Index: content/renderer/media/media_stream_audio_processor_unittest.cc |
diff --git a/content/renderer/media/media_stream_audio_processor_unittest.cc b/content/renderer/media/media_stream_audio_processor_unittest.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..2d20dede8dc10b52bff7beba246c6637e16b3be6 |
--- /dev/null |
+++ b/content/renderer/media/media_stream_audio_processor_unittest.cc |
@@ -0,0 +1,166 @@ |
+// Copyright 2013 The Chromium Authors. All rights reserved. |
+// Use of this source code is governed by a BSD-style license that can be |
+// found in the LICENSE file. |
+ |
+#include "base/command_line.h" |
+#include "base/file_util.h" |
+#include "base/files/file_path.h" |
+#include "base/logging.h" |
+#include "base/path_service.h" |
+#include "base/time/time.h" |
+#include "content/public/common/content_switches.h" |
+#include "content/renderer/media/media_stream_audio_processor.h" |
+#include "content/renderer/media/rtc_media_constraints.h" |
+#include "media/audio/audio_parameters.h" |
+#include "media/base/audio_bus.h" |
+#include "testing/gmock/include/gmock/gmock.h" |
+#include "testing/gtest/include/gtest/gtest.h" |
+#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
+ |
+using ::testing::_; |
+using ::testing::AnyNumber; |
+using ::testing::AtLeast; |
+using ::testing::Return; |
+ |
+namespace content { |
+ |
+namespace { |
+ |
+#if defined(ANDROID) |
+const int kAudioProcessingSampleRate = 16000; |
+#else |
+const int kAudioProcessingSampleRate = 32000; |
+#endif |
+const int kAudioProcessingNumberOfChannel = 1; |
+ |
+// The number of packers used for testing. |
+const int kNumberOfPacketsForTest = 100; |
+ |
+void ReadDataFromSpeechFile(char* data, int length) { |
+ base::FilePath file; |
+ CHECK(PathService::Get(base::DIR_SOURCE_ROOT, &file)); |
+ file = file.Append(FILE_PATH_LITERAL("media")) |
+ .Append(FILE_PATH_LITERAL("test")) |
+ .Append(FILE_PATH_LITERAL("data")) |
+ .Append(FILE_PATH_LITERAL("speech_16b_stereo_48kHz.raw")); |
+ DCHECK(base::PathExists(file)); |
+ int64 data_file_size64 = 0; |
+ DCHECK(file_util::GetFileSize(file, &data_file_size64)); |
+ EXPECT_EQ(length, file_util::ReadFile(file, data, length)); |
+ DCHECK(data_file_size64 > length); |
+} |
+ |
+void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) { |
+ // Constant constraint keys which enables default audio constraints on |
+ // mediastreams with audio. |
+ struct { |
+ const char* key; |
+ const char* value; |
+ } static const kDefaultAudioConstraints[] = { |
+ { webrtc::MediaConstraintsInterface::kEchoCancellation, |
+ webrtc::MediaConstraintsInterface::kValueTrue }, |
+ #if defined(OS_CHROMEOS) || defined(OS_MACOSX) |
+ // Enable the extended filter mode AEC on platforms with known echo issues. |
+ { webrtc::MediaConstraintsInterface::kExperimentalEchoCancellation, |
+ webrtc::MediaConstraintsInterface::kValueTrue }, |
+ #endif |
+ { webrtc::MediaConstraintsInterface::kAutoGainControl, |
+ webrtc::MediaConstraintsInterface::kValueTrue }, |
+ { webrtc::MediaConstraintsInterface::kExperimentalAutoGainControl, |
+ webrtc::MediaConstraintsInterface::kValueTrue }, |
+ { webrtc::MediaConstraintsInterface::kNoiseSuppression, |
+ webrtc::MediaConstraintsInterface::kValueTrue }, |
+ { webrtc::MediaConstraintsInterface::kHighpassFilter, |
+ webrtc::MediaConstraintsInterface::kValueTrue }, |
+ }; |
+ |
