Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 70 insertions, 147 deletions

diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
index 811e9070fa..a489d255c8 100644
--- a/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -9,35 +9,50 @@
  */
 
 #include <stdio.h>
+#include <iostream>
 #include <sstream>
 #include <string>
+#include <utility>
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/audio_file_processor.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 
-DEFINE_string(dump, "", "The name of the debug dump file to read from.");
-DEFINE_string(i, "", "The name of the input file to read from.");
-DEFINE_string(i_rev, "", "The name of the reverse input file to read from.");
-DEFINE_string(o, "out.wav", "Name of the output file to write to.");
-DEFINE_string(o_rev,
-              "out_rev.wav",
-              "Name of the reverse output file to write to.");
-DEFINE_int32(out_channels, 0, "Number of output channels. Defaults to input.");
-DEFINE_int32(out_sample_rate, 0,
-             "Output sample rate in Hz. Defaults to input.");
+namespace {
+
+bool ValidateOutChannels(const char* flagname, int32_t value) {
+  return value >= 0;
+}
+
+}  // namespace
+
+DEFINE_string(dump, "", "Name of the aecdump debug file to read from.");
+DEFINE_string(i, "", "Name of the capture input stream file to read from.");
+DEFINE_string(
+    o,
+    "out.wav",
+    "Name of the output file to write the processed capture stream to.");
+DEFINE_int32(out_channels, 1, "Number of output channels.");
+const bool out_channels_dummy =
+    google::RegisterFlagValidator(&FLAGS_out_channels, &ValidateOutChannels);
+DEFINE_int32(out_sample_rate, 48000, "Output sample rate in Hz.");
 DEFINE_string(mic_positions, "",
     "Space delimited cartesian coordinates of microphones in meters. "
     "The coordinates of each point are contiguous. "
     "For a two element array: \"x1 y1 z1 x2 y2 z2\"");
-DEFINE_double(target_angle_degrees, 90, "The azimuth of the target in radians");
+DEFINE_double(
+    target_angle_degrees,
+    90,
+    "The azimuth of the target in degrees. Only applies to beamforming.");
 
 DEFINE_bool(aec, false, "Enable echo cancellation.");
 DEFINE_bool(agc, false, "Enable automatic gain control.");
@@ -64,15 +79,6 @@ const char kUsage[] =
     "All components are disabled by default. If any bi-directional components\n"
     "are enabled, only debug dump files are permitted.";
 
-// Returns a StreamConfig corresponding to wav_file if it's non-nullptr.
-// Otherwise returns a default initialized StreamConfig.
-StreamConfig MakeStreamConfig(const WavFile* wav_file) {
-  if (wav_file) {
-    return {wav_file->sample_rate(), wav_file->num_channels()};
-  }
-  return {};
-}
-
 }  // namespace
 
 int main(int argc, char* argv[]) {
@@ -84,158 +90,75 @@ int main(int argc, char* argv[]) {
             "An input file must be specified with either -i or -dump.\n");
     return 1;
   }
-  if (!FLAGS_dump.empty()) {
-    fprintf(stderr, "FIXME: the -dump option is not yet implemented.\n");
+  if (FLAGS_dump.empty() && (FLAGS_aec || FLAGS_ie)) {
+    fprintf(stderr, "-aec and -ie require a -dump file.\n");
+    return 1;
+  }
+  if (FLAGS_ie) {
+    fprintf(stderr,
+            "FIXME(ajm): The intelligibility enhancer output is not dumped.\n");
     return 1;
   }
 
   test::TraceToStderr trace_to_stderr(true);
-  WavReader in_file(FLAGS_i);
-  // If the output format is uninitialized, use the input format.
-  const int out_channels =
-      FLAGS_out_channels ? FLAGS_out_channels : in_file.num_channels();
-  const int out_sample_rate =
-      FLAGS_out_sample_rate ? FLAGS_out_sample_rate : in_file.sample_rate();
-  WavWriter out_file(FLAGS_o, out_sample_rate, out_channels);
-
   Config config;
-  config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
-  config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all));
-
   if (FLAGS_bf || FLAGS_all) {
-    const size_t num_mics = in_file.num_channels();
-    const std::vector<Point> array_geometry =
-        ParseArrayGeometry(FLAGS_mic_positions, num_mics);
-    RTC_CHECK_EQ(array_geometry.size(), num_mics);
-
+    if (FLAGS_mic_positions.empty()) {
+      fprintf(stderr, "-mic_positions must be specified when -bf is used.\n");
+      return 1;
+    }
     config.Set<Beamforming>(new Beamforming(
-        true, array_geometry,
+        true, ParseArrayGeometry(FLAGS_mic_positions),
         SphericalPointf(DegreesToRadians(FLAGS_target_angle_degrees), 0.f,
                         1.f)));
   }
+  config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
+  config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all));
 
   rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
-  if (!FLAGS_dump.empty()) {
-    RTC_CHECK_EQ(kNoErr,
-                 ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
-  } else if (FLAGS_aec) {
-    fprintf(stderr, "-aec requires a -dump file.\n");
-    return -1;
-  }
-  bool process_reverse = !FLAGS_i_rev.empty();
+  RTC_CHECK_EQ(kNoErr, ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
   RTC_CHECK_EQ(kNoErr, ap->gain_control()->Enable(FLAGS_agc || FLAGS_all));
-  RTC_CHECK_EQ(kNoErr,
-               ap->gain_control()->set_mode(GainControl::kFixedDigital));
   RTC_CHECK_EQ(kNoErr, ap->high_pass_filter()->Enable(FLAGS_hpf || FLAGS_all));
   RTC_CHECK_EQ(kNoErr, ap->noise_suppression()->Enable(FLAGS_ns || FLAGS_all));
-  if (FLAGS_ns_level != -1)
+  if (FLAGS_ns_level != -1) {
     RTC_CHECK_EQ(kNoErr,
                  ap->noise_suppression()->set_level(
                      static_cast<NoiseSuppression::Level>(FLAGS_ns_level)));
-
-  printf("Input file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
-         FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
-  printf("Output file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
-         FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
-
-  ChannelBuffer<float> in_buf(
-      rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond),
-      in_file.num_channels());
-  ChannelBuffer<float> out_buf(
-      rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond),
-      out_file.num_channels());
-
-  std::vector<float> in_interleaved(in_buf.size());
-  std::vector<float> out_interleaved(out_buf.size());
-
-  rtc::scoped_ptr<WavReader> in_rev_file;
-  rtc::scoped_ptr<WavWriter> out_rev_file;
-  rtc::scoped_ptr<ChannelBuffer<float>> in_rev_buf;
-  rtc::scoped_ptr<ChannelBuffer<float>> out_rev_buf;
-  std::vector<float> in_rev_interleaved;
-  std::vector<float> out_rev_interleaved;
-  if (process_reverse) {
-    in_rev_file.reset(new WavReader(FLAGS_i_rev));
-    out_rev_file.reset(new WavWriter(FLAGS_o_rev, in_rev_file->sample_rate(),
-                                     in_rev_file->num_channels()));
-    printf("In rev file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
-           FLAGS_i_rev.c_str(), in_rev_file->num_channels(),
-           in_rev_file->sample_rate());
-    printf("Out rev file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
-           FLAGS_o_rev.c_str(), out_rev_file->num_channels(),
-           out_rev_file->sample_rate());
-    in_rev_buf.reset(new ChannelBuffer<float>(
-        rtc::CheckedDivExact(in_rev_file->sample_rate(), kChunksPerSecond),
-        in_rev_file->num_channels()));
-    in_rev_interleaved.resize(in_rev_buf->size());
-    out_rev_buf.reset(new ChannelBuffer<float>(
-        rtc::CheckedDivExact(out_rev_file->sample_rate(), kChunksPerSecond),
-        out_rev_file->num_channels()));
-    out_rev_interleaved.resize(out_rev_buf->size());
+  }
+  ap->set_stream_key_pressed(FLAGS_ts);
+
+  rtc::scoped_ptr<AudioFileProcessor> processor;
+  auto out_file = rtc_make_scoped_ptr(new WavWriter(
+      FLAGS_o, FLAGS_out_sample_rate, static_cast<size_t>(FLAGS_out_channels)));
+  std::cout << FLAGS_o << ": " << out_file->FormatAsString() << std::endl;
+  if (FLAGS_dump.empty()) {
+    auto in_file = rtc_make_scoped_ptr(new WavReader(FLAGS_i));
+    std::cout << FLAGS_i << ": " << in_file->FormatAsString() << std::endl;
+    processor.reset(new WavFileProcessor(std::move(ap), std::move(in_file),
+                                         std::move(out_file)));
+
+  } else {
+    processor.reset(new AecDumpFileProcessor(
+        std::move(ap), fopen(FLAGS_dump.c_str(), "rb"), std::move(out_file)));
   }
 
