| 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..a8b76a5bc9e2b5a390dd22863ece84da4cc73d0b 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;
|
| + 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, 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
|
|
|