Adding upstream version 115.7.0esr.upstream/115.7.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 439 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/audio_receive_stream_unittest.cc b/third_party/libwebrtc/audio/audio_receive_stream_unittest.cc
new file mode 100644
index 0000000000..2cee6a4bae
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_receive_stream_unittest.cc
@@ -0,0 +1,439 @@
+/*
+ *  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 "audio/audio_receive_stream.h"
+
+#include <map>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/test/mock_audio_mixer.h"
+#include "api/test/mock_frame_decryptor.h"
+#include "audio/conversion.h"
+#include "audio/mock_voe_channel_proxy.h"
+#include "call/rtp_stream_receiver_controller.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/time_utils.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder_factory.h"
+#include "test/mock_transport.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using ::testing::_;
+using ::testing::FloatEq;
+using ::testing::NiceMock;
+using ::testing::Return;
+
+AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
+  AudioDecodingCallStats audio_decode_stats;
+  audio_decode_stats.calls_to_silence_generator = 234;
+  audio_decode_stats.calls_to_neteq = 567;
+  audio_decode_stats.decoded_normal = 890;
+  audio_decode_stats.decoded_neteq_plc = 123;
+  audio_decode_stats.decoded_codec_plc = 124;
+  audio_decode_stats.decoded_cng = 456;
+  audio_decode_stats.decoded_plc_cng = 789;
+  audio_decode_stats.decoded_muted_output = 987;
+  return audio_decode_stats;
+}
+
+const uint32_t kRemoteSsrc = 1234;
+const uint32_t kLocalSsrc = 5678;
+const int kAudioLevelId = 3;
+const int kTransportSequenceNumberId = 4;
+const int kJitterBufferDelay = -7;
+const int kPlayoutBufferDelay = 302;
+const unsigned int kSpeechOutputLevel = 99;
+const double kTotalOutputEnergy = 0.25;
+const double kTotalOutputDuration = 0.5;
+const int64_t kPlayoutNtpTimestampMs = 5678;
+
+const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 78, 890, 123};
+const std::pair<int, SdpAudioFormat> kReceiveCodec = {
+    123,
+    {"codec_name_recv", 96000, 0}};
+const NetworkStatistics kNetworkStats = {
+    /*currentBufferSize=*/123,
+    /*preferredBufferSize=*/456,
+    /*jitterPeaksFound=*/false,
+    /*totalSamplesReceived=*/789012,
+    /*concealedSamples=*/3456,
+    /*silentConcealedSamples=*/123,
+    /*concealmentEvents=*/456,
+    /*jitterBufferDelayMs=*/789,
+    /*jitterBufferEmittedCount=*/543,
+    /*jitterBufferTargetDelayMs=*/123,
+    /*jitterBufferMinimumDelayMs=*/222,
+    /*insertedSamplesForDeceleration=*/432,
+    /*removedSamplesForAcceleration=*/321,
+    /*fecPacketsReceived=*/123,
+    /*fecPacketsDiscarded=*/101,
+    /*packetsDiscarded=*/989,
+    /*currentExpandRate=*/789,
+    /*currentSpeechExpandRate=*/12,
+    /*currentPreemptiveRate=*/345,
+    /*currentAccelerateRate =*/678,
+    /*currentSecondaryDecodedRate=*/901,
+    /*currentSecondaryDiscardedRate=*/0,
+    /*meanWaitingTimeMs=*/-1,
+    /*maxWaitingTimeMs=*/-1,
+    /*packetBufferFlushes=*/0,
+    /*delayedPacketOutageSamples=*/0,
+    /*relativePacketArrivalDelayMs=*/135,
+    /*interruptionCount=*/-1,
+    /*totalInterruptionDurationMs=*/-1};
+const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
+
+struct ConfigHelper {
+  explicit ConfigHelper(bool use_null_audio_processing)
+      : ConfigHelper(rtc::make_ref_counted<MockAudioMixer>(),
+                     use_null_audio_processing) {}
+
+  ConfigHelper(rtc::scoped_refptr<MockAudioMixer> audio_mixer,
+               bool use_null_audio_processing)
+      : audio_mixer_(audio_mixer) {
+    using ::testing::Invoke;
+
+    AudioState::Config config;
+    config.audio_mixer = audio_mixer_;
+    config.audio_processing =
+        use_null_audio_processing
+            ? nullptr
+            : rtc::make_ref_counted<NiceMock<MockAudioProcessing>>();
