Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 270 insertions, 0 deletions

diff --git a/webrtc/modules/audio_coding/main/test/delay_test.cc b/webrtc/modules/audio_coding/main/test/delay_test.cc
new file mode 100644
index 0000000000..6186d67fc9
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/test/delay_test.cc
@@ -0,0 +1,270 @@
+/*
+ *  Copyright (c) 2013 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 <assert.h>
+#include <math.h>
+
+#include <iostream>
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/test/Channel.h"
+#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+DEFINE_string(codec, "isac", "Codec Name");
+DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
+DEFINE_int32(num_channels, 1, "Number of Channels.");
+DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
+DEFINE_int32(delay, 0, "Delay in millisecond.");
+DEFINE_int32(init_delay, 0, "Initial delay in millisecond.");
+DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
+DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
+DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
+
+namespace webrtc {
+
+namespace {
+
+struct CodecSettings {
+  char name[50];
+  int sample_rate_hz;
+  int num_channels;
+};
+
+struct AcmSettings {
+  bool dtx;
+  bool fec;
+};
+
+struct TestSettings {
+  CodecSettings codec;
+  AcmSettings acm;
+  bool packet_loss;
+};
+
+}  // namespace
+
+class DelayTest {
+ public:
+  DelayTest()
+      : acm_a_(AudioCodingModule::Create(0)),
+        acm_b_(AudioCodingModule::Create(1)),
+        channel_a2b_(new Channel),
+        test_cntr_(0),
+        encoding_sample_rate_hz_(8000) {}
+
+  ~DelayTest() {
+    if (channel_a2b_ != NULL) {
+      delete channel_a2b_;
+      channel_a2b_ = NULL;
+    }
+    in_file_a_.Close();
+  }
+
+  void Initialize() {
+    test_cntr_ = 0;
+    std::string file_name = webrtc::test::ResourcePath(
+        "audio_coding/testfile32kHz", "pcm");
+    if (FLAGS_input_file.size() > 0)
+      file_name = FLAGS_input_file;
+    in_file_a_.Open(file_name, 32000, "rb");
+    ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
+        "Couldn't initialize receiver.\n";
+    ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
+        "Couldn't initialize receiver.\n";
+    if (FLAGS_init_delay > 0) {
+      ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay)) <<
+          "Failed to set initial delay.\n";
+    }
+
+    if (FLAGS_delay > 0) {
+      ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
+          "Failed to set minimum delay.\n";
+    }
+
+    int num_encoders = acm_a_->NumberOfCodecs();
+    CodecInst my_codec_param;
+    for (int n = 0; n < num_encoders; n++) {
+      EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
+          "Failed to get codec.";
+      if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
+        my_codec_param.channels = 1;
+      else if (my_codec_param.channels > 1)
+        continue;
+      if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
+          my_codec_param.plfreq == 48000)
+        continue;
+      if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
+        continue;
+      ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
+          "Couldn't register receive codec.\n";
+    }
+
+    // Create and connect the channel
+    ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
+        "Couldn't register Transport callback.\n";
+    channel_a2b_->RegisterReceiverACM(acm_b_.get());
+  }
+
+  void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
+               const char* output_prefix) {
+    for (size_t n = 0; n < num_tests; ++n) {
+      ApplyConfig(config[n]);
+      Run(duration_sec, output_prefix);
+    }
+  }
+
+ private:
+  void ApplyConfig(const TestSettings& config) {
+    printf("====================================\n");
+    printf("Test %d \n"
+           "Codec: %s, %d kHz, %d channel(s)\n"
+           "ACM: DTX %s, FEC %s\n"
+           "Channel: %s\n",
+           ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
+           config.codec.num_channels, config.acm.dtx ? "on" : "off",
+           config.acm.fec ? "on" : "off",
+           config.packet_loss ? "with packet-loss" : "no packet-loss");
+    SendCodec(config.codec);
+    ConfigAcm(config.acm);
