Move string_builder.h to webrtc namespace

Bug: webrtc:42232595
Change-Id: Iad12b11767c3bbaddcf0e87357e8e6037608defb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43926}

api/audio/audio_device_defines.h

@@ -154,7 +154,7 @@ class AudioParameters {
   }
   std::string ToString() const {
     char ss_buf[1024];
-    rtc::SimpleStringBuilder ss(ss_buf);
+    SimpleStringBuilder ss(ss_buf);
     ss << "AudioParameters: ";
     ss << "sample_rate=" << sample_rate() << ", channels=" << channels();
     ss << ", frames_per_buffer=" << frames_per_buffer();

api/audio/audio_processing.cc

@@ -128,7 +128,7 @@ operator==(const AudioProcessing::Config::CaptureLevelAdjustment::
 
 std::string AudioProcessing::Config::ToString() const {
   char buf[2048];
-  rtc::SimpleStringBuilder builder(buf);
+  SimpleStringBuilder builder(buf);
   builder << "AudioProcessing::Config{ "
              "pipeline: { "
              "maximum_internal_processing_rate: "

api/neteq/neteq.cc

@@ -23,7 +23,7 @@ NetEq::Config& NetEq::Config::operator=(Config&&) = default;
 
 std::string NetEq::Config::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "sample_rate_hz=" << sample_rate_hz
      << ", max_packets_in_buffer=" << max_packets_in_buffer
      << ", min_delay_ms=" << min_delay_ms << ", enable_fast_accelerate="

api/rtp_parameters.cc

@@ -128,7 +128,7 @@ RtpParameters::~RtpParameters() = default;
 
 std::string RtpExtension::ToString() const {
   char buf[256];
-  rtc::SimpleStringBuilder sb(buf);
+  SimpleStringBuilder sb(buf);
   sb << "{uri: " << uri;
   sb << ", id: " << id;
   if (encrypt) {

api/test/metrics/stdout_metrics_exporter.cc

@@ -30,7 +30,7 @@ int64_t IntegralPart(double value) {
 
 void AppendWithPrecision(double value,
                          int digits_after_comma,
-                         rtc::StringBuilder& out) {
+                         StringBuilder& out) {
   int64_t multiplier = std::lround(std::pow(10, digits_after_comma));
   int64_t integral_part = IntegralPart(value);
   double decimal_part = std::abs(value) - integral_part;
@@ -78,7 +78,7 @@ bool StdoutMetricsExporter::Export(rtc::ArrayView<const Metric> metrics) {
 }
 
 void StdoutMetricsExporter::PrintMetric(const Metric& metric) {
-  rtc::StringBuilder value_stream;
+  StringBuilder value_stream;
   value_stream << metric.test_case << " / " << metric.name << "= {mean=";
   if (metric.stats.mean.has_value()) {
     AppendWithPrecision(*metric.stats.mean, 8, value_stream);

api/test/pclf/media_configuration.cc

@@ -44,7 +44,7 @@ absl::string_view SpecToString(VideoResolution::Spec spec) {
 }
 
 void AppendResolution(const VideoResolution& resolution,
-                      rtc::StringBuilder& builder) {
+                      StringBuilder& builder) {
   builder << "_" << resolution.width() << "x" << resolution.height() << "_"
           << resolution.fps();
 }
@@ -95,7 +95,7 @@ bool VideoResolution::IsRegular() const {
   return spec_ == Spec::kNone;
 }
 std::string VideoResolution::ToString() const {
-  rtc::StringBuilder out;
+  StringBuilder out;
   out << "{ width=" << width_ << ", height=" << height_ << ", fps=" << fps_
       << ", spec=" << SpecToString(spec_) << " }";
   return out.Release();
@@ -162,7 +162,7 @@ VideoDumpOptions::Y4mVideoFrameWriterFactory(
 std::string VideoDumpOptions::GetInputDumpFileName(
     absl::string_view stream_label,
     const VideoResolution& resolution) const {
-  rtc::StringBuilder file_name;
+  StringBuilder file_name;
   file_name << stream_label;
   AppendResolution(resolution, file_name);
   return test::JoinFilename(output_directory_, file_name.Release());
@@ -181,7 +181,7 @@ std::string VideoDumpOptions::GetOutputDumpFileName(
     absl::string_view stream_label,
     absl::string_view receiver,
     const VideoResolution& resolution) const {
-  rtc::StringBuilder file_name;
+  StringBuilder file_name;
   file_name << stream_label << "_" << receiver;
   AppendResolution(resolution, file_name);
   return test::JoinFilename(output_directory_, file_name.Release());
@@ -199,7 +199,7 @@ std::optional<std::string> VideoDumpOptions::GetOutputFrameIdsDumpFileName(
 }
 
 std::string VideoDumpOptions::ToString() const {
-  rtc::StringBuilder out;
+  StringBuilder out;
   out << "{ output_directory_=" << output_directory_
       << ", sampling_modulo_=" << sampling_modulo_
       << ", export_frame_ids_=" << export_frame_ids_ << " }";
@@ -298,7 +298,7 @@ std::vector<std::string> VideoSubscription::GetSubscribedPeers() const {
 }
 
