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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-09-05 13:14:37 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-09-05 13:14:37 +0000
commitfd3b2704efc2b206784615c1a23eb25501842259 (patch)
tree61ba3a8af2a0ae2ac9ec362bbf18b038f5dc0448 /oss-fuzz/encoder_v2.cc
parentInitial commit. (diff)
downloadflac-fd3b2704efc2b206784615c1a23eb25501842259.tar.xz
flac-fd3b2704efc2b206784615c1a23eb25501842259.zip
Adding upstream version 1.4.3+ds.upstream/1.4.3+dsupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'oss-fuzz/encoder_v2.cc')
-rw-r--r--oss-fuzz/encoder_v2.cc352
1 files changed, 352 insertions, 0 deletions
diff --git a/oss-fuzz/encoder_v2.cc b/oss-fuzz/encoder_v2.cc
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+/* fuzzer_encoder_v2
+ * Copyright (C) 2022-2023 Xiph.Org Foundation
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include <cstdlib>
+#include <cstring> /* for memcpy */
+#include "FLAC/stream_encoder.h"
+#include "FLAC/metadata.h"
+extern "C" {
+#include "share/private.h"
+}
+#include "common.h"
+
+/* This C++ fuzzer uses the FLAC and not FLAC++ because the latter lacks a few
+ * hidden functions like FLAC__stream_encoder_disable_constant_subframes. It
+ * is still processed by a C++ compiler because that's what oss-fuzz expects */
+
+
+static FLAC__StreamEncoderWriteStatus write_callback(const FLAC__StreamEncoder *encoder, const FLAC__byte buffer[], size_t bytes, uint32_t samples, uint32_t current_frame, void *client_data)
+{
+ (void)encoder, (void)buffer, (void)bytes, (void)samples, (void)current_frame, (void)client_data;
+ return FLAC__STREAM_ENCODER_WRITE_STATUS_OK;
+}
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size)
+{
+ FLAC__bool encoder_valid = true;
+ FLAC__StreamEncoder *encoder = 0;
+ FLAC__StreamEncoderState state;
+ FLAC__StreamMetadata *metadata[16] = {NULL};
+ unsigned num_metadata = 0;
+ FLAC__StreamMetadata_VorbisComment_Entry VorbisCommentField;
+
+ unsigned sample_rate, channels, bps;
+ uint64_t samples_estimate, samples_in_input;
+ unsigned compression_level, input_data_width, blocksize, max_lpc_order, qlp_coeff_precision, min_residual_partition_order, max_residual_partition_order, metadata_mask, instruction_set_disable_mask;
+ FLAC__bool ogg, write_to_file, interleaved;
+
+ FLAC__bool data_bools[24];
+
+ /* Set alloc threshold. This check was added later and no spare config
+ * bytes were left, so we're reusing the sample rate as that of little
+ * consequence to the encoder and decoder except reading the frame header */
+
+ if(size < 3)
+ return 0;
+ alloc_check_threshold = data[2];
+ alloc_check_counter = 0;
+
+ /* allocate the encoder */
+ if((encoder = FLAC__stream_encoder_new()) == NULL) {
+ fprintf(stderr, "ERROR: allocating encoder\n");
+ return 1;
+ }
+
+ /* Use first 20 byte for configuration */
+ if(size < 20){
+ FLAC__stream_encoder_delete(encoder);
+ return 0;
+ }
+
+ /* First 3 byte for sample rate, 4th byte for channels, 5th byte for bps */
+ sample_rate = ((unsigned)data[0] << 16) + ((unsigned)data[1] << 8) + data[2];
+ channels = data[3];
+ bps = data[4];
+
+ /* Number of samples estimate, format accepts 36-bit max */
+ samples_estimate = ((uint64_t)data[5] << 32) + ((unsigned)data[6] << 24) + ((unsigned)data[7] << 16) + ((unsigned)data[8] << 8) + data[9];
+
+ compression_level = data[10]&0b1111;
+ input_data_width = 1 + (data[10]>>4)%4;
+ samples_in_input = (size-20)/input_data_width;
+ blocksize = ((unsigned)data[11] << 8) + (unsigned)data[12];
+ max_lpc_order = data[13];
+ qlp_coeff_precision = data[14];
+ min_residual_partition_order = data[15] & 0b1111;
