Index: webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc |
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc b/webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..8252c6a250460fcf47b348a291951443efc1b490 |
--- /dev/null |
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc |
@@ -0,0 +1,64 @@ |
+/* |
+ * 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/mock/mock_audio_encoder.h" |
+ |
+using ::testing::_; |
+using ::testing::Invoke; |
+using ::testing::Return; |
+ |
+namespace webrtc { |
+ |
+TEST(AudioEncoderTest, EncodeInternalRedirectsOk) { |
+ const size_t kPayloadSize = 16; |
+ const uint8_t payload[kPayloadSize] = |
+ {0xf, 0xe, 0xd, 0xc, 0xb, 0xa, 0x9, 0x8, |
+ 0x7, 0x6, 0x5, 0x4, 0x3, 0x2, 0x1, 0x0}; |
+ |
+ MockAudioEncoderDeprecated old_impl; |
+ MockAudioEncoder new_impl; |
+ MockAudioEncoderBase* impls[] = { &old_impl, &new_impl }; |
+ for (auto* impl : impls) { |
+ EXPECT_CALL(*impl, Die()); |
+ EXPECT_CALL(*impl, MaxEncodedBytes()) |
+ .WillRepeatedly(Return(kPayloadSize * 2)); |
+ EXPECT_CALL(*impl, NumChannels()).WillRepeatedly(Return(1)); |
+ EXPECT_CALL(*impl, SampleRateHz()).WillRepeatedly(Return(8000)); |
+ } |
+ |
+ EXPECT_CALL(old_impl, EncodeInternal(_, _, _, _)).WillOnce( |
+ Invoke(MockAudioEncoderDeprecated::CopyEncoding(payload))); |
+ |
+ EXPECT_CALL(new_impl, EncodeInternal(_, _, _)).WillOnce( |
+ Invoke(MockAudioEncoder::CopyEncoding(payload))); |
+ |
+ int16_t audio[80]; |
+ uint8_t output_array[kPayloadSize * 2]; |
+ rtc::Buffer output_buffer; |
+ |
+ AudioEncoder* old_encoder = &old_impl; |
+ AudioEncoder* new_encoder = &new_impl; |
+ auto old_info = old_encoder->Encode(0, audio, &output_buffer); |
+ auto new_info = new_encoder->DEPRECATED_Encode(0, audio, |
+ kPayloadSize * 2, |
+ output_array); |
+ |
+ EXPECT_EQ(old_info.encoded_bytes, kPayloadSize); |
+ EXPECT_EQ(new_info.encoded_bytes, kPayloadSize); |
+ EXPECT_EQ(output_buffer.size(), kPayloadSize); |
+ |
+ for (size_t i = 0; i != kPayloadSize; ++i) { |
+ EXPECT_EQ(output_buffer.data()[i], payload[i]); |
+ EXPECT_EQ(output_array[i], payload[i]); |
+ } |
+} |
+ |
+} // namespace webrtc |