February 2024 - ffmpeg-devel - lists.ffmpeg.org

[PATCH v1] avcodec/cbs_vp8: Use little endian in fixed()
by Dai, Jianhui J 27 Mar '24

27 Mar '24

This commit adds value range checks to cbs_vp8_read_unsigned_le, migrates fixed() to use it, and enforces little-endian consistency for all read methods. Signed-off-by: Jianhui Dai <jianhui.j.dai(a)intel.com> --- libavcodec/cbs_vp8.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) diff --git a/libavcodec/cbs_vp8.c b/libavcodec/cbs_vp8.c index 065156c248..7a0752019b 100644 --- a/libavcodec/cbs_vp8.c +++ b/libavcodec/cbs_vp8.c @@ -181,9 +181,11 @@ static int cbs_vp8_bool_decoder_read_signed( return 0; } -static int cbs_vp8_read_unsigned_le(CodedBitstreamContext *ctx, GetBitContext *gbc, - int width, const char *name, - const int *subscripts, uint32_t *write_to) +static int cbs_vp8_read_unsigned_le(CodedBitstreamContext *ctx, + GetBitContext *gbc, int width, + const char *name, const int *subscripts, + uint32_t *write_to, uint32_t range_min, + uint32_t range_max) { int32_t value; @@ -200,6 +202,14 @@ static int cbs_vp8_read_unsigned_le(CodedBitstreamContext *ctx, GetBitContext *g CBS_TRACE_READ_END(); + if (value < range_min || value > range_max) { + av_log(ctx->log_ctx, AV_LOG_ERROR, + "%s out of range: " + "%" PRIu32 ", but must be in [%" PRIu32 ",%" PRIu32 "].\n", + name, value, range_min, range_max); + return AVERROR_INVALIDDATA; + } + *write_to = value; return 0; } @@ -246,15 +256,16 @@ static int cbs_vp8_read_unsigned_le(CodedBitstreamContext *ctx, GetBitContext *g do { \ uint32_t value; \ CHECK(cbs_vp8_read_unsigned_le(ctx, rw, width, #name, \ - SUBSCRIPTS(subs, __VA_ARGS__), &value)); \ + SUBSCRIPTS(subs, __VA_ARGS__), &value, \ + 0, MAX_UINT_BITS(width))); \ current->name = value; \ } while (0) #define fixed(width, name, value) \ do { \ uint32_t fixed_value; \ - CHECK(ff_cbs_read_unsigned(ctx, rw, width, #name, 0, &fixed_value, \ - value, value)); \ + CHECK(cbs_vp8_read_unsigned_le(ctx, rw, width, #name, 0, &fixed_value, \ + value, value)); \ } while (0) #define bc_unsigned_subs(width, prob, enable_trace, name, subs, ...) \ -- 2.25.1

2 3

[PATCH 1/3] avformat/tmv: Check video chunk size
by Michael Niedermayer 26 Mar '24

26 Mar '24

This check matches the audio chunk check Fixes: Timeout Fixes: 62681/clusterfuzz-testcase-minimized-ffmpeg_dem_TMV_fuzzer-5299107876700160 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael(a)niedermayer.cc> --- libavformat/tmv.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/libavformat/tmv.c b/libavformat/tmv.c index ea39954190..18105f764e 100644 --- a/libavformat/tmv.c +++ b/libavformat/tmv.c @@ -103,6 +103,10 @@ static int tmv_read_header(AVFormatContext *s) char_cols = avio_r8(pb); char_rows = avio_r8(pb); tmv->video_chunk_size = char_cols * char_rows * 2; + if (!tmv->video_chunk_size) { + av_log(s, AV_LOG_ERROR, "invalid video chunk size\n"); + return AVERROR_INVALIDDATA; + } features = avio_r8(pb); if (features & ~(TMV_PADDING | TMV_STEREO)) { -- 2.17.1

5 10

[PATCH 1/3] avcodec/osq: Implement flush()
by Michael Niedermayer 26 Mar '24

26 Mar '24

Fixes: out of array access Fixes: 62164/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_OSQ_fuzzer-6227491892887552 Fixes: 62164/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_OSQ_fuzzer-6268561729126400 Fixes: 62164/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_OSQ_fuzzer-6414805046788096 Fixes: 62164/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_OSQ_fuzzer-6538151088488448 Fixes: 62164/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_OSQ_fuzzer-6608131540779008 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael(a)niedermayer.cc> --- libavcodec/osq.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/libavcodec/osq.c b/libavcodec/osq.c index 435f3ea7f69..abe15c97f18 100644 --- a/libavcodec/osq.c +++ b/libavcodec/osq.c @@ -61,6 +61,15 @@ typedef struct OSQContext { int pkt_offset; } OSQContext; +static void osq_flush(AVCodecContext *avctx) +{ + OSQContext *s = avctx->priv_data; + + s->bitstream_size = 0; + s->pkt_offset = 0; + av_packet_unref(s->pkt); +} + static av_cold int osq_close(AVCodecContext *avctx) { OSQContext *s = avctx->priv_data; @@ -478,4 +487,5 @@ const FFCodec ff_osq_decoder = { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE }, + .flush = osq_flush, }; -- 2.17.1

