619a211562
We detect an unreasonable state (caused by a bad encoded stream) before it can lead to problems, and handle it by resetting the decoder. NOPRESUBMIT=true BUG=chromium:617124 Review-Url: https://codereview.webrtc.org/2255203002 Cr-Commit-Position: refs/heads/master@{#13888}
58 lines
2.4 KiB
C++
58 lines
2.4 KiB
C++
/*
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* Copyright (c) 2016 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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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
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#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
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namespace webrtc {
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TEST(IlbcTest, BadPacket) {
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// Get a good packet.
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AudioEncoderIlbc::Config config;
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config.frame_size_ms = 20; // We need 20 ms rather than the default 30 ms;
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// otherwise, all possible values of cb_index[2]
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// are valid.
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AudioEncoderIlbc encoder(config);
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std::vector<int16_t> samples(encoder.SampleRateHz() / 100, 4711);
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rtc::Buffer packet;
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int num_10ms_chunks = 0;
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while (packet.size() == 0) {
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encoder.Encode(0, samples, &packet);
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num_10ms_chunks += 1;
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}
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// Break the packet by setting all bits of the unsigned 7-bit number
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// cb_index[2] to 1, giving it a value of 127. For a 20 ms packet, this is
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// too large.
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EXPECT_EQ(38u, packet.size());
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rtc::Buffer bad_packet(packet.data(), packet.size());
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bad_packet[29] |= 0x3f; // Bits 1-6.
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bad_packet[30] |= 0x80; // Bit 0.
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// Decode the bad packet. We expect the decoder to respond by returning -1.
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AudioDecoderIlbc decoder;
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std::vector<int16_t> decoded_samples(num_10ms_chunks * samples.size());
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AudioDecoder::SpeechType speech_type;
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EXPECT_EQ(-1, decoder.Decode(bad_packet.data(), bad_packet.size(),
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encoder.SampleRateHz(),
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sizeof(int16_t) * decoded_samples.size(),
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decoded_samples.data(), &speech_type));
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// Decode the good packet. This should work, because the failed decoding
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// should not have left the decoder in a broken state.
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EXPECT_EQ(static_cast<int>(decoded_samples.size()),
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decoder.Decode(packet.data(), packet.size(), encoder.SampleRateHz(),
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sizeof(int16_t) * decoded_samples.size(),
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decoded_samples.data(), &speech_type));
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}
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} // namespace webrtc
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