Adding DebugDumpReplayer.

webrtc/modules/audio_processing/test/debug_dump_replayer.cc

Index: webrtc/modules/audio_processing/test/debug_dump_replayer.cc
diff --git a/webrtc/modules/audio_processing/test/debug_dump_replayer.cc b/webrtc/modules/audio_processing/test/debug_dump_replayer.cc
new file mode 100644
index 0000000000000000000000000000000000000000..69afec62e472890d22d13fc6e1917980e3f8c69f
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/debug_dump_replayer.cc
@@ -0,0 +1,266 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/test/debug_dump_replayer.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
+
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
+                      const StreamConfig& config) {
+  auto& buffer_ref = *buffer;
+  if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
+      buffer_ref->num_channels() != config.num_channels()) {
+    buffer_ref.reset(new ChannelBuffer<float>(config.num_frames(),
+                                             config.num_channels()));
+  }
+}
+
+}  // namespace
+
+DebugDumpReplayer::DebugDumpReplayer()
+    : input_(nullptr),  // will be created upon usage.
+      reverse_(nullptr),
+      output_(nullptr),
+      apm_(nullptr),
+      debug_file_(nullptr) {}
+
+DebugDumpReplayer::~DebugDumpReplayer() {
+  if (debug_file_)
+    fclose(debug_file_);
+}
+
+bool DebugDumpReplayer::SetDumpFile(const std::string& filename) {
+  debug_file_ = fopen(filename.c_str(), "rb");
+  LoadNextMessage();
+  return debug_file_;
+}
+
+// Get next event that has not run.
+rtc::Optional<audioproc::Event> DebugDumpReplayer::GetNextEvent() const {
+  if (!has_next_event_)
+    return rtc::Optional<audioproc::Event>();
+  else
+    return rtc::Optional<audioproc::Event>(next_event_);
+}
+
+// Run the next event. Returns the event type.
+bool DebugDumpReplayer::RunNextEvent() {
+  if (!has_next_event_)
+    return false;
+  switch (next_event_.type()) {
+    case audioproc::Event::INIT:
+      OnInitEvent(next_event_.init());
+      break;
+    case audioproc::Event::STREAM:
+      OnStreamEvent(next_event_.stream());
+      break;
+    case audioproc::Event::REVERSE_STREAM:
+      OnReverseStreamEvent(next_event_.reverse_stream());
+      break;
+    case audioproc::Event::CONFIG:
+      OnConfigEvent(next_event_.config());
+      break;
+    case audioproc::Event::UNKNOWN_EVENT:
+      // We do not expect receive UNKNOWN event.

[hlundin-webrtc, 2016/03/16 13:30:07]

