December 2016 - ffmpeg-cvslog

lavfi/selectivecolor: simplify crazy mid val computations
by git＠videolan.org 27 Dec '16

27 Dec '16

ffmpeg | branch: master | Clément Bœsch <u(a)pkh.me> | Tue Dec 27 23:43:59 2016 +0100| [afaaf8db1847ddd7d291c218edcba5cd4bf2441e] | committer: Clément Bœsch lavfi/selectivecolor: simplify crazy mid val computations > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=afaaf8db1847ddd7d… --- libavfilter/vf_selectivecolor.c | 33 +++------------------------------ 1 file changed, 3 insertions(+), 30 deletions(-) diff --git a/libavfilter/vf_selectivecolor.c b/libavfilter/vf_selectivecolor.c index a5cb533..26fd8eb 100644 --- a/libavfilter/vf_selectivecolor.c +++ b/libavfilter/vf_selectivecolor.c @@ -28,6 +28,7 @@ #include "libavutil/intreadwrite.h" #include "libavutil/opt.h" #include "libavutil/pixdesc.h" +#include "libavcodec/mathops.h" // for mid_pred(), which is a macro so no link dependency #include "avfilter.h" #include "drawutils.h" #include "formats.h" @@ -112,42 +113,14 @@ static const AVOption selectivecolor_options[] = { AVFILTER_DEFINE_CLASS(selectivecolor); -static inline int get_mid_val(int r, int g, int b) -{ - if ((r < g && r > b) || (r < b && r > g)) return r; - if ((g < r && g > b) || (g < b && g > r)) return g; - if ((b < r && b > g) || (b < g && b > r)) return b; - return -1; -} - static int get_rgb_adjust_range(int r, int g, int b, int min_val, int max_val) { - // max - mid - const int mid_val = get_mid_val(r, g, b); - if (mid_val == -1) { - // XXX: can be simplified - if ((r != min_val && g == min_val && b == min_val) || - (r == min_val && g != min_val && b == min_val) || - (r == min_val && g == min_val && b != min_val)) - return max_val - min_val; - return 0; - } - return max_val - mid_val; + return max_val - mid_pred(r, g, b); } static int get_cmy_adjust_range(int r, int g, int b, int min_val, int max_val) { - // mid - min - const int mid_val = get_mid_val(r, g, b); - if (mid_val == -1) { - // XXX: refactor with rgb - if ((r != max_val && g == max_val && b == max_val) || - (r == max_val && g != max_val && b == max_val) || - (r == max_val && g == max_val && b != max_val)) - return max_val - min_val; - return 0; - } - return mid_val - min_val; + return mid_pred(r, g, b) - min_val; } #define DECLARE_ADJUST_RANGE_FUNCS(nbits) \

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lavfi/transpose: add missing const options flags
by git＠videolan.org 27 Dec '16

27 Dec '16

ffmpeg | branch: master | Clément Bœsch <u(a)pkh.me> | Tue Dec 27 09:36:18 2016 +0100| [571a36015738b081dea1893fd7e5f193a51d6438] | committer: Clément Bœsch lavfi/transpose: add missing const options flags Fixes ffmpeg -h filter=transpose > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=571a36015738b081d… --- libavfilter/vf_transpose.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/libavfilter/vf_transpose.c b/libavfilter/vf_transpose.c index 28226c3..75b4dda 100644 --- a/libavfilter/vf_transpose.c +++ b/libavfilter/vf_transpose.c @@ -260,10 +260,10 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) static const AVOption transpose_options[] = { { "dir", "set transpose direction", OFFSET(dir), AV_OPT_TYPE_INT, { .i64 = TRANSPOSE_CCLOCK_FLIP }, 0, 7, FLAGS, "dir" }, - { "cclock_flip", "rotate counter-clockwise with vertical flip", 0, AV_OPT_TYPE_CONST, { .i64 = TRANSPOSE_CCLOCK_FLIP }, .unit = "dir" }, - { "clock", "rotate clockwise", 0, AV_OPT_TYPE_CONST, { .i64 = TRANSPOSE_CLOCK }, .unit = "dir" }, - { "cclock", "rotate counter-clockwise", 0, AV_OPT_TYPE_CONST, { .i64 = TRANSPOSE_CCLOCK }, .unit = "dir" }, - { "clock_flip", "rotate clockwise with vertical flip", 0, AV_OPT_TYPE_CONST, { .i64 = TRANSPOSE_CLOCK_FLIP }, .unit = "dir" }, + { "cclock_flip", "rotate counter-clockwise with vertical flip", 0, AV_OPT_TYPE_CONST, { .i64 = TRANSPOSE_CCLOCK_FLIP }, .flags=FLAGS, .unit = "dir" }, + { "clock", "rotate clockwise", 0, AV_OPT_TYPE_CONST, { .i64 = TRANSPOSE_CLOCK }, .flags=FLAGS, .unit = "dir" }, + { "cclock", "rotate counter-clockwise", 0, AV_OPT_TYPE_CONST, { .i64 = TRANSPOSE_CCLOCK }, .flags=FLAGS, .unit = "dir" }, + { "clock_flip", "rotate clockwise with vertical flip", 0, AV_OPT_TYPE_CONST, { .i64 = TRANSPOSE_CLOCK_FLIP }, .flags=FLAGS, .unit = "dir" }, { "passthrough", "do not apply transposition if the input matches the specified geometry", OFFSET(passthrough), AV_OPT_TYPE_INT, {.i64=TRANSPOSE_PT_TYPE_NONE}, 0, INT_MAX, FLAGS, "passthrough" },

