- Removes voe_conference_test.
- Adds a new AudioStatsTest, with better coverage of the same features, based on call_test.
- Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses.
BUG=webrtc:4690
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/3008273002 .
Cr-Original-Commit-Position: refs/heads/master@{#19833}
Cr-Mirrored-From: https://webrtc.googlesource.com/src
Cr-Mirrored-Commit: 73276ad7ed1e70ab764cd02d7189ed5839fadc20
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 42b74ff..890de51 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -58,6 +58,27 @@
]
}
if (rtc_include_tests) {
+ rtc_source_set("audio_end_to_end_test") {
+ testonly = true
+
+ sources = [
+ "test/audio_end_to_end_test.cc",
+ "test/audio_end_to_end_test.h",
+ ]
+ deps = [
+ ":audio",
+ "../system_wrappers:system_wrappers",
+ "../test:fake_audio_device",
+ "../test:test_common",
+ "../test:test_support",
+ ]
+
+ if (!build_with_chromium && is_clang) {
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+ }
+
rtc_source_set("audio_tests") {
testonly = true
@@ -80,6 +101,7 @@
]
deps = [
":audio",
+ ":audio_end_to_end_test",
"../api:mock_audio_mixer",
"../call:rtp_receiver",
"../modules/audio_device:mock_audio_device",
@@ -96,6 +118,11 @@
"//testing/gtest",
]
+ if (!rtc_use_memcheck) {
+ # This test is timing dependent, which rules out running on memcheck bots.
+ sources += [ "test/audio_stats_test.cc" ]
+ }
+
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@@ -108,10 +135,10 @@
sources = [
"test/low_bandwidth_audio_test.cc",
- "test/low_bandwidth_audio_test.h",
]
deps = [
+ ":audio_end_to_end_test",
"../common_audio",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
diff --git a/audio/test/audio_end_to_end_test.cc b/audio/test/audio_end_to_end_test.cc
new file mode 100644
index 0000000..5d4cbf0
--- /dev/null
+++ b/audio/test/audio_end_to_end_test.cc
@@ -0,0 +1,105 @@
+/*
+ * 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 <algorithm>
+
+#include "webrtc/audio/test/audio_end_to_end_test.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/fake_audio_device.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+// Wait half a second between stopping sending and stopping receiving audio.
+constexpr int kExtraRecordTimeMs = 500;
+
+constexpr int kSampleRate = 48000;
+} // namespace
+
+AudioEndToEndTest::AudioEndToEndTest()
+ : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
+
+FakeNetworkPipe::Config AudioEndToEndTest::GetNetworkPipeConfig() const {
+ return FakeNetworkPipe::Config();
+}
+
+size_t AudioEndToEndTest::GetNumVideoStreams() const {
+ return 0;
+}
+
+size_t AudioEndToEndTest::GetNumAudioStreams() const {
+ return 1;
+}
+
+size_t AudioEndToEndTest::GetNumFlexfecStreams() const {
+ return 0;
+}
+
+std::unique_ptr<test::FakeAudioDevice::Capturer>
+ AudioEndToEndTest::CreateCapturer() {
+ return test::FakeAudioDevice::CreatePulsedNoiseCapturer(32000, kSampleRate);
+}
+
+std::unique_ptr<test::FakeAudioDevice::Renderer>
+ AudioEndToEndTest::CreateRenderer() {
+ return test::FakeAudioDevice::CreateDiscardRenderer(kSampleRate);
+}
+
+void AudioEndToEndTest::OnFakeAudioDevicesCreated(
+ test::FakeAudioDevice* send_audio_device,
+ test::FakeAudioDevice* recv_audio_device) {
+ send_audio_device_ = send_audio_device;
+}
+
+test::PacketTransport* AudioEndToEndTest::CreateSendTransport(
+ SingleThreadedTaskQueueForTesting* task_queue,
+ Call* sender_call) {
+ return new test::PacketTransport(
+ task_queue, sender_call, this, test::PacketTransport::kSender,
+ test::CallTest::payload_type_map_, GetNetworkPipeConfig());
+}
+
+test::PacketTransport* AudioEndToEndTest::CreateReceiveTransport(
+ SingleThreadedTaskQueueForTesting* task_queue) {
+ return new test::PacketTransport(
+ task_queue, nullptr, this, test::PacketTransport::kReceiver,
+ test::CallTest::payload_type_map_, GetNetworkPipeConfig());
+}
+
+void AudioEndToEndTest::ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) {
+ // Large bitrate by default.
+ const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2,
+ {{"stereo", "1"}});
+ send_config->send_codec_spec =
+ rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+ {test::CallTest::kAudioSendPayloadType, kDefaultFormat});
+}
+
+void AudioEndToEndTest::OnAudioStreamsCreated(
+ AudioSendStream* send_stream,
+ const std::vector<AudioReceiveStream*>& receive_streams) {
+ ASSERT_NE(nullptr, send_stream);
+ ASSERT_EQ(1u, receive_streams.size());
+ ASSERT_NE(nullptr, receive_streams[0]);
+ send_stream_ = send_stream;
+ receive_stream_ = receive_streams[0];
+}
+
+void AudioEndToEndTest::PerformTest() {
+ // Wait until the input audio file is done...
+ send_audio_device_->WaitForRecordingEnd();
+ // and some extra time to account for network delay.
+ SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
+}
+} // namespace test
+} // namespace webrtc
diff --git a/audio/test/low_bandwidth_audio_test.h b/audio/test/audio_end_to_end_test.h
similarity index 64%
rename from audio/test/low_bandwidth_audio_test.h
rename to audio/test/audio_end_to_end_test.h
index ae75707..d14b7a1 100644
--- a/audio/test/low_bandwidth_audio_test.h
+++ b/audio/test/audio_end_to_end_test.h
@@ -7,28 +7,28 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
-#define WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
+#ifndef WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
+#define WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
#include <memory>
#include <string>
#include <vector>
#include "webrtc/test/call_test.h"
-#include "webrtc/test/fake_audio_device.h"
namespace webrtc {
namespace test {
-class AudioQualityTest : public test::EndToEndTest {
+class AudioEndToEndTest : public test::EndToEndTest {
public:
- AudioQualityTest();
+ AudioEndToEndTest();
protected:
- virtual std::string AudioInputFile();
- virtual std::string AudioOutputFile();
+ test::FakeAudioDevice* send_audio_device() { return send_audio_device_; }
+ const AudioSendStream* send_stream() const { return send_stream_; }
+ const AudioReceiveStream* receive_stream() const { return receive_stream_; }
- virtual FakeNetworkPipe::Config GetNetworkPipeConfig();
+ virtual FakeNetworkPipe::Config GetNetworkPipeConfig() const;
size_t GetNumVideoStreams() const override;
size_t GetNumAudioStreams() const override;
@@ -50,15 +50,19 @@
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override;
+ void OnAudioStreamsCreated(
+ AudioSendStream* send_stream,
+ const std::vector<AudioReceiveStream*>& receive_streams) override;
void PerformTest() override;
- void OnTestFinished() override;
private:
- test::FakeAudioDevice* send_audio_device_;
+ test::FakeAudioDevice* send_audio_device_ = nullptr;
+ AudioSendStream* send_stream_ = nullptr;
+ AudioReceiveStream* receive_stream_ = nullptr;
};
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
+#endif // WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
diff --git a/audio/test/audio_stats_test.cc b/audio/test/audio_stats_test.cc
new file mode 100644
index 0000000..57dfbed
--- /dev/null
+++ b/audio/test/audio_stats_test.cc
@@ -0,0 +1,118 @@
+/*
+ * 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/audio/test/audio_end_to_end_test.h"
+#include "webrtc/rtc_base/safe_compare.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+bool IsNear(int reference, int v) {
+ // Margin is 10%.
+ const int error = reference / 10 + 1;
+ return std::abs(reference - v) <= error;
+}
+
+class NoLossTest : public AudioEndToEndTest {
+ public:
+ const int kTestDurationMs = 8000;
+ const int kBytesSent = 69351;
+ const int32_t kPacketsSent = 400;
+ const int64_t kRttMs = 100;
+
+ NoLossTest() = default;
+
+ FakeNetworkPipe::Config GetNetworkPipeConfig() const override {
+ FakeNetworkPipe::Config pipe_config;
+ pipe_config.queue_delay_ms = kRttMs / 2;
+ return pipe_config;
+ }
+
+ void PerformTest() override {
+ SleepMs(kTestDurationMs);
+ send_audio_device()->StopRecording();
+ AudioEndToEndTest::PerformTest();
+ }
+
+ void OnStreamsStopped() override {
+ AudioSendStream::Stats send_stats = send_stream()->GetStats();
+ EXPECT_PRED2(IsNear, kBytesSent, send_stats.bytes_sent);
+ EXPECT_PRED2(IsNear, kPacketsSent, send_stats.packets_sent);
+ EXPECT_EQ(0, send_stats.packets_lost);
+ EXPECT_EQ(0.0f, send_stats.fraction_lost);
+ EXPECT_EQ("opus", send_stats.codec_name);
+ // send_stats.jitter_ms
+ EXPECT_PRED2(IsNear, kRttMs, send_stats.rtt_ms);
+ // Send level is 0 because it is cleared in TransmitMixer::StopSend().
+ EXPECT_EQ(0, send_stats.audio_level);
+ // send_stats.total_input_energy
+ // send_stats.total_input_duration
+ EXPECT_EQ(-1.0f, send_stats.aec_quality_min);
+ EXPECT_EQ(-1, send_stats.echo_delay_median_ms);
+ EXPECT_EQ(-1, send_stats.echo_delay_std_ms);
+ EXPECT_EQ(-100, send_stats.echo_return_loss);
+ EXPECT_EQ(-100, send_stats.echo_return_loss_enhancement);
+ EXPECT_EQ(0.0f, send_stats.residual_echo_likelihood);
+ EXPECT_EQ(0.0f, send_stats.residual_echo_likelihood_recent_max);
+ EXPECT_EQ(false, send_stats.typing_noise_detected);
+
+ AudioReceiveStream::Stats recv_stats = receive_stream()->GetStats();
+ EXPECT_PRED2(IsNear, kBytesSent, recv_stats.bytes_rcvd);
+ EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd);
+ EXPECT_EQ(0u, recv_stats.packets_lost);
+ EXPECT_EQ(0.0f, recv_stats.fraction_lost);
+ EXPECT_EQ("opus", send_stats.codec_name);
+ // recv_stats.jitter_ms
+ // recv_stats.jitter_buffer_ms
+ EXPECT_EQ(20u, recv_stats.jitter_buffer_preferred_ms);
+ // recv_stats.delay_estimate_ms
+ // Receive level is 0 because it is cleared in Channel::StopPlayout().
+ EXPECT_EQ(0, recv_stats.audio_level);
+ // recv_stats.total_output_energy
+ // recv_stats.total_samples_received
+ // recv_stats.total_output_duration
+ // recv_stats.concealed_samples
+ // recv_stats.expand_rate
+ // recv_stats.speech_expand_rate
+ EXPECT_EQ(0.0, recv_stats.secondary_decoded_rate);
+ EXPECT_EQ(0.0, recv_stats.secondary_discarded_rate);
+ EXPECT_EQ(0.0, recv_stats.accelerate_rate);
+ EXPECT_EQ(0.0, recv_stats.preemptive_expand_rate);
+ EXPECT_EQ(0, recv_stats.decoding_calls_to_silence_generator);
+ // recv_stats.decoding_calls_to_neteq
+ // recv_stats.decoding_normal
+ // recv_stats.decoding_plc
+ EXPECT_EQ(0, recv_stats.decoding_cng);
+ // recv_stats.decoding_plc_cng
+ // recv_stats.decoding_muted_output
+ // Capture start time is -1 because we do not have an associated send stream
+ // on the receiver side.
+ EXPECT_EQ(-1, recv_stats.capture_start_ntp_time_ms);
+
+ // Match these stats between caller and receiver.
+ EXPECT_EQ(send_stats.local_ssrc, recv_stats.remote_ssrc);
+ EXPECT_EQ(*send_stats.codec_payload_type, *recv_stats.codec_payload_type);
+ EXPECT_TRUE(rtc::SafeEq(send_stats.ext_seqnum, recv_stats.ext_seqnum));
+ }
+};
+} // namespace
+
+using AudioStatsTest = CallTest;
+
+TEST_F(AudioStatsTest, NoLoss) {
+ NoLossTest test;
+ RunBaseTest(&test);
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/audio/test/low_bandwidth_audio_test.cc b/audio/test/low_bandwidth_audio_test.cc
index ea0cdf0..8bbadfb 100644
--- a/audio/test/low_bandwidth_audio_test.cc
+++ b/audio/test/low_bandwidth_audio_test.cc
@@ -8,16 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <algorithm>
-
-#include "webrtc/audio/test/low_bandwidth_audio_test.h"
-#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/audio/test/audio_end_to_end_test.h"
#include "webrtc/rtc_base/flags.h"
#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
-
DEFINE_int(sample_rate_hz, 16000,
"Sample rate (Hz) of the produced audio files.");
@@ -25,122 +20,59 @@
"Don't do the full audio recording. "
"Used to quickly check that the test runs without crashing.");
+namespace webrtc {
+namespace test {
namespace {
-// Wait half a second between stopping sending and stopping receiving audio.
