- 30 May, 2020 2 commits
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Paul B Mahol authored
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Paul B Mahol authored
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- 29 May, 2020 4 commits
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Dale Curtis authored
Avoids overflow from fuzzed skip_samples values. Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Dale Curtis authored
Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Michael Niedermayer authored
Fixes: overread by 1 Fixes: 21880/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ADPCM_IMA_CUNNING_fuzzer-5717917221257216.fuzz Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpegSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Lynne authored
We need at least a few bits of entropy to determine the start index of each queue, in order to let filters run in parallel as much as possible, and rand() is not thread safe and disrupts any external API's usage of rand, so instead replace it with av_get_random_seed. While it has more overhead than rand, we only run it once per filter upon init.
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- 28 May, 2020 14 commits
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Andreas Rheinhardt authored
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Andreas Rheinhardt authored
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Andreas Rheinhardt authored
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Andreas Rheinhardt authored
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Andreas Rheinhardt authored
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Andreas Rheinhardt authored
Up until now, the HLS muxer uses av_strtok() to split an input string controlling parameters of the VariantStreams and then duplicates parts of this string containing parameters such as the language or the name of the VariantStream. But these parts are proper zero-terminated strings of their own that are never modified lateron, so one can simply use the substring as-is without creating a copy. This commit implements this. The same also happened for the string controlling the closed caption groups. Furthermore, add const to indicate that the pointers to these substrings are not used to modify them and also to indicate that these strings are not allocated on their own. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Andreas Rheinhardt authored
Up until now, the HLS muxer duplicated a string for every VariantStream, although neither the original nor the copies are ever modified. So use the original directly and stop copying. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Andreas Rheinhardt authored
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
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Jun Zhao authored
commit 32aeba12 missed coding style fix. Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
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Jun Zhao authored
commit 4ed3a01d missed coding style fix. Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
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Limin Wang authored
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Ting Fu authored
Signed-off-by: Ting Fu <ting.fu@intel.com> Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
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Ting Fu authored
more math unary operations will be added here It can be tested with the model file generated with below python scripy: import tensorflow as tf import numpy as np import imageio in_img = imageio.imread('input.jpeg') in_img = in_img.astype(np.float32)/255.0 in_data = in_img[np.newaxis, :] x = tf.placeholder(tf.float32, shape=[1, None, None, 3], name='dnn_in') x1 = tf.subtract(x, 0.5) x2 = tf.abs(x1) y = tf.identity(x2, name='dnn_out') sess=tf.Session() sess.run(tf.global_variables_initializer()) graph_def = tf.graph_util.convert_variables_to_constants(sess, sess.graph_def, ['dnn_out']) tf.train.write_graph(graph_def, '.', 'image_process.pb', as_text=False) print("image_process.pb generated, please use \ path_to_ffmpeg/tools/python/convert.py to generate image_process.model\n") output = sess.run(y, feed_dict={x: in_data}) imageio.imsave("out.jpg", np.squeeze(output)) Signed-off-by: Ting Fu <ting.fu@intel.com> Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
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James Almer authored
Regression since a1133db3Found-by: comex <comexk@gmail.com> Signed-off-by: James Almer <jamrial@gmail.com>
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- 27 May, 2020 20 commits
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Michael Niedermayer authored
Fixes: Timeout (170sec -> 6sec) Fixes: 20956/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HAP_fuzzer-5713643025203200 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpegSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Michael Niedermayer authored
high resolutions with only small blocks appear to be rather slow with the fuzzer + sanitizers. A solution which makes this run faster is welcome. Fixes: Timeout (did not wait -> 17sec) Fixes: 21006/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HEVC_fuzzer-6002552539971584 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpegSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Michael Niedermayer authored
This combination skips allocating large padding which can read out of array Fixes: 20978/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_H264_fuzzer-5746381832847360 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpegSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Marton Balint authored
The old resync logic had some bugs, for example the packet size could stuck into 192 bytes, because pos47_full was not updated for every packet, and for unseekable inputs the resync logic simply skipped some 0x47 sync bytes, therefore the calculated distance between sync bytes was a multiple of 188 bytes. AVIO only buffers a single packet (for UDP/mpegts, that usually means 1316 bytes), so for every ten consecutive 188-byte MPEGTS packets there was always a seek failure, and that caused the old code to not find the 188 byte pattern across 10 consecutive packets. This patch changes the custom logic to the one which is used when probing to determine the packet size. This was already proposed as a FIXME a long time ago...
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Zane van Iperen authored
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Zane van Iperen authored
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Zane van Iperen authored
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com> Reviewed-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Dale Curtis authored
Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Zane van Iperen authored
Uses ff_get_wav_header() in riffdec.c Signed-off-by: Zane van Iperen <zane@zanevaniperen.com> Reviewed-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Limin Wang authored
then ff_h264_free_tables() and h264_decode_end() can be removed in h264_decode_init() if it's failed. The FF_CODEC_CAP_INIT_CLEANUP flag is need for single thread, For multithread, it'll be cleanup still by AV_CODEC_CAP_FRAME_THREADS flag if have. Reviewed-by: Anton Khirnov <anton@khirnov.net> Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Limin Wang authored
then ff_mpv_encode_end() will be unnecessary in ff_mpv_encode_init() if it's failed. The FF_CODEC_CAP_INIT_CLEANUP flag is need for single thread, For multithread, it'll be cleanup still by AV_CODEC_CAP_FRAME_THREADS flag if have. Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Limin Wang authored
then we can remove adpcm_encode_close() in adpcm_encode_init() if have failed. so the goto error lable will be unnecessary and can be removed later. Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Paul B Mahol authored
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Limin Wang authored
Merge the short lines after the last commit Reviewed-by: Marton Balint <cus@passwd.hu> Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Limin Wang authored
Reviewed-by: Marton Balint <cus@passwd.hu> Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
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Anton Khirnov authored
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Anton Khirnov authored
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Anton Khirnov authored
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Dale Curtis authored
7546ac2f made it so that the start_time for mp3 files is adjusted for skip_samples. However, this appears incorrect because subsequent packet timestamps are not adjusted and skip_samples are applied by deleting data from a packet without changing the timestamp. E.g., we are told the start_time is ~25ms and we get a packet with a timestamp of 0 that has had the skip_samples discarded from it. As such rendering engines may incorrectly discard everything prior to the 25ms thinking that is where playback should officially start. Since the samples were deleted without adjusting timestamps though, the true start_time is still 0. Other formats like MP4 with edit lists will adjust both the start time and the timestamps of subsequent packets to avoid this issue. Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Linjie Fu authored
This is accidentally missed while rebasing. Signed-off-by: Linjie Fu <linjie.fu@intel.com>
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