Index: webrtc/modules/audio_mixer/frame_combiner_unittest.cc |
diff --git a/webrtc/modules/audio_mixer/frame_combiner_unittest.cc b/webrtc/modules/audio_mixer/frame_combiner_unittest.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..13c66012f99f27c4fd30d820e19c0166fcf3d1f6 |
--- /dev/null |
+++ b/webrtc/modules/audio_mixer/frame_combiner_unittest.cc |
@@ -0,0 +1,132 @@ |
+/* |
+ * Copyright (c) 2017 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 "webrtc/modules/audio_mixer/frame_combiner.h" |
+ |
+#include <numeric> |
+#include <sstream> |
+#include <string> |
+ |
+#include "webrtc/base/checks.h" |
+#include "webrtc/test/gtest.h" |
+ |
+namespace webrtc { |
+ |
+namespace { |
+std::string ProduceDebugText(int sample_rate_hz, |
+ int number_of_channels, |
+ int number_of_sources) { |
+ std::ostringstream ss; |
+ ss << "Sample rate: " << sample_rate_hz << " "; |
+ ss << "Number of channels: " << number_of_channels << " "; |
+ ss << "Number of sources: " << number_of_sources; |
+ return ss.str(); |
+} |
+ |
+AudioFrame frame1; |
+AudioFrame frame2; |
+AudioFrame audio_frame_for_mixing; |
+ |
+void SetUpFrames(int sample_rate_hz, int number_of_channels) { |
+ for (auto* frame : {&frame1, &frame2}) { |
+ frame->UpdateFrame(-1, 0, nullptr, |
+ rtc::CheckedDivExact(sample_rate_hz, 100), |
+ sample_rate_hz, AudioFrame::kNormalSpeech, |
+ AudioFrame::kVadActive, number_of_channels); |
+ } |
+} |
+} // namespace |
+ |
+TEST(FrameCombiner, BasicApiCallsLimiter) { |
+ FrameCombiner combiner(true); |
+ for (const int rate : {8000, 16000, 32000, 48000}) { |
+ for (const int number_of_channels : {1, 2}) { |
+ const std::vector<AudioFrame*> all_frames = {&frame1, &frame2}; |
+ SetUpFrames(rate, number_of_channels); |
+ |
+ for (const int number_of_frames : {0, 1, 2}) { |
+ SCOPED_TRACE( |
+ ProduceDebugText(rate, number_of_channels, number_of_frames)); |
+ const std::vector<AudioFrame*> frames_to_combine( |
+ all_frames.begin(), all_frames.begin() + number_of_frames); |
+ combiner.Combine(frames_to_combine, number_of_channels, rate, |
+ &audio_frame_for_mixing); |
+ } |
+ } |
+ } |
+} |
+ |
+// No APM limiter means no AudioProcessing::NativeRate restriction |
+// on rate. The rate has to be divisible by 100 since we use |
+// 10 ms frames, though. |
+TEST(FrameCombiner, BasicApiCallsNoLimiter) { |
+ FrameCombiner combiner(false); |
+ for (const int rate : {8000, 10000, 11000, 32000, 44100}) { |
+ for (const int number_of_channels : {1, 2}) { |
+ const std::vector<AudioFrame*> all_frames = {&frame1, &frame2}; |
+ SetUpFrames(rate, number_of_channels); |
+ |
+ for (const int number_of_frames : {0, 1, 2}) { |
+ SCOPED_TRACE( |
+ ProduceDebugText(rate, number_of_channels, number_of_frames)); |
+ const std::vector<AudioFrame*> frames_to_combine( |
+ all_frames.begin(), all_frames.begin() + number_of_frames); |
+ combiner.Combine(frames_to_combine, number_of_channels, rate, |
+ &audio_frame_for_mixing); |
+ } |
+ } |
+ } |
+} |
+ |
+TEST(FrameCombiner, CombiningZeroFramesShouldProduceSilence) { |
+ FrameCombiner combiner(false); |
+ for (const int rate : {8000, 10000, 11000, 32000, 44100}) { |
+ for (const int number_of_channels : {1, 2}) { |
+ SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 0)); |
+ |
+ const std::vector<AudioFrame*> frames_to_combine; |
+ combiner.Combine(frames_to_combine, number_of_channels, rate, |
+ &audio_frame_for_mixing); |
+ |
+ const std::vector<int16_t> mixed_data( |
+ audio_frame_for_mixing.data_, |
+ audio_frame_for_mixing.data_ + number_of_channels * rate / 100); |
+ |
+ const std::vector<int16_t> expected(number_of_channels * rate / 100, 0); |
+ EXPECT_EQ(mixed_data, expected); |
+ } |
+ } |
+} |
+ |
+TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) { |
+ FrameCombiner combiner(false); |
+ for (const int rate : {8000, 10000, 11000, 32000, 44100}) { |
+ for (const int number_of_channels : {1, 2}) { |
+ SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1)); |
+ |
+ SetUpFrames(rate, number_of_channels); |
+ std::iota(frame1.data_, frame1.data_ + number_of_channels * rate / 100, |
+ 0); |
+ const std::vector<AudioFrame*> frames_to_combine = {&frame1}; |
+ combiner.Combine(frames_to_combine, number_of_channels, rate, |
+ &audio_frame_for_mixing); |
+ |
+ const std::vector<int16_t> mixed_data( |
+ audio_frame_for_mixing.data_, |
+ audio_frame_for_mixing.data_ + number_of_channels * rate / 100); |
+ |
+ std::vector<int16_t> expected(number_of_channels * rate / 100); |
+ std::iota(expected.begin(), expected.end(), 0); |
+ EXPECT_EQ(mixed_data, expected); |
+ } |
+ } |
+} |
+ |
+} // namespace webrtc |