Adding debug dump to audio network adaptor.

webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc

+/*
+ *  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_coding/audio_network_adaptor/debug_dump_writer.h"
+
+#include "webrtc/base/checks.h"
+
+#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
+#else
+#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
+#endif
+#endif
+
+namespace webrtc {
+
+#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+namespace {
+
+using audio_network_adaptor::debug_dump::Event;
+using audio_network_adaptor::debug_dump::NetworkMetrics;
+using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
+
+void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
+  RTC_CHECK(dump_file->is_open());
+  std::string dump_data;
+  event.SerializeToString(&dump_data);
+  int32_t size = event.ByteSize();
+  dump_file->Write(&size, sizeof(size));
+  dump_file->Write(dump_data.data(), dump_data.length());
+}
+
+}  // namespace
+#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+
+class DebugDumpWriterImpl final : public DebugDumpWriter {
+ public:
+  explicit DebugDumpWriterImpl(FILE* file_handle);
+  ~DebugDumpWriterImpl() override;
+
+  void DumpEncoderRuntimeConfig(
+      const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
+      int64_t timestamp) override;
+
+  void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
+                          int64_t timestamp) override;
+
+ private:
+  std::unique_ptr<FileWrapper> dump_file_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(DebugDumpWriterImpl);

[kwiberg-webrtc, 2016/09/22 15:54:16]

You can skip this line. Everyone will be handling your class with base class
pointers, so this line isn't needed (which is good, because it also doesn't do
anything, since it isn't in the .h file where the users can see it).

[minyue-webrtc, 2016/09/22 20:40:31]

Done.

+};
+
+DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle)
+    : dump_file_(FileWrapper::Create()) {
+#ifndef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+  RTC_DCHECK(false);
+#endif
+  dump_file_->OpenFromFileHandle(file_handle);
+  RTC_CHECK(dump_file_->is_open());
+}
+
+DebugDumpWriterImpl::~DebugDumpWriterImpl() = default;

[kwiberg-webrtc, 2016/09/22 15:54:15]

Now that this entire class is in the .cc file, there's no longer any reason to
avoid virtual inline methods. You may still find it more readable to have the
long methods non-inlined, but tiny ones like this don't have to be.

[minyue-webrtc, 2016/09/22 20:40:31]

Done.

+
+void DebugDumpWriterImpl::DumpNetworkMetrics(
+    const Controller::NetworkMetrics& metrics,
+    int64_t timestamp) {
+#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+  Event event;
+  event.set_timestamp(timestamp);
+  event.set_type(Event::NETWORK_METRICS);
+  auto dump_metrics = event.mutable_network_metrics();
+
+  if (metrics.uplink_bandwidth_bps)
+    dump_metrics->set_uplink_bandwidth_bps(*metrics.uplink_bandwidth_bps);
+
+  if (metrics.uplink_packet_loss_fraction) {
+    dump_metrics->set_uplink_packet_loss_fraction(
+        *metrics.uplink_packet_loss_fraction);
+  }
+
+  if (metrics.target_audio_bitrate_bps) {
+    dump_metrics->set_target_audio_bitrate_bps(
+        *metrics.target_audio_bitrate_bps);
+  }
+
+  if (metrics.rtt_ms)
+    dump_metrics->set_rtt_ms(*metrics.rtt_ms);
+
+  DumpEventToFile(event, dump_file_.get());
+#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+}
+
+void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
+    const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
+    int64_t timestamp) {
+#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+  Event event;
+  event.set_timestamp(timestamp);
+  event.set_type(Event::ENCODER_RUNTIME_CONFIG);
+  auto dump_config = event.mutable_encoder_runtime_config();
+
+  if (config.bitrate_bps)
+    dump_config->set_bitrate_bps(*config.bitrate_bps);
+
+  if (config.frame_length_ms)
+    dump_config->set_frame_length_ms(*config.frame_length_ms);
+
+  if (config.uplink_packet_loss_fraction) {
+    dump_config->set_uplink_packet_loss_fraction(
+        *config.uplink_packet_loss_fraction);
+  }
+
+  if (config.enable_fec)
+    dump_config->set_enable_fec(*config.enable_fec);
+
+  if (config.enable_dtx)
+    dump_config->set_enable_dtx(*config.enable_dtx);
+
+  if (config.num_channels)
+    dump_config->set_num_channels(*config.num_channels);
+
+  DumpEventToFile(event, dump_file_.get());
+#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+}
+
+std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) {
+  std::unique_ptr<DebugDumpWriter> writer(new DebugDumpWriterImpl(file_handle));
+  return writer;

[kwiberg-webrtc, 2016/09/22 15:54:15]

return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle));

[minyue-webrtc, 2016/09/22 20:40:31]

Done.

+}
+
+}  // namespace webrtc