1 0

MAINTAINERS: update
by git＠videolan.org 27 Dec '16

27 Dec '16

ffmpeg | branch: master | Christophe Gisquet <christophe.gisquet(a)gmail.com> | Tue Dec 27 09:18:51 2016 +0000| [e3312b3746ef364c3101381bada57f8ef75f175a] | committer: Michael Niedermayer MAINTAINERS: update > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e3312b3746ef364c3… --- MAINTAINERS | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index e718463..cb9516d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -176,7 +176,7 @@ Codecs: h263* Michael Niedermayer h264* Loren Merritt, Michael Niedermayer hap* Tom Butterworth - huffyuv* Michael Niedermayer, Christophe Gisquet + huffyuv* Michael Niedermayer idcinvideo.c Mike Melanson interplayvideo.c Mike Melanson jni*, ffjni* Matthieu Bouron @@ -223,7 +223,6 @@ Codecs: rpza.c Roberto Togni rtjpeg.c, rtjpeg.h Reimar Doeffinger rv10.c Michael Niedermayer - rv4* Christophe Gisquet s3tc* Ivo van Poorten smc.c Mike Melanson smvjpegdec.c Ash Hughes @@ -237,7 +236,6 @@ Codecs: tta.c Alex Beregszaszi, Jaikrishnan Menon ttaenc.c Paul B Mahol txd.c Ivo van Poorten - vc1* Christophe Gisquet vc2* Rostislav Pehlivanov vcr1.c Michael Niedermayer vda_h264_dec.c Xidorn Quan

1 0

wmavoice: protect against zero-energy in adaptive gain control.
by git＠videolan.org 27 Dec '16

27 Dec '16

ffmpeg | branch: master | Ronald S. Bultje <rsbultje(a)gmail.com> | Tue Dec 20 18:01:05 2016 -0500| [33d7f822f8ed2d1870babc1d04d4d48cf8b6f240] | committer: Ronald S. Bultje wmavoice: protect against zero-energy in adaptive gain control. Otherwise the scale factor becomes NaN, resulting in corrupt output. Fixes #5426. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=33d7f822f8ed2d187… --- libavcodec/wmavoice.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index 90dfe20..cd5958c 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -512,7 +512,8 @@ static void adaptive_gain_control(float *out, const float *in, speech_energy += fabsf(speech_synth[i]); postfilter_energy += fabsf(in[i]); } - gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy; + gain_scale_factor = postfilter_energy == 0.0 ? 0.0 : + (1.0 - alpha) * speech_energy / postfilter_energy; for (i = 0; i < size; i++) { mem = alpha * mem + gain_scale_factor;

