_data; + if (avctx->channels != 1) { av_log_ask_for_sample(avctx, "unsupported number of channels\n"); return AVERROR(EINVAL); } avctx->sample_fmt = AV_SAMPLE_FMT_U8; + + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + return 0; } static int ws_snd_decode_frame(AVCodecContext *avctx, void *data, - int *data_size, AVPacket *avpkt) + int *got_frame_ptr, AVPacket *avpkt) { + WSSndContext *s = avctx->priv_data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; - int in_size, out_size; + int in_size, out_size, ret; int sample = 128; - uint8_t *samples = data; + uint8_t *samples; uint8_t *samples_end; if (!buf_size) @@ -71,19 +82,24 @@ static int ws_snd_decode_frame(AVCodecContext *avctx, void *data, in_size = AV_RL16(&buf[2]); buf += 4; - if (out_size > *data_size) { - av_log(avctx, AV_LOG_ERROR, "Frame is too large to fit in buffer\n"); - return -1; - } if (in_size > buf_size) { av_log(avctx, AV_LOG_ERROR, "Frame data is larger than input buffer\n"); return -1; } + + /* get output buffer */ + s->frame.nb_samples = out_size; + if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + samples = s->frame.data[0]; samples_end = samples + out_size; if (in_size == out_size) { memcpy(samples, buf, out_size); - *data_size = out_size; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; return buf_size; } @@ -159,7 +175,9 @@ static int ws_snd_decode_frame(AVCodecContext *avctx, void *data, } } - *data_size = samples - (uint8_t *)data; + s->frame.nb_samples = samples - s->frame.data[0]; + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; return buf_size; } @@ -168,7 +186,9 @@ AVCodec ff_ws_snd1_decoder = { .name = "ws_snd1", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_WESTWOOD_SND1, + .priv_data_size = sizeof(WSSndContext), .init = ws_snd_decode_init, .decode = ws_snd_decode_frame, + .capabilities = CODEC_CAP_DR1, .long_name = NULL_IF_CONFIG_SMALL("Westwood Audio (SND1)"), }; diff --git a/libavformat/adtsenc.c b/libavformat/adtsenc.c index ed1d913347..18a055a1e5 100644 --- a/libavformat/adtsenc.c +++ b/libavformat/adtsenc.c @@ -37,7 +37,7 @@ int ff_adts_decode_extradata(AVFormatContext *s, ADTSContext *adts, uint8_t *buf int off; init_get_bits(&gb, buf, size * 8); - off = avpriv_mpeg4audio_get_config(&m4ac, buf, size); + off = avpriv_mpeg4audio_get_config(&m4ac, buf, size * 8, 1); if (off < 0) return off; skip_bits_long(&gb, off); diff --git a/libavformat/asfdec.c b/libavformat/asfdec.c index 167ab03e8c..0bb210bcd6 100644 --- a/libavformat/asfdec.c +++ b/libavformat/asfdec.c @@ -1182,7 +1182,7 @@ static int64_t asf_read_pts(AVFormatContext *s, int stream_index, int64_t *ppos, return AV_NOPTS_VALUE; } - pts= pkt->dts; + pts = pkt->dts; av_free_packet(pkt); if(pkt->flags&AV_PKT_FLAG_KEY){ diff --git a/libavformat/flvdec.c b/libavformat/flvdec.c index 4bca6a6580..65c30da1ba 100644 --- a/libavformat/flvdec.c +++ b/libavformat/flvdec.c @@ -550,7 +550,7 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt) if (st->codec->codec_id == CODEC_ID_AAC) { MPEG4AudioConfig cfg; avpriv_mpeg4audio_get_config(&cfg, st->codec->extradata, - st->codec->extradata_size); + st->codec->extradata_size * 8, 1); st->codec->channels = cfg.channels; if (cfg.ext_sample_rate) st->codec->sample_rate = cfg.ext_sample_rate; diff --git a/libavformat/isom.c b/libavformat/isom.c index 457cc117e2..8e5e14514d 100644 --- a/libavformat/isom.c +++ b/libavformat/isom.c @@ -438,7 +438,7 @@ int ff_mp4_read_dec_config_descr(AVFormatContext *fc, AVStream *st, AVIOContext if (st->codec->codec_id == CODEC_ID_AAC) { MPEG4AudioConfig cfg; avpriv_mpeg4audio_get_config(&cfg, st->codec->extradata, - st->codec->extradata_size); + st->codec->extradata_size * 8, 1); st->codec->channels = cfg.channels; if (cfg.object_type == 29 && cfg.sampling_index < 3) // old mp3on4 st->codec->sample_rate = avpriv_mpa_freq_tab[cfg.sampling_index]; diff --git a/libavformat/latmenc.c b/libavformat/latmenc.c index 56522106d6..c71f78b78a 100644 --- a/libavformat/latmenc.c +++ b/libavformat/latmenc.c @@ -55,7 +55,7 @@ static int latm_decode_extradata(LATMContext *ctx, uint8_t *buf, int size) MPEG4AudioConfig m4ac; init_get_bits(&gb, buf, size * 8); - ctx->off = avpriv_mpeg4audio_get_config(&m4ac, buf, size); + ctx->off = avpriv_mpeg4audio_get_config(&m4ac, buf, size * 8, 1); if (ctx->off < 0) return ctx->off; skip_bits_long(&gb, ctx->off); diff --git a/libavformat/matroskaenc.c b/libavformat/matroskaenc.c index 332d1ca572..b8c4667a4e 100644 --- a/libavformat/matroskaenc.c +++ b/libavformat/matroskaenc.c @@ -448,7 +448,8 @@ static void get_aac_sample_rates(AVFormatContext *s, AVCodecContext *codec, int { MPEG4AudioConfig mp4ac; - if (avpriv_mpeg4audio_get_config(&mp4ac, codec->extradata, codec->extradata_size) < 0) { + if (avpriv_mpeg4audio_get_config(&mp4ac, codec->extradata, + codec->extradata_size * 8, 1) < 0) { av_log(s, AV_LOG_WARNING, "Error parsing AAC extradata, unable to determine samplerate.\n"); return; } diff --git a/libavformat/nullenc.c b/libavformat/nullenc.c index 3f4107923f..9edbf20fc8 100644 --- a/libavformat/nullenc.c +++ b/libavformat/nullenc.c @@ -32,5 +32,5 @@ AVOutputFormat ff_null_muxer = { .audio_codec = AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE), .video_codec = CODEC_ID_RAWVIDEO, .write_packet = null_write_packet, - .flags = AVFMT_NOFILE | AVFMT_NOTIMESTAMPS, + .flags = AVFMT_NOFILE | AVFMT_NOTIMESTAMPS | AVFMT_RAWPICTURE, }; diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index 24be912045..46315a923a 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -1934,6 +1934,7 @@ static int rtp_read_header(AVFormatContext *s, struct sockaddr_storage addr; AVIOContext pb; socklen_t addrlen = sizeof(addr); + RTSPState *rt = s->priv_data; if (!ff_network_init()) return AVERROR(EIO); @@ -1997,6 +1998,8 @@ static int rtp_read_header(AVFormatContext *s, /* sdp_read_header initializes this again */ ff_network_close(); + rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1; + ret = sdp_read_header(s, ap); s->pb = NULL; return ret; |