-constexpr int kExtraRecordTimeMs = 500;
-
std::string FileSampleRateSuffix() {
return std::to_string(FLAG_sample_rate_hz / 1000);
}
-} // namespace
+class AudioQualityTest : public AudioEndToEndTest {
+ public:
+ AudioQualityTest() = default;
-namespace webrtc {
-namespace test {
-
-AudioQualityTest::AudioQualityTest()
- : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
-
-size_t AudioQualityTest::GetNumVideoStreams() const {
- return 0;
-}
-size_t AudioQualityTest::GetNumAudioStreams() const {
- return 1;
-}
-size_t AudioQualityTest::GetNumFlexfecStreams() const {
- return 0;
-}
-
-std::string AudioQualityTest::AudioInputFile() {
- return test::ResourcePath("voice_engine/audio_tiny" + FileSampleRateSuffix(),
- "wav");
-}
-
-std::string AudioQualityTest::AudioOutputFile() {
- const ::testing::TestInfo* const test_info =
- ::testing::UnitTest::GetInstance()->current_test_info();
- return webrtc::test::OutputPath() + "LowBandwidth_" + test_info->name() +
- "_" + FileSampleRateSuffix() + ".wav";
-}
-
-std::unique_ptr<test::FakeAudioDevice::Capturer>
- AudioQualityTest::CreateCapturer() {
- return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile());
-}
-
-std::unique_ptr<test::FakeAudioDevice::Renderer>
- AudioQualityTest::CreateRenderer() {
- return test::FakeAudioDevice::CreateBoundedWavFileWriter(
- AudioOutputFile(), FLAG_sample_rate_hz);
-}
-
-void AudioQualityTest::OnFakeAudioDevicesCreated(
- test::FakeAudioDevice* send_audio_device,
- test::FakeAudioDevice* recv_audio_device) {
- send_audio_device_ = send_audio_device;
-}
-
-FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() {
- return FakeNetworkPipe::Config();
-}
-
-test::PacketTransport* AudioQualityTest::CreateSendTransport(
- SingleThreadedTaskQueueForTesting* task_queue,
- Call* sender_call) {
- return new test::PacketTransport(
- task_queue, sender_call, this, test::PacketTransport::kSender,
- test::CallTest::payload_type_map_, GetNetworkPipeConfig());
-}
-
-test::PacketTransport* AudioQualityTest::CreateReceiveTransport(
- SingleThreadedTaskQueueForTesting* task_queue) {
- return new test::PacketTransport(
- task_queue, nullptr, this, test::PacketTransport::kReceiver,
- test::CallTest::payload_type_map_, GetNetworkPipeConfig());
-}
-
-void AudioQualityTest::ModifyAudioConfigs(
- AudioSendStream::Config* send_config,
- std::vector<AudioReceiveStream::Config>* receive_configs) {
- // Large bitrate by default.
- const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2,
- {{"stereo", "1"}});
- send_config->send_codec_spec =
- rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
- {test::CallTest::kAudioSendPayloadType, kDefaultFormat});
-}
-
-void AudioQualityTest::PerformTest() {
- if (FLAG_quick) {
- // Let the recording run for a small amount of time to check if it works.
- SleepMs(1000);
- } else {
- // Wait until the input audio file is done...
- send_audio_device_->WaitForRecordingEnd();
- // and some extra time to account for network delay.
- SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
+ private:
+ std::string AudioInputFile() const {
+ return test::ResourcePath(
+ "voice_engine/audio_tiny" + FileSampleRateSuffix(), "wav");
}
-}
-void AudioQualityTest::OnTestFinished() {
- const ::testing::TestInfo* const test_info =
- ::testing::UnitTest::GetInstance()->current_test_info();
+ std::string AudioOutputFile() const {
+ const ::testing::TestInfo* const test_info =
+ ::testing::UnitTest::GetInstance()->current_test_info();
+ return webrtc::test::OutputPath() + "LowBandwidth_" + test_info->name() +
+ "_" + FileSampleRateSuffix() + ".wav";
+ }
- // Output information about the input and output audio files so that further
- // processing can be done by an external process.
- printf("TEST %s %s %s\n", test_info->name(),
- AudioInputFile().c_str(), AudioOutputFile().c_str());
-}
+ std::unique_ptr<test::FakeAudioDevice::Capturer> CreateCapturer() override {
+ return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile());
+ }
+ std::unique_ptr<test::FakeAudioDevice::Renderer> CreateRenderer() override {
+ return test::FakeAudioDevice::CreateBoundedWavFileWriter(
+ AudioOutputFile(), FLAG_sample_rate_hz);
+ }
-using LowBandwidthAudioTest = CallTest;
+ void PerformTest() override {
+ if (FLAG_quick) {
+ // Let the recording run for a small amount of time to check if it works.
+ SleepMs(1000);
+ } else {
+ AudioEndToEndTest::PerformTest();
+ }
+ }
-TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) {
- AudioQualityTest test;
- RunBaseTest(&test);
-}
+ void OnStreamsStopped() override {
+ const ::testing::TestInfo* const test_info =
+ ::testing::UnitTest::GetInstance()->current_test_info();
+ // Output information about the input and output audio files so that further
+ // processing can be done by an external process.
