Sync upstream GStreamer into ignis-main #11
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,187 @@ | ||
| /* | ||
| * Copyright 2026 Google Inc. | ||
| * author: Arthur SC Chan <arthur.chan@adalogics.com> | ||
| * | ||
| * Licensed under the Apache License, Version 2.0 (the "License"); | ||
| * you may not use this file except in compliance with the License. | ||
| * You may obtain a copy of the License at | ||
| * | ||
| * http://www.apache.org/licenses/LICENSE-2.0 | ||
| * | ||
| * Unless required by applicable law or agreed to in writing, software | ||
| * distributed under the License is distributed on an "AS IS" BASIS, | ||
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. | ||
| * See the License for the specific language governing permissions and | ||
| * limitations under the License. | ||
| */ | ||
|
|
||
| /* Audio converter/resampler/quantize fuzzing target | ||
| * | ||
| * Exercises: | ||
| * gst-libs/gst/audio/audio-converter.c | ||
| * gst-libs/gst/audio/audio-resampler.c | ||
| * gst-libs/gst/audio/audio-quantize.c | ||
| */ | ||
|
|
||
| #ifdef HAVE_CONFIG_H | ||
| #include "config.h" | ||
| #endif | ||
|
|
||
| #include <stdlib.h> | ||
| #include <string.h> | ||
| #include <glib.h> | ||
| #include <gst/gst.h> | ||
| #include <gst/audio/audio.h> | ||
|
|
||
| #define HEADER_SIZE 8 | ||
| #define MAX_FRAMES 4096 | ||
| #define MAX_RESAMPLE_BYTES (16 * 1024 * 1024) | ||
|
|
||
| /* All formats exercised by the converter path */ | ||
| static const GstAudioFormat all_formats[] = { | ||
| GST_AUDIO_FORMAT_S8, GST_AUDIO_FORMAT_U8, | ||
| GST_AUDIO_FORMAT_S16LE, GST_AUDIO_FORMAT_U16LE, | ||
| GST_AUDIO_FORMAT_S24LE, GST_AUDIO_FORMAT_S32LE, | ||
| GST_AUDIO_FORMAT_U32LE, GST_AUDIO_FORMAT_F32LE, | ||
| GST_AUDIO_FORMAT_F64LE, | ||
| }; | ||
|
|
||
| #define N_ALL_FORMATS G_N_ELEMENTS(all_formats) | ||
|
|
||
| /* Resampler only accepts these four formats */ | ||
| static const GstAudioFormat resamp_formats[] = { | ||
| GST_AUDIO_FORMAT_S16, GST_AUDIO_FORMAT_S32, | ||
| GST_AUDIO_FORMAT_F32, GST_AUDIO_FORMAT_F64, | ||
| }; | ||
|
|
||
| #define N_RESAMP_FORMATS G_N_ELEMENTS(resamp_formats) | ||
|
|
||
| static const gint rates[] = { | ||
| 100, 997, 7919, | ||
| 8000, 11025, 16000, 22050, | ||
| 32000, 44100, 48000, 96000, | ||
| 8009, 22051, 44101 | ||
| }; | ||
|
|
||
| #define N_RATES G_N_ELEMENTS(rates) | ||
|
|
||
| /* Channel counts the audio converter's mix matrix supports out of the box. */ | ||
| static const gint mixable_channels[] = { 1, 2, 3, 4, 5, 6, 7, 8 }; | ||
|
|
||
| #define N_MIXABLE_CHANNELS G_N_ELEMENTS(mixable_channels) | ||
|
|
||
| static void | ||
| custom_logger (const gchar * log_domain, | ||
| GLogLevelFlags log_level, const gchar * message, gpointer unused_data) | ||
| { | ||
| if (log_level & G_LOG_LEVEL_CRITICAL) { | ||
| g_printerr ("CRITICAL ERROR : %s\n", message); | ||
| abort (); | ||
| } else if (log_level & G_LOG_LEVEL_WARNING) { | ||
| g_printerr ("WARNING : %s\n", message); | ||
| } | ||
| } | ||
|
|
||
| int | ||
| LLVMFuzzerTestOneInput (const guint8 * data, size_t size) | ||
| { | ||
| static gboolean initialized = FALSE; | ||
|
|
||
| if (!initialized) { | ||
| g_log_set_always_fatal (G_LOG_LEVEL_CRITICAL); | ||
| g_log_set_default_handler (custom_logger, NULL); | ||
| gst_init (NULL, NULL); | ||
| initialized = TRUE; | ||
| } | ||
|
|
||
| if (size < HEADER_SIZE) | ||
| return 0; | ||
|
|
||
| GstAudioFormat in_fmt = all_formats[data[0] % N_ALL_FORMATS]; | ||
| GstAudioFormat out_fmt = all_formats[data[1] % N_ALL_FORMATS]; | ||
|
|
||
| /* Picked from the supported mix-matrix set so the converter actually runs. */ | ||
| gint in_ch = mixable_channels[data[2] % N_MIXABLE_CHANNELS]; | ||
| gint in_rate = rates[data[3] % N_RATES]; | ||
| gint out_rate = rates[data[4] % N_RATES]; | ||
| GstAudioResamplerMethod rm = (GstAudioResamplerMethod) ((data[5] >> 4) % 5); | ||
| GstAudioFormat resamp_fmt = resamp_formats[data[5] % N_RESAMP_FORMATS]; | ||
| GstAudioDitherMethod dither = (GstAudioDitherMethod) ((data[6] >> 4) % 4); | ||
| GstAudioNoiseShapingMethod ns = | ||
| (GstAudioNoiseShapingMethod) ((data[6] & 0x0f) % 5); | ||
| guint quantizer = (guint) data[6] + 1; | ||
|
|
||
| const guint8 *audio_data = data + HEADER_SIZE; | ||
| gsize audio_size = size - HEADER_SIZE; | ||
|
|
||
| /* audio converter */ | ||
| { | ||
| gint out_ch = mixable_channels[data[7] % N_MIXABLE_CHANNELS]; | ||
| GstAudioInfo in_info, out_info; | ||
| GstAudioConverter *conv; | ||
|
|
||
| gst_audio_info_set_format (&in_info, in_fmt, in_rate, in_ch, NULL); | ||
| gst_audio_info_set_format (&out_info, out_fmt, out_rate, out_ch, NULL); | ||
| conv = gst_audio_converter_new (0, &in_info, &out_info, NULL); | ||
| if (conv) { | ||
| if (audio_size > 0) { | ||
| gpointer out = NULL; | ||
| gsize out_size = 0; | ||
| gst_audio_converter_convert (conv, 0, (gpointer) audio_data, | ||
| audio_size, &out, &out_size); | ||
| g_free (out); | ||
| } | ||
| gst_audio_converter_free (conv); | ||
| } | ||
| } | ||
|
There was a problem hiding this comment. Converter fuzzer can exhaust memoryMedium Severity The audio converter path calls   Reviewed by Cursor Bugbot for commit 36c4354. Configure here. |
||
|
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| /* audio resampler */ | ||
| { | ||
| GstAudioResampler *resampler; | ||
|
|
||
| resampler = gst_audio_resampler_new (rm, 0, resamp_fmt, in_ch, | ||
| in_rate, out_rate, NULL); | ||
| if (resampler) { | ||
| const GstAudioFormatInfo *finfo = gst_audio_format_get_info (resamp_fmt); | ||
| gsize frame_size = (finfo->width / 8) * in_ch; | ||
| if (frame_size > 0 && audio_size >= frame_size) { | ||
| gsize in_frames = MIN (audio_size / frame_size, MAX_FRAMES); | ||
| gsize out_frames = | ||
| gst_audio_resampler_get_out_frames (resampler, in_frames); | ||
| if (out_frames > 0) { | ||
| gsize out_buf_size = out_frames * (finfo->width / 8) * in_ch; | ||
| /* Cap allocation to keep the fuzzer alive on extreme rate ratios. */ | ||
| if (out_buf_size > 0 && out_buf_size <= MAX_RESAMPLE_BYTES) { | ||
| gpointer out_buf = g_malloc0 (out_buf_size); | ||
| gpointer in_ptrs[1] = { (gpointer) audio_data }; | ||
| gpointer out_ptrs[1] = { out_buf }; | ||
| gst_audio_resampler_resample (resampler, in_ptrs, in_frames, | ||
| out_ptrs, out_frames); | ||
| g_free (out_buf); | ||
| } | ||
| } | ||
| } | ||
| gst_audio_resampler_free (resampler); | ||
| } | ||
| } | ||
|
|
||
| /* audio quantize */ | ||
| { | ||
| GstAudioQuantize *quant = gst_audio_quantize_new (dither, ns, 0, | ||
| GST_AUDIO_FORMAT_S32, 1, quantizer); | ||
| if (quant) { | ||
| const gsize frame_size = sizeof (gint32); | ||
| if (audio_size >= frame_size) { | ||
| guint frames = (guint) MIN (audio_size / frame_size, MAX_FRAMES); | ||
| gpointer out_buf = g_malloc0 (frames * frame_size); | ||
| gpointer in_ptrs[1] = { (gpointer) audio_data }; | ||
| gpointer out_ptrs[1] = { out_buf }; | ||
| gst_audio_quantize_samples (quant, in_ptrs, out_ptrs, frames); | ||
| g_free (out_buf); | ||
| } | ||
| gst_audio_quantize_free (quant); | ||
| } | ||
| } | ||
|
|
||
| return 0; | ||
| } | ||
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Cerbero security skip rule order
Medium Severity
The new
gstreamer-securitywhen: neverrule oncerbero triggersits after- if: '$MESON_COMMIT'. GitLab applies the first matching rule, so a non-emptyMESON_COMMITstill starts Cerbero ongstreamer-security, unlike Windows/macOS jobs that check the project name first.Reviewed by Cursor Bugbot for commit a53b91e. Configure here.