3 7

[PATCH 1/6] avformat/avidec: support huge durations
by Michael Niedermayer 25 Mar '24

25 Mar '24

Fixes: signed integer overflow: 109817402400 * 301990077 cannot be represented in type 'long long' Fixes: 51896/clusterfuzz-testcase-minimized-ffmpeg_dem_AVI_fuzzer-6706191715139584 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael(a)niedermayer.cc> --- libavformat/avidec.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/libavformat/avidec.c b/libavformat/avidec.c index 00bd7a98a9d..cfc693842b7 100644 --- a/libavformat/avidec.c +++ b/libavformat/avidec.c @@ -27,6 +27,7 @@ #include "libavutil/avstring.h" #include "libavutil/opt.h" #include "libavutil/dict.h" +#include "libavutil/integer.h" #include "libavutil/internal.h" #include "libavutil/intreadwrite.h" #include "libavutil/mathematics.h" @@ -476,7 +477,7 @@ static int calculate_bitrate(AVFormatContext *s) AVStream *st = s->streams[i]; FFStream *const sti = ffstream(st); int64_t duration; - int64_t bitrate; + int64_t bitrate = 0; for (j = 0; j < sti->nb_index_entries; j++) len += sti->index_entries[j].size; @@ -484,7 +485,14 @@ static int calculate_bitrate(AVFormatContext *s) if (sti->nb_index_entries < 2 || st->codecpar->bit_rate > 0) continue; duration = sti->index_entries[j-1].timestamp - sti->index_entries[0].timestamp; - bitrate = av_rescale(8*len, st->time_base.den, duration * st->time_base.num); + if (INT64_MAX / duration >= st->time_base.num) { + bitrate = av_rescale(8*len, st->time_base.den, duration * st->time_base.num); + } else { + AVInteger bitrate_i = av_div_i(av_mul_i(av_int2i(8*len), av_int2i(st->time_base.den)), + av_mul_i(av_int2i(duration), av_int2i(st->time_base.num))); + if (av_cmp_i(bitrate_i, av_int2i(INT64_MAX)) <= 0) + bitrate = av_i2int(bitrate_i); + } if (bitrate > 0) { st->codecpar->bit_rate = bitrate; } -- 2.17.1

5 19

[PATCH 01/31] fftools/ffmpeg_dec: split Decoder into a private and public part
by Anton Khirnov 24 Mar '24

