Adding NetEq performance test to webrtc_perf_tests
The performance test is based on the neteq4_speed_test application. The
bulk of the test code is extracted into a test class, and included into
the neteq_unittest_tools target. The actual gtest that runs the
performance test is implemented in neteq_performance_unittest.cc, and
built as a part of webrtc_perf_tests.
The old stand-alone test application is now made dependent on the new
test class, to avoid code duplication.
BUG=2397
R=andrew@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5362 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/neteq.gypi b/webrtc/modules/audio_coding/neteq/neteq.gypi
index 27e5c37..a8acfb3 100644
--- a/webrtc/modules/audio_coding/neteq/neteq.gypi
+++ b/webrtc/modules/audio_coding/neteq/neteq.gypi
@@ -167,6 +167,7 @@
'PCM16B',
'neteq_unittest_tools',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ '<(webrtc_root)/test/test.gyp:test_support_main',
],
'sources': [
'test/neteq_speed_test.cc',
diff --git a/webrtc/modules/audio_coding/neteq4/neteq.gypi b/webrtc/modules/audio_coding/neteq4/neteq.gypi
index 41fdb31..4660109 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq.gypi
+++ b/webrtc/modules/audio_coding/neteq4/neteq.gypi
@@ -162,7 +162,7 @@
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
- '<(webrtc_root)/test/test.gyp:test_support_main',
+ 'PCM16B', # Needed by neteq_performance_test.
],
'direct_dependent_settings': {
'include_dirs': [
@@ -177,6 +177,8 @@
'tools/audio_loop.h',
'tools/input_audio_file.cc',
'tools/input_audio_file.h',
+ 'tools/neteq_performance_test.cc',
+ 'tools/neteq_performance_test.h',
'tools/rtp_generator.cc',
'tools/rtp_generator.h',
],
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi b/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi
index a2b9265..c74466d 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi
+++ b/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi
@@ -146,6 +146,7 @@
'neteq_unittest_tools',
'PCM16B',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ '<(webrtc_root)/test/test.gyp:test_support_main',
],
'sources': [
'test/neteq_speed_test.cc',
diff --git a/webrtc/modules/audio_coding/neteq4/test/neteq_performance_unittest.cc b/webrtc/modules/audio_coding/neteq4/test/neteq_performance_unittest.cc
new file mode 100644
index 0000000..aa33649
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq4/test/neteq_performance_unittest.cc
@@ -0,0 +1,25 @@
+/*
+ * Copyright (c) 2014 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/neteq4/tools/neteq_performance_test.h"
+#include "webrtc/test/testsupport/perf_test.h"
+#include "webrtc/typedefs.h"
+
+TEST(NetEqPerformanceTest, Run) {
+ const int kSimulationTimeMs = 1000000;
+ const int kLossPeriod = 10; // Drop every 10th packet.
+ const double kDriftFactor = 0.1;
+ int64_t runtime = webrtc::test::NetEqPerformanceTest::Run(
+ kSimulationTimeMs, kLossPeriod, kDriftFactor);
+ ASSERT_GT(runtime, 0);
+ webrtc::test::PrintResult(
+ "neteq4-runtime", "", "", runtime, "ms", true);
+}
diff --git a/webrtc/modules/audio_coding/neteq4/test/neteq_speed_test.cc b/webrtc/modules/audio_coding/neteq4/test/neteq_speed_test.cc
index 34f0de4..cecd48b 100644
--- a/webrtc/modules/audio_coding/neteq4/test/neteq_speed_test.cc
+++ b/webrtc/modules/audio_coding/neteq4/test/neteq_speed_test.cc
@@ -13,18 +13,9 @@
#include <iostream>
#include "gflags/gflags.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
-#include "webrtc/modules/audio_coding/neteq4/tools/audio_loop.h"
-#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/modules/audio_coding/neteq4/tools/neteq_performance_test.h"
#include "webrtc/typedefs.h"
-using webrtc::NetEq;
-using webrtc::test::AudioLoop;
-using webrtc::test::RtpGenerator;
-using webrtc::WebRtcRTPHeader;
-
// Flag validators.
static bool ValidateRuntime(const char* flagname, int value) {
if (value > 0) // Value is ok.
@@ -59,15 +50,6 @@
google::RegisterFlagValidator(&FLAGS_drift, &ValidateDriftfactor);
int main(int argc, char* argv[]) {
- static const int kMaxChannels = 1;
- static const int kMaxSamplesPerMs = 48000 / 1000;
- static const int kOutputBlockSizeMs = 10;
- const std::string kInputFileName =
- webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
- const int kSampRateHz = 32000;
- const webrtc::NetEqDecoder kDecoderType = webrtc::kDecoderPCM16Bswb32kHz;
- const int kPayloadType = 95;
-
std::string program_name = argv[0];
std::string usage = "Tool for measuring the speed of NetEq.\n"
"Usage: " + program_name + " [options]\n\n"
@@ -84,101 +66,15 @@
return 0;
}
- // Initialize NetEq instance.
