blob: c23cbc44e808a60185b30a6ad93d18778fdbe160 [file] [log] [blame]
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
namespace webrtc {
TEST(IlbcTest, BadPacket) {
// Get a good packet.
AudioEncoderIlbc::Config config;
config.frame_size_ms = 20; // We need 20 ms rather than the default 30 ms;
// otherwise, all possible values of cb_index[2]
// are valid.
AudioEncoderIlbc encoder(config);
std::vector<int16_t> samples(encoder.SampleRateHz() / 100, 4711);
rtc::Buffer packet;
int num_10ms_chunks = 0;
while (packet.size() == 0) {
encoder.Encode(0, samples, &packet);
num_10ms_chunks += 1;
// Break the packet by setting all bits of the unsigned 7-bit number
// cb_index[2] to 1, giving it a value of 127. For a 20 ms packet, this is
// too large.
EXPECT_EQ(38u, packet.size());
rtc::Buffer bad_packet(, packet.size());
bad_packet[29] |= 0x3f; // Bits 1-6.
bad_packet[30] |= 0x80; // Bit 0.
// Decode the bad packet. We expect the decoder to respond by returning -1.
AudioDecoderIlbc decoder;
std::vector<int16_t> decoded_samples(num_10ms_chunks * samples.size());
AudioDecoder::SpeechType speech_type;
EXPECT_EQ(-1, decoder.Decode(, bad_packet.size(),
sizeof(int16_t) * decoded_samples.size(),, &speech_type));
// Decode the good packet. This should work, because the failed decoding
// should not have left the decoder in a broken state.
decoder.Decode(, packet.size(), encoder.SampleRateHz(),
sizeof(int16_t) * decoded_samples.size(),, &speech_type));
} // namespace webrtc