24 Mar '24

Similar to what is currently done for other components, e.g. (de)muxers. There is nothing in the public part currently, but that will change in future commits. --- fftools/ffmpeg.h | 4 +- fftools/ffmpeg_dec.c | 193 ++++++++++++++++++++++--------------------- 2 files changed, 104 insertions(+), 93 deletions(-) diff --git a/fftools/ffmpeg.h b/fftools/ffmpeg.h index 3014a626b4..429c1aba85 100644 --- a/fftools/ffmpeg.h +++ b/fftools/ffmpeg.h @@ -280,7 +280,9 @@ typedef struct FilterGraph { int nb_outputs; } FilterGraph; -typedef struct Decoder Decoder; +typedef struct Decoder { + char dummy; +} Decoder; typedef struct InputStream { const AVClass *class; diff --git a/fftools/ffmpeg_dec.c b/fftools/ffmpeg_dec.c index 5d4ef7909c..fd697c5f1e 100644 --- a/fftools/ffmpeg_dec.c +++ b/fftools/ffmpeg_dec.c @@ -34,7 +34,9 @@ #include "ffmpeg_utils.h" #include "thread_queue.h" -struct Decoder { +typedef struct DecoderPriv { + Decoder dec; + AVFrame *frame; AVPacket *pkt; @@ -55,7 +57,12 @@ struct Decoder { Scheduler *sch; unsigned sch_idx; -}; +} DecoderPriv; + +static DecoderPriv *dp_from_dec(Decoder *d) +{ + return (DecoderPriv*)d; +} // data that is local to the decoder thread and not visible outside of it typedef struct DecThreadContext { @@ -66,61 +73,63 @@ typedef struct DecThreadContext { void dec_free(Decoder **pdec) { Decoder *dec = *pdec; + DecoderPriv *dp; if (!dec) return; + dp = dp_from_dec(dec); - av_frame_free(&dec->frame); - av_packet_free(&dec->pkt); + av_frame_free(&dp->frame); + av_packet_free(&dp->pkt); - for (int i = 0; i < FF_ARRAY_ELEMS(dec->sub_prev); i++) - av_frame_free(&dec->sub_prev[i]); - av_frame_free(&dec->sub_heartbeat); + for (int i = 0; i < FF_ARRAY_ELEMS(dp->sub_prev); i++) + av_frame_free(&dp->sub_prev[i]); + av_frame_free(&dp->sub_heartbeat); av_freep(pdec); } static int dec_alloc(Decoder **pdec) { - Decoder *dec; + DecoderPriv *dp; *pdec = NULL; - dec = av_mallocz(sizeof(*dec)); - if (!dec) + dp = av_mallocz(sizeof(*dp)); + if (!dp) return AVERROR(ENOMEM); - dec->frame = av_frame_alloc(); - if (!dec->frame) + dp->frame = av_frame_alloc(); + if (!dp->frame) goto fail; - dec->pkt = av_packet_alloc(); - if (!dec->pkt) + dp->pkt = av_packet_alloc(); + if (!dp->pkt) goto fail; - dec->last_filter_in_rescale_delta = AV_NOPTS_VALUE; - dec->last_frame_pts = AV_NOPTS_VALUE; - dec->last_frame_tb = (AVRational){ 1, 1 }; - dec->hwaccel_pix_fmt = AV_PIX_FMT_NONE; + dp->last_filter_in_rescale_delta = AV_NOPTS_VALUE; + dp->last_frame_pts = AV_NOPTS_VALUE; + dp->last_frame_tb = (AVRational){ 1, 1 }; + dp->hwaccel_pix_fmt = AV_PIX_FMT_NONE; - *pdec = dec; + *pdec = &dp->dec; return 0; fail: - dec_free(&dec); + dec_free((Decoder**)dp); return AVERROR(ENOMEM); } -static AVRational audio_samplerate_update(void *logctx, Decoder *d, +static AVRational audio_samplerate_update(void *logctx, DecoderPriv *dp, const AVFrame *frame) { - const int prev = d->last_frame_tb.den; + const int prev = dp->last_frame_tb.den; const int sr = frame->sample_rate; AVRational tb_new; int64_t gcd; - if (frame->sample_rate == d->last_frame_sample_rate) + if (frame->sample_rate == dp->last_frame_sample_rate) goto finish; gcd = av_gcd(prev, sr); @@ -141,20 +150,20 @@ static AVRational audio_samplerate_update(void *logctx, Decoder *d, !(frame->time_base.den % tb_new.den)) tb_new = frame->time_base; - if (d->last_frame_pts != AV_NOPTS_VALUE) - d->last_frame_pts = av_rescale_q(d->last_frame_pts, - d->last_frame_tb, tb_new); - d->last_frame_duration_est = av_rescale_q(d->last_frame_duration_est, - d->last_frame_tb, tb_new); + if (dp->last_frame_pts != AV_NOPTS_VALUE) + dp->last_frame_pts = av_rescale_q(dp->last_frame_pts, + dp->last_frame_tb, tb_new); + dp->last_frame_duration_est = av_rescale_q(dp->last_frame_duration_est, + dp->last_frame_tb, tb_new); - d->last_frame_tb = tb_new; - d->last_frame_sample_rate = frame->sample_rate; + dp->last_frame_tb = tb_new; + dp->last_frame_sample_rate = frame->sample_rate; finish: - return d->last_frame_tb; + return dp->last_frame_tb; } -static void audio_ts_process(void *logctx, Decoder *d, AVFrame *frame) +static void audio_ts_process(void *logctx, DecoderPriv *dp, AVFrame *frame) { AVRational tb_filter = (AVRational){1, frame->sample_rate}; AVRational tb; @@ -163,27 +172,27 @@ static void audio_ts_process(void *logctx, Decoder *d, AVFrame *frame) // on samplerate change, choose a new internal timebase for timestamp // generation that can represent timestamps from all the samplerates // seen so far - tb = audio_samplerate_update(logctx, d, frame); - pts_pred = d->last_frame_pts == AV_NOPTS_VALUE ? 0 : - d->last_frame_pts + d->last_frame_duration_est; + tb = audio_samplerate_update(logctx, dp, frame); + pts_pred = dp->last_frame_pts == AV_NOPTS_VALUE ? 0 : + dp->last_frame_pts + dp->last_frame_duration_est; if (frame->pts == AV_NOPTS_VALUE) { frame->pts = pts_pred; frame->time_base = tb; - } else if (d->last_frame_pts != AV_NOPTS_VALUE && + } else if (dp->last_frame_pts != AV_NOPTS_VALUE && frame->pts > av_rescale_q_rnd(pts_pred, tb, frame->time_base, AV_ROUND_UP)) { // there was a gap in timestamps, reset conversion state - d->last_filter_in_rescale_delta = AV_NOPTS_VALUE; + dp->last_filter_in_rescale_delta = AV_NOPTS_VALUE; } frame->pts = av_rescale_delta(frame->time_base, frame->pts, tb, frame->nb_samples, - &d->last_filter_in_rescale_delta, tb); + &dp->last_filter_in_rescale_delta, tb); - d->last_frame_pts = frame->pts; - d->last_frame_duration_est = av_rescale_q(frame->nb_samples, - tb_filter, tb); + dp->last_frame_pts = frame->pts; + dp->last_frame_duration_est = av_rescale_q(frame->nb_samples, + tb_filter, tb); // finally convert to filtering timebase frame->pts = av_rescale_q(frame->pts, tb, tb_filter); @@ -193,7 +202,7 @@ static void audio_ts_process(void *logctx, Decoder *d, AVFrame *frame) static int64_t video_duration_estimate(const InputStream *ist, const AVFrame *frame) { - const Decoder *d = ist->decoder; + const DecoderPriv *dp = dp_from_dec(ist->decoder); const InputFile *ifile = ist->file; int64_t codec_duration = 0; @@ -221,9 +230,9 @@ static int64_t video_duration_estimate(const InputStream *ist, const AVFrame *fr // when timestamps are available, repeat last frame's actual duration // (i.e. pts difference between this and last frame) - if (frame->pts != AV_NOPTS_VALUE && d->last_frame_pts != AV_NOPTS_VALUE && - frame->pts > d->last_frame_pts) - return frame->pts - d->last_frame_pts; + if (frame->pts != AV_NOPTS_VALUE && dp->last_frame_pts != AV_NOPTS_VALUE && + frame->pts > dp->last_frame_pts) + return frame->pts - dp->last_frame_pts; // try frame/codec duration if (frame->duration > 0) @@ -240,12 +249,12 @@ static int64_t video_duration_estimate(const InputStream *ist, const AVFrame *fr } // last resort is last frame's estimated duration, and 1 - return FFMAX(d->last_frame_duration_est, 1); + return FFMAX(dp->last_frame_duration_est, 1); } static int video_frame_process(InputStream *ist, AVFrame *frame) { - Decoder *d = ist->decoder; + DecoderPriv *dp = dp_from_dec(ist->decoder); // The following line may be required in some cases where there is no parser // or the parser does not has_b_frames correctly @@ -281,7 +290,7 @@ static int video_frame_process(InputStream *ist, AVFrame *frame) } #endif - if (frame->format == d->hwaccel_pix_fmt) { + if (frame->format == dp->hwaccel_pix_fmt) { int err = hwaccel_retrieve_data(ist->dec_ctx, frame); if (err < 0) return err; @@ -298,13 +307,13 @@ static int video_frame_process(InputStream *ist, AVFrame *frame) // no timestamp available - extrapolate from previous frame duration if (frame->pts == AV_NOPTS_VALUE) - frame->pts = d->last_frame_pts == AV_NOPTS_VALUE ? 0 : - d->last_frame_pts + d->last_frame_duration_est; + frame->pts = dp->last_frame_pts == AV_NOPTS_VALUE ? 