| 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
|
|
|