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
+
+DebugDumpWriterImpl::DebugDumpWriterImpl(std::string ana_dump_file_name)
+    : dump_file_(FileWrapper::Create()) {
+  dump_file_->OpenFile(ana_dump_file_name.c_str(), false);

[minyue-webrtc, 2016/09/21 09:16:30]

Per offline discussion, Michael suggested adding 

#ifndef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
  RTC_DCHECK(false);
#endif

That is, no DebugDumpWriterImpl should be created if protobuf is not allowed.

I agree. This will be added together with other comments, if any.

[kwiberg-webrtc, 2016/09/21 10:54:09]

Not sure I understand how that code would give the effect you say. But I'll
review the code when you upload a new patch set.

[minyue-webrtc, 2016/09/22 15:11:49]

Done.

+  RTC_CHECK(dump_file_->is_open());
+}
+
+DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle)
+    : dump_file_(FileWrapper::Create()) {
+  dump_file_->OpenFromFileHandle(file_handle);
+  RTC_CHECK(dump_file_->is_open());
+}
+
+DebugDumpWriterImpl::~DebugDumpWriterImpl() = default;
+
+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
+}
+
+}  // namespace webrtc