1 0

wmavoice: reindent.
by git＠videolan.org 27 Dec '16

27 Dec '16

ffmpeg | branch: master | Ronald S. Bultje <rsbultje(a)gmail.com> | Tue Dec 20 17:14:41 2016 -0500| [b011bb5f8b2ce5f21bc4f07f50a56a26310383af] | committer: Ronald S. Bultje wmavoice: reindent. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b011bb5f8b2ce5f21… --- libavcodec/wmavoice.c | 72 +++++++++++++++++++++++++-------------------------- 1 file changed, 36 insertions(+), 36 deletions(-) diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index ae100fb..f31c9d2 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -1915,29 +1915,29 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, /* If the packet header specifies a s->spillover_nbits, then we want * to push out all data of the previous packet (+ spillover) before * continuing to parse new superframes in the current packet. */ - if (s->sframe_cache_size > 0) { - int cnt = get_bits_count(gb); - 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, got_frame_ptr)) == 0 && - *got_frame_ptr) { - cnt += s->spillover_nbits; - s->skip_bits_next = cnt & 7; - res = cnt >> 3; - if (res > avpkt->size) { - av_log(ctx, AV_LOG_ERROR, - "Trying to skip %d bytes in packet of size %d\n", - res, avpkt->size); - return AVERROR_INVALIDDATA; - } - return res; - } else - skip_bits_long (gb, s->spillover_nbits - cnt + - get_bits_count(gb)); // resync - } else if (s->spillover_nbits) { - skip_bits_long(gb, s->spillover_nbits); // resync - } + if (s->sframe_cache_size > 0) { + int cnt = get_bits_count(gb); + 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, got_frame_ptr)) == 0 && + *got_frame_ptr) { + cnt += s->spillover_nbits; + s->skip_bits_next = cnt & 7; + res = cnt >> 3; + if (res > avpkt->size) { + av_log(ctx, AV_LOG_ERROR, + "Trying to skip %d bytes in packet of size %d\n", + res, avpkt->size); + return AVERROR_INVALIDDATA; + } + return res; + } else + skip_bits_long (gb, s->spillover_nbits - cnt + + get_bits_count(gb)); // resync + } else if (s->spillover_nbits) { + skip_bits_long(gb, s->spillover_nbits); // resync + } } else if (s->skip_bits_next) skip_bits(gb, s->skip_bits_next); @@ -1949,20 +1949,20 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, *got_frame_ptr = 0; return size; } else if (s->nb_superframes > 0) { - if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) { - return res; - } else if (*got_frame_ptr) { - int cnt = get_bits_count(gb); - s->skip_bits_next = cnt & 7; - res = cnt >> 3; - if (res > avpkt->size) { - av_log(ctx, AV_LOG_ERROR, - "Trying to skip %d bytes in packet of size %d\n", - res, avpkt->size); - return AVERROR_INVALIDDATA; + if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) { + return res; + } else if (*got_frame_ptr) { + int cnt = get_bits_count(gb); + s->skip_bits_next = cnt & 7; + res = cnt >> 3; + if (res > avpkt->size) { + av_log(ctx, AV_LOG_ERROR, + "Trying to skip %d bytes in packet of size %d\n", + res, avpkt->size); + return AVERROR_INVALIDDATA; + } + return res; } - return res; - } } else if ((s->sframe_cache_size = pos) > 0) { /* ... cache it for spillover in next packet */ init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);

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wmavoice: move overflow handling to common code.
by git＠videolan.org 27 Dec '16

27 Dec '16

ffmpeg | branch: master | Ronald S. Bultje <rsbultje(a)gmail.com> | Tue Dec 20 17:10:33 2016 -0500| [7b27dd5c16de785297ce4de4b88afa0b6685f61d] | committer: Ronald S. Bultje wmavoice: move overflow handling to common code. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7b27dd5c16de78529… --- libavcodec/wmavoice.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index f31c9d2..90dfe20 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -1800,6 +1800,11 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, skip_bits(gb, 10 * (res + 1)); } + if (get_bits_left(gb) < 0) { + wmavoice_flush(ctx); + return AVERROR_INVALIDDATA; + } + *got_frame_ptr = 1; /* Update history */ @@ -1925,12 +1930,6 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, cnt += s->spillover_nbits; s->skip_bits_next = cnt & 7; res = cnt >> 3; - if (res > avpkt->size) { - av_log(ctx, AV_LOG_ERROR, - "Trying to skip %d bytes in packet of size %d\n", - res, avpkt->size); - return AVERROR_INVALIDDATA; - } return res; } else skip_bits_long (gb, s->spillover_nbits - cnt + @@ -1955,12 +1954,6 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, int cnt = get_bits_count(gb); s->skip_bits_next = cnt & 7; res = cnt >> 3; - if (res > avpkt->size) { - av_log(ctx, AV_LOG_ERROR, - "Trying to skip %d bytes in packet of size %d\n", - res, avpkt->size); - return AVERROR_INVALIDDATA; - } return res; } } else if ((s->sframe_cache_size = pos) > 0) {

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wmavoice: disable bitstream checking.
by git＠videolan.org 27 Dec '16