expect to receive

+      return false;
+  }
+  LoadNextMessage();
+  return true;
+}
+
+const ChannelBuffer<float>* DebugDumpReplayer::GetOutput() const {
+  return output_.get();
+}
+
+StreamConfig DebugDumpReplayer::GetOutputConfig() const {
+  return output_config_;
+}
+
+// OnInitEvent reset the input/output/reserve channel format.
+void DebugDumpReplayer::OnInitEvent(const audioproc::Init& msg) {
+  RTC_CHECK(msg.has_num_input_channels());
+  RTC_CHECK(msg.has_output_sample_rate());
+  RTC_CHECK(msg.has_num_output_channels());
+  RTC_CHECK(msg.has_reverse_sample_rate());
+  RTC_CHECK(msg.has_num_reverse_channels());
+
+  input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
+  output_config_ =
+      StreamConfig(msg.output_sample_rate(), msg.num_output_channels());
+  reverse_config_ =
+      StreamConfig(msg.reverse_sample_rate(), msg.num_reverse_channels());
+
+  MaybeResetBuffer(&input_, input_config_);
+  MaybeResetBuffer(&output_, output_config_);
+  MaybeResetBuffer(&reverse_, reverse_config_);
+}
+
+// OnStreamEvent replays an input signal and verifies the output.
+void DebugDumpReplayer::OnStreamEvent(const audioproc::Stream& msg) {
+  // APM should have been created.
+  RTC_CHECK(apm_.get());
+
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->gain_control()->set_stream_analog_level(msg.level()));
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->set_stream_delay_ms(msg.delay()));
+
+  apm_->echo_cancellation()->set_stream_drift_samples(msg.drift());
+  if (msg.has_keypress()) {
+    apm_->set_stream_key_pressed(msg.keypress());
+  } else {
+    apm_->set_stream_key_pressed(true);
+  }
+
+  RTC_CHECK_EQ(input_config_.num_channels(),
+               static_cast<size_t>(msg.input_channel_size()));
+  RTC_CHECK_EQ(input_config_.num_frames() * sizeof(float),
+               msg.input_channel(0).size());
+
+  for (int i = 0; i < msg.input_channel_size(); ++i) {
+    memcpy(input_->channels()[i], msg.input_channel(i).data(),
+           msg.input_channel(i).size());
+  }
+
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->ProcessStream(input_->channels(), input_config_,
+                                   output_config_, output_->channels()));
+}
+
+void DebugDumpReplayer::OnReverseStreamEvent(
+    const audioproc::ReverseStream& msg) {
+  // APM should have been created.
+  RTC_CHECK(apm_.get());
+
+  RTC_CHECK_GT(msg.channel_size(), 0);
+  RTC_CHECK_EQ(reverse_config_.num_channels(),
+               static_cast<size_t>(msg.channel_size()));
+  RTC_CHECK_EQ(reverse_config_.num_frames() * sizeof(float),
+               msg.channel(0).size());
+
+  for (int i = 0; i < msg.channel_size(); ++i) {
+    memcpy(reverse_->channels()[i], msg.channel(i).data(),
+           msg.channel(i).size());
+  }
+
+  RTC_CHECK_EQ(
+      AudioProcessing::kNoError,
+      apm_->ProcessReverseStream(reverse_->channels(), reverse_config_,
+                                 reverse_config_, reverse_->channels()));
+}
+
+void DebugDumpReplayer::OnConfigEvent(const audioproc::Config& msg) {
+  MaybeRecreateApm(msg);
+  ConfigureApm(msg);
+}
+
+void DebugDumpReplayer::MaybeRecreateApm(const audioproc::Config& msg) {
+  // These configurations cannot be changed on the fly.
+  Config config;
+  RTC_CHECK(msg.has_aec_delay_agnostic_enabled());
+  config.Set<DelayAgnostic>(
+      new DelayAgnostic(msg.aec_delay_agnostic_enabled()));
+
+  RTC_CHECK(msg.has_noise_robust_agc_enabled());
+  config.Set<ExperimentalAgc>(
+      new ExperimentalAgc(msg.noise_robust_agc_enabled()));
+
+  RTC_CHECK(msg.has_transient_suppression_enabled());
+  config.Set<ExperimentalNs>(
+      new ExperimentalNs(msg.transient_suppression_enabled()));
+
+  RTC_CHECK(msg.has_aec_extended_filter_enabled());
+  config.Set<ExtendedFilter>(
+      new ExtendedFilter(msg.aec_extended_filter_enabled()));
+
+  // We only create APM once, since changes on these fields should not
+  // happen in current implementation.
+  if (!apm_.get()) {
+    apm_.reset(AudioProcessing::Create(config));
+  }
+}
+
+void DebugDumpReplayer::ConfigureApm(const audioproc::Config& msg) {
+  // AEC configs.
+  RTC_CHECK(msg.has_aec_enabled());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->echo_cancellation()->Enable(msg.aec_enabled()));
+
+  RTC_CHECK(msg.has_aec_drift_compensation_enabled());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->echo_cancellation()->enable_drift_compensation(
+                   msg.aec_drift_compensation_enabled()));
+
+  RTC_CHECK(msg.has_aec_suppression_level());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->echo_cancellation()->set_suppression_level(
+                   static_cast<EchoCancellation::SuppressionLevel>(
+                       msg.aec_suppression_level())));
+
+  // AECM configs.
+  RTC_CHECK(msg.has_aecm_enabled());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->echo_control_mobile()->Enable(msg.aecm_enabled()));
+
+  RTC_CHECK(msg.has_aecm_comfort_noise_enabled());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->echo_control_mobile()->enable_comfort_noise(
+                   msg.aecm_comfort_noise_enabled()));
+
+  RTC_CHECK(msg.has_aecm_routing_mode());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->echo_control_mobile()->set_routing_mode(
+                   static_cast<EchoControlMobile::RoutingMode>(
+                       msg.aecm_routing_mode())));
+
+  // AGC configs.
+  RTC_CHECK(msg.has_agc_enabled());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->gain_control()->Enable(msg.agc_enabled()));
+
+  RTC_CHECK(msg.has_agc_mode());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->gain_control()->set_mode(
+                   static_cast<GainControl::Mode>(msg.agc_mode())));
+
+  RTC_CHECK(msg.has_agc_limiter_enabled());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->gain_control()->enable_limiter(msg.agc_limiter_enabled()));
+
+  // HPF configs.
+  RTC_CHECK(msg.has_hpf_enabled());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->high_pass_filter()->Enable(msg.hpf_enabled()));
+
+  // NS configs.
+  RTC_CHECK(msg.has_ns_enabled());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->noise_suppression()->Enable(msg.ns_enabled()));
+
+  RTC_CHECK(msg.has_ns_level());
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->noise_suppression()->set_level(
+                   static_cast<NoiseSuppression::Level>(msg.ns_level())));
+}
+
+void DebugDumpReplayer::LoadNextMessage() {
+  has_next_event_ =
+      debug_file_ && ReadMessageFromFile(debug_file_, &next_event_);
+}
+
+}  // namespace test
+}  // namespace webrtc