 std::string VideoSubscription::ToString() const {
-  rtc::StringBuilder out;
+  StringBuilder out;
   out << "{ default_resolution_=[";
   if (default_resolution_.has_value()) {
     out << default_resolution_->ToString();

api/test/videocodec_test_stats.cc

@@ -27,7 +27,7 @@ VideoCodecTestStats::FrameStatistics::FrameStatistics(size_t frame_number,
       spatial_idx(spatial_idx) {}
 
 std::string VideoCodecTestStats::FrameStatistics::ToString() const {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   for (const auto& entry : ToMap()) {
     if (ss.size() > 0) {
       ss << " ";
@@ -66,7 +66,7 @@ std::map<std::string, std::string> VideoCodecTestStats::FrameStatistics::ToMap()
 
 std::string VideoCodecTestStats::VideoStatistics::ToString(
     std::string prefix) const {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   for (const auto& entry : ToMap()) {
     if (ss.size() > 0) {
       ss << "\n";

api/units/data_rate.cc

@@ -19,7 +19,7 @@ namespace webrtc {
 
 std::string ToString(DataRate value) {
   char buf[64];
-  rtc::SimpleStringBuilder sb(buf);
+  SimpleStringBuilder sb(buf);
   if (value.IsPlusInfinity()) {
     sb << "+inf bps";
   } else if (value.IsMinusInfinity()) {

api/units/data_size.cc

@@ -19,7 +19,7 @@ namespace webrtc {
 
 std::string ToString(DataSize value) {
   char buf[64];
-  rtc::SimpleStringBuilder sb(buf);
+  SimpleStringBuilder sb(buf);
   if (value.IsPlusInfinity()) {
     sb << "+inf bytes";
   } else if (value.IsMinusInfinity()) {

api/units/frequency.cc

@@ -17,7 +17,7 @@
 namespace webrtc {
 std::string ToString(Frequency value) {
   char buf[64];
-  rtc::SimpleStringBuilder sb(buf);
+  SimpleStringBuilder sb(buf);
   if (value.IsPlusInfinity()) {
     sb << "+inf Hz";
   } else if (value.IsMinusInfinity()) {

api/units/time_delta.cc

@@ -19,7 +19,7 @@ namespace webrtc {
 
 std::string ToString(TimeDelta value) {
   char buf[64];
-  rtc::SimpleStringBuilder sb(buf);
+  SimpleStringBuilder sb(buf);
   if (value.IsPlusInfinity()) {
     sb << "+inf ms";
   } else if (value.IsMinusInfinity()) {

api/units/timestamp.cc

@@ -18,7 +18,7 @@
 namespace webrtc {
 std::string ToString(Timestamp value) {
   char buf[64];
-  rtc::SimpleStringBuilder sb(buf);
+  SimpleStringBuilder sb(buf);
   if (value.IsPlusInfinity()) {
     sb << "+inf ms";
   } else if (value.IsMinusInfinity()) {

api/video/color_space.cc

@@ -139,7 +139,7 @@ const HdrMetadata* ColorSpace::hdr_metadata() const {
 
 std::string ColorSpace::AsString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{primaries:";
   switch (primaries_) {
     PRINT_ENUM_CASE(PrimaryID, kBT709)

api/video/video_adaptation_counters.cc

@@ -35,7 +35,7 @@ VideoAdaptationCounters VideoAdaptationCounters::operator+(
 }
 
 std::string VideoAdaptationCounters::ToString() const {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "{ res=" << resolution_adaptations << " fps=" << fps_adaptations
      << " }";
   return ss.Release();

api/video/video_bitrate_allocation.cc

@@ -147,7 +147,7 @@ std::string VideoBitrateAllocation::ToString() const {
   // Max string length in practice is 260, but let's have some overhead and
   // round up to nearest power of two.
   char string_buf[512];
-  rtc::SimpleStringBuilder ssb(string_buf);
+  SimpleStringBuilder ssb(string_buf);
 
   ssb << "VideoBitrateAllocation [";
   uint32_t spatial_cumulator = 0;

api/video/video_timing.cc

@@ -89,7 +89,7 @@ std::string TimingFrameInfo::ToString() const {
   }
 
   char buf[1024];
-  rtc::SimpleStringBuilder sb(buf);
+  SimpleStringBuilder sb(buf);
 
   sb << rtp_timestamp << ',' << capture_time_ms << ',' << encode_start_ms << ','
      << encode_finish_ms << ',' << packetization_finish_ms << ','

api/video_codecs/libaom_av1_encoder_factory.cc

@@ -515,7 +515,7 @@ aom_svc_ref_frame_config_t GetSvcRefFrameConfig(
   }
 
   char buf[256];
-  rtc::SimpleStringBuilder sb(buf);
+  SimpleStringBuilder sb(buf);
   sb << " spatial_id=" << settings.spatial_id;
   sb << "  ref_idx=[ ";
   for (auto r : ref_frame_config.ref_idx) {
@@ -618,7 +618,7 @@ aom_svc_params_t GetSvcParams(
   }
 