27 Dec '16

ffmpeg | branch: master | Ronald S. Bultje <rsbultje(a)gmail.com> | Tue Dec 20 17:14:26 2016 -0500| [3deb4b54a24f8cddce463d9f5751b01efeb976af] | committer: Ronald S. Bultje wmavoice: disable bitstream checking. The checked bitstream reader does that already. To allow parsing of superframes split over a packet boundary, we always decode the last superframe in each packet at the start of the next packet, even if theoretically we could have decoded it. The last superframe in the last packet is decoded using AV_CODEC_CAP_DELAY. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=3deb4b54a24f8cddc… --- libavcodec/wmavoice.c | 144 ++++++++++---------------------------------------- 1 file changed, 29 insertions(+), 115 deletions(-) diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index 4b3ab43..ae100fb 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -251,6 +251,7 @@ typedef struct WMAVoiceContext { int frame_cntr; ///< current frame index [0 - 0xFFFE]; is ///< only used for comfort noise in #pRNG() + int nb_superframes; ///< number of superframes in current packet float gain_pred_err[6]; ///< cache for gain prediction float excitation_history[MAX_SIGNAL_HISTORY]; ///< cache of the signal of previous @@ -875,7 +876,6 @@ static void dequant_lsps(double *lsps, int num, /** * @name LSP dequantization routines * LSP dequantization routines, for 10/16LSPs and independent/residual coding. - * @note we assume enough bits are available, caller should check. * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. * @{ @@ -1419,7 +1419,6 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, /** * Parse data in a single block. - * @note we assume enough bits are available, caller should check. * * @param s WMA Voice decoding context private data * @param gb bit I/O context @@ -1463,7 +1462,6 @@ static void synth_block(WMAVoiceContext *s, GetBitContext *gb, /** * Synthesize output samples for a single frame. - * @note we assume enough bits are available, caller should check. * * @param ctx WMA Voice decoder context * @param gb bit I/O context (s->gb or one for cross-packet superframes) @@ -1682,83 +1680,6 @@ static void stabilize_lsps(double *lsps, int num) } /** - * Test if there's enough bits to read 1 superframe. - * - * @param orig_gb bit I/O context used for reading. This function - * does not modify the state of the bitreader; it - * only uses it to copy the current stream position - * @param s WMA Voice decoding context private data - * @return < 0 on error, 1 on not enough bits or 0 if OK. - */ -static int check_bits_for_superframe(GetBitContext *orig_gb, - WMAVoiceContext *s) -{ - GetBitContext s_gb, *gb = &s_gb; - int n, need_bits, bd_idx; - const struct frame_type_desc *frame_desc; - - /* initialize a copy */ - init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits); - skip_bits_long(gb, get_bits_count(orig_gb)); - av_assert1(get_bits_left(gb) == get_bits_left(orig_gb)); - - /* superframe header */ - if (get_bits_left(gb) < 14) - return 1; - if (!get_bits1(gb)) - return AVERROR(ENOSYS); // WMAPro-in-WMAVoice superframe - if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe - if (s->has_residual_lsps) { // residual LSPs (for all frames) - if (get_bits_left(gb) < s->sframe_lsp_bitsize) - return 1; - skip_bits_long(gb, s->sframe_lsp_bitsize); - } - - /* frames */ - for (n = 0; n < MAX_FRAMES; n++) { - int aw_idx_is_ext = 0; - - if (!s->has_residual_lsps) { // independent LSPs (per-frame) - if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1; - skip_bits_long(gb, s->frame_lsp_bitsize); - } - bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)]; - if (bd_idx < 0) - return AVERROR_INVALIDDATA; // invalid frame type VLC code - frame_desc = &frame_descs[bd_idx]; - if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { - if (get_bits_left(gb) < s->pitch_nbits) - return 1; - skip_bits_long(gb, s->pitch_nbits); - } - if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { - skip_bits(gb, 8); - } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { - int tmp = get_bits(gb, 6); - if (tmp >= 0x36) { - skip_bits(gb, 2); - aw_idx_is_ext = 1; - } - } - - /* blocks */ - if (frame_desc->acb_type == ACB_TYPE_HAMMING) { - need_bits = s->block_pitch_nbits + - (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits; - } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { - need_bits = 2 * !aw_idx_is_ext; - } else - need_bits = 0; - need_bits += frame_desc->frame_size; - if (get_bits_left(gb) < need_bits) - return 1; - skip_bits_long(gb, need_bits); - } - - return 0; -} - -/** * Synthesize output samples for a single superframe. If we have any data * cached in s->sframe_cache, that will be used instead of whatever is loaded * in s->gb. @@ -1780,7 +1701,7 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, { WMAVoiceContext *s = ctx->priv_data; GetBitContext *gb = &s->gb, s_gb; - int n, res, n_samples = 480; + int n, res, n_samples = MAX_SFRAMESIZE; 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]; @@ -1799,12 +1720,6 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, s->sframe_cache_size = 0; } - if ((res = check_bits_for_superframe(gb, s)) == 1) { - *got_frame_ptr = 0; - return 1; - } else if (res < 0) - return res; - /* First bit is speech/music bit, it differentiates between WMAVoice * speech samples (the actual codec) and WMAVoice music samples, which * are really WMAPro-in-WMAVoice-superframes. I've never seen those in @@ -1816,13 +1731,14 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ if (get_bits1(gb)) { - if ((n_samples = get_bits(gb, 12)) > 480) { + if ((n_samples = get_bits(gb, 12)) > MAX_SFRAMESIZE) { av_log(ctx, AV_LOG_ERROR, - "Superframe encodes >480 samples (%d), not allowed\n", - n_samples); + "Superframe encodes > %d samples (%d), not allowed\n", + MAX_SFRAMESIZE, n_samples); return AVERROR_INVALIDDATA; } } + /* Parse LSPs, if global for the superframe (can also be per-frame). */ if (s->has_residual_lsps) { double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; @@ -1845,7 +1761,7 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, } /* get output buffer */ - frame->nb_samples = 480; + frame->nb_samples = MAX_SFRAMESIZE; if ((res = ff_get_buffer(ctx, frame, 0)) < 0) return res; frame->nb_samples = n_samples; @@ -1905,26 +1821,23 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, * decoder). * * @param s WMA Voice decoding context private data - * @return 1 if not enough bits were available, or 0 on success. + * @return <0 on error, nb_superframes on success. */ static int parse_packet_header(WMAVoiceContext *s) { GetBitContext *gb = &s->gb; - unsigned int res; + unsigned int res, n_superframes = 0; - if (get_bits_left(gb) < 11) - return 1; skip_bits(gb, 4); // packet sequence number s->has_residual_lsps = get_bits1(gb); do { res = get_bits(gb, 6); // number of superframes per packet // (minus first one if there is spillover) - if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize) - return 1; + n_superframes += res; } while (res == 0x3F); s->spillover_nbits = get_bits(gb, s->spillover_bitsize); - return 0; + return get_bits_left(gb) >= 0 ? n_superframes : AVERROR_INVALIDDATA; } /** @@ -1984,23 +1897,24 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, * in a single "muxer" packet, so we artificially emulate that by * capping the packet size at ctx->block_align. */ for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); - if (!size) { - *got_frame_ptr = 0; - return 0; - } init_get_bits(&s->gb, avpkt->data, size << 3); /* size == ctx->block_align is used to indicate whether we are dealing with * a new packet or a packet of which we already read the packet header * previously. */ - if (size == ctx->block_align) { // new packet header - if ((res = parse_packet_header(s)) < 0) - return res; + if (!(size % ctx->block_align)) { // new packet header + if (!size) { + s->spillover_nbits = 0; + s->nb_superframes = 0; + } else { + if ((res = parse_packet_header(s)) < 0) + return res; + s->nb_superframes = res; + } /* If the packet header specifies a s->spillover_nbits, then we want * to push out all data of the previous packet (+ spillover) before * continuing to parse new superframes in the current packet. */ - if (s->spillover_nbits > 0) { if (s->sframe_cache_size > 0) { int cnt = get_bits_count(gb); copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); @@ -2021,9 +1935,9 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, } else skip_bits_long (gb, s->spillover_nbits - cnt + get_bits_count(gb)); // resync - } else + } else if (s->spillover_nbits) { skip_bits_long(gb, s->spillover_nbits); // resync - } + } } else if (s->skip_bits_next) skip_bits(gb, s->skip_bits_next); @@ -2031,6 +1945,10 @@ 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 (s->nb_superframes-- == 0) { + *got_frame_ptr = 0; + return size; + } else if (s->nb_superframes > 0) { if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) { return res; } else if (*got_frame_ptr) { @@ -2044,13 +1962,9 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, return AVERROR_INVALIDDATA; } return res; + } } else if ((s->sframe_cache_size = pos) > 0) { - /* rewind bit reader to start of last (incomplete) superframe... */ - init_get_bits(gb, avpkt->data, size << 3); - skip_bits_long(gb, (size << 3) - pos); - av_assert1(get_bits_left(gb) == pos); - - /* ...and cache it for spillover in next packet */ + /* ... cache it for spillover in next packet */ init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); // FIXME bad - just copy bytes as whole and add use the @@ -2084,6 +1998,6 @@ AVCodec ff_wmavoice_decoder = { .init_static_data = wmavoice_init_static_data, .close = wmavoice_decode_end, .decode = wmavoice_decode_packet, - .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1, + .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY, .flush = wmavoice_flush, };