+    config.audio_device_module =
+        rtc::make_ref_counted<testing::NiceMock<MockAudioDeviceModule>>();
+    audio_state_ = AudioState::Create(config);
+
+    channel_receive_ = new ::testing::StrictMock<MockChannelReceive>();
+    EXPECT_CALL(*channel_receive_, SetNACKStatus(true, 15)).Times(1);
+    EXPECT_CALL(*channel_receive_,
+                RegisterReceiverCongestionControlObjects(&packet_router_))
+        .Times(1);
+    EXPECT_CALL(*channel_receive_, ResetReceiverCongestionControlObjects())
+        .Times(1);
+    EXPECT_CALL(*channel_receive_, SetAssociatedSendChannel(nullptr)).Times(1);
+    EXPECT_CALL(*channel_receive_, SetReceiveCodecs(_))
+        .WillRepeatedly(Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
+          EXPECT_THAT(codecs, ::testing::IsEmpty());
+        }));
+    EXPECT_CALL(*channel_receive_, SetSourceTracker(_));
+    EXPECT_CALL(*channel_receive_, GetLocalSsrc())
+        .WillRepeatedly(Return(kLocalSsrc));
+
+    stream_config_.rtp.local_ssrc = kLocalSsrc;
+    stream_config_.rtp.remote_ssrc = kRemoteSsrc;
+    stream_config_.rtp.nack.rtp_history_ms = 300;
+    stream_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
+    stream_config_.rtp.extensions.push_back(RtpExtension(
+        RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
+    stream_config_.rtcp_send_transport = &rtcp_send_transport_;
+    stream_config_.decoder_factory =
+        rtc::make_ref_counted<MockAudioDecoderFactory>();
+  }
+
+  std::unique_ptr<AudioReceiveStreamImpl> CreateAudioReceiveStream() {
+    auto ret = std::make_unique<AudioReceiveStreamImpl>(
+        Clock::GetRealTimeClock(), &packet_router_, stream_config_,
+        audio_state_, &event_log_,
+        std::unique_ptr<voe::ChannelReceiveInterface>(channel_receive_));
+    ret->RegisterWithTransport(&rtp_stream_receiver_controller_);
+    return ret;
+  }
+
+  AudioReceiveStreamInterface::Config& config() { return stream_config_; }
+  rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; }
+  MockChannelReceive* channel_receive() { return channel_receive_; }
+
+  void SetupMockForGetStats() {
+    using ::testing::DoAll;
+    using ::testing::SetArgPointee;
+
+    ASSERT_TRUE(channel_receive_);
+    EXPECT_CALL(*channel_receive_, GetRTCPStatistics())
+        .WillOnce(Return(kCallStats));
+    EXPECT_CALL(*channel_receive_, GetDelayEstimate())
+        .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
+    EXPECT_CALL(*channel_receive_, GetSpeechOutputLevelFullRange())
+        .WillOnce(Return(kSpeechOutputLevel));
+    EXPECT_CALL(*channel_receive_, GetTotalOutputEnergy())
+        .WillOnce(Return(kTotalOutputEnergy));
+    EXPECT_CALL(*channel_receive_, GetTotalOutputDuration())
+        .WillOnce(Return(kTotalOutputDuration));
+    EXPECT_CALL(*channel_receive_, GetNetworkStatistics(_))
+        .WillOnce(Return(kNetworkStats));
+    EXPECT_CALL(*channel_receive_, GetDecodingCallStatistics())
+        .WillOnce(Return(kAudioDecodeStats));
+    EXPECT_CALL(*channel_receive_, GetReceiveCodec())
+        .WillOnce(Return(kReceiveCodec));
+    EXPECT_CALL(*channel_receive_, GetCurrentEstimatedPlayoutNtpTimestampMs(_))
+        .WillOnce(Return(kPlayoutNtpTimestampMs));
+  }
+
+ private:
+  PacketRouter packet_router_;
+  MockRtcEventLog event_log_;
+  rtc::scoped_refptr<AudioState> audio_state_;
+  rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
+  AudioReceiveStreamInterface::Config stream_config_;
+  ::testing::StrictMock<MockChannelReceive>* channel_receive_ = nullptr;
+  RtpStreamReceiverController rtp_stream_receiver_controller_;
+  MockTransport rtcp_send_transport_;
+};
+
+const std::vector<uint8_t> CreateRtcpSenderReport() {
+  std::vector<uint8_t> packet;
+  const size_t kRtcpSrLength = 28;  // In bytes.
+  packet.resize(kRtcpSrLength);
+  packet[0] = 0x80;  // Version 2.
+  packet[1] = 0xc8;  // PT = 200, SR.
+  // Length in number of 32-bit words - 1.