   char buf[512];
-  rtc::SimpleStringBuilder sb(buf);
+  SimpleStringBuilder sb(buf);
   sb << "GetSvcParams" << " layer bitrates kbps";
   for (int s = 0; s < svc_params.number_spatial_layers; ++s) {
     sb << " S" << s << "=[ ";

api/video_codecs/sdp_video_format.cc

@@ -160,7 +160,7 @@ SdpVideoFormat& SdpVideoFormat::operator=(SdpVideoFormat&&) = default;
 SdpVideoFormat::~SdpVideoFormat() = default;
 
 std::string SdpVideoFormat::ToString() const {
-  rtc::StringBuilder builder;
+  StringBuilder builder;
   builder << "Codec name: " << name << ", parameters: {";
   for (const auto& kv : parameters) {
     builder << " " << kv.first << "=" << kv.second;

api/video_codecs/video_codec.cc

@@ -79,7 +79,7 @@ VideoCodec::VideoCodec()
 
 std::string VideoCodec::ToString() const {
   char string_buf[2048];
-  rtc::SimpleStringBuilder ss(string_buf);
+  SimpleStringBuilder ss(string_buf);
 
   ss << "VideoCodec {" << "type: " << CodecTypeToPayloadString(codecType)
      << ", mode: "

api/video_codecs/video_decoder.cc

@@ -44,7 +44,7 @@ const char* VideoDecoder::ImplementationName() const {
 
 std::string VideoDecoder::DecoderInfo::ToString() const {
   char string_buf[2048];
-  rtc::SimpleStringBuilder oss(string_buf);
+  SimpleStringBuilder oss(string_buf);
 
   oss << "DecoderInfo { "
       << "prefers_late_decoding = " << "implementation_name = '"

api/video_codecs/video_encoder.cc

@@ -120,7 +120,7 @@ VideoEncoder::EncoderInfo::EncoderInfo(const EncoderInfo&) = default;
 VideoEncoder::EncoderInfo::~EncoderInfo() = default;
 
 std::string VideoEncoder::EncoderInfo::ToString() const {
-  rtc::StringBuilder oss;
+  StringBuilder oss;
   oss << "EncoderInfo { "
          "ScalingSettings { ";
   if (scaling_settings.thresholds) {

audio/audio_receive_stream.cc

@@ -49,7 +49,7 @@ namespace webrtc {
 
 std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const {
   char ss_buf[1024];
-  rtc::SimpleStringBuilder ss(ss_buf);
+  SimpleStringBuilder ss(ss_buf);
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", local_ssrc: " << local_ssrc;
   ss << ", nack: " << nack.ToString();
@@ -63,7 +63,7 @@ std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const {
 
 std::string AudioReceiveStreamInterface::Config::ToString() const {
   char ss_buf[1024];
-  rtc::SimpleStringBuilder ss(ss_buf);
+  SimpleStringBuilder ss(ss_buf);
   ss << "{rtp: " << rtp.ToString();
   ss << ", rtcp_send_transport: "
      << (rtcp_send_transport ? "(Transport)" : "null");

audio/channel_receive.cc

@@ -749,7 +749,7 @@ void ChannelReceive::ReceivePacket(const uint8_t* packet,
     // Asynchronously transform the received payload. After the payload is
     // transformed, the delegate will call OnReceivedPayloadData to handle it.
     char buf[1024];
-    rtc::SimpleStringBuilder mime_type(buf);
+    SimpleStringBuilder mime_type(buf);
     auto it = payload_type_map_.find(header.payloadType);
     mime_type << MediaTypeToString(cricket::MEDIA_TYPE_AUDIO) << "/"
               << (it != payload_type_map_.end() ? it->second.name

audio/channel_send.cc

@@ -384,7 +384,7 @@ int32_t ChannelSend::SendData(AudioFrameType frameType,
     // Asynchronously transform the payload before sending it. After the payload
     // is transformed, the delegate will call SendRtpAudio to send it.
     char buf[1024];
-    rtc::SimpleStringBuilder mime_type(buf);
+    SimpleStringBuilder mime_type(buf);
     mime_type << MediaTypeToString(cricket::MEDIA_TYPE_AUDIO) << "/"
               << encoder_format_.name;
     frame_transformer_delegate_->Transform(

audio/utility/channel_mixer_unittest.cc

@@ -163,7 +163,7 @@ TEST_F(ChannelMixerTest, ConstructAllPossibleLayouts) {
         continue;
       }
 
-      rtc::StringBuilder ss;
+      StringBuilder ss;
       ss << "Input Layout: " << input_layout
          << ", Output Layout: " << output_layout;
       SCOPED_TRACE(ss.str());

audio/utility/channel_mixing_matrix_unittest.cc

@@ -43,7 +43,7 @@ TEST(ChannelMixingMatrixTest, ConstructAllPossibleLayouts) {
         continue;
       }
 
