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/*
* 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 "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
#include "test/gtest.h"
namespace webrtc {
// TODO(bugs.webrtc.org/345525069): Either fix/enable or remove iLBC.
#if defined(__has_feature) && __has_feature(undefined_behavior_sanitizer)
TEST(IlbcTest, DISABLED_BadPacket) {
#else
TEST(IlbcTest, BadPacket) {
#endif
// Get a good packet.
AudioEncoderIlbcConfig 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.
AudioEncoderIlbcImpl encoder(config, 102);
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.data(), 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.
AudioDecoderIlbcImpl decoder;
std::vector<int16_t> decoded_samples(num_10ms_chunks * samples.size());
AudioDecoder::SpeechType speech_type;
EXPECT_EQ(-1, decoder.Decode(bad_packet.data(), bad_packet.size(),
encoder.SampleRateHz(),
sizeof(int16_t) * decoded_samples.size(),
decoded_samples.data(), &speech_type));
// Decode the good packet. This should work, because the failed decoding
// should not have left the decoder in a broken state.
EXPECT_EQ(static_cast<int>(decoded_samples.size()),
decoder.Decode(packet.data(), packet.size(), encoder.SampleRateHz(),
sizeof(int16_t) * decoded_samples.size(),
decoded_samples.data(), &speech_type));
}
class SplitIlbcTest : public ::testing::TestWithParam<std::pair<int, int> > {
protected:
virtual void SetUp() {
const std::pair<int, int> parameters = GetParam();
num_frames_ = parameters.first;
frame_length_ms_ = parameters.second;
frame_length_bytes_ = (frame_length_ms_ == 20) ? 38 : 50;
}
size_t num_frames_;
int frame_length_ms_;
size_t frame_length_bytes_;
};
TEST_P(SplitIlbcTest, NumFrames) {
AudioDecoderIlbcImpl decoder;
const size_t frame_length_samples = frame_length_ms_ * 8;
const auto generate_payload = [](size_t payload_length_bytes) {
rtc::Buffer payload(payload_length_bytes);
// Fill payload with increasing integers {0, 1, 2, ...}.
for (size_t i = 0; i < payload.size(); ++i) {
payload[i] = static_cast<uint8_t>(i);
}
return payload;
};
const auto results = decoder.ParsePayload(
generate_payload(frame_length_bytes_ * num_frames_), 0);
EXPECT_EQ(num_frames_, results.size());
size_t frame_num = 0;
uint8_t payload_value = 0;
for (const auto& result : results) {
EXPECT_EQ(frame_length_samples * frame_num, result.timestamp);
const LegacyEncodedAudioFrame* frame =
static_cast<const LegacyEncodedAudioFrame*>(result.frame.get());
const rtc::Buffer& payload = frame->payload();
EXPECT_EQ(frame_length_bytes_, payload.size());
for (size_t i = 0; i < payload.size(); ++i, ++payload_value) {
EXPECT_EQ(payload_value, payload[i]);
}
++frame_num;
}
}
// Test 1 through 5 frames of 20 and 30 ms size.
// Also test the maximum number of frames in one packet for 20 and 30 ms.
// The maximum is defined by the largest payload length that can be uniquely
// resolved to a frame size of either 38 bytes (20 ms) or 50 bytes (30 ms).
INSTANTIATE_TEST_SUITE_P(
IlbcTest,
SplitIlbcTest,
::testing::Values(std::pair<int, int>(1, 20), // 1 frame, 20 ms.
std::pair<int, int>(2, 20), // 2 frames, 20 ms.
std::pair<int, int>(3, 20), // And so on.
std::pair<int, int>(4, 20),
std::pair<int, int>(5, 20),
std::pair<int, int>(24, 20),
std::pair<int, int>(1, 30),
std::pair<int, int>(2, 30),
std::pair<int, int>(3, 30),
std::pair<int, int>(4, 30),
std::pair<int, int>(5, 30),
std::pair<int, int>(18, 30)));
// Test too large payload size.
TEST(IlbcTest, SplitTooLargePayload) {
AudioDecoderIlbcImpl decoder;
constexpr size_t kPayloadLengthBytes = 950;
const auto results =
decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0);
EXPECT_TRUE(results.empty());
}
// Payload not an integer number of frames.
TEST(IlbcTest, SplitUnevenPayload) {
AudioDecoderIlbcImpl decoder;
constexpr size_t kPayloadLengthBytes = 39; // Not an even number of frames.
const auto results =
decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0);
EXPECT_TRUE(results.empty());
}
} // namespace webrtc