- NetEq* neteq = NetEq::Create(kSampRateHz);
- // Register decoder in |neteq|.
- int error;
- error = neteq->RegisterPayloadType(kDecoderType, kPayloadType);
- if (error) {
- std::cerr << "Cannot register decoder." << std::endl;
- exit(1);
- }
-
- // Set up AudioLoop object.
- AudioLoop audio_loop;
- const size_t kMaxLoopLengthSamples = kSampRateHz * 10; // 10 second loop.
- const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000; // 60 ms.
- if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
- kInputBlockSizeSamples)) {
- std::cerr << "Cannot initialize AudioLoop object." << std::endl;
- exit(1);
- }
-
- int32_t time_now_ms = 0;
-
- // Get first input packet.
- WebRtcRTPHeader rtp_header;
- RtpGenerator rtp_gen(kSampRateHz / 1000);
- // Start with positive drift first half of simulation.
- double drift_factor = 0.1;
- rtp_gen.set_drift_factor(drift_factor);
- bool drift_flipped = false;
- int32_t packet_input_time_ms =
- rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
- const int16_t* input_samples = audio_loop.GetNextBlock();
- if (!input_samples) exit(1);
- uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
- int payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
- kInputBlockSizeSamples,
- input_payload);
- assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
-
- // Main loop.
- while (time_now_ms < FLAGS_runtime_ms) {
- while (packet_input_time_ms <= time_now_ms) {
- // Drop every N packets, where N = FLAGS_lossrate.
- bool lost = false;
- if (FLAGS_lossrate > 0) {
- lost = ((rtp_header.header.sequenceNumber - 1) % FLAGS_lossrate) == 0;
- }
- if (!lost) {
- // Insert packet.
- int error = neteq->InsertPacket(
- rtp_header, input_payload, payload_len,
- packet_input_time_ms * kSampRateHz / 1000);
- if (error != NetEq::kOK) {
- std::cerr << "InsertPacket returned error code " <<
- neteq->LastError() << std::endl;
- exit(1);
- }
- }
-
- // Get next packet.
- packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType,
- kInputBlockSizeSamples,
- &rtp_header);
- input_samples = audio_loop.GetNextBlock();
- if (!input_samples) exit(1);
- payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
- kInputBlockSizeSamples,
- input_payload);
- assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
- }
-
- // Get output audio, but don't do anything with it.
- static const int kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs *
- kMaxChannels;
- int16_t out_data[kOutDataLen];
- int num_channels;
- int samples_per_channel;
- int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
- &num_channels, NULL);
- if (error != NetEq::kOK) {
- std::cerr << "GetAudio returned error code " <<
- neteq->LastError() << std::endl;
- exit(1);
- }
- assert(samples_per_channel == kSampRateHz * 10 / 1000);
-
- time_now_ms += kOutputBlockSizeMs;
- if (time_now_ms >= FLAGS_runtime_ms / 2 && !drift_flipped) {
- // Apply negative drift second half of simulation.
- rtp_gen.set_drift_factor(-drift_factor);
- drift_flipped = true;
- }
+ int64_t result =
+ webrtc::test::NetEqPerformanceTest::Run(FLAGS_runtime_ms, FLAGS_lossrate,
+ FLAGS_drift);
+ if (result <= 0) {
+ std::cout << "There was an error" << std::endl;
+ return -1;
}
std::cout << "Simulation done" << std::endl;
- delete neteq;
+ std::cout << "Runtime = " << result << " ms" << std::endl;
return 0;
}
diff --git a/webrtc/modules/audio_coding/neteq4/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq4/tools/neteq_performance_test.cc
new file mode 100644
index 0000000..203ea04
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq4/tools/neteq_performance_test.cc
@@ -0,0 +1,130 @@
+/*
+ * Copyright (c) 2014 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_coding/neteq4/tools/neteq_performance_test.h"
+
+#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq4/tools/audio_loop.h"
+#include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/typedefs.h"
+
+using webrtc::NetEq;
+using webrtc::test::AudioLoop;
+using webrtc::test::RtpGenerator;
+using webrtc::WebRtcRTPHeader;
+
+namespace webrtc {
+namespace test {
+
+int64_t NetEqPerformanceTest::Run(int runtime_ms,
+ int lossrate,
+ double drift_factor) {
+ const std::string kInputFileName =
+ webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+ const int kSampRateHz = 32000;
+ const webrtc::NetEqDecoder kDecoderType = webrtc::kDecoderPCM16Bswb32kHz;
+ const int kPayloadType = 95;
+
+ // Initialize NetEq instance.