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wmavoice: move wmavoice_flush() up.
by git＠videolan.org 27 Dec '16

27 Dec '16

ffmpeg | branch: master | Ronald S. Bultje <rsbultje(a)gmail.com> | Tue Dec 20 17:03:46 2016 -0500| [992cb15e671332650ddd5020c00cf08a40bb7cf0] | committer: Ronald S. Bultje wmavoice: move wmavoice_flush() up. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=992cb15e671332650… --- libavcodec/wmavoice.c | 56 +++++++++++++++++++++++++-------------------------- 1 file changed, 28 insertions(+), 28 deletions(-) diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index ceac61f..4b3ab43 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -337,6 +337,34 @@ static av_cold void wmavoice_init_static_data(AVCodec *codec) bits, 1, 1, codes, 2, 2, 132); } +static av_cold void wmavoice_flush(AVCodecContext *ctx) +{ + WMAVoiceContext *s = ctx->priv_data; + int n; + + s->postfilter_agc = 0; + s->sframe_cache_size = 0; + s->skip_bits_next = 0; + for (n = 0; n < s->lsps; n++) + s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); + memset(s->excitation_history, 0, + sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); + memset(s->synth_history, 0, + sizeof(*s->synth_history) * MAX_LSPS); + memset(s->gain_pred_err, 0, + sizeof(s->gain_pred_err)); + + if (s->do_apf) { + memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, + sizeof(*s->synth_filter_out_buf) * s->lsps); + memset(s->dcf_mem, 0, + sizeof(*s->dcf_mem) * 2); + memset(s->zero_exc_pf, 0, + sizeof(*s->zero_exc_pf) * s->history_nsamples); + memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); + } +} + /** * Set up decoder with parameters from demuxer (extradata etc.). */ @@ -2046,34 +2074,6 @@ static av_cold int wmavoice_decode_end(AVCodecContext *ctx) return 0; } -static av_cold void wmavoice_flush(AVCodecContext *ctx) -{ - WMAVoiceContext *s = ctx->priv_data; - int n; - - s->postfilter_agc = 0; - s->sframe_cache_size = 0; - s->skip_bits_next = 0; - for (n = 0; n < s->lsps; n++) - s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); - memset(s->excitation_history, 0, - sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); - memset(s->synth_history, 0, - sizeof(*s->synth_history) * MAX_LSPS); - memset(s->gain_pred_err, 0, - sizeof(s->gain_pred_err)); - - if (s->do_apf) { - memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, - sizeof(*s->synth_filter_out_buf) * s->lsps); - memset(s->dcf_mem, 0, - sizeof(*s->dcf_mem) * 2); - memset(s->zero_exc_pf, 0, - sizeof(*s->zero_exc_pf) * s->history_nsamples); - memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); - } -} - AVCodec ff_wmavoice_decoder = { .name = "wmavoice", .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),

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checkasm/vp9: benchmark all sub-IDCTs (but not WHT or ADST).
by git＠videolan.org 27 Dec '16

27 Dec '16

ffmpeg | branch: master | Ronald S. Bultje <rsbultje(a)gmail.com> | Thu Nov 17 16:08:20 2016 -0500| [1c8fbd7b90469f69fe3a3f78ba7886195d97c34f] | committer: Ronald S. Bultje checkasm/vp9: benchmark all sub-IDCTs (but not WHT or ADST). > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=1c8fbd7b90469f69f… --- tests/checkasm/vp9dsp.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/tests/checkasm/vp9dsp.c b/tests/checkasm/vp9dsp.c index 441041c..f32b97c 100644 --- a/tests/checkasm/vp9dsp.c +++ b/tests/checkasm/vp9dsp.c @@ -331,15 +331,20 @@ static void check_itxfm(void) int n_txtps = tx < TX_32X32 ? N_TXFM_TYPES : 1; for (txtp = 0; txtp < n_txtps; txtp++) { - if (check_func(dsp.itxfm_add[tx][txtp], "vp9_inv_%s_%dx%d_add_%d", - tx == 4 ? "wht_wht" : txtp_types[txtp], sz, sz, - bit_depth)) { - randomize_buffers(); - ftx(coef, tx, txtp, sz, bit_depth); - - for (sub = (txtp == 0) ? 1 : 2; sub <= sz; sub <<= 1) { + // skip testing sub-IDCTs for WHT or ADST since they don't + // implement it in any of the SIMD functions. If they do, + // consider changing this to ensure we have complete test + // coverage + for (sub = (txtp == 0 && tx < 4) ? 1 : sz; sub <= sz; sub <<= 1) { + if (check_func(dsp.itxfm_add[tx][txtp], + "vp9_inv_%s_%dx%d_sub%d_add_%d", + tx == 4 ? "wht_wht" : txtp_types[txtp], + sz, sz, sub, bit_depth)) { int eob; + randomize_buffers(); + ftx(coef, tx, txtp, sz, bit_depth); + if (sub < sz) { eob = copy_subcoefs(subcoef0, coef, tx, txtp, sz, sub, bit_depth); @@ -357,8 +362,9 @@ static void check_itxfm(void) !iszero(subcoef0, sz * sz * SIZEOF_COEF) || !iszero(subcoef1, sz * sz * SIZEOF_COEF)) fail(); + + bench_new(dst, sz * SIZEOF_PIXEL, coef, eob); } - bench_new(dst, sz * SIZEOF_PIXEL, coef, sz * sz); } } }