0 : + dp->last_frame_pts + dp->last_frame_duration_est; // update timestamp history - d->last_frame_duration_est = video_duration_estimate(ist, frame); - d->last_frame_pts = frame->pts; - d->last_frame_tb = frame->time_base; + dp->last_frame_duration_est = video_duration_estimate(ist, frame); + dp->last_frame_pts = frame->pts; + dp->last_frame_tb = frame->time_base; if (debug_ts) { av_log(ist, AV_LOG_INFO, @@ -330,13 +339,13 @@ static int video_frame_process(InputStream *ist, AVFrame *frame) static int process_subtitle(InputStream *ist, AVFrame *frame) { - Decoder *d = ist->decoder; + DecoderPriv *dp = dp_from_dec(ist->decoder); const AVSubtitle *subtitle = (AVSubtitle*)frame->buf[0]->data; int ret = 0; if (ist->fix_sub_duration) { - AVSubtitle *sub_prev = d->sub_prev[0]->buf[0] ? - (AVSubtitle*)d->sub_prev[0]->buf[0]->data : NULL; + AVSubtitle *sub_prev = dp->sub_prev[0]->buf[0] ? + (AVSubtitle*)dp->sub_prev[0]->buf[0]->data : NULL; int end = 1; if (sub_prev) { end = av_rescale(subtitle->pts - sub_prev->pts, @@ -350,13 +359,13 @@ static int process_subtitle(InputStream *ist, AVFrame *frame) } } - av_frame_unref(d->sub_prev[1]); - av_frame_move_ref(d->sub_prev[1], frame); + av_frame_unref(dp->sub_prev[1]); + av_frame_move_ref(dp->sub_prev[1], frame); - frame = d->sub_prev[0]; + frame = dp->sub_prev[0]; subtitle = frame->buf[0] ? (AVSubtitle*)frame->buf[0]->data : NULL; - FFSWAP(AVFrame*, d->sub_prev[0], d->sub_prev[1]); + FFSWAP(AVFrame*, dp->sub_prev[0], dp->sub_prev[1]); if (end <= 0) return 0; @@ -365,7 +374,7 @@ static int process_subtitle(InputStream *ist, AVFrame *frame) if (!subtitle) return 0; - ret = sch_dec_send(d->sch, d->sch_idx, frame); + ret = sch_dec_send(dp->sch, dp->sch_idx, frame); if (ret < 0) av_frame_unref(frame); @@ -374,31 +383,31 @@ static int process_subtitle(InputStream *ist, AVFrame *frame) static int fix_sub_duration_heartbeat(InputStream *ist, int64_t signal_pts) { - Decoder *d = ist->decoder; + DecoderPriv *dp = dp_from_dec(ist->decoder); int ret = AVERROR_BUG; - AVSubtitle *prev_subtitle = d->sub_prev[0]->buf[0] ? - (AVSubtitle*)d->sub_prev[0]->buf[0]->data : NULL; + AVSubtitle *prev_subtitle = dp->sub_prev[0]->buf[0] ? + (AVSubtitle*)dp->sub_prev[0]->buf[0]->data : NULL; AVSubtitle *subtitle; if (!ist->fix_sub_duration || !prev_subtitle || !prev_subtitle->num_rects || signal_pts <= prev_subtitle->pts) return 0; - av_frame_unref(d->sub_heartbeat); - ret = subtitle_wrap_frame(d->sub_heartbeat, prev_subtitle, 1); + av_frame_unref(dp->sub_heartbeat); + ret = subtitle_wrap_frame(dp->sub_heartbeat, prev_subtitle, 1); if (ret < 0) return ret; - subtitle = (AVSubtitle*)d->sub_heartbeat->buf[0]->data; + subtitle = (AVSubtitle*)dp->sub_heartbeat->buf[0]->data; subtitle->pts = signal_pts; - return process_subtitle(ist, d->sub_heartbeat); + return process_subtitle(ist, dp->sub_heartbeat); } static int transcode_subtitles(InputStream *ist, const AVPacket *pkt, AVFrame *frame) { - Decoder *d = ist->decoder; + DecoderPriv *dp = dp_from_dec(ist->decoder); AVPacket *flush_pkt = NULL; AVSubtitle subtitle; int got_output; @@ -409,7 +418,7 @@ static int transcode_subtitles(InputStream *ist, const AVPacket *pkt, frame->time_base = pkt->time_base; frame->opaque = (void*)(intptr_t)FRAME_OPAQUE_SUB_HEARTBEAT; - ret = sch_dec_send(d->sch, d->sch_idx, frame); + ret = sch_dec_send(dp->sch, dp->sch_idx, frame); return ret == AVERROR_EOF ? AVERROR_EXIT : ret; } else if (pkt && (intptr_t)pkt->opaque == PKT_OPAQUE_FIX_SUB_DURATION) { return fix_sub_duration_heartbeat(ist, av_rescale_q(pkt->pts, pkt->time_base, @@ -457,7 +466,7 @@ static int transcode_subtitles(InputStream *ist, const AVPacket *pkt, static int packet_decode(InputStream *ist, AVPacket *pkt, AVFrame *frame) { const InputFile *ifile = ist->file; - Decoder *d = ist->decoder; + DecoderPriv *dp = dp_from_dec(ist->decoder); AVCodecContext *dec = ist->dec_ctx; const char *type_desc = av_get_media_type_string(dec->codec_type); int ret; @@ -553,7 +562,7 @@ static int packet_decode(InputStream *ist, AVPacket *pkt, AVFrame *frame) if (dec->codec_type == AVMEDIA_TYPE_AUDIO) { ist->samples_decoded += frame->nb_samples; - audio_ts_process(ist, ist->decoder, frame); + audio_ts_process(ist, dp, frame); } else { ret = video_frame_process(ist, frame); if (ret < 0) { @@ -565,7 +574,7 @@ static int packet_decode(InputStream *ist, AVPacket *pkt, AVFrame *frame) ist->frames_decoded++; - ret = sch_dec_send(d->sch, d->sch_idx, frame); + ret = sch_dec_send(dp->sch, dp->sch_idx, frame); if (ret < 0) { av_frame_unref(frame); return ret == AVERROR_EOF ? AVERROR_EXIT : ret; @@ -611,7 +620,7 @@ fail: void *decoder_thread(void *arg) { InputStream *ist = arg; - Decoder *d = ist->decoder; + DecoderPriv *dp = dp_from_dec(ist->decoder); DecThreadContext dt; int ret = 0, input_status = 0; @@ -624,7 +633,7 @@ void *decoder_thread(void *arg) while (!input_status) { int flush_buffers, have_data; - input_status = sch_dec_receive(d->sch, d->sch_idx, dt.pkt); + input_status = sch_dec_receive(dp->sch, dp->sch_idx, dt.pkt); have_data = input_status >= 0 && (dt.pkt->buf || dt.pkt->side_data_elems || (intptr_t)dt.pkt->opaque == PKT_OPAQUE_SUB_HEARTBEAT || @@ -656,8 +665,8 @@ void *decoder_thread(void *arg) /* report last frame duration to the scheduler */ if (ist->dec->type == AVMEDIA_TYPE_AUDIO) { - dt.pkt->pts = d->last_frame_pts + d->last_frame_duration_est; - dt.pkt->time_base = d->last_frame_tb; + dt.pkt->pts = dp->last_frame_pts + dp->last_frame_duration_est; + dt.pkt->time_base = dp->last_frame_tb; } avcodec_flush_buffers(ist->dec_ctx); @@ -679,11 +688,11 @@ void *decoder_thread(void *arg) av_frame_unref(dt.frame); dt.frame->opaque = (void*)(intptr_t)FRAME_OPAQUE_EOF; - dt.frame->pts = d->last_frame_pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE : - d->last_frame_pts + d->last_frame_duration_est; - dt.frame->time_base = d->last_frame_tb; + dt.frame->pts = dp->last_frame_pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE : + dp->last_frame_pts + dp->last_frame_duration_est; + dt.frame->time_base = dp->last_frame_tb; - ret = sch_dec_send(d->sch, d->sch_idx, dt.frame); + ret = sch_dec_send(dp->sch, dp->sch_idx, dt.frame); if (ret < 0 && ret != AVERROR_EOF) { av_log(NULL, AV_LOG_FATAL, "Error signalling EOF timestamp: %s\n", av_err2str(ret)); @@ -710,7 +719,7 @@ finish: static enum AVPixelFormat get_format(AVCodecContext *s, const enum AVPixelFormat *pix_fmts) { InputStream *ist = s->opaque; - Decoder *d = ist->decoder; + DecoderPriv *dp = dp_from_dec(ist->decoder); const enum AVPixelFormat *p; for (p = pix_fmts; *p != AV_PIX_FMT_NONE; p++) { @@ -735,7 +744,7 @@ static enum AVPixelFormat get_format(AVCodecContext *s, const enum AVPixelFormat } } if (config && config->device_type == ist->hwaccel_device_type) { - d->hwaccel_pix_fmt = *p; + dp->hwaccel_pix_fmt = *p; break; } } @@ -890,7 +899,7 @@ static int hw_device_setup_for_decode(InputStream *ist) int dec_open(InputStream *ist, Scheduler *sch, unsigned sch_idx) { - Decoder *d; + DecoderPriv *dp; const AVCodec *codec = ist->dec; int ret; @@ -904,19 +913,19 @@ int dec_open(InputStream *ist, Scheduler *sch, unsigned sch_idx) ret = dec_alloc(&ist->decoder); if (ret < 0) return ret; - d = ist->decoder; + dp = dp_from_dec(ist->decoder); - d->sch = sch; - d->sch_idx = sch_idx; + dp->sch = sch; + dp->sch_idx = sch_idx; if (codec->type == AVMEDIA_TYPE_SUBTITLE && ist->fix_sub_duration) { - for (int i = 0; i < FF_ARRAY_ELEMS(d->sub_prev); i++) { - d->sub_prev[i] = av_frame_alloc(); - if (!d->sub_prev[i]) + for (int i = 0; i < FF_ARRAY_ELEMS(dp->sub_prev); i++) { + dp->sub_prev[i] = av_frame_alloc(); + if (!dp->sub_prev[i]) return AVERROR(ENOMEM); } - d->sub_heartbeat = av_frame_alloc(); - if (!d->sub_heartbeat) + dp->sub_heartbeat = av_frame_alloc(); + if (!dp->sub_heartbeat) return AVERROR(ENOMEM); } -- 2.42.0

