Index: webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc |
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc |
index 7770e65a269ec8e246d982e647f9ad3bc2d23f23..54398e05c89c4d82831ac481ffeed714f273a2a4 100644 |
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc |
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc |
@@ -105,34 +105,38 @@ void DebugDumpWriterImpl::DumpEncoderRuntimeConfig( |
Event event; |
event.set_timestamp(timestamp); |
event.set_type(Event::ENCODER_RUNTIME_CONFIG); |
- auto dump_config = event.mutable_encoder_runtime_config(); |
+ ConvertEncoderConfigToDumpEntry(config, |
+ event.mutable_encoder_runtime_config()); |
+ DumpEventToFile(event, dump_file_.get()); |
+#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP |
+} |
+ |
+std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) { |
+ return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle)); |
+} |
+void DebugDumpWriter::ConvertEncoderConfigToDumpEntry( |
+ const AudioNetworkAdaptor::EncoderRuntimeConfig& config, |
+ audio_network_adaptor::debug_dump::EncoderRuntimeConfig* dump_entry) { |
if (config.bitrate_bps) |
- dump_config->set_bitrate_bps(*config.bitrate_bps); |
+ dump_entry->set_bitrate_bps(*config.bitrate_bps); |
if (config.frame_length_ms) |
- dump_config->set_frame_length_ms(*config.frame_length_ms); |
+ dump_entry->set_frame_length_ms(*config.frame_length_ms); |
if (config.uplink_packet_loss_fraction) { |
- dump_config->set_uplink_packet_loss_fraction( |
+ dump_entry->set_uplink_packet_loss_fraction( |
*config.uplink_packet_loss_fraction); |
} |
if (config.enable_fec) |
- dump_config->set_enable_fec(*config.enable_fec); |
+ dump_entry->set_enable_fec(*config.enable_fec); |
if (config.enable_dtx) |
- dump_config->set_enable_dtx(*config.enable_dtx); |
+ dump_entry->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) { |
- return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle)); |
+ dump_entry->set_num_channels(*config.num_channels); |
} |
} // namespace webrtc |