-      rtc::StringBuilder ss;
+      StringBuilder ss;
       ss << "Input Layout: " << input_layout
          << ", Output Layout: " << output_layout;
       SCOPED_TRACE(ss.str());

call/adaptation/resource_adaptation_processor.cc

@@ -230,7 +230,7 @@ ResourceAdaptationProcessor::OnResourceUnderuse(
   // How can this stream be adapted up?
   Adaptation adaptation = stream_adapter_->GetAdaptationUp();
   if (adaptation.status() != Adaptation::Status::kValid) {
-    rtc::StringBuilder message;
+    StringBuilder message;
     message << "Not adapting up because VideoStreamAdapter returned "
             << Adaptation::StatusToString(adaptation.status());
     return MitigationResultAndLogMessage(MitigationResult::kRejectedByAdapter,
@@ -251,7 +251,7 @@ ResourceAdaptationProcessor::OnResourceUnderuse(
     // adaptation.
     if (absl::c_find(most_limited_resources, reason_resource) ==
         most_limited_resources.end()) {
-      rtc::StringBuilder message;
+      StringBuilder message;
       message << "Resource \"" << reason_resource->Name()
               << "\" was not the most limited resource.";
       return MitigationResultAndLogMessage(
@@ -263,7 +263,7 @@ ResourceAdaptationProcessor::OnResourceUnderuse(
       // before the adaptation is applied.
       UpdateResourceLimitations(reason_resource, adaptation.restrictions(),
                                 adaptation.counters());
-      rtc::StringBuilder message;
+      StringBuilder message;
       message << "Resource \"" << reason_resource->Name()
               << "\" was not the only most limited resource.";
       return MitigationResultAndLogMessage(
@@ -272,7 +272,7 @@ ResourceAdaptationProcessor::OnResourceUnderuse(
   }
   // Apply adaptation.
   stream_adapter_->ApplyAdaptation(adaptation, reason_resource);
-  rtc::StringBuilder message;
+  StringBuilder message;
   message << "Adapted up successfully. Unfiltered adaptations: "
           << stream_adapter_->adaptation_counters().ToString();
   return MitigationResultAndLogMessage(MitigationResult::kAdaptationApplied,
@@ -293,7 +293,7 @@ ResourceAdaptationProcessor::OnResourceOveruse(
                               restrictions.counters);
   }
   if (adaptation.status() != Adaptation::Status::kValid) {
-    rtc::StringBuilder message;
+    StringBuilder message;
     message << "Not adapting down because VideoStreamAdapter returned "
             << Adaptation::StatusToString(adaptation.status());
     return MitigationResultAndLogMessage(MitigationResult::kRejectedByAdapter,
@@ -303,7 +303,7 @@ ResourceAdaptationProcessor::OnResourceOveruse(
   UpdateResourceLimitations(reason_resource, adaptation.restrictions(),
                             adaptation.counters());
   stream_adapter_->ApplyAdaptation(adaptation, reason_resource);
-  rtc::StringBuilder message;
+  StringBuilder message;
   message << "Adapted down successfully. Unfiltered adaptations: "
           << stream_adapter_->adaptation_counters().ToString();
   return MitigationResultAndLogMessage(MitigationResult::kAdaptationApplied,

call/adaptation/video_source_restrictions.cc

@@ -39,7 +39,7 @@ VideoSourceRestrictions::VideoSourceRestrictions(
 }
 
 std::string VideoSourceRestrictions::ToString() const {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "{";
   if (max_frame_rate_)
     ss << " max_fps=" << max_frame_rate_.value();

call/audio_send_stream.cc

@@ -31,7 +31,7 @@ AudioSendStream::Config::Config(Transport* send_transport)
 AudioSendStream::Config::~Config() = default;
 
 std::string AudioSendStream::Config::ToString() const {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "{rtp: " << rtp.ToString();
   ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms;
   ss << ", send_transport: " << (send_transport ? "(Transport)" : "null");
@@ -52,7 +52,7 @@ AudioSendStream::Config::Rtp::~Rtp() = default;
 
 std::string AudioSendStream::Config::Rtp::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{ssrc: " << ssrc;
   if (!rid.empty()) {
     ss << ", rid: " << rid;
@@ -82,7 +82,7 @@ AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default;
 
 std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
   ss << ", enable_non_sender_rtt: "
      << (enable_non_sender_rtt ? "true" : "false");

call/call.cc

@@ -516,7 +516,7 @@ class Call final : public webrtc::Call,
 
 std::string Call::Stats::ToString(int64_t time_ms) const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "Call stats: " << time_ms << ", {";
   ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
   ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";

call/flexfec_receive_stream_impl.cc

@@ -31,7 +31,7 @@ namespace webrtc {
 
 std::string FlexfecReceiveStream::Config::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{payload_type: " << payload_type;
   ss << ", remote_ssrc: " << rtp.remote_ssrc;
   ss << ", local_ssrc: " << rtp.local_ssrc;