+ printf("TEST %s %s %s\n", test_info->name(),
+ AudioInputFile().c_str(), AudioOutputFile().c_str());
+ }
+};
class Mobile2GNetworkTest : public AudioQualityTest {
void ModifyAudioConfigs(AudioSendStream::Config* send_config,
@@ -156,7 +88,7 @@
{"stereo", "1"}}}});
}
- FakeNetworkPipe::Config GetNetworkPipeConfig() override {
+ FakeNetworkPipe::Config GetNetworkPipeConfig() const override {
FakeNetworkPipe::Config pipe_config;
pipe_config.link_capacity_kbps = 12;
pipe_config.queue_length_packets = 1500;
@@ -164,11 +96,18 @@
return pipe_config;
}
};
+} // namespace
+
+using LowBandwidthAudioTest = CallTest;
+
+TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) {
+ AudioQualityTest test;
+ RunBaseTest(&test);
+}
TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
Mobile2GNetworkTest test;
RunBaseTest(&test);
}
-
} // namespace test
} // namespace webrtc
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 641be6f..6760b14 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -1914,6 +1914,33 @@
EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0);
}
+// Test that we can get capture start ntp time.
+TEST_F(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
+ ASSERT_TRUE(CreatePeerConnectionWrappers());
+ ConnectFakeSignaling();
+ caller()->AddAudioOnlyMediaStream();
+
+ auto audio_track = callee()->CreateLocalAudioTrack();
+ callee()->AddMediaStreamFromTracks(audio_track, nullptr);
+
+ // Do offer/answer, wait for the callee to receive some frames.
+ caller()->CreateAndSetAndSignalOffer();
+ ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+ // Get the remote audio track created on the receiver, so they can be used as
+ // GetStats filters.
+ StreamCollectionInterface* remote_streams = callee()->remote_streams();
+ ASSERT_EQ(1u, remote_streams->count());
+ ASSERT_EQ(1u, remote_streams->at(0)->GetAudioTracks().size());
+ MediaStreamTrackInterface* remote_audio_track =
+ remote_streams->at(0)->GetAudioTracks()[0];
+
+ // Get the audio output level stats. Note that the level is not available
+ // until an RTCP packet has been received.
+ EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track)->
+ CaptureStartNtpTime() > 0, 2 * kMaxWaitForFramesMs);
+}
+
// Test that we can get stats (using the new stats implemnetation) for
// unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in
// SDP.
diff --git a/pc/test/mockpeerconnectionobservers.h b/pc/test/mockpeerconnectionobservers.h
index 5367eeb..84c80f8 100644
--- a/pc/test/mockpeerconnectionobservers.h
+++ b/pc/test/mockpeerconnectionobservers.h
@@ -125,6 +125,8 @@
&stats_.bytes_received);
GetIntValue(r, StatsReport::kStatsValueNameBytesSent,
&stats_.bytes_sent);
+ GetInt64Value(r, StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
+ &stats_.capture_start_ntp_time);
} else if (r->type() == StatsReport::kStatsReportTypeBwe) {
stats_.timestamp = r->timestamp();
GetIntValue(r, StatsReport::kStatsValueNameAvailableReceiveBandwidth,
@@ -163,6 +165,11 @@
return stats_.bytes_sent;
}
+ int64_t CaptureStartNtpTime() const {
+ RTC_CHECK(called_);
+ return stats_.capture_start_ntp_time;
+ }
+
int AvailableReceiveBandwidth() const {
RTC_CHECK(called_);
return stats_.available_receive_bandwidth;
@@ -190,6 +197,17 @@
return v != nullptr;
}
+ bool GetInt64Value(const StatsReport* report,
+ StatsReport::StatsValueName name,
+ int64_t* value) {
+ const StatsReport::Value* v = report->FindValue(name);
+ if (v) {
+ // TODO(tommi): We should really just be using an int here :-/
+ *value = rtc::FromString<int64_t>(v->ToString());
+ }
+ return v != nullptr;
+ }
+
bool GetStringValue(const StatsReport* report,
StatsReport::StatsValueName name,
std::string* value) {
@@ -208,6 +226,7 @@
audio_input_level = 0;
bytes_received = 0;
bytes_sent = 0;
+ capture_start_ntp_time = 0;
available_receive_bandwidth = 0;
dtls_cipher.clear();
srtp_cipher.clear();
@@ -219,6 +238,7 @@
int audio_input_level;
int bytes_received;
int bytes_sent;
+ int64_t capture_start_ntp_time;
int available_receive_bandwidth;
std::string dtls_cipher;
std::string srtp_cipher;
diff --git a/test/call_test.cc b/test/call_test.cc
index b5d7236..d4084d5 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -153,8 +153,9 @@
test->PerformTest();
- task_queue_.SendTask([this]() {
+ task_queue_.SendTask([this, test]() {
Stop();
+ test->OnStreamsStopped();
DestroyStreams();
send_transport_.reset();
receive_transport_.reset();
@@ -162,8 +163,6 @@
if (num_audio_streams_ > 0)
DestroyVoiceEngines();
});
-
- test->OnTestFinished();
}
void CallTest::CreateCalls(const Call::Config& sender_config,
@@ -223,7 +222,7 @@
audio_send_config_.rtp.ssrc = kAudioSendSsrc;
audio_send_config_.send_codec_spec =
rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
- {kAudioSendPayloadType, {"OPUS", 48000, 2, {{"stereo", "1"}}}});
+ {kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}}});
audio_send_config_.encoder_factory = encoder_factory_;
}
@@ -590,7 +589,7 @@
FrameGeneratorCapturer* frame_generator_capturer) {
}
-void BaseTest::OnTestFinished() {
+void BaseTest::OnStreamsStopped() {
}
SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
diff --git a/test/call_test.h b/test/call_test.h
index b0ae3f6..3372015 100644
--- a/test/call_test.h
+++ b/test/call_test.h
@@ -223,7 +223,7 @@
virtual void OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer);
- virtual void OnTestFinished();
+ virtual void OnStreamsStopped();
std::unique_ptr<webrtc::RtcEventLog> event_log_;
};
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index 809333b..627f412 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -266,10 +266,6 @@
sources = [
"test/auto_test/automated_mode.cc",
- "test/auto_test/fakes/conference_transport.cc",
- "test/auto_test/fakes/conference_transport.h",
- "test/auto_test/fakes/loudest_filter.cc",
- "test/auto_test/fakes/loudest_filter.h",
"test/auto_test/fixtures/after_initialization_fixture.cc",
"test/auto_test/fixtures/after_initialization_fixture.h",
"test/auto_test/fixtures/after_streaming_fixture.cc",
@@ -284,7 +280,6 @@
"test/auto_test/standard/rtp_rtcp_before_streaming_test.cc",
"test/auto_test/standard/rtp_rtcp_extensions.cc",
"test/auto_test/standard/rtp_rtcp_test.cc",
- "test/auto_test/voe_conference_test.cc",
"test/auto_test/voe_standard_test.cc",
"test/auto_test/voe_standard_test.h",
"test/auto_test/voe_test_defines.h",
diff --git a/voice_engine/test/auto_test/fakes/conference_transport.cc b/voice_engine/test/auto_test/fakes/conference_transport.cc
deleted file mode 100644
index 305b236..0000000
--- a/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ /dev/null
@@ -1,307 +0,0 @@
