11d6bd3f86
Original change's description: > Check vector sizes when crossfading from CNG/expand to normal. > > This fixes a potential out-of-bounds write. > > Bug: chromium:502661101 > Change-Id: I6f03b522643d7a55040d7f5403f342b32d47f0c7 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/464500 > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#47447} (cherry picked from commit ff1be2e8524fe63069fc2406d5e3f3ddccafee1a) Bug: chromium:504500514,chromium:502661101 Change-Id: I6f03b522643d7a55040d7f5403f342b32d47f0c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/465261 Commit-Queue: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Auto-Submit: Chrome Cherry Picker <chrome-cherry-picker@chops-service-accounts.iam.gserviceaccount.com> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/branch-heads/7680@{#7} Cr-Branched-From: d1972add2a63b2a528a6471d447f82e0010b5215-refs/heads/main@{#46853}
73 lines
2.3 KiB
C++
73 lines
2.3 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
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#define MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
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#include <stdint.h>
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#include <string.h> // Access to size_t.
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#include "api/neteq/neteq.h"
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#include "modules/audio_coding/neteq/statistics_calculator.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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// Forward declarations.
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class AudioMultiVector;
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class BackgroundNoise;
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class DecoderDatabase;
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class Expand;
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// This class provides the "Normal" DSP operation, that is performed when
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// there is no data loss, no need to stretch the timing of the signal, and
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// no other "special circumstances" are at hand.
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class Normal {
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public:
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Normal(int fs_hz,
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DecoderDatabase* decoder_database,
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const BackgroundNoise& background_noise,
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Expand* expand,
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StatisticsCalculator* statistics)
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: fs_hz_(fs_hz),
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decoder_database_(decoder_database),
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background_noise_(background_noise),
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expand_(expand),
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samples_per_ms_(CheckedDivExact(fs_hz_, 1000)),
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statistics_(statistics) {}
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virtual ~Normal() {}
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Normal(const Normal&) = delete;
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Normal& operator=(const Normal&) = delete;
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// Performs the "Normal" operation. The decoder data is supplied in `input`,
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// having `length` samples in total for all channels (interleaved). The
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// result is written to `output`. The number of channels allocated in
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// `output` defines the number of channels that will be used when
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// de-interleaving `input`. `last_mode` contains the mode used in the previous
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// GetAudio call (i.e., not the current one).
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int Process(const int16_t* input,
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size_t length,
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NetEq::Mode last_mode,
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AudioMultiVector* output);
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private:
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int fs_hz_;
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DecoderDatabase* decoder_database_;
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const BackgroundNoise& background_noise_;
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Expand* expand_;
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const size_t samples_per_ms_;
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StatisticsCalculator* const statistics_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
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