-  TickTime processing_start_time;
-  TickInterval accumulated_time;
   int num_chunks = 0;
-
-  const auto input_config = MakeStreamConfig(&in_file);
-  const auto output_config = MakeStreamConfig(&out_file);
-  const auto reverse_input_config = MakeStreamConfig(in_rev_file.get());
-  const auto reverse_output_config = MakeStreamConfig(out_rev_file.get());
-
-  while (in_file.ReadSamples(in_interleaved.size(),
-                             &in_interleaved[0]) == in_interleaved.size()) {
-    // Have logs display the file time rather than wallclock time.
+  while (processor->ProcessChunk()) {
     trace_to_stderr.SetTimeSeconds(num_chunks * 1.f / kChunksPerSecond);
-    FloatS16ToFloat(&in_interleaved[0], in_interleaved.size(),
-                    &in_interleaved[0]);
-    Deinterleave(&in_interleaved[0], in_buf.num_frames(),
-                 in_buf.num_channels(), in_buf.channels());
-    if (process_reverse) {
-      in_rev_file->ReadSamples(in_rev_interleaved.size(),
-                               in_rev_interleaved.data());
-      FloatS16ToFloat(in_rev_interleaved.data(), in_rev_interleaved.size(),
-                      in_rev_interleaved.data());
-      Deinterleave(in_rev_interleaved.data(), in_rev_buf->num_frames(),
-                   in_rev_buf->num_channels(), in_rev_buf->channels());
-    }
-
-    if (FLAGS_perf) {
-      processing_start_time = TickTime::Now();
-    }
-    RTC_CHECK_EQ(kNoErr, ap->ProcessStream(in_buf.channels(), input_config,
-                                           output_config, out_buf.channels()));
-    if (process_reverse) {
-      RTC_CHECK_EQ(kNoErr, ap->ProcessReverseStream(
-                               in_rev_buf->channels(), reverse_input_config,
-                               reverse_output_config, out_rev_buf->channels()));
-    }
-    if (FLAGS_perf) {
-      accumulated_time += TickTime::Now() - processing_start_time;
-    }
-
-    Interleave(out_buf.channels(), out_buf.num_frames(),
-               out_buf.num_channels(), &out_interleaved[0]);
-    FloatToFloatS16(&out_interleaved[0], out_interleaved.size(),
-                    &out_interleaved[0]);
-    out_file.WriteSamples(&out_interleaved[0], out_interleaved.size());
-    if (process_reverse) {
-      Interleave(out_rev_buf->channels(), out_rev_buf->num_frames(),
-                 out_rev_buf->num_channels(), out_rev_interleaved.data());
-      FloatToFloatS16(out_rev_interleaved.data(), out_rev_interleaved.size(),
-                      out_rev_interleaved.data());
-      out_rev_file->WriteSamples(out_rev_interleaved.data(),
-                                 out_rev_interleaved.size());
-    }
-    num_chunks++;
+    ++num_chunks;
   }
+
   if (FLAGS_perf) {
-    int64_t execution_time_ms = accumulated_time.Milliseconds();
-    printf("\nExecution time: %.3f s\nFile time: %.2f s\n"
-           "Time per chunk: %.3f ms\n",
-           execution_time_ms * 0.001f, num_chunks * 1.f / kChunksPerSecond,
-           execution_time_ms * 1.f / num_chunks);
+    const auto& proc_time = processor->proc_time();
+    int64_t exec_time_us = proc_time.sum.Microseconds();
+    printf(
+        "\nExecution time: %.3f s, File time: %.2f s\n"
+        "Time per chunk (mean, max, min):\n%.0f us, %.0f us, %.0f us\n",
+        exec_time_us * 1e-6, num_chunks * 1.f / kChunksPerSecond,
+        exec_time_us * 1.f / num_chunks, 1.f * proc_time.max.Microseconds(),
+        1.f * proc_time.min.Microseconds());
   }
+
   return 0;
 }