+    ConfigChannel(config.packet_loss);
+  }
+
+  void SendCodec(const CodecSettings& config) {
+    CodecInst my_codec_param;
+    ASSERT_EQ(0, AudioCodingModule::Codec(
+              config.name, &my_codec_param, config.sample_rate_hz,
+              config.num_channels)) << "Specified codec is not supported.\n";
+
+    encoding_sample_rate_hz_ = my_codec_param.plfreq;
+    ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
+        "Failed to register send-codec.\n";
+  }
+
+  void ConfigAcm(const AcmSettings& config) {
+    ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
+        "Failed to set VAD.\n";
+    ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
+        "Failed to set RED.\n";
+  }
+
+  void ConfigChannel(bool packet_loss) {
+    channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
+  }
+
+  void OpenOutFile(const char* output_id) {
+    std::stringstream file_stream;
+    file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
+        << "Hz" << "_" << FLAGS_init_delay << "ms_" << FLAGS_delay << "ms.pcm";
+    std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
+    std::string file_name = webrtc::test::OutputPath() + file_stream.str();
+    out_file_b_.Open(file_name.c_str(), 32000, "wb");
+  }
+
+  void Run(int duration_sec, const char* output_prefix) {
+    OpenOutFile(output_prefix);
+    AudioFrame audio_frame;
+    uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
+
+    int num_frames = 0;
+    int in_file_frames = 0;
+    uint32_t playout_ts;
+    uint32_t received_ts;
+    double average_delay = 0;
+    double inst_delay_sec = 0;
+    while (num_frames < (duration_sec * 100)) {
+      if (in_file_a_.EndOfFile()) {
+        in_file_a_.Rewind();
+      }
+
+      // Print delay information every 16 frame
+      if ((num_frames & 0x3F) == 0x3F) {
+        NetworkStatistics statistics;
+        acm_b_->GetNetworkStatistics(&statistics);
+        fprintf(stdout, "delay: min=%3d  max=%3d  mean=%3d  median=%3d"
+                " ts-based average = %6.3f, "
+                "curr buff-lev = %4u opt buff-lev = %4u \n",
+                statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
+                statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
+                average_delay, statistics.currentBufferSize,
+                statistics.preferredBufferSize);
+        fflush (stdout);
+      }
+
+      in_file_a_.Read10MsData(audio_frame);
+      ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
+      ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+      out_file_b_.Write10MsData(
+          audio_frame.data_,
+          audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+      acm_b_->PlayoutTimestamp(&playout_ts);
+      received_ts = channel_a2b_->LastInTimestamp();
+      inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
+          / static_cast<double>(encoding_sample_rate_hz_);
+
+      if (num_frames > 10)
+        average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
+
+      ++num_frames;
+      ++in_file_frames;
+    }
+    out_file_b_.Close();
+  }
+
+  rtc::scoped_ptr<AudioCodingModule> acm_a_;
+  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+
+  Channel* channel_a2b_;
+
+  PCMFile in_file_a_;
+  PCMFile out_file_b_;
+  int test_cntr_;
+  int encoding_sample_rate_hz_;
+};
+
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  google::ParseCommandLineFlags(&argc, &argv, true);
+  webrtc::TestSettings test_setting;
+  strcpy(test_setting.codec.name, FLAGS_codec.c_str());
+
+  if (FLAGS_sample_rate_hz != 8000 &&
+      FLAGS_sample_rate_hz != 16000 &&
+      FLAGS_sample_rate_hz != 32000 &&
+      FLAGS_sample_rate_hz != 48000) {
+    std::cout << "Invalid sampling rate.\n";
+    return 1;
+  }
+  test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
+  if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
+    std::cout << "Only mono and stereo are supported.\n";
+    return 1;
+  }
+  test_setting.codec.num_channels = FLAGS_num_channels;
+  test_setting.acm.dtx = FLAGS_dtx;
+  test_setting.acm.fec = FLAGS_fec;
+  test_setting.packet_loss = FLAGS_packet_loss;
+
+  webrtc::DelayTest delay_test;
+  delay_test.Initialize();
+  delay_test.Perform(&test_setting, 1, 240, "delay_test");
+  return 0;
+}