-/*
- * 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 "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
-
-#include <string>
-
-#include "webrtc/rtc_base/byteorder.h"
-#include "webrtc/rtc_base/timeutils.h"
-#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/voice_engine/channel_proxy.h"
-#include "webrtc/voice_engine/voice_engine_impl.h"
-
-namespace webrtc {
-namespace voetest {
-
-namespace {
-
-static const unsigned int kReflectorSsrc = 0x0000;
-static const unsigned int kLocalSsrc = 0x0001;
-static const unsigned int kFirstRemoteSsrc = 0x0002;
-static const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
-static const int kAudioLevelHeaderId = 1;
-
-static unsigned int ParseRtcpSsrc(const void* data, size_t len) {
- const size_t ssrc_pos = 4;
- unsigned int ssrc = 0;
- if (len >= (ssrc_pos + sizeof(ssrc))) {
- ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
- }
- return ssrc;
-}
-
-} // namespace
-
-ConferenceTransport::ConferenceTransport()
- : packet_event_(webrtc::EventWrapper::Create()),
- thread_(Run, this, "ConferenceTransport"),
- rtt_ms_(0),
- stream_count_(0),
- rtp_header_parser_(webrtc::RtpHeaderParser::Create()) {
- rtp_header_parser_->
- RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
- kAudioLevelHeaderId);
-
- local_voe_ = webrtc::VoiceEngine::Create();
- local_base_ = webrtc::VoEBase::GetInterface(local_voe_);
- local_network_ = webrtc::VoENetwork::GetInterface(local_voe_);
- local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_);
-
- local_apm_ = webrtc::AudioProcessing::Create();
- local_base_->Init(nullptr, local_apm_.get(), nullptr);
-
- // In principle, we can use one VoiceEngine to achieve the same goal. Well, in
- // here, we use two engines to make it more like reality.
- remote_voe_ = webrtc::VoiceEngine::Create();
- remote_base_ = webrtc::VoEBase::GetInterface(remote_voe_);
- remote_codec_ = webrtc::VoECodec::GetInterface(remote_voe_);
- remote_network_ = webrtc::VoENetwork::GetInterface(remote_voe_);
- remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_);
- remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_);
-
- remote_apm_ = webrtc::AudioProcessing::Create();
- remote_base_->Init(nullptr, remote_apm_.get(), nullptr);
-
- local_sender_ = local_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
- ->GetChannelProxy(local_sender_)
- ->RegisterLegacyReceiveCodecs();
- EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
- EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
- EXPECT_EQ(0, local_rtp_rtcp_->
- SetSendAudioLevelIndicationStatus(local_sender_, true,
- kAudioLevelHeaderId));
- EXPECT_EQ(0, local_base_->StartSend(local_sender_));
-
- reflector_ = remote_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
- ->GetChannelProxy(reflector_)
- ->RegisterLegacyReceiveCodecs();
- EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
- EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));
-
- thread_.Start();
- thread_.SetPriority(rtc::kHighPriority);
-}
-
-ConferenceTransport::~ConferenceTransport() {
- // Must stop sending, otherwise DispatchPackets() cannot quit.
- EXPECT_EQ(0, remote_network_->DeRegisterExternalTransport(reflector_));
- EXPECT_EQ(0, local_network_->DeRegisterExternalTransport(local_sender_));
-
- while (!streams_.empty()) {
- auto stream = streams_.begin();
- RemoveStream(stream->first);
- }
-
- thread_.Stop();
-
- remote_file_->Release();
- remote_rtp_rtcp_->Release();
- remote_network_->Release();
- remote_base_->Release();
-
- local_rtp_rtcp_->Release();
- local_network_->Release();
- local_base_->Release();
-
- EXPECT_TRUE(webrtc::VoiceEngine::Delete(remote_voe_));
- EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_));
-}
-
-bool ConferenceTransport::SendRtp(const uint8_t* data,
- size_t len,
- const webrtc::PacketOptions& options) {
- StorePacket(Packet::Rtp, data, len);
- return true;
-}
-
-bool ConferenceTransport::SendRtcp(const uint8_t* data, size_t len) {
- StorePacket(Packet::Rtcp, data, len);
- return true;
-}
-
-int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc)
- const {
- rtc::CritScope lock(&stream_crit_);
- auto it = streams_.find(sender_ssrc);
- if (it != streams_.end()) {
- return it->second.second;
- }
- return -1;
-}
-
-void ConferenceTransport::StorePacket(Packet::Type type,
- const void* data,
- size_t len) {
- {
- rtc::CritScope lock(&pq_crit_);
- packet_queue_.push_back(Packet(type, data, len, rtc::TimeMillis()));
- }
- packet_event_->Set();
-}
-
-// This simulates the flow of RTP and RTCP packets. Complications like that
-// a packet is first sent to the reflector, and then forwarded to the receiver
-// are simplified, in this particular case, to a direct link between the sender
-// and the receiver.
-void ConferenceTransport::SendPacket(const Packet& packet) {
- int destination = -1;
-
- switch (packet.type_) {
- case Packet::Rtp: {
- webrtc::RTPHeader rtp_header;
- rtp_header_parser_->Parse(packet.data_, packet.len_, &rtp_header);
- if (rtp_header.ssrc == kLocalSsrc) {
- remote_network_->ReceivedRTPPacket(reflector_, packet.data_,
- packet.len_, webrtc::PacketTime());
- } else {
- if (loudest_filter_.ForwardThisPacket(rtp_header)) {
- destination = GetReceiverChannelForSsrc(rtp_header.ssrc);
- if (destination != -1) {
- local_network_->ReceivedRTPPacket(destination, packet.data_,
- packet.len_,
- webrtc::PacketTime());
- }
- }
- }
- break;
- }
- case Packet::Rtcp: {
- unsigned int sender_ssrc = ParseRtcpSsrc(packet.data_, packet.len_);
- if (sender_ssrc == kLocalSsrc) {
- remote_network_->ReceivedRTCPPacket(reflector_, packet.data_,
- packet.len_);
- } else if (sender_ssrc == kReflectorSsrc) {
- local_network_->ReceivedRTCPPacket(local_sender_, packet.data_,
- packet.len_);
- } else {
- destination = GetReceiverChannelForSsrc(sender_ssrc);
- if (destination != -1) {
- local_network_->ReceivedRTCPPacket(destination, packet.data_,
- packet.len_);
- }
- }
- break;
- }
- }
-}
-
-bool ConferenceTransport::DispatchPackets() {
- switch (packet_event_->Wait(1000)) {
- case webrtc::kEventSignaled:
- break;
- case webrtc::kEventTimeout:
- return true;
- case webrtc::kEventError:
- ADD_FAILURE() << "kEventError encountered.";
- return true;
- }
-
- while (true) {
- Packet packet;
- {
- rtc::CritScope lock(&pq_crit_);
- if (packet_queue_.empty())
- break;
- packet = packet_queue_.front();
- packet_queue_.pop_front();
- }
-
- int32_t elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_);
- int32_t sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;
- if (sleep_ms > 0) {
- // Every packet should be delayed by half of RTT.
