Add aecdump support to audioproc_f.

webrtc/modules/audio_processing/test/audio_file_processor.cc

Index: webrtc/modules/audio_processing/test/audio_file_processor.cc
diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.cc b/webrtc/modules/audio_processing/test/audio_file_processor.cc
new file mode 100644
index 0000000000000000000000000000000000000000..26c25c95338c4be8211fe545cea274b51232f26a
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/audio_file_processor.cc
@@ -0,0 +1,187 @@
+/*
+ *  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 "webrtc/modules/audio_processing/test/audio_file_processor.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
+
+using rtc::scoped_ptr;
+using rtc::CheckedDivExact;
+using std::vector;
+using webrtc::audioproc::Event;
+using webrtc::audioproc::Init;
+using webrtc::audioproc::ReverseStream;
+using webrtc::audioproc::Stream;
+
+namespace webrtc {
+namespace {
+
+// Returns a StreamConfig corresponding to file.
+StreamConfig GetStreamConfig(const WavFile& file) {
+  return StreamConfig(file.sample_rate(), file.num_channels());
+}
+
+// Returns a ChannelBuffer corresponding to file.
+ChannelBuffer<float> GetChannelBuffer(const WavFile& file) {
+  return ChannelBuffer<float>(
+      CheckedDivExact(file.sample_rate(), AudioFileProcessor::kChunksPerSecond),
+      file.num_channels());
+}
+
+}  // namespace
+
+WavFileProcessor::WavFileProcessor(scoped_ptr<AudioProcessing> ap,
+                                   scoped_ptr<WavReader> in_file,
+                                   scoped_ptr<WavWriter> out_file)
+    : ap_(ap.Pass()),
+      in_file_(in_file.Pass()),
+      in_buf_(GetChannelBuffer(*in_file_)),
+      out_buf_(GetChannelBuffer(*out_file)),
+      in_interleaved_(in_buf_.size()),
+      input_config_(GetStreamConfig(*in_file_)),
+      output_config_(GetStreamConfig(*out_file)),
+      buffer_writer_(out_file.Pass()) {}
+
+bool WavFileProcessor::ProcessAndWriteChunk() {
+  if (in_file_->ReadSamples(in_interleaved_.size(), &in_interleaved_[0]) !=
+      in_interleaved_.size()) {
+    return false;
+  }
+
+  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());
+
+  {
+    const auto st = ScopedTimer(mutable_proc_time());
+    RTC_CHECK_EQ(kNoErr,
+                 ap_->ProcessStream(in_buf_.channels(), input_config_,
+                                    output_config_, out_buf_.channels()));
+  }
+
+  buffer_writer_.Write(out_buf_);
+  return true;
+}
+
+AecDumpFileProcessor::AecDumpFileProcessor(scoped_ptr<AudioProcessing> ap,

[peah-webrtc, 2015/10/23 07:42:40]

One more comment on this: Would it be possible to split the AecDumpFileProcessor
into a separate file? In that way it could be reused by other files.

+                                           FILE* dump_file,
+                                           scoped_ptr<WavWriter> out_file)
+    : ap_(ap.Pass()),
+      dump_file_(dump_file),
+      out_buf_(GetChannelBuffer(*out_file)),
+      output_config_(GetStreamConfig(*out_file)),
+      buffer_writer_(out_file.Pass()) {
+  RTC_CHECK(dump_file_) << "Could not open dump file for reading.";
+}
+
+AecDumpFileProcessor::~AecDumpFileProcessor() {
+  fclose(dump_file_);
+}
+
+bool AecDumpFileProcessor::ProcessAndWriteChunk() {
+  Event event_msg;
+
+  // Continue until we process our first Stream message.
+  do {
+    if (!ReadMessageFromFile(dump_file_, &event_msg)) {
+      return false;
+    }
+
+    if (event_msg.type() == Event::INIT) {
+      RTC_CHECK(event_msg.has_init());
+      HandleMessage(event_msg.init());
+
+    } else if (event_msg.type() == Event::STREAM) {
+      RTC_CHECK(event_msg.has_stream());
+      HandleMessage(event_msg.stream());
+      break;
+
+    } else if (event_msg.type() == Event::REVERSE_STREAM) {
+      RTC_CHECK(event_msg.has_reverse_stream());
+      HandleMessage(event_msg.reverse_stream());
+    }
+  } while (event_msg.type() != Event::STREAM);
+
+  return true;
+}
+
+void AecDumpFileProcessor::HandleMessage(const Init& msg) {
+  RTC_CHECK(msg.has_sample_rate());
+  RTC_CHECK(msg.has_num_input_channels());
+  RTC_CHECK(msg.has_num_reverse_channels());
+
+  in_buf_.reset(new ChannelBuffer<float>(
+      CheckedDivExact(msg.sample_rate(), kChunksPerSecond),
+      msg.num_input_channels()));
+  const int reverse_sample_rate = msg.has_reverse_sample_rate()
+                                      ? msg.reverse_sample_rate()
+                                      : msg.sample_rate();
+  reverse_buf_.reset(new ChannelBuffer<float>(
+      CheckedDivExact(reverse_sample_rate, kChunksPerSecond),
+      msg.num_reverse_channels()));
+  input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
+  reverse_config_ =
+      StreamConfig(msg.reverse_sample_rate(), msg.num_reverse_channels());
+
+  const ProcessingConfig config = {
+      {input_config_, output_config_, reverse_config_, reverse_config_}};
+  RTC_CHECK_EQ(kNoErr, ap_->Initialize(config));
+}
+
+void AecDumpFileProcessor::HandleMessage(const Stream& msg) {
+  RTC_CHECK(!msg.has_input_data());
+  RTC_CHECK_EQ(in_buf_->num_channels(), msg.input_channel_size());
+
+  for (int i = 0; i < msg.input_channel_size(); ++i) {
+    RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+                 msg.input_channel(i).size());
+    std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
+                msg.input_channel(i).size());
+  }
+  {
+    const auto st = ScopedTimer(mutable_proc_time());
+    RTC_CHECK_EQ(kNoErr, ap_->set_stream_delay_ms(msg.delay()));
+    ap_->echo_cancellation()->set_stream_drift_samples(msg.drift());
+    if (msg.has_keypress()) {
+      ap_->set_stream_key_pressed(msg.keypress());
+    }
+    RTC_CHECK_EQ(kNoErr,
+                 ap_->ProcessStream(in_buf_->channels(), input_config_,
+                                    output_config_, out_buf_.channels()));
+  }
+
+  buffer_writer_.Write(out_buf_);
+}
+
+void AecDumpFileProcessor::HandleMessage(const ReverseStream& msg) {
+  RTC_CHECK(!msg.has_data());
+  RTC_CHECK_EQ(reverse_buf_->num_channels(), msg.channel_size());
+
+  for (int i = 0; i < msg.channel_size(); ++i) {
+    RTC_CHECK_EQ(reverse_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+                 msg.channel(i).size());
+    std::memcpy(reverse_buf_->channels()[i], msg.channel(i).data(),
+                msg.channel(i).size());
+  }
+  {
+    const auto st = ScopedTimer(mutable_proc_time());
+    // TODO(ajm): This currently discards the processed output, which is needed
+    // for e.g. intelligibility enhancement.
+    RTC_CHECK_EQ(kNoErr, ap_->ProcessReverseStream(
+                             reverse_buf_->channels(), reverse_config_,
+                             reverse_config_, reverse_buf_->channels()));
+  }
+}
+
+}  // namespace webrtc