Chromium Code Reviews| Index: webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc |
| diff --git a/webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc b/webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc |
| index c23cbc44e808a60185b30a6ad93d18778fdbe160..31483693356913264327c3970cde8373f24ccca5 100644 |
| --- a/webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc |
| +++ b/webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc |
| @@ -11,6 +11,7 @@ |
| #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" |
| +#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h" |
| namespace webrtc { |
| @@ -54,4 +55,85 @@ TEST(IlbcTest, BadPacket) { |
| 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) { |
| + AudioDecoderIlbc decoder; |
| + const size_t frame_length_samples = frame_length_ms_ * 8; |
| + auto generate_payload = [] (size_t payload_length_bytes) { |
|
kwiberg-webrtc
2016/09/16 00:48:08
const auto
ossu
2016/09/16 11:46:00
Acknowledged.
|
| + 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, true); |
| + 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_CASE_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) { |
| + AudioDecoderIlbc decoder; |
| + constexpr size_t kPayloadLengthBytes = 950; |
| + const auto results = |
| + decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0, true); |
| + EXPECT_TRUE(results.empty()); |
| +} |
| + |
| +// Payload not an integer number of frames. |
| +TEST(IlbcTest, SplitUnevenPayload) { |
| + AudioDecoderIlbc decoder; |
| + constexpr size_t kPayloadLengthBytes = 39; // Not an even number of frames. |
| + const auto results = |
| + decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0, true); |
| + EXPECT_TRUE(results.empty()); |
| +} |
| + |
| } // namespace webrtc |