call/rtp_config.cc

@@ -48,7 +48,7 @@ std::string LntfConfig::ToString() const {
 
 std::string NackConfig::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{rtp_history_ms: " << rtp_history_ms;
   ss << '}';
   return ss.str();
@@ -56,7 +56,7 @@ std::string NackConfig::ToString() const {
 
 std::string UlpfecConfig::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
   ss << ", red_payload_type: " << red_payload_type;
   ss << ", red_rtx_payload_type: " << red_rtx_payload_type;
@@ -72,7 +72,7 @@ bool UlpfecConfig::operator==(const UlpfecConfig& other) const {
 
 std::string RtpStreamConfig::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{ssrc: " << ssrc;
   ss << ", rid: " << rid;
   ss << ", payload_name: " << payload_name;
@@ -87,7 +87,7 @@ std::string RtpStreamConfig::ToString() const {
 
 std::string RtpStreamConfig::Rtx::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{ssrc: " << ssrc;
   ss << ", payload_type: " << payload_type;
   ss << '}';
@@ -104,7 +104,7 @@ RtpConfig::Flexfec::~Flexfec() = default;
 
 std::string RtpConfig::ToString() const {
   char buf[2 * 1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{ssrcs: [";
   for (size_t i = 0; i < ssrcs.size(); ++i) {
     ss << ssrcs[i];
@@ -168,7 +168,7 @@ RtpConfig::Rtx::~Rtx() = default;
 
 std::string RtpConfig::Rtx::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{ssrcs: [";
   for (size_t i = 0; i < ssrcs.size(); ++i) {
     ss << ssrcs[i];

call/rtp_demuxer.cc

@@ -70,7 +70,7 @@ bool RtpDemuxerCriteria::operator!=(const RtpDemuxerCriteria& other) const {
 }
 
 std::string RtpDemuxerCriteria::ToString() const {
-  rtc::StringBuilder sb;
+  StringBuilder sb;
   sb << "{mid: " << (mid_.empty() ? "<empty>" : mid_)
      << ", rsid: " << (rsid_.empty() ? "<empty>" : rsid_) << ", ssrcs: [";
 
@@ -90,7 +90,7 @@ std::string RtpDemuxerCriteria::ToString() const {
 
 // static
 std::string RtpDemuxer::DescribePacket(const RtpPacketReceived& packet) {
-  rtc::StringBuilder sb;
+  StringBuilder sb;
   sb << "PT=" << packet.PayloadType() << " SSRC=" << packet.Ssrc();
   std::string mid;
   if (packet.GetExtension<RtpMid>(&mid)) {

call/video_receive_stream.cc

@@ -37,7 +37,7 @@ bool VideoReceiveStreamInterface::Decoder::operator==(
 
 std::string VideoReceiveStreamInterface::Decoder::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{payload_type: " << payload_type;
   ss << ", payload_name: " << video_format.name;
   ss << ", codec_params: {";
@@ -60,7 +60,7 @@ VideoReceiveStreamInterface::Stats::~Stats() = default;
 std::string VideoReceiveStreamInterface::Stats::ToString(
     int64_t time_ms) const {
   char buf[2048];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "VideoReceiveStreamInterface stats: " << time_ms << ", {ssrc: " << ssrc
      << ", ";
   ss << "total_bps: " << total_bitrate_bps << ", ";
@@ -120,7 +120,7 @@ VideoReceiveStreamInterface::Config::Config::~Config() = default;
 
 std::string VideoReceiveStreamInterface::Config::ToString() const {
   char buf[4 * 1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{decoders: [";
   for (size_t i = 0; i < decoders.size(); ++i) {
     ss << decoders[i].ToString();
@@ -144,7 +144,7 @@ VideoReceiveStreamInterface::Config::Rtp::~Rtp() = default;
 
 std::string VideoReceiveStreamInterface::Config::Rtp::ToString() const {
   char buf[2 * 1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", local_ssrc: " << local_ssrc;
   ss << ", rtcp_mode: "

call/video_send_stream.cc

@@ -43,7 +43,7 @@ VideoSendStream::StreamStats::~StreamStats() = default;
 
 std::string VideoSendStream::StreamStats::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "type: " << StreamTypeToString(type);
   if (referenced_media_ssrc.has_value())
     ss << " (for: " << referenced_media_ssrc.value() << ")";
@@ -72,7 +72,7 @@ VideoSendStream::Stats::~Stats() = default;
 
 std::string VideoSendStream::Stats::ToString(int64_t time_ms) const {
   char buf[2048];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "VideoSendStream stats: " << time_ms << ", {";
   ss << "input_fps: " << StringFormat("%.1f", input_frame_rate) << ", ";
   ss << "encode_fps: " << encode_frame_rate << ", ";
@@ -114,7 +114,7 @@ VideoSendStream::Config::Config::~Config() = default;
 
 std::string VideoSendStream::Config::ToString() const {
   char buf[2 * 1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   ss << "{encoder_settings: { experiment_cpu_load_estimator: "
      << (encoder_settings.experiment_cpu_load_estimator ? "on" : "off") << "}}";
   ss << ", rtp: " << rtp.ToString();