+ NetEq* neteq = NetEq::Create(kSampRateHz);
+ // Register decoder in |neteq|.
+ if (neteq->RegisterPayloadType(kDecoderType, kPayloadType) != 0)
+ return -1;
+
+ // Set up AudioLoop object.
+ AudioLoop audio_loop;
+ const size_t kMaxLoopLengthSamples = kSampRateHz * 10; // 10 second loop.
+ const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000; // 60 ms.
+ if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+ kInputBlockSizeSamples))
+ return -1;
+
+ int32_t time_now_ms = 0;
+
+ // Get first input packet.
+ WebRtcRTPHeader rtp_header;
+ RtpGenerator rtp_gen(kSampRateHz / 1000);
+ // Start with positive drift first half of simulation.
+ rtp_gen.set_drift_factor(drift_factor);
+ bool drift_flipped = false;
+ int32_t packet_input_time_ms =
+ rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
+ const int16_t* input_samples = audio_loop.GetNextBlock();
+ if (!input_samples) exit(1);
+ uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
+ int payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
+ kInputBlockSizeSamples,
+ input_payload);
+ assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
+
+ // Main loop.
+ webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
+ int64_t start_time_ms = clock->TimeInMilliseconds();
+ while (time_now_ms < runtime_ms) {
+ while (packet_input_time_ms <= time_now_ms) {
+ // Drop every N packets, where N = FLAGS_lossrate.
+ bool lost = false;
+ if (lossrate > 0) {
+ lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0;
+ }
+ if (!lost) {
+ // Insert packet.
+ int error = neteq->InsertPacket(
+ rtp_header, input_payload, payload_len,
+ packet_input_time_ms * kSampRateHz / 1000);
+ if (error != NetEq::kOK)
+ return -1;
+ }
+
+ // Get next packet.
+ packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType,
+ kInputBlockSizeSamples,
+ &rtp_header);
+ input_samples = audio_loop.GetNextBlock();
+ if (!input_samples) return -1;
+ payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
+ kInputBlockSizeSamples,
+ input_payload);
+ assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
+ }
+
+ // Get output audio, but don't do anything with it.
+ static const int kMaxChannels = 1;
+ static const int kMaxSamplesPerMs = 48000 / 1000;
+ static const int kOutputBlockSizeMs = 10;
+ static const int kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs *
+ kMaxChannels;
+ int16_t out_data[kOutDataLen];
+ int num_channels;
+ int samples_per_channel;
+ int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
+ &num_channels, NULL);
+ if (error != NetEq::kOK)
+ return -1;
+
+ assert(samples_per_channel == kSampRateHz * 10 / 1000);
+
+ time_now_ms += kOutputBlockSizeMs;
+ if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {
+ // Apply negative drift second half of simulation.
+ rtp_gen.set_drift_factor(-drift_factor);
+ drift_flipped = true;
+ }
+ }
+ int64_t end_time_ms = clock->TimeInMilliseconds();
+ delete neteq;
+ return end_time_ms - start_time_ms;
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/tools/neteq_performance_test.h b/webrtc/modules/audio_coding/neteq4/tools/neteq_performance_test.h
new file mode 100644
index 0000000..1b205c0
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq4/tools/neteq_performance_test.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2014 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqPerformanceTest {
+ public:
+ // Runs a performance test with parameters as follows:
+ // |runtime_ms|: the simulation time, i.e., the duration of the audio data.
+ // |lossrate|: drop one out of |lossrate| packets, e.g., one out of 10.
+ // |drift_factor|: clock drift in [0, 1].
+ // Returns the runtime in ms.
+ static int64_t Run(int runtime_ms, int lossrate, double drift_factor);
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_TOOLS_NETEQ_PERFORMANCE_TEST_H_
diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi
index b60660a..57fd907 100644
--- a/webrtc/webrtc_tests.gypi
+++ b/webrtc/webrtc_tests.gypi
@@ -54,6 +54,7 @@
'target_name': 'webrtc_perf_tests',
'type': '<(gtest_target_type)',
'sources': [
+ 'modules/audio_coding/neteq4/test/neteq_performance_unittest.cc',
'test/test_main.cc',
'video/call_perf_tests.cc',
'video/full_stack.cc',
@@ -62,6 +63,7 @@
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ 'modules/modules.gyp:neteq_unittest_tools', # Needed by neteq_performance_unittest.
'modules/modules.gyp:rtp_rtcp',
'test/webrtc_test_common.gyp:webrtc_test_common',
'webrtc',