- webrtc::SleepMs(sleep_ms);
- }
-
- SendPacket(packet);
- }
- return true;
-}
-
-void ConferenceTransport::SetRtt(unsigned int rtt_ms) {
- rtt_ms_ = rtt_ms;
-}
-
-unsigned int ConferenceTransport::AddStream(std::string file_name,
- webrtc::FileFormats format) {
- const int new_sender = remote_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
- ->GetChannelProxy(new_sender)
- ->RegisterLegacyReceiveCodecs();
- EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));
-
- const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
- EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc));
- EXPECT_EQ(0, remote_rtp_rtcp_->
- SetSendAudioLevelIndicationStatus(new_sender, true, kAudioLevelHeaderId));
-
- EXPECT_EQ(0, remote_codec_->SetSendCodec(new_sender, kCodecInst));
- EXPECT_EQ(0, remote_base_->StartSend(new_sender));
- EXPECT_EQ(0, remote_file_->StartPlayingFileAsMicrophone(
- new_sender, file_name.c_str(), true, false, format, 1.0));
-
- const int new_receiver = local_base_->CreateChannel();
- static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
- ->GetChannelProxy(new_receiver)
- ->RegisterLegacyReceiveCodecs();
- EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));
-
- EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));
- // Receive channels have to have the same SSRC in order to send receiver
- // reports with this SSRC.
- EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc));
-
- {
- rtc::CritScope lock(&stream_crit_);
- streams_[remote_ssrc] = std::make_pair(new_sender, new_receiver);
- }
- return remote_ssrc; // remote ssrc used as stream id.
-}
-
-bool ConferenceTransport::RemoveStream(unsigned int id) {
- rtc::CritScope lock(&stream_crit_);
- auto it = streams_.find(id);
- if (it == streams_.end()) {
- return false;
- }
- EXPECT_EQ(0, remote_network_->
- DeRegisterExternalTransport(it->second.second));
- EXPECT_EQ(0, local_network_->
- DeRegisterExternalTransport(it->second.first));
- EXPECT_EQ(0, remote_base_->DeleteChannel(it->second.second));
- EXPECT_EQ(0, local_base_->DeleteChannel(it->second.first));
- streams_.erase(it);
- return true;
-}
-
-bool ConferenceTransport::StartPlayout(unsigned int id) {
- int dst = GetReceiverChannelForSsrc(id);
- if (dst == -1) {
- return false;
- }
- EXPECT_EQ(0, local_base_->StartPlayout(dst));
- return true;
-}
-
-bool ConferenceTransport::GetReceiverStatistics(unsigned int id,
- webrtc::CallStatistics* stats) {
- int dst = GetReceiverChannelForSsrc(id);
- if (dst == -1) {
- return false;
- }
- EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
- return true;
-}
-
-} // namespace voetest
-} // namespace webrtc
diff --git a/voice_engine/test/auto_test/fakes/conference_transport.h b/voice_engine/test/auto_test/fakes/conference_transport.h
deleted file mode 100644
index a0acd9e..0000000
--- a/voice_engine/test/auto_test/fakes/conference_transport.h
+++ /dev/null
@@ -1,168 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
-#define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
-
-#include <deque>
-#include <map>
-#include <memory>
-#include <utility>
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/platform_thread.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_file.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-#include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
-
-namespace webrtc {
-namespace voetest {
-
-static const size_t kMaxPacketSizeByte = 1500;
-
-// This class is to simulate a conference call. There are two Voice Engines, one
-// for local channels and the other for remote channels. There is a simulated
-// reflector, which exchanges RTCP with local channels. For simplicity, it
-// also uses the Voice Engine for remote channels. One can add streams by
-// calling AddStream(), which creates a remote sender channel and a local
-// receive channel. The remote sender channel plays a file as microphone in a
-// looped fashion. Received streams are mixed and played.
-
-class ConferenceTransport: public webrtc::Transport {
- public:
- ConferenceTransport();
- virtual ~ConferenceTransport();
-
- /* SetRtt()
- * Set RTT between local channels and reflector.
- *
- * Input:
- * rtt_ms : RTT in milliseconds.
- */
- void SetRtt(unsigned int rtt_ms);
-
- /* AddStream()
- * Adds a stream in the conference.
- *
- * Input:
- * file_name : name of the file to be added as microphone input.
- * format : format of the input file.
- *
- * Returns stream id.
- */
- unsigned int AddStream(std::string file_name, webrtc::FileFormats format);
-
- /* RemoveStream()
- * Removes a stream with specified ID from the conference.
- *
- * Input:
- * id : stream id.
- *
- * Returns false if the specified stream does not exist, true if succeeds.
- */
- bool RemoveStream(unsigned int id);
-
- /* StartPlayout()
- * Starts playing out the stream with specified ID, using the default device.
- *
- * Input:
- * id : stream id.
- *
- * Returns false if the specified stream does not exist, true if succeeds.
- */
- bool StartPlayout(unsigned int id);
-
- /* GetReceiverStatistics()
- * Gets RTCP statistics of the stream with specified ID.
- *
- * Input:
- * id : stream id;
- * stats : pointer to a CallStatistics to store the result.
- *
- * Returns false if the specified stream does not exist, true if succeeds.
- */
- bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats);
-
- // Inherit from class webrtc::Transport.
- bool SendRtp(const uint8_t* data,
- size_t len,
- const webrtc::PacketOptions& options) override;
- bool SendRtcp(const uint8_t *data, size_t len) override;
-
- private:
- struct Packet {
- enum Type { Rtp, Rtcp, } type_;
-
- Packet() : len_(0) {}
- Packet(Type type, const void* data, size_t len, int64_t time_ms)
- : type_(type), len_(len), send_time_ms_(time_ms) {
- EXPECT_LE(len_, kMaxPacketSizeByte);
- memcpy(data_, data, len_);
- }
-
- uint8_t data_[kMaxPacketSizeByte];
- size_t len_;
- int64_t send_time_ms_;
- };
-
- static bool Run(void* transport) {
- return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
- }
-
- int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
- void StorePacket(Packet::Type type, const void* data, size_t len);
- void SendPacket(const Packet& packet);
- bool DispatchPackets();
-
- rtc::CriticalSection pq_crit_;
- rtc::CriticalSection stream_crit_;
- const std::unique_ptr<webrtc::EventWrapper> packet_event_;
- rtc::PlatformThread thread_;
-
- unsigned int rtt_ms_;
- unsigned int stream_count_;
-
- std::map<unsigned int, std::pair<int, int>> streams_
- RTC_GUARDED_BY(stream_crit_);
- std::deque<Packet> packet_queue_ RTC_GUARDED_BY(pq_crit_);
-
- int local_sender_; // Channel Id of local sender
- int reflector_;
-
- webrtc::VoiceEngine* local_voe_;
- webrtc::VoEBase* local_base_;
- webrtc::VoERTP_RTCP* local_rtp_rtcp_;
- webrtc::VoENetwork* local_network_;
- rtc::scoped_refptr<webrtc::AudioProcessing> local_apm_;
-
- webrtc::VoiceEngine* remote_voe_;
- webrtc::VoEBase* remote_base_;
- webrtc::VoECodec* remote_codec_;
- webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
- webrtc::VoENetwork* remote_network_;
- webrtc::VoEFile* remote_file_;
- rtc::scoped_refptr<webrtc::AudioProcessing> remote_apm_;
- LoudestFilter loudest_filter_;
-
- const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
-};
-
-} // namespace voetest
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
diff --git a/voice_engine/test/auto_test/fakes/loudest_filter.cc b/voice_engine/test/auto_test/fakes/loudest_filter.cc
deleted file mode 100644
index ec1d667..0000000
--- a/voice_engine/test/auto_test/fakes/loudest_filter.cc
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * 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 "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
-
-#include "webrtc/rtc_base/checks.h"
-
-namespace webrtc {
-namespace voetest {
-
-void LoudestFilter::RemoveTimeoutStreams(int64_t time_ms) {
- auto it = stream_levels_.begin();
- while (it != stream_levels_.end()) {
- if (rtc::TimeDiff(time_ms, it->second.last_time_ms) > kStreamTimeOutMs) {
- stream_levels_.erase(it++);
- } else {
- ++it;
- }
- }
-}
-
-unsigned int LoudestFilter::FindQuietestStream() {
- int quietest_level = kInvalidAudioLevel;
- unsigned int quietest_ssrc = 0;
- for (auto stream : stream_levels_) {
- // A smaller value if audio level corresponds to a louder sound.
- if (quietest_level == kInvalidAudioLevel ||
- stream.second.audio_level > quietest_level) {
- quietest_level = stream.second.audio_level;
- quietest_ssrc = stream.first;
- }
- }
- return quietest_ssrc;
-}
-
-bool LoudestFilter::ForwardThisPacket(const webrtc::RTPHeader& rtp_header) {
- int64_t time_now_ms = rtc::TimeMillis();
- RemoveTimeoutStreams(time_now_ms);
-
- int source_ssrc = rtp_header.ssrc;
- int audio_level = rtp_header.extension.hasAudioLevel ?
- rtp_header.extension.audioLevel : kInvalidAudioLevel;
-
- if (audio_level == kInvalidAudioLevel) {
- // Always forward streams with unknown audio level, and don't keep their
- // states.
- return true;
- }
-
- auto it = stream_levels_.find(source_ssrc);
- if (it != stream_levels_.end()) {
- // Stream has been forwarded. Update and continue to forward.
- it->second.audio_level = audio_level;
- it->second.last_time_ms = time_now_ms;
- return true;
- }
-
- if (stream_levels_.size() < kMaxMixSize) {
- stream_levels_[source_ssrc].Set(audio_level, time_now_ms);
- return true;
- }
-
- unsigned int quietest_ssrc = FindQuietestStream();
- RTC_CHECK_NE(0, quietest_ssrc);
- // A smaller value if audio level corresponds to a louder sound.
- if (audio_level < stream_levels_[quietest_ssrc].audio_level) {
- stream_levels_.erase(quietest_ssrc);
- stream_levels_[source_ssrc].Set(audio_level, time_now_ms);
- return true;
- }
- return false;
-}
-
-} // namespace voetest
-} // namespace webrtc
diff --git a/voice_engine/test/auto_test/fakes/loudest_filter.h b/voice_engine/test/auto_test/fakes/loudest_filter.h
deleted file mode 100644
index dd5a426..0000000
--- a/voice_engine/test/auto_test/fakes/loudest_filter.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
-#define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
-
-#include <map>
-#include "webrtc/common_types.h"
-#include "webrtc/rtc_base/timeutils.h"
-
-namespace webrtc {
-namespace voetest {
-
-class LoudestFilter {
- public:
- /* ForwardThisPacket()
- * Decide whether to forward a RTP packet, given its header.
- *
- * Input:
- * rtp_header : Header of the RTP packet of interest.
- */
- bool ForwardThisPacket(const webrtc::RTPHeader& rtp_header);
-
- private:
- struct Status {
- void Set(int audio_level, int64_t last_time_ms) {
- this->audio_level = audio_level;
- this->last_time_ms = last_time_ms;
- }
- int audio_level;
- int64_t last_time_ms;
- };
-
- void RemoveTimeoutStreams(int64_t time_ms);
- unsigned int FindQuietestStream();
-
- // Keeps the streams being forwarded in pair<SSRC, Status>.
- std::map<unsigned int, Status> stream_levels_;
-
- const int32_t kStreamTimeOutMs = 5000;
- const size_t kMaxMixSize = 3;
- const int kInvalidAudioLevel = 128;
-};
-
-} // namespace voetest
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
diff --git a/voice_engine/test/auto_test/voe_conference_test.cc b/voice_engine/test/auto_test/voe_conference_test.cc
deleted file mode 100644
index 9e466af..0000000
--- a/voice_engine/test/auto_test/voe_conference_test.cc
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- * 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 <queue>
-
-#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/rtc_base/timeutils.h"
-#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
-
-namespace webrtc {
-namespace {
-
-const int kRttMs = 25;
-
-bool IsNear(int ref, int comp, int error) {
- return (ref - comp <= error) && (comp - ref >= -error);
-}
-
-void CreateSilenceFile(const std::string& silence_file, int sample_rate_hz) {
- FILE* fid = fopen(silence_file.c_str(), "wb");
- int16_t zero = 0;
- for (int i = 0; i < sample_rate_hz; ++i) {
- // Write 1 second, but it does not matter since the file will be looped.
- fwrite(&zero, sizeof(int16_t), 1, fid);
- }
- fclose(fid);
-}
-
-} // namespace
-
-namespace voetest {
-
-TEST(VoeConferenceTest, RttAndStartNtpTime) {
- struct Stats {
- Stats(int64_t rtt_receiver_1, int64_t rtt_receiver_2, int64_t ntp_delay)
- : rtt_receiver_1_(rtt_receiver_1),
- rtt_receiver_2_(rtt_receiver_2),
- ntp_delay_(ntp_delay) {
- }
- int64_t rtt_receiver_1_;
- int64_t rtt_receiver_2_;
- int64_t ntp_delay_;
- };
-
- const std::string input_file =
- webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
- const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
-
- const int kDelayMs = 987;
- ConferenceTransport trans;
- trans.SetRtt(kRttMs);
-
- unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
- unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
-
- EXPECT_TRUE(trans.StartPlayout(id_1));
- // Start NTP time is the time when a stream is played out, rather than
- // when it is added.
- webrtc::SleepMs(kDelayMs);
- EXPECT_TRUE(trans.StartPlayout(id_2));
-
- const int kMaxRunTimeMs = 25000;
- const int kNeedSuccessivePass = 3;
- const int kStatsRequestIntervalMs = 1000;
- const int kStatsBufferSize = 3;
-
- int64_t deadline = rtc::TimeAfter(kMaxRunTimeMs);
- // Run the following up to |kMaxRunTimeMs| milliseconds.
- int successive_pass = 0;
- webrtc::CallStatistics stats_1;
- webrtc::CallStatistics stats_2;
- std::queue<Stats> stats_buffer;
-
- while (rtc::TimeMillis() < deadline &&
- successive_pass < kNeedSuccessivePass) {
- webrtc::SleepMs(kStatsRequestIntervalMs);
-
- EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
- EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
-
- // It is not easy to verify the NTP time directly. We verify it by testing
- // the difference of two start NTP times.
- int64_t captured_start_ntp_delay = stats_2.capture_start_ntp_time_ms_ -
- stats_1.capture_start_ntp_time_ms_;
-
- // For the checks of RTT and start NTP time, We allow 10% accuracy.
- if (IsNear(kRttMs, stats_1.rttMs, kRttMs / 10 + 1) &&
- IsNear(kRttMs, stats_2.rttMs, kRttMs / 10 + 1) &&
- IsNear(kDelayMs, captured_start_ntp_delay, kDelayMs / 10 + 1)) {
- successive_pass++;
- } else {
- successive_pass = 0;
- }
- if (stats_buffer.size() >= kStatsBufferSize) {
- stats_buffer.pop();
- }
- stats_buffer.push(Stats(stats_1.rttMs, stats_2.rttMs,
- captured_start_ntp_delay));
- }
-
- EXPECT_GE(successive_pass, kNeedSuccessivePass) << "Expected to get RTT and"
- " start NTP time estimate within 10% of the correct value over "
- << kStatsRequestIntervalMs * kNeedSuccessivePass / 1000
- << " seconds.";
- if (successive_pass < kNeedSuccessivePass) {
- printf("The most recent values (RTT for receiver 1, RTT for receiver 2, "
- "NTP delay between receiver 1 and 2) are (from oldest):\n");
- while (!stats_buffer.empty()) {
- Stats stats = stats_buffer.front();
- printf("(%" PRId64 ", %" PRId64 ", %" PRId64 ")\n", stats.rtt_receiver_1_,
- stats.rtt_receiver_2_, stats.ntp_delay_);
- stats_buffer.pop();
- }
- }
-}
-
-
-TEST(VoeConferenceTest, ReceivedPackets) {
- const int kPackets = 50;
- const int kPacketDurationMs = 20; // Correspond to Opus.
-
- const std::string input_file =
- webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
- const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
-
- const std::string silence_file =
- webrtc::test::TempFilename(webrtc::test::OutputPath(), "silence");
- CreateSilenceFile(silence_file, 32000);
-
- {
- ConferenceTransport trans;
- // Add silence to stream 0, so that it will be filtered out.
- unsigned int id_0 = trans.AddStream(silence_file, kInputFormat);
- unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
- unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
- unsigned int id_3 = trans.AddStream(input_file, kInputFormat);
-
- EXPECT_TRUE(trans.StartPlayout(id_0));
- EXPECT_TRUE(trans.StartPlayout(id_1));
- EXPECT_TRUE(trans.StartPlayout(id_2));
- EXPECT_TRUE(trans.StartPlayout(id_3));
-
- webrtc::SleepMs(kPacketDurationMs * kPackets);
-
- webrtc::CallStatistics stats_0;
- webrtc::CallStatistics stats_1;
- webrtc::CallStatistics stats_2;
- webrtc::CallStatistics stats_3;
- EXPECT_TRUE(trans.GetReceiverStatistics(id_0, &stats_0));
- EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
- EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
- EXPECT_TRUE(trans.GetReceiverStatistics(id_3, &stats_3));
-
- // We expect stream 0 to be filtered out totally, but since it may join the
- // call earlier than other streams and the beginning packets might have got
- // through. So we only expect |packetsReceived| to be close to zero.
- EXPECT_NEAR(stats_0.packetsReceived, 0, 2);
- // We expect |packetsReceived| to match |kPackets|, but the actual value
- // depends on the sleep timer. So we allow a small off from |kPackets|.
- EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2);
- EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2);
- EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2);
- }
-
- remove(silence_file.c_str());
-}
-
-} // namespace voetest
-} // namespace webrtc