+ for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) { |
+ constraints->AddMandatory(kDefaultAudioConstraints[i].key, |
+ kDefaultAudioConstraints[i].value, false); |
+ } |
+} |
+ |
+} // namespace |
+ |
+class MediaStreamAudioProcessorTest : public ::testing::Test { |
+ public: |
+ MediaStreamAudioProcessorTest() |
+ : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
+ media::CHANNEL_LAYOUT_STEREO, 48000, 16, 512) { |
+ CommandLine::ForCurrentProcess()->AppendSwitch( |
+ switches::kEnableAudioTrackProcessing); |
+ } |
+ |
+ protected: |
+ // Helper method to save duplicated code. |
+ void ProcessDataAndVerifyFormat(MediaStreamAudioProcessor* audio_processor, |
+ int expected_output_sample_rate, |
+ int expected_output_channels, |
+ int expected_output_buffer_size) { |
+ // Read the audio data from a file. |
+ const int packet_size = |
+ params_.frames_per_buffer() * 2 * params_.channels(); |
+ const size_t length = packet_size * kNumberOfPacketsForTest; |
+ scoped_ptr<char[]> capture_data(new char[length]); |
+ ReadDataFromSpeechFile(capture_data.get(), length); |
+ const int16* data_ptr = reinterpret_cast<const int16*>(capture_data.get()); |
+ scoped_ptr<media::AudioBus> data_bus = media::AudioBus::Create( |
+ params_.channels(), params_.frames_per_buffer()); |
+ for (int i = 0; i < kNumberOfPacketsForTest; ++i) { |
+ data_bus->FromInterleaved(data_ptr, data_bus->frames(), 2); |
+ audio_processor->PushCaptureData(data_bus.get()); |
+ |
+ // |audio_processor| does nothing when the audio processing is off in |
+ // the processor. |
+ audio_processor->PushRenderData( |
+ data_ptr, |
+ params_.sample_rate(), params_.channels(), |
+ params_.frames_per_buffer(), base::TimeDelta::FromMilliseconds(10)); |
+ |
+ int16* output = NULL; |
+ while(audio_processor->ProcessAndConsumeData( |
+ base::TimeDelta::FromMilliseconds(10), 255, false, &output)) { |
+ EXPECT_TRUE(output != NULL); |
+ EXPECT_EQ(audio_processor->OutputFormat().sample_rate(), |
+ expected_output_sample_rate); |
+ EXPECT_EQ(audio_processor->OutputFormat().channels(), |
+ expected_output_channels); |
+ EXPECT_EQ(audio_processor->OutputFormat().frames_per_buffer(), |
+ expected_output_buffer_size); |
+ } |
+ |
+ data_ptr += params_.frames_per_buffer() * params_.channels(); |
+ } |
+ } |
+ |
+ media::AudioParameters params_; |
+}; |
+ |
+TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) { |
+ // Setup the audio processor with empty constraint. |
+ RTCMediaConstraints constraints; |
+ MediaStreamAudioProcessor audio_processor(&constraints); |
+ audio_processor.SetCaptureFormat(params_); |
+ EXPECT_FALSE(audio_processor.has_audio_processing()); |
+ |
+ ProcessDataAndVerifyFormat(&audio_processor, |
+ params_.sample_rate(), |
+ params_.channels(), |
+ params_.sample_rate() / 100); |
+} |
+ |
+TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) { |
+ // Setup the audio processor with default constraint. |
+ RTCMediaConstraints constraints; |
+ ApplyFixedAudioConstraints(&constraints); |
+ MediaStreamAudioProcessor audio_processor(&constraints); |
+ audio_processor.SetCaptureFormat(params_); |
+ EXPECT_TRUE(audio_processor.has_audio_processing()); |
+ |
+ ProcessDataAndVerifyFormat(&audio_processor, |
+ kAudioProcessingSampleRate, |
+ kAudioProcessingNumberOfChannel, |
+ kAudioProcessingSampleRate / 100); |
+} |
+ |
+} // namespace content |