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-rw-r--r--third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.cc250
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diff --git a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.cc b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.cc
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index 0000000000..2f483f5077
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.cc
@@ -0,0 +1,250 @@
+/*
+ * 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 "modules/audio_processing/test/debug_dump_replayer.h"
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+#include "modules/audio_processing/test/protobuf_utils.h"
+#include "modules/audio_processing/test/runtime_setting_util.h"
+#include "rtc_base/checks.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(absl::string_view filename) {
+ debug_file_ = fopen(std::string(filename).c_str(), "rb");
+ LoadNextMessage();
+ return debug_file_;
+}
+
+// Get next event that has not run.
+absl::optional<audioproc::Event> DebugDumpReplayer::GetNextEvent() const {
+ if (!has_next_event_)
+ return absl::nullopt;
+ else
+ return 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::RUNTIME_SETTING:
+ OnRuntimeSettingEvent(next_event_.runtime_setting());
+ break;
+ case audioproc::Event::UNKNOWN_EVENT:
+ // We do not expect to receive UNKNOWN event.
+ RTC_CHECK_NOTREACHED();
+ }
+ 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());
+
+ if (msg.has_applied_input_volume()) {
+ apm_->set_stream_analog_level(msg.applied_input_volume());
+ }
+ RTC_CHECK_EQ(AudioProcessing::kNoError,
+ apm_->set_stream_delay_ms(msg.delay()));
+
+ 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::OnRuntimeSettingEvent(
+ const audioproc::RuntimeSetting& msg) {
+ RTC_CHECK(apm_.get());
+ ReplayRuntimeSetting(apm_.get(), msg);
+}
+
+void DebugDumpReplayer::MaybeRecreateApm(const audioproc::Config& msg) {
+ // These configurations cannot be changed on the fly.
+ RTC_CHECK(msg.has_aec_delay_agnostic_enabled());
+ RTC_CHECK(msg.has_aec_extended_filter_enabled());
+
+ // We only create APM once, since changes on these fields should not
+ // happen in current implementation.
+ if (!apm_.get()) {
+ apm_ = AudioProcessingBuilderForTesting().Create();
+ }
+}
+
+void DebugDumpReplayer::ConfigureApm(const audioproc::Config& msg) {
+ AudioProcessing::Config apm_config;
+
+ // AEC2/AECM configs.
+ RTC_CHECK(msg.has_aec_enabled());
+ RTC_CHECK(msg.has_aecm_enabled());
+ apm_config.echo_canceller.enabled = msg.aec_enabled() || msg.aecm_enabled();
+ apm_config.echo_canceller.mobile_mode = msg.aecm_enabled();
+
+ // HPF configs.
+ RTC_CHECK(msg.has_hpf_enabled());
+ apm_config.high_pass_filter.enabled = msg.hpf_enabled();
+
+ // Preamp configs.
+ RTC_CHECK(msg.has_pre_amplifier_enabled());
+ apm_config.pre_amplifier.enabled = msg.pre_amplifier_enabled();
+ apm_config.pre_amplifier.fixed_gain_factor =
+ msg.pre_amplifier_fixed_gain_factor();
+
+ // NS configs.
+ RTC_CHECK(msg.has_ns_enabled());
+ RTC_CHECK(msg.has_ns_level());
+ apm_config.noise_suppression.enabled = msg.ns_enabled();
+ apm_config.noise_suppression.level =
+ static_cast<AudioProcessing::Config::NoiseSuppression::Level>(
+ msg.ns_level());
+
+ // TS configs.
+ RTC_CHECK(msg.has_transient_suppression_enabled());
+ apm_config.transient_suppression.enabled =
+ msg.transient_suppression_enabled();
+
+ // AGC configs.
+ RTC_CHECK(msg.has_agc_enabled());
+ RTC_CHECK(msg.has_agc_mode());
+ RTC_CHECK(msg.has_agc_limiter_enabled());
+ apm_config.gain_controller1.enabled = msg.agc_enabled();
+ apm_config.gain_controller1.mode =
+ static_cast<AudioProcessing::Config::GainController1::Mode>(
+ msg.agc_mode());
+ apm_config.gain_controller1.enable_limiter = msg.agc_limiter_enabled();
+ RTC_CHECK(msg.has_noise_robust_agc_enabled());
+ apm_config.gain_controller1.analog_gain_controller.enabled =
+ msg.noise_robust_agc_enabled();
+
+ apm_->ApplyConfig(apm_config);
+}
+
+void DebugDumpReplayer::LoadNextMessage() {
+ has_next_event_ =
+ debug_file_ && ReadMessageFromFile(debug_file_, &next_event_);
+}
+
+} // namespace test
+} // namespace webrtc
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