common_audio/resampler/resampler_unittest.cc

@@ -66,7 +66,7 @@ void ResamplerTest::TearDown() {}
 void ResamplerTest::ResetIfNeededAndPush(int in_rate,
                                          int out_rate,
                                          int num_channels) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Input rate: " << in_rate << ", output rate: " << out_rate
      << ", channel count: " << num_channels;
   SCOPED_TRACE(ss.str());
@@ -92,7 +92,7 @@ TEST_F(ResamplerTest, Reset) {
   for (size_t i = 0; i < kRatesSize; ++i) {
     for (size_t j = 0; j < kRatesSize; ++j) {
       for (size_t k = 0; k < kNumChannelsSize; ++k) {
-        rtc::StringBuilder ss;
+        StringBuilder ss;
         ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j]
            << ", channels: " << kNumChannels[k];
         SCOPED_TRACE(ss.str());
@@ -111,7 +111,7 @@ TEST_F(ResamplerTest, Mono) {
   const int kChannels = 1;
   for (size_t i = 0; i < kRatesSize; ++i) {
     for (size_t j = 0; j < kRatesSize; ++j) {
-      rtc::StringBuilder ss;
+      StringBuilder ss;
       ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
       SCOPED_TRACE(ss.str());
 
@@ -133,7 +133,7 @@ TEST_F(ResamplerTest, Stereo) {
   const int kChannels = 2;
   for (size_t i = 0; i < kRatesSize; ++i) {
     for (size_t j = 0; j < kRatesSize; ++j) {
-      rtc::StringBuilder ss;
+      StringBuilder ss;
       ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
       SCOPED_TRACE(ss.str());
 

common_audio/signal_processing/signal_processing_unittest.cc

@@ -128,7 +128,7 @@ TEST(SplTest, AddSubSatW32) {
           INT32_MIN, std::min<int64_t>(INT32_MAX, static_cast<int64_t>(a) + b));
       const int64_t diff = std::max<int64_t>(
           INT32_MIN, std::min<int64_t>(INT32_MAX, static_cast<int64_t>(a) - b));
-      rtc::StringBuilder ss;
+      webrtc::StringBuilder ss;
       ss << a << " +/- " << b << ": sum " << sum << ", diff " << diff;
       SCOPED_TRACE(ss.str());
       EXPECT_EQ(sum, WebRtcSpl_AddSatW32(a, b));

media/engine/simulcast_encoder_adapter.cc

@@ -1005,7 +1005,7 @@ VideoEncoder::EncoderInfo SimulcastEncoderAdapter::GetEncoderInfo() const {
   }
 
   if (!encoder_names.empty()) {
-    rtc::StringBuilder implementation_name_builder(" (");
+    StringBuilder implementation_name_builder(" (");
     implementation_name_builder << StrJoin(encoder_names, ", ");
     implementation_name_builder << ")";
     encoder_info.implementation_name += implementation_name_builder.Release();

media/sctp/dcsctp_transport.cc

@@ -158,7 +158,7 @@ DcSctpTransport::DcSctpTransport(
           }) {
   RTC_DCHECK_RUN_ON(network_thread_);
   static std::atomic<int> instance_count = 0;
-  rtc::StringBuilder sb;
+  StringBuilder sb;
   sb << debug_name_ << instance_count++;
   debug_name_ = sb.Release();
   ConnectTransportSignals();

modules/audio_coding/neteq/buffer_level_filter_unittest.cc

@@ -31,7 +31,7 @@ TEST(BufferLevelFilter, ConvergenceTest) {
     for (int value = 100; value <= 200; value += 10) {
       filter.Reset();
       filter.SetTargetBufferLevel(20);  // Makes filter coefficient 251/256.
-      rtc::StringBuilder ss;
+      StringBuilder ss;
       ss << "times = " << times << ", value = " << value;
       SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
       for (int i = 0; i < times; ++i) {

modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc

@@ -34,7 +34,7 @@ class DtmfToneGeneratorTest : public ::testing::Test {
     AudioMultiVector signal(channels);
 
     for (int event = 0; event <= 15; ++event) {
-      rtc::StringBuilder ss;
+      StringBuilder ss;
       ss << "Checking event " << event << " at sample rate " << fs_hz;
       SCOPED_TRACE(ss.str());
       const int kAttenuation = 0;
@@ -73,7 +73,7 @@ class DtmfToneGeneratorTest : public ::testing::Test {
       EXPECT_EQ(kNumSamples, tone_gen_.Generate(kNumSamples, &ref_signal));
       // Test every 5 steps (to save time).
       for (int attenuation = 1; attenuation <= 63; attenuation += 5) {
-        rtc::StringBuilder ss;
+        StringBuilder ss;
         ss << "Checking event " << event << " at sample rate " << fs_hz;
         ss << "; attenuation " << attenuation;
         SCOPED_TRACE(ss.str());

modules/audio_coding/neteq/neteq_stereo_unittest.cc

@@ -198,7 +198,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
       EXPECT_EQ(num_channels_, output_multi_channel_.num_channels_);
       EXPECT_EQ(output_size_samples_,
                 output_multi_channel_.samples_per_channel_);
-      rtc::StringBuilder ss;
+      StringBuilder ss;
       ss << "Lap number " << k << ".";
       SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
       // Compare mono and multi-channel.