+  ByteWriter<uint16_t>::WriteBigEndian(&packet[2], 6);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[4], kLocalSsrc);
+  return packet;
+}
+}  // namespace
+
+TEST(AudioReceiveStreamTest, ConfigToString) {
+  AudioReceiveStreamInterface::Config config;
+  config.rtp.remote_ssrc = kRemoteSsrc;
+  config.rtp.local_ssrc = kLocalSsrc;
+  config.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
+  EXPECT_EQ(
+      "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, nack: "
+      "{rtp_history_ms: 0}, extensions: [{uri: "
+      "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, "
+      "rtcp_send_transport: null}",
+      config.ToString());
+}
+
+TEST(AudioReceiveStreamTest, ConstructDestruct) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+    std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
+    EXPECT_CALL(*helper.channel_receive(),
+                ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size()))
+        .WillOnce(Return());
+    recv_stream->DeliverRtcp(&rtcp_packet[0], rtcp_packet.size());
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, GetStats) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+    helper.SetupMockForGetStats();
+    AudioReceiveStreamInterface::Stats stats =
+        recv_stream->GetStats(/*get_and_clear_legacy_stats=*/true);
+    EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
+    EXPECT_EQ(kCallStats.payload_bytes_rcvd, stats.payload_bytes_rcvd);
+    EXPECT_EQ(kCallStats.header_and_padding_bytes_rcvd,
+              stats.header_and_padding_bytes_rcvd);
+    EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
+              stats.packets_rcvd);
+    EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
+    EXPECT_EQ(kReceiveCodec.second.name, stats.codec_name);
+    EXPECT_EQ(
+        kCallStats.jitterSamples / (kReceiveCodec.second.clockrate_hz / 1000),
+        stats.jitter_ms);
+    EXPECT_EQ(kNetworkStats.currentBufferSize, stats.jitter_buffer_ms);
+    EXPECT_EQ(kNetworkStats.preferredBufferSize,
+              stats.jitter_buffer_preferred_ms);
+    EXPECT_EQ(static_cast<uint32_t>(kJitterBufferDelay + kPlayoutBufferDelay),
+              stats.delay_estimate_ms);
+    EXPECT_EQ(static_cast<int32_t>(kSpeechOutputLevel), stats.audio_level);
+    EXPECT_EQ(kTotalOutputEnergy, stats.total_output_energy);
+    EXPECT_EQ(kNetworkStats.totalSamplesReceived, stats.total_samples_received);
+    EXPECT_EQ(kTotalOutputDuration, stats.total_output_duration);
+    EXPECT_EQ(kNetworkStats.concealedSamples, stats.concealed_samples);
+    EXPECT_EQ(kNetworkStats.concealmentEvents, stats.concealment_events);
+    EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferDelayMs) /
+                  static_cast<double>(rtc::kNumMillisecsPerSec),
+              stats.jitter_buffer_delay_seconds);
+    EXPECT_EQ(kNetworkStats.jitterBufferEmittedCount,
+              stats.jitter_buffer_emitted_count);
+    EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferTargetDelayMs) /
+                  static_cast<double>(rtc::kNumMillisecsPerSec),
+              stats.jitter_buffer_target_delay_seconds);
+    EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferMinimumDelayMs) /
+                  static_cast<double>(rtc::kNumMillisecsPerSec),
+              stats.jitter_buffer_minimum_delay_seconds);
+    EXPECT_EQ(kNetworkStats.insertedSamplesForDeceleration,
+              stats.inserted_samples_for_deceleration);
+    EXPECT_EQ(kNetworkStats.removedSamplesForAcceleration,
+              stats.removed_samples_for_acceleration);
+    EXPECT_EQ(kNetworkStats.fecPacketsReceived, stats.fec_packets_received);
+    EXPECT_EQ(kNetworkStats.fecPacketsDiscarded, stats.fec_packets_discarded);
+    EXPECT_EQ(kNetworkStats.packetsDiscarded, stats.packets_discarded);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate),
+              stats.speech_expand_rate);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDecodedRate),
+              stats.secondary_decoded_rate);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDiscardedRate),
+              stats.secondary_discarded_rate);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentAccelerateRate),
+              stats.accelerate_rate);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate),
+              stats.preemptive_expand_rate);
+    EXPECT_EQ(kNetworkStats.packetBufferFlushes, stats.jitter_buffer_flushes);
+    EXPECT_EQ(kNetworkStats.delayedPacketOutageSamples,
+              stats.delayed_packet_outage_samples);
+    EXPECT_EQ(static_cast<double>(kNetworkStats.relativePacketArrivalDelayMs) /
+                  static_cast<double>(rtc::kNumMillisecsPerSec),
+              stats.relative_packet_arrival_delay_seconds);
+    EXPECT_EQ(kNetworkStats.interruptionCount, stats.interruption_count);