2 33

Add protocol for Android content providers
by Matthieu Bouron 23 Mar '24

23 Mar '24

Hi, On Android, content providers are used for accessing files through shared mechanisms. One typical case would be an app willing to open a video from Google Photos, gallery apps, TikTok, Instagram or some other providers. A content URI looks something like "content://authority/path/id", see: https://developer.android.com/reference/android/content/ContentUris https://developer.android.com/guide/topics/providers/content-provider-basics It can currently be somehow managed through clumsy means such as using a "fd:" filename and crafting a special AVOption, which also has the drawback of requiring the third party to carry around opened file descriptors (with the multiple opened file limitations implied). Custom AVIOContexts are also an option. Both options will have to deal with the JNI though and end users will have to re-implement the same exact thing. This patchset addresses this by adding a content provider protocol, which has an API fairly similar to fopen. Android 11 appears to provide something transparent within fopen(), but FFmpeg doesn't use it in the file protocol, and Android < 11 are still widely used. The first part move the JNI infrastructure from avcodec to avutil (it remains internally shared, there is little user implication), and then the URLProtocol is added, along with a few cleanups. Regards, -- Matthieu

5 50

[PATCH v1] avcodec/cbs_vp8: Improve the bitstream position check
by Dai, Jianhui J 19 Mar '24

19 Mar '24

The VP8 compressed header may not be byte-aligned due to boolean coding. Use bitwise comparison to prevent the potential overread. Signed-off-by: Jianhui Dai <jianhui.j.dai(a)intel.com> --- libavcodec/cbs_vp8.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/libavcodec/cbs_vp8.c b/libavcodec/cbs_vp8.c index 065156c248..13acad3724 100644 --- a/libavcodec/cbs_vp8.c +++ b/libavcodec/cbs_vp8.c @@ -327,9 +327,10 @@ static int cbs_vp8_read_unit(CodedBitstreamContext *ctx, if (err < 0) return err; + // Position may not be byte-aligned after compressed header; using bits + // count comparison for accuracy. pos = get_bits_count(&gbc); - pos /= 8; - av_assert0(pos <= unit->data_size); + av_assert0(pos <= unit->data_size * 8); frame->data_ref = av_buffer_ref(unit->data_ref); if (!frame->data_ref) -- 2.25.1

2 4

[RFC PATCH] avformat/rtpdec: Audio level RTP extension RFC6464
by jon＠jonb.org 18 Mar '24