+ max_residual_partition_order = data[15] & 0b11110000;
+ metadata_mask = data[16];
+ instruction_set_disable_mask = data[17];
+
+ /* Get array of bools from configuration */
+ for(int i = 0; i < 16; i++)
+ data_bools[i] = data[18+i/8] & (1 << (i % 8));
+
+ ogg = data_bools[0];
+ interleaved = data_bools[1];
+ write_to_file = data_bools[13];
+
+ /* Set input and process parameters */
+ encoder_valid &= FLAC__stream_encoder_set_verify(encoder, data_bools[2]);
+ encoder_valid &= FLAC__stream_encoder_set_channels(encoder, channels);
+ encoder_valid &= FLAC__stream_encoder_set_bits_per_sample(encoder, bps);
+ encoder_valid &= FLAC__stream_encoder_set_sample_rate(encoder, sample_rate);
+ encoder_valid &= FLAC__stream_encoder_set_total_samples_estimate(encoder, samples_estimate);
+ encoder_valid &= FLAC__stream_encoder_disable_instruction_set(encoder, instruction_set_disable_mask);
+ encoder_valid &= FLAC__stream_encoder_set_limit_min_bitrate(encoder, data_bools[15]);
+
+ /* Set compression related parameters */
+ encoder_valid &= FLAC__stream_encoder_set_compression_level(encoder, compression_level);
+ if(data_bools[3]){
+ /* Bias towards regular compression levels */
+ encoder_valid &= FLAC__stream_encoder_set_blocksize(encoder, blocksize);
+ encoder_valid &= FLAC__stream_encoder_set_max_lpc_order(encoder, max_lpc_order);
+ encoder_valid &= FLAC__stream_encoder_set_qlp_coeff_precision(encoder, qlp_coeff_precision);
+ encoder_valid &= FLAC__stream_encoder_set_min_residual_partition_order(encoder, min_residual_partition_order);
+
+ /* With large inputs and expensive options enabled, the fuzzer can get *really* slow.
+ * Some combinations can make the fuzzer timeout (>60 seconds). However, while combining
+ * options makes the fuzzer slower, most options do not expose new code when combined.
+ * Therefore, combining slow options is disabled for large inputs. Any input containing
+ * more than 65536 * 2 samples (max blocksize, stereo) is considered large
+ */
+ if(samples_in_input < (2*65536)) {
+ encoder_valid &= FLAC__stream_encoder_set_streamable_subset(encoder, data_bools[4]);
+ encoder_valid &= FLAC__stream_encoder_set_do_qlp_coeff_prec_search(encoder, data_bools[5]);
+ encoder_valid &= FLAC__stream_encoder_set_do_escape_coding(encoder, data_bools[6]);
+ encoder_valid &= FLAC__stream_encoder_set_do_exhaustive_model_search(encoder, data_bools[7]);
+ /* Combining model search, precision search and a high residual partition order is especially
+ * expensive, so limit that even further. This high partition order can only be set on
+ * large blocksize and with streamable subset disabled */
+ if(samples_in_input < (2 * 4609) || data_bools[4] || !data_bools[7] || !data_bools[5] || max_residual_partition_order < 9 || blocksize < 4609)
+ encoder_valid &= FLAC__stream_encoder_set_max_residual_partition_order(encoder, max_residual_partition_order);
+ }
+ else {
+ if(!data_bools[4])
+ encoder_valid &= FLAC__stream_encoder_set_streamable_subset(encoder, false);
+ else if(data_bools[6])
+ encoder_valid &= FLAC__stream_encoder_set_do_escape_coding(encoder, true);
+ else if(data_bools[7])
+ encoder_valid &= FLAC__stream_encoder_set_do_exhaustive_model_search(encoder, true);
+ else if(data_bools[5])
+ encoder_valid &= FLAC__stream_encoder_set_do_qlp_coeff_prec_search(encoder, true);
+ }
+ encoder_valid &= FLAC__stream_encoder_set_do_mid_side_stereo(encoder, data_bools[8]);
+ encoder_valid &= FLAC__stream_encoder_set_loose_mid_side_stereo(encoder, data_bools[9]);
+
+ encoder_valid &= FLAC__stream_encoder_disable_constant_subframes(encoder, data_bools[10]);
+ encoder_valid &= FLAC__stream_encoder_disable_fixed_subframes(encoder, data_bools[11]);