+    EXPECT_EQ(kNetworkStats.totalInterruptionDurationMs,
+              stats.total_interruption_duration_ms);
+
+    EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator,
+              stats.decoding_calls_to_silence_generator);
+    EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
+    EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
+    EXPECT_EQ(kAudioDecodeStats.decoded_neteq_plc, stats.decoding_plc);
+    EXPECT_EQ(kAudioDecodeStats.decoded_codec_plc, stats.decoding_codec_plc);
+    EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
+    EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
+    EXPECT_EQ(kAudioDecodeStats.decoded_muted_output,
+              stats.decoding_muted_output);
+    EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
+              stats.capture_start_ntp_time_ms);
+    EXPECT_EQ(kPlayoutNtpTimestampMs, stats.estimated_playout_ntp_timestamp_ms);
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, SetGain) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+    EXPECT_CALL(*helper.channel_receive(),
+                SetChannelOutputVolumeScaling(FloatEq(0.765f)));
+    recv_stream->SetGain(0.765f);
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, StreamsShouldBeAddedToMixerOnceOnStart) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper1(use_null_audio_processing);
+    ConfigHelper helper2(helper1.audio_mixer(), use_null_audio_processing);
+    auto recv_stream1 = helper1.CreateAudioReceiveStream();
+    auto recv_stream2 = helper2.CreateAudioReceiveStream();
+
+    EXPECT_CALL(*helper1.channel_receive(), StartPlayout()).Times(1);
+    EXPECT_CALL(*helper2.channel_receive(), StartPlayout()).Times(1);
+    EXPECT_CALL(*helper1.channel_receive(), StopPlayout()).Times(1);
+    EXPECT_CALL(*helper2.channel_receive(), StopPlayout()).Times(1);
+    EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream1.get()))
+        .WillOnce(Return(true));
+    EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream2.get()))
+        .WillOnce(Return(true));
+    EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream1.get()))
+        .Times(1);
+    EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream2.get()))
+        .Times(1);
+
+    recv_stream1->Start();
+    recv_stream2->Start();
+
+    // One more should not result in any more mixer sources added.
+    recv_stream1->Start();
+
+    // Stop stream before it is being destructed.
+    recv_stream2->Stop();
+
+    recv_stream1->UnregisterFromTransport();
+    recv_stream2->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+
+    auto new_config = helper.config();
+
+    new_config.rtp.extensions.clear();
+    new_config.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId + 1));
+    new_config.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                     kTransportSequenceNumberId + 1));
+
+    MockChannelReceive& channel_receive = *helper.channel_receive();
+
+    // TODO(tommi, nisse): This applies new extensions to the internal config,
+    // but there's nothing that actually verifies that the changes take effect.
+    // In fact Call manages the extensions separately in Call::ReceiveRtpConfig
+    // and changing this config value (there seem to be a few copies), doesn't
+    // affect that logic.
+    recv_stream->ReconfigureForTesting(new_config);
+
+    new_config.decoder_map.emplace(1, SdpAudioFormat("foo", 8000, 1));
+    EXPECT_CALL(channel_receive, SetReceiveCodecs(new_config.decoder_map));
+    recv_stream->SetDecoderMap(new_config.decoder_map);
+
+    EXPECT_CALL(channel_receive, SetNACKStatus(true, 15 + 1)).Times(1);
+    recv_stream->SetNackHistory(300 + 20);
+
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, ReconfigureWithFrameDecryptor) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+
+    auto new_config_0 = helper.config();
+    rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_0(
+        rtc::make_ref_counted<MockFrameDecryptor>());
+    new_config_0.frame_decryptor = mock_frame_decryptor_0;
+
+    // TODO(tommi): While this changes the internal config value, it doesn't
+    // actually change what frame_decryptor is used. WebRtcAudioReceiveStream
+    // recreates the whole instance in order to change this value.
+    // So, it's not clear if changing this post initialization needs to be
+    // supported.
+    recv_stream->ReconfigureForTesting(new_config_0);
+
+    auto new_config_1 = helper.config();
+    rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_1(
+        rtc::make_ref_counted<MockFrameDecryptor>());
+    new_config_1.frame_decryptor = mock_frame_decryptor_1;
+    new_config_1.crypto_options.sframe.require_frame_encryption = true;
+    recv_stream->ReconfigureForTesting(new_config_1);
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc