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Unified Diff: webrtc/voice_engine/test/auto_test/voe_output_test.cc

Issue 1415173005: Prevent Opus DTX from generating intermittent noise during silence (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: on Fredrik's comments Created 5 years, 1 month ago
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Index: webrtc/voice_engine/test/auto_test/voe_output_test.cc
diff --git a/webrtc/voice_engine/test/auto_test/voe_output_test.cc b/webrtc/voice_engine/test/auto_test/voe_output_test.cc
new file mode 100644
index 0000000000000000000000000000000000000000..6a842b80b9a0819a3f2af0675387886aa32a8a71
--- /dev/null
+++ b/webrtc/voice_engine/test/auto_test/voe_output_test.cc
@@ -0,0 +1,198 @@
+/*
+ * Copyright (c) 2015 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/base/scoped_ptr.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/random.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
+
+namespace {
+
+const char kIp[] = "127.0.0.1";
+const int kPort = 1234;
+const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
+
+} // namespace
+
+namespace voetest {
+
+using webrtc::test::Random;
+using webrtc::test::VoiceChannelTransport;
+
+// This test allows a check on the output signal in an end-to-end call.
+class OutputTest {
+ public:
+ OutputTest(int16_t lower_bound, int16_t upper_bound);
+ ~OutputTest();
+
+ void Start();
+
+ void EnableOutputCheck();
+ void DisableOutputCheck();
+ void SetOutputBound(int16_t lower_bound, int16_t upper_bound);
+ void Mute();
+ void Unmute();
+ void SetBitRate(int rate);
+
+ private:
+ // This class checks all output values and count the number of samples that
+ // go out of a defined range.
+ class VoEOutputCheckMediaProcess : public VoEMediaProcess {
+ public:
+ VoEOutputCheckMediaProcess(int16_t lower_bound, int16_t upper_bound);
+
+ void set_enabled(bool enabled) { enabled_ = enabled; }
+ void Process(int channel,
+ ProcessingTypes type,
+ int16_t audio10ms[],
+ size_t length,
+ int samplingFreq,
+ bool isStereo) override;
+
+ private:
+ bool enabled_;
+ int16_t lower_bound_;
+ int16_t upper_bound_;
+ };
+
+ VoETestManager manager_;
+ VoEOutputCheckMediaProcess output_checker_;
+
+ int channel_;
+};
+
+OutputTest::OutputTest(int16_t lower_bound, int16_t upper_bound)
+ : output_checker_(lower_bound, upper_bound) {
+ EXPECT_TRUE(manager_.Init());
+ manager_.GetInterfaces();
+
+ VoEBase* base = manager_.BasePtr();
+ VoECodec* codec = manager_.CodecPtr();
+ VoENetwork* network = manager_.NetworkPtr();
+
+ EXPECT_EQ(0, base->Init());
+
+ channel_ = base->CreateChannel();
+
+ // |network| will take care of the life time of |transport|.
+ VoiceChannelTransport* transport =
+ new VoiceChannelTransport(network, channel_);
+
+ EXPECT_EQ(0, transport->SetSendDestination(kIp, kPort));
+ EXPECT_EQ(0, transport->SetLocalReceiver(kPort));
+
+ EXPECT_EQ(0, codec->SetSendCodec(channel_, kCodecInst));
+ EXPECT_EQ(0, codec->SetOpusDtx(channel_, true));
+
+ EXPECT_EQ(0, manager_.VolumeControlPtr()->SetSpeakerVolume(255));
+
+ manager_.ExternalMediaPtr()->RegisterExternalMediaProcessing(
+ channel_, ProcessingTypes::kPlaybackPerChannel, output_checker_);
+}
+
+OutputTest::~OutputTest() {
+ EXPECT_EQ(0, manager_.NetworkPtr()->DeRegisterExternalTransport(channel_));
+ EXPECT_EQ(0, manager_.ReleaseInterfaces());
+}
+
+void OutputTest::Start() {
+ const std::string file_name =
+ webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+ const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
+
+ ASSERT_EQ(0, manager_.FilePtr()->StartPlayingFileAsMicrophone(
+ channel_, file_name.c_str(), true, false, kInputFormat, 1.0));
+
+ VoEBase* base = manager_.BasePtr();
+ ASSERT_EQ(0, base->StartPlayout(channel_));
+ ASSERT_EQ(0, base->StartSend(channel_));
+}
+
+void OutputTest::EnableOutputCheck() {
+ output_checker_.set_enabled(true);
+}
+
+void OutputTest::DisableOutputCheck() {
+ output_checker_.set_enabled(false);
+}
+
+void OutputTest::Mute() {
+ manager_.VolumeControlPtr()->SetInputMute(channel_, true);
+}
+
+void OutputTest::Unmute() {
+ manager_.VolumeControlPtr()->SetInputMute(channel_, false);
+}
+
+void OutputTest::SetBitRate(int rate) {
+ manager_.CodecPtr()->SetBitRate(channel_, rate);
+}
+
+OutputTest::VoEOutputCheckMediaProcess::VoEOutputCheckMediaProcess(
+ int16_t lower_bound, int16_t upper_bound)
+ : enabled_(false),
+ lower_bound_(-lower_bound),
+ upper_bound_(upper_bound) {}
+
+void OutputTest::VoEOutputCheckMediaProcess::Process(int channel,
+ ProcessingTypes type,
+ int16_t* audio10ms,
+ size_t length,
+ int samplingFreq,
+ bool isStereo) {
+ if (!enabled_)
+ return;
+ const int num_channels = isStereo ? 2 : 1;
+ for (size_t i = 0; i < length; ++i) {
+ for (int c = 0; c < num_channels; ++c) {
+ ASSERT_GE(audio10ms[i * num_channels + c], lower_bound_);
+ ASSERT_LE(audio10ms[i * num_channels + c], upper_bound_);
+ }
+ }
+}
+
+TEST(OutputTest, OpusDtxHasNoNoisePump) {
+ const int kRuntimeMs = 20000;
+ const uint32_t kUnmuteTimeMs = 1000;
+ const int kCheckAfterMute = 2000;
+ const uint32_t kCheckTimeMs = 2000;
+ const int kMinOpusRate = 6000;
+ const int kMaxOpusRate = 64000;
+
+#if defined(OPUS_FIXED_POINT)
+ const int16_t kDtxBoundForSilence = 20;
+#else
+ const int16_t kDtxBoundForSilence = 2;
+#endif
+
+ OutputTest test(-kDtxBoundForSilence, kDtxBoundForSilence);
+ Random random(1234ull);
+
+ uint32_t start_time = rtc::Time();
+ test.Start();
+ while (rtc::TimeSince(start_time) < kRuntimeMs) {
+ webrtc::SleepMs(random.Rand(kUnmuteTimeMs - kUnmuteTimeMs / 10,
+ kUnmuteTimeMs + kUnmuteTimeMs / 10));
+ test.Mute();
+ webrtc::SleepMs(kCheckAfterMute);
+ test.EnableOutputCheck();
+ webrtc::SleepMs(random.Rand(kCheckTimeMs - kCheckTimeMs / 10,
+ kCheckTimeMs + kCheckTimeMs / 10));
+ test.DisableOutputCheck();
+ test.SetBitRate(random.Rand(kMinOpusRate, kMaxOpusRate));
+ test.Unmute();
+ }
+}
+
+} // namespace voetest
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