+ encoder_valid &= FLAC__stream_encoder_disable_verbatim_subframes(encoder, data_bools[12]);
+ }
+
+ /* Disable alloc check if requested */
+ if(encoder_valid && data_bools[14])
+ alloc_check_threshold = INT32_MAX;
+
+ /* add metadata */
+ if(encoder_valid && (metadata_mask & 1)) {
+ if((metadata[num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_STREAMINFO)) == NULL)
+ encoder_valid = false;
+ else
+ num_metadata++;
+ }
+ if(encoder_valid && (metadata_mask & 2) && size > 21){
+ if((metadata[num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_PADDING)) == NULL)
+ encoder_valid = false;
+ else {
+ metadata[num_metadata++]->length = (((unsigned)data[20]) << 8) + (unsigned)(data[21]);
+ }
+ }
+ if(encoder_valid && (metadata_mask & 4) && size > 20){
+ FLAC__byte * application_data = (FLAC__byte *)malloc(size-20);
+ if(0 != application_data && ((metadata[num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_APPLICATION)) == NULL))
+ encoder_valid = false;
+ else {
+ memcpy(application_data,data+20,size-20);
+ FLAC__metadata_object_application_set_data(metadata[num_metadata++], application_data, size-20, 0);
+ }
+ }
+ if(encoder_valid && (metadata_mask & 8) && size > 25){
+ if((metadata[num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_SEEKTABLE)) == NULL)
+ encoder_valid = false;
+ else {
+ unsigned seekpoint_spacing = ((unsigned)data[22] << 8) + data[23];
+ unsigned total_samples_for_seekpoints = ((unsigned)data[24] << 8) + data[25];
+ FLAC__metadata_object_seektable_template_append_spaced_points_by_samples(metadata[num_metadata++], seekpoint_spacing, total_samples_for_seekpoints);
+ }
+ }
+ if(encoder_valid && (metadata_mask & 16)){
+ if((metadata[num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT)) != NULL) {
+ bool vorbiscomment_valid = true;
+ /* Append a vorbis comment */
+ if(!FLAC__metadata_object_vorbiscomment_entry_from_name_value_pair(&VorbisCommentField, "COMMENTARY", "Nothing to 🤔 report"))
+ vorbiscomment_valid = false;
+ else {
+ if(FLAC__metadata_object_vorbiscomment_append_comment(metadata[num_metadata], VorbisCommentField, false)) {
+
+ /* Insert a vorbis comment at the first index */
+ if(!FLAC__metadata_object_vorbiscomment_entry_from_name_value_pair(&VorbisCommentField, "COMMENTARY", "Still nothing to report 🤔🤣"))
+ vorbiscomment_valid = false;
+ else
+ if(!FLAC__metadata_object_vorbiscomment_insert_comment(metadata[num_metadata], 0, VorbisCommentField, false)) {
+ free(VorbisCommentField.entry);
+ vorbiscomment_valid = false;
+ }
+ }
+ else {
+ free(VorbisCommentField.entry);
+ vorbiscomment_valid = false;
+ }
+ }
+ if(!vorbiscomment_valid) {
+ FLAC__metadata_object_delete(metadata[num_metadata]);
+ metadata[num_metadata] = 0;
+ }
+ else
+ num_metadata++;
+ }
+ }
+ if(encoder_valid && (metadata_mask & 32)){
+ if((metadata[num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_CUESHEET)) != NULL) {
+ if(!FLAC__metadata_object_cuesheet_insert_blank_track(metadata[num_metadata],0)) {
+ FLAC__metadata_object_delete(metadata[num_metadata]);
+ metadata[num_metadata] = 0;
+ }
+ else {
+ if(!FLAC__metadata_object_cuesheet_track_insert_blank_index(metadata[num_metadata],0,0)) {
+ FLAC__metadata_object_delete(metadata[num_metadata]);
+ metadata[num_metadata] = 0;
+ }
+ else {
+ metadata[num_metadata]->data.cue_sheet.tracks[0].number = 1;
+ num_metadata++;
+ }
+ }
+ }
+ }
+ if(encoder_valid && (metadata_mask & 64)){
+ if((metadata[num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_PICTURE)) != NULL) {
+ num_metadata++;
+ }
+ }
+ if(encoder_valid && (metadata_mask & 128)){
+ if((metadata[num_metadata] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_UNDEFINED)) != NULL) {
+ metadata[num_metadata]->length = 24;
+ metadata[num_metadata]->data.unknown.data = (FLAC__byte *)calloc(24, 1);
+ num_metadata++;
+ }
+ }
+
+ if(num_metadata && encoder_valid)
+ encoder_valid = FLAC__stream_encoder_set_metadata(encoder, metadata, num_metadata);
+
+ /* initialize encoder */
+ if(encoder_valid) {
+ FLAC__StreamEncoderInitStatus init_status;
+ if(ogg)
+ if(write_to_file)
+ init_status = FLAC__stream_encoder_init_ogg_file(encoder, "/tmp/tmp.flac", NULL, NULL);
+ else
+ init_status = FLAC__stream_encoder_init_ogg_stream(encoder, NULL, write_callback, NULL, NULL, NULL, NULL);
+ else
+ if(write_to_file)
+ init_status = FLAC__stream_encoder_init_file(encoder, "/tmp/tmp.flac", NULL, NULL);
+ else
+ init_status = FLAC__stream_encoder_init_stream(encoder, write_callback, NULL, NULL, NULL, NULL);
+ if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
+ encoder_valid = false;
+ }
+ }
+
+
+ /* send samples to encoder */
+ if(encoder_valid && size > (input_data_width*channels+26)) {
+ unsigned samples = (size - 26)/input_data_width/channels;
+ const uint8_t * pcm_data = data + 26;
+ int32_t * data_as_int32 = (int32_t *)malloc(4*samples*channels);
+ if(0 != data_as_int32){
+ for(unsigned i = 0; i < samples*channels; i++)
+ if(input_data_width == 1)
+ data_as_int32[i] = (int32_t)pcm_data[i] - 0x80;
+ else if(input_data_width == 2)
+ data_as_int32[i] = (((int32_t)pcm_data[i*2] << 8) + pcm_data[i*2+1]) - 0x8000;
+ else if(input_data_width == 3)
+ data_as_int32[i] = (((int32_t)pcm_data[i*3] << 16) + ((int32_t)pcm_data[i*3+1] << 8) + pcm_data[i*3+2]) - 0x800000;
+ else if(input_data_width == 4)
+ data_as_int32[i] = (((int64_t)pcm_data[i*4] << 24) + ((int32_t)pcm_data[i*4+1] << 16) + ((int32_t)pcm_data[i*4+2] << 8) + pcm_data[i*4+3]) - 0x80000000;
+
+ /* feed samples to encoder */
+ if(interleaved)
+ encoder_valid = FLAC__stream_encoder_process_interleaved(encoder, data_as_int32, samples);
+ else {
+ encoder_valid = FLAC__stream_encoder_process(encoder, (const int32_t*[]){data_as_int32,
+ data_as_int32+samples,
+ data_as_int32+samples*2,
+ data_as_int32+samples*3,
+ data_as_int32+samples*4, data_as_int32+samples*5, data_as_int32+samples*6, data_as_int32+samples*7}, samples);
+ }
+ free(data_as_int32);
+ }
+ else {
+ encoder_valid = false;
+ }
+ }
+
+ state = FLAC__stream_encoder_get_state(encoder);
+ if(!(state == FLAC__STREAM_ENCODER_OK ||
+ state == FLAC__STREAM_ENCODER_UNINITIALIZED ||
+ state == FLAC__STREAM_ENCODER_CLIENT_ERROR ||
+ ((state == FLAC__STREAM_ENCODER_MEMORY_ALLOCATION_ERROR ||
+ state == FLAC__STREAM_ENCODER_FRAMING_ERROR ||
+ (state == FLAC__STREAM_ENCODER_VERIFY_DECODER_ERROR &&
+ FLAC__stream_encoder_get_verify_decoder_state(encoder) == FLAC__STREAM_DECODER_MEMORY_ALLOCATION_ERROR)) &&
+ alloc_check_threshold < INT32_MAX))) {
+ fprintf(stderr,"-----\nERROR: stream encoder returned %s\n-----\n",FLAC__stream_encoder_get_resolved_state_string(encoder));
+ if(state == FLAC__STREAM_ENCODER_VERIFY_MISMATCH_IN_AUDIO_DATA) {
+ uint32_t frame_number, channel, sample_number;
+ FLAC__int32 expected, got;
+ FLAC__stream_encoder_get_verify_decoder_error_stats(encoder, NULL, &frame_number, &channel, &sample_number, &expected, &got);
+ fprintf(stderr,"Frame number %d\nChannel %d\n Sample number %d\nExpected value %d\nGot %d\n", frame_number, channel, sample_number, expected, got);
+ }
+ abort();
+ }
+
+ FLAC__stream_encoder_finish(encoder);
+
+ /* now that encoding is finished, the metadata can be freed */
+ for(unsigned i = 0; i < 16; i++)
+ if(0 != metadata[i])
+ FLAC__metadata_object_delete(metadata[i]);
+
+ FLAC__stream_encoder_delete(encoder);
+
+ return 0;
+}
+