1 0

avformat/hlsenc: strftime identifiers and segment index
by git＠videolan.org 27 Dec '16

27 Dec '16

ffmpeg | branch: master | Bodecs Bela <bodecsb(a)vivanet.hu> | Tue Dec 27 10:40:35 2016 +0800| [0ff8c6b6d5243becfa73ff6960348a949055617b] | committer: Bodecs Bela avformat/hlsenc: strftime identifiers and segment index in filenames Putting date/time values into segment filenames is very usefull. But to produce non-conflicting segment filenames with -use_localtime option with date/time values in hls_segment_filename option, sometimes is not enough. Like in cases when multiple segments produced in the same second. But hlsenc currently does not make possible to use segment index (%d) at the same time whe use_localtime is in effect, due to identifier conflict. This patch makes possible to use strftime identifiers and still put segment index (%d) at same time in segment filenames by introducing second_level_segment_index flag. When -use_localtime is active, identifier %d is for month day index, so %%d is the segment index placeholder. This enhanced behaviour only exists when new second_level_segment_index flag is specified. For instance putting 'segment_%Y%m%d%H%M%S_%%05d.ts' value into -hls_segment_filename option and specifing -hls_flags second_level_segment_index and -use_localtime 1, may produce segment filename as 'segment_20161230235758_00002.ts' An example: ffmpeg -loglevel info -y -f lavfi -i color=c=red:size=640x480:r=25 -f lavfi -i anullsrc=r=44100:cl=stereo -c:v mpeg2video -g 25 -acodec aac -cutoff 20000 -ac 2 -ar 44100 -ab 192k -f hls -hls_time 3 -hls_list_size 5 -hls_flags delete_segments+second_level_segment_index -use_localtime 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%05d.ts" stream.m3u8 will produce segments filenames: .... segment_20161227005902_00013.ts segment_20161227005902_00014.ts segment_20161227005902_00015.ts segment_20161227005903_00016.ts segment_20161227005903_00017.ts segment_20161227005903_00018.ts segment_20161227005903_00019.ts segment_20161227005903_00020.ts .... Signed-off-by: Bela Bodecs <bodecsb(a)vivanet.hu> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=0ff8c6b6d5243becf… --- doc/muxers.texi | 11 ++++++++++- libavformat/hlsenc.c | 22 ++++++++++++++++++++-- 2 files changed, 30 insertions(+), 3 deletions(-) diff --git a/doc/muxers.texi b/doc/muxers.texi index 075b8d3..3520120 100644 --- a/doc/muxers.texi +++ b/doc/muxers.texi @@ -443,12 +443,18 @@ This example will produce the playlist, @file{out.m3u8}, and segment files: @item use_localtime Use strftime on @var{filename} to expand the segment filename with localtime. -The segment number (%d) is not available in this mode. +The segment number is also available in this mode, but to use it, you need to specify second_level_segment_index +hls_flag and %%d will be the specifier. @example ffmpeg -i in.nut -use_localtime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8 @end example This example will produce the playlist, @file{out.m3u8}, and segment files: @file{file-20160215-1455569023.ts}, @file{file-20160215-1455569024.ts}, etc. +@example +ffmpeg -i in.nut -use_localtime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8 +@end example +This example will produce the playlist, @file{out.m3u8}, and segment files: +(a)file{file-20160215-0001.ts}, @file{file-20160215-0002.ts}, etc. @item use_localtime_mkdir Used together with -use_localtime, it will create