modules/audio_coding/neteq/neteq_unittest.cc

@@ -261,7 +261,7 @@ TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
   static const int kExpectedOutputLength = 160;  // 10 ms at 16 kHz sample rate.
   const int16_t* const_out_frame_data = out_frame_.data();
   for (int i = 0; i < kExpectedOutputLength; ++i) {
-    rtc::StringBuilder ss;
+    StringBuilder ss;
     ss << "i = " << i;
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     EXPECT_EQ(0, const_out_frame_data[i]);
@@ -283,7 +283,7 @@ TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
       kInitSampleRateHz / 100;  // 10 ms at initial sample rate.
   const int16_t* const_out_frame_data = out_frame_.data();
   for (int i = 0; i < kExpectedOutputLength; ++i) {
-    rtc::StringBuilder ss;
+    StringBuilder ss;
     ss << "i = " << i;
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     EXPECT_EQ(0, const_out_frame_data[i]);
@@ -799,7 +799,7 @@ TEST_F(NetEqDecodingTestTwoInstances, CompareMutedStateOnOff) {
   AudioFrame out_frame1, out_frame2;
   bool muted;
   for (int i = 0; i < 1000; ++i) {
-    rtc::StringBuilder ss;
+    StringBuilder ss;
     ss << "i = " << i;
     SCOPED_TRACE(ss.str());  // Print out the loop iterator on failure.
     EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
@@ -825,7 +825,7 @@ TEST_F(NetEqDecodingTestTwoInstances, CompareMutedStateOnOff) {
   int counter = 0;
   while (out_frame1.speech_type_ != AudioFrame::kNormalSpeech) {
     ASSERT_LT(counter++, 1000) << "Test timed out";
-    rtc::StringBuilder ss;
+    StringBuilder ss;
     ss << "counter = " << counter;
     SCOPED_TRACE(ss.str());  // Print out the loop iterator on failure.
     EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));

modules/audio_coding/neteq/test/neteq_decoding_test.cc

@@ -154,7 +154,7 @@ void NetEqDecodingTest::DecodeAndCompare(
   uint64_t last_concealed_samples = 0;
   uint64_t last_total_samples_received = 0;
   while (packet_) {
-    rtc::StringBuilder ss;
+    StringBuilder ss;
     ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     ASSERT_NO_FATAL_FAILURE(Process());

modules/audio_coding/neteq/tools/neteq_input.cc

@@ -19,7 +19,7 @@ NetEqInput::PacketData::PacketData() = default;
 NetEqInput::PacketData::~PacketData() = default;
 
 std::string NetEqInput::PacketData::ToString() const {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "{"
         "time_ms: "
      << static_cast<int64_t>(time_ms)

modules/audio_coding/neteq/tools/neteq_rtpplay.cc

@@ -285,13 +285,13 @@ std::optional<std::string> CreateOptionalOutputFileName(
   }
   if (!basename.empty()) {
     // Override the automatic assignment.
-    rtc::StringBuilder sb(basename);
+    webrtc::StringBuilder sb(basename);
     sb << suffix;
     return sb.str();
   }
   if (!output_audio_filename.empty()) {
     // Automatically assign name.
-    rtc::StringBuilder sb(output_audio_filename);
+    webrtc::StringBuilder sb(output_audio_filename);
     sb << suffix;
     return sb.str();
   }

modules/audio_coding/neteq/tools/neteq_stats_getter.cc

@@ -23,7 +23,7 @@ namespace test {
 
 std::string NetEqStatsGetter::ConcealmentEvent::ToString() const {
   char ss_buf[256];
-  rtc::SimpleStringBuilder ss(ss_buf);
+  SimpleStringBuilder ss(ss_buf);
   ss << "ConcealmentEvent duration_ms:" << duration_ms
      << " event_number:" << concealment_event_number
      << " time_from_previous_event_end_ms:" << time_from_previous_event_end_ms;

modules/audio_coding/test/Channel.cc

@@ -219,7 +219,7 @@ Channel::Channel(int16_t chID)
   }
   if (chID >= 0) {
     _saveBitStream = true;
-    rtc::StringBuilder ss;
+    StringBuilder ss;
     ss.AppendFormat("bitStream_%d.dat", chID);
     _bitStreamFile = fopen(ss.str().c_str(), "wb");
   } else {

modules/audio_coding/test/EncodeDecodeTest.cc

@@ -136,7 +136,7 @@ void Receiver::Setup(NetEq* neteq,
 
   int playSampFreq;
   std::string file_name;
-  rtc::StringBuilder file_stream;
+  StringBuilder file_stream;
   file_stream << webrtc::test::OutputPath() << out_file_name << file_num
               << ".pcm";
   file_name = file_stream.str();