18 Mar '24

From: Jonathan Baudanza <jon(a)jonb.org> libwebrtc will add audio level (in decibels) and VAD status to each RTP packet. This patch will add both values to the packet sidedata. I've been using this patch in production for about a year on live audio RTP streams to detect when users are speaking without needing to decode the audio data. --- libavcodec/avpacket.c | 1 + libavcodec/defs.h | 15 ++++++++ libavcodec/packet.h | 5 +++ libavformat/rtpdec.c | 87 +++++++++++++++++++++++++++++++++++++++++++ libavformat/rtpdec.h | 5 +++ libavformat/rtsp.c | 16 ++++++++ libavformat/rtsp.h | 2 + 7 files changed, 131 insertions(+) diff --git a/libavcodec/avpacket.c b/libavcodec/avpacket.c index e118bbaad1..73e0341bf7 100644 --- a/libavcodec/avpacket.c +++ b/libavcodec/avpacket.c @@ -305,6 +305,7 @@ const char *av_packet_side_data_name(enum AVPacketSideDataType type) case AV_PKT_DATA_IAMF_MIX_GAIN_PARAM: return "IAMF Mix Gain Parameter Data"; case AV_PKT_DATA_IAMF_DEMIXING_INFO_PARAM: return "IAMF Demixing Info Parameter Data"; case AV_PKT_DATA_IAMF_RECON_GAIN_INFO_PARAM: return "IAMF Recon Gain Info Parameter Data"; + case AV_PKT_DATA_SSRC_AUDIO_LEVEL: return "RTP SSRC Audio Level"; } return NULL; } diff --git a/libavcodec/defs.h b/libavcodec/defs.h index 00d840ec19..87e8814760 100644 --- a/libavcodec/defs.h +++ b/libavcodec/defs.h @@ -323,6 +323,21 @@ typedef struct AVProducerReferenceTime { int flags; } AVProducerReferenceTime; +/** + * Audio level structure from the ssrc-audio-level RTP header extension. + */ +typedef struct AVAudioLevel { + /** + * Audio level for this packet, measured in dBov: -127 - 0 + */ + int8_t level; + + /** + * Set to 1 if the encoder believes this packet contains voice. + */ + int voice; +} AVAudioLevel; + /** * Encode extradata length to a buffer. Used by xiph codecs. * diff --git a/libavcodec/packet.h b/libavcodec/packet.h index 8558ae849e..f7f1deb6e0 100644 --- a/libavcodec/packet.h +++ b/libavcodec/packet.h @@ -330,6 +330,11 @@ enum AVPacketSideDataType { */ AV_PKT_DATA_AMBIENT_VIEWING_ENVIRONMENT, + /** + * Audio Level and VAD data from the RTP header extension as defined by RFC 6464. + */ + AV_PKT_DATA_SSRC_AUDIO_LEVEL, + /** * The number of side data types. * This is not part of the public API/ABI in the sense that it may diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index fa7544cc07..479ea2e245 100644 --- a/libavformat/rtpdec.c +++ b/libavformat/rtpdec.c @@ -694,6 +694,79 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam s->base_timestamp; } + +static const uint8_t* find_header_ext_data(int id, const uint8_t *buf, uint8_t *len) { + int buflen = (AV_RB16(buf + 2)) * 4; + + const uint8_t *p = buf + 4; + int idx = 0; + int this_id; + int this_len; + + // This is a one-byte extention format, as defined by RFC rfc5285 + if (buf[0] == 0xbe && buf[1] == 0xde) { + while (idx + 1 < buflen) { + if (p[idx] == 0) { + idx++; // skip padding + } else { + this_id = p[idx] >> 4; + this_len = (p[idx] & 0xf) + 1; + + // spec says 15 is reserved + if (this_id == 15) { + break; // reject + } + + if (this_id == id) { + if (this_len > buflen - idx - 1) { + break; // reject + } + + if (len != NULL) + *len = this_len; + + return p + idx + 1; + } + + idx += 1 + this_len; + } + } + } else if (buf[0] == 0x10 && (buf[1] & 0xff) == 0) { + // This is a two-byte extention format + while (idx + 1 < buflen) { + if (p[idx] == 0) { + idx++; // Skip padding + } else { + this_id = p[idx]; + this_len = p[idx + 1]; + + // spec says 15 is reserved + if (this_id == 15) { + break; // reject + } + + if (this_id == id) { + if (this_len > buflen - idx - 2) { + break; // reject + } + + if (len != NULL) + *len = this_len; + return p + idx + 2; + } + + idx += 2 + this_len; + } + } + } + + if (len != NULL) + *len = 0; + + return NULL; +} + + static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, const uint8_t *buf, int len) { @@ -703,6 +776,7 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, AVStream *st; uint32_t timestamp; int rv = 0; + const uint8_t *audio_level_data = NULL; csrc = buf[0] & 0x0f; ext = buf[0] & 0x10; @@ -753,6 +827,11 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, if (len < ext) return -1; + + if (s->ssrc_audio_level_ext_id) { + audio_level_data = find_header_ext_data(s->ssrc_audio_level_ext_id, buf, NULL); + } + // skip past RTP header extension len -= ext; buf += ext; @@ -774,6 +853,14 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, // now perform timestamp things.... finalize_packet(s, pkt, timestamp); + if (audio_level_data) { + AVAudioLevel *side_data = (struct AVAudioLevel *)av_packet_new_side_data(pkt, AV_PKT_DATA_SSRC_AUDIO_LEVEL, sizeof(AVAudioLevel)); + if (side_data) { + side_data->voice = ((*audio_level_data & 0x80) == 0x80); + side_data->level = -(*audio_level_data & 0x7f); + } + } + return rv; } diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h index 5a02e72dc2..91a338200a 100644 --- a/libavformat/rtpdec.h +++ b/libavformat/rtpdec.h @@ -188,6 +188,11 @@ struct RTPDemuxContext { /* dynamic payload stuff */ const RTPDynamicProtocolHandler *handler; PayloadContext *dynamic_protocol_context; + + /** + * RFC 6464 header extension id + */ + int ssrc_audio_level_ext_id; }; /** diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index c7d9b48684..63bc67fdf7 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -691,6 +691,21 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, } } } + } else if (av_strstart(p, "extmap:", &p)) { + char *end; + int id; + id = strtol(p, &end, 10); + if (p == end) { + break; + } + p = end; + + get_word(buf1, sizeof(buf1), &p); + + if (!strcmp(buf1, "urn:ietf:params:rtp-hdrext:ssrc-audio-level")) { + rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; + rtsp_st->ssrc_audio_level_ext_id = id; + } } else { if (rt->server_type == RTSP_SERVER_WMS) ff_wms_parse_sdp_a_line(s, p); @@ -868,6 +883,7 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st) s->iformat) { RTPDemuxContext *rtpctx = rtsp_st->transport_priv; rtpctx->ssrc = rtsp_st->ssrc; + rtpctx->ssrc_audio_level_ext_id = rtsp_st->ssrc_audio_level_ext_id; if (rtsp_st->dynamic_handler) { ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv, rtsp_st->dynamic_protocol_context, diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 83b2e3f4fb..4315bbe2c8 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -483,6 +483,8 @@ typedef struct RTSPStream { char crypto_suite[40]; char crypto_params[100]; + + int ssrc_audio_level_ext_id; } RTSPStream; void ff_rtsp_parse_line(AVFormatContext *s, -- 2.41.0

2 1

[PATCH 1/3] lavc/vp8dsp: R-V V put_bilin_h
by flow gg 17 Mar '24

17 Mar '24

2 8

[PATCH] vvcdec: Mark as experimental
by Kieran Kunhya 16 Mar '24

16 Mar '24

$subj As discussed at FOSDEM. Kieran

7 17