up to one subdirectory which @@ -556,6 +562,9 @@ seeking. This flag should be used with the @code{hls_time} option. @item hls_flags program_date_time Generate @code{EXT-X-PROGRAM-DATE-TIME} tags. +@item hls_flags second_level_segment_index +Makes it possible to use segment indexes as %%d besides date/time values when use_localtime is on. + @item hls_playlist_type event Emit @code{#EXT-X-PLAYLIST-TYPE:EVENT} in the m3u8 header. Forces @option{hls_list_size} to 0; the playlist can only be appended to. diff --git a/libavformat/hlsenc.c b/libavformat/hlsenc.c index 648e775..cf4d4bd 100644 --- a/libavformat/hlsenc.c +++ b/libavformat/hlsenc.c @@ -66,6 +66,7 @@ typedef enum HLSFlags { HLS_SPLIT_BY_TIME = (1 << 5), HLS_APPEND_LIST = (1 << 6), HLS_PROGRAM_DATE_TIME = (1 << 7), + HLS_SECOND_LEVEL_SEGMENT_INDEX = (1 << 8), // include segment index in segment filenames when use_localtime e.g.: %%03d } HLSFlags; typedef enum { @@ -717,7 +718,19 @@ static int hls_start(AVFormatContext *s) av_log(oc, AV_LOG_ERROR, "Could not get segment filename with use_localtime\n"); return AVERROR(EINVAL); } - + if (c->flags & HLS_SECOND_LEVEL_SEGMENT_INDEX) { + char * filename = av_strdup(oc->filename); // %%d will be %d after strftime + if (!filename) + return AVERROR(ENOMEM); + if (av_get_frame_filename2(oc->filename, sizeof(oc->filename), + filename, c->wrap ? c->sequence % c->wrap : c->sequence, + AV_FRAME_FILENAME_FLAGS_MULTIPLE) < 0) { + av_log(c, AV_LOG_ERROR, "Invalid second level segment filename template '%s', you can try to remove second_level_segment_index flag\n", filename); + av_free(filename); + return AVERROR(EINVAL); + } + av_free(filename); + } if (find_segment_by_filename(c->segments, oc->filename) || find_segment_by_filename(c->old_segments, oc->filename)) { av_log(c, AV_LOG_WARNING, "Duplicated segment filename detected: %s\n", oc->filename); @@ -900,7 +913,11 @@ static int hls_write_header(AVFormatContext *s) av_strlcat(hls->basename, pattern, basename_size); } } - + if (!hls->use_localtime && (hls->flags & HLS_SECOND_LEVEL_SEGMENT_INDEX)) { + av_log(hls, AV_LOG_ERROR, "second_level_segment_index hls_flag requires use_localtime to be true\n"); + ret = AVERROR(EINVAL); + goto fail; + } if(hls->has_subtitle) { if (hls->flags & HLS_SINGLE_FILE) @@ -1156,6 +1173,7 @@ static const AVOption options[] = { {"split_by_time", "split the hls segment by time which user set by hls_time", 0, AV_OPT_TYPE_CONST, {.i64 = HLS_SPLIT_BY_TIME }, 0, UINT_MAX, E, "flags"}, {"append_list", "append the new segments into old hls segment list", 0, AV_OPT_TYPE_CONST, {.i64 = HLS_APPEND_LIST }, 0, UINT_MAX, E, "flags"}, {"program_date_time", "add EXT-X-PROGRAM-DATE-TIME", 0, AV_OPT_TYPE_CONST, {.i64 = HLS_PROGRAM_DATE_TIME }, 0, UINT_MAX, E, "flags"}, + {"second_level_segment_index", "include segment index in segment filenames when use_localtime", 0, AV_OPT_TYPE_CONST, {.i64 = HLS_SECOND_LEVEL_SEGMENT_INDEX }, 0, UINT_MAX, E, "flags"}, {"use_localtime", "set filename expansion with strftime at segment creation", OFFSET(use_localtime), AV_OPT_TYPE_BOOL, {.i64 = 0 }, 0, 1, E }, {"use_localtime_mkdir", "create last directory component in strftime-generated filename", OFFSET(use_localtime_mkdir), AV_OPT_TYPE_BOOL, {.i64 = 0 }, 0, 1, E }, {"hls_playlist_type", "set the HLS playlist type", OFFSET(pl_type), AV_OPT_TYPE_INT, {.i64 = PLAYLIST_TYPE_NONE }, 0, PLAYLIST_TYPE_NB-1, E, "pl_type" },

1 0