modules/audio_coding/test/PacketLossTest.cc

@@ -41,7 +41,7 @@ void ReceiverWithPacketLoss::Setup(NetEq* neteq,
   loss_rate_ = loss_rate;
   burst_length_ = burst_length;
   burst_lost_counter_ = burst_length_;  // To prevent first packet gets lost.
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
   Receiver::Setup(neteq, rtpStream, ss.str(), channels, file_num);
 }

modules/audio_coding/test/TestAllCodecs.cc

@@ -373,7 +373,7 @@ void TestAllCodecs::Run(TestPack* channel) {
 
 void TestAllCodecs::OpenOutFile(int test_number) {
   std::string filename = webrtc::test::OutputPath();
-  rtc::StringBuilder test_number_str;
+  StringBuilder test_number_str;
   test_number_str << test_number;
   filename += "testallcodecs_out_";
   filename += test_number_str.str();

modules/audio_coding/test/TestRedFec.cc

@@ -192,7 +192,7 @@ void TestRedFec::Run() {
 
 void TestRedFec::OpenOutFile(int16_t test_number) {
   std::string file_name;
-  rtc::StringBuilder file_stream;
+  StringBuilder file_stream;
   file_stream << webrtc::test::OutputPath();
   file_stream << "TestRedFec_outFile_";
   file_stream << test_number << ".pcm";

modules/audio_coding/test/TestStereo.cc

@@ -603,7 +603,7 @@ void TestStereo::Run(TestPackStereo* channel,
 
 void TestStereo::OpenOutFile(int16_t test_number) {
   std::string file_name;
-  rtc::StringBuilder file_stream;
+  StringBuilder file_stream;
   file_stream << webrtc::test::OutputPath() << "teststereo_out_" << test_number
               << ".pcm";
   file_name = file_stream.str();

modules/audio_coding/test/TestVADDTX.cc

@@ -205,7 +205,7 @@ void TestWebRtcVadDtx::Test(bool new_outfile, bool expect_dtx_enabled) {
   if (new_outfile) {
     output_file_num_++;
   }
-  rtc::StringBuilder out_filename;
+  StringBuilder out_filename;
   out_filename << webrtc::test::OutputPath() << "testWebRtcVadDtx_outFile_"
                << output_file_num_ << ".pcm";
   Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,

modules/audio_mixer/audio_mixer_impl_unittest.cc

@@ -61,7 +61,7 @@ void ResetFrame(AudioFrame* frame) {
 std::string ProduceDebugText(int sample_rate_hz,
                              int number_of_channels,
                              int number_of_sources) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz << " ";
   ss << "Number of channels: " << number_of_channels << " ";
   ss << "Number of sources: " << number_of_sources;

modules/audio_mixer/audio_mixer_test.cc

@@ -78,7 +78,7 @@ class FilePlayingSource : public AudioMixer::Source {
   bool FileHasEnded() const { return file_has_ended_; }
 
   std::string ToString() const {
-    rtc::StringBuilder ss;
+    StringBuilder ss;
     ss << "{rate: " << sample_rate_hz_ << ", channels: " << number_of_channels_
        << ", samples_tot: " << wav_reader_->num_samples() << "}";
     return ss.Release();

modules/audio_mixer/frame_combiner_unittest.cc

@@ -48,7 +48,7 @@ struct FrameCombinerConfig {
 std::string ProduceDebugText(int sample_rate_hz,
                              int number_of_channels,
                              int number_of_sources) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz << " ,";
   ss << "number of channels: " << number_of_channels << " ,";
   ss << "number of sources: " << number_of_sources;
@@ -56,7 +56,7 @@ std::string ProduceDebugText(int sample_rate_hz,
 }
 
 std::string ProduceDebugText(const FrameCombinerConfig& config) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << config.sample_rate_hz << " ,";
   ss << "number of channels: " << config.number_of_channels << " ,";
   ss << "limiter active: " << (config.use_limiter ? "on" : "off") << " ,";

modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc

@@ -44,7 +44,7 @@ namespace aec3 {
 namespace {
 
 std::string ProduceDebugText(size_t num_render_channels, size_t delay) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "delay: " << delay << ", ";
   ss << "num_render_channels:" << num_render_channels;
   return ss.Release();

modules/audio_processing/aec3/alignment_mixer_unittest.cc

@@ -28,7 +28,7 @@ std::string ProduceDebugText(bool initial_silence,
                              bool prefer_first_two_channels,
                              int num_channels,
                              int strongest_ch) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << ", Initial silence: " << initial_silence;
   ss << ", Huge activity threshold: " << huge_activity_threshold;
   ss << ", Prefer first two channels: " << prefer_first_two_channels;

modules/audio_processing/aec3/block_delay_buffer_unittest.cc

@@ -39,7 +39,7 @@ void PopulateInputFrame(size_t frame_length,
 
 std::string ProduceDebugText(int sample_rate_hz, size_t delay) {
   char log_stream_buffer[8 * 1024];
-  rtc::SimpleStringBuilder ss(log_stream_buffer);
+  SimpleStringBuilder ss(log_stream_buffer);
   ss << "Sample rate: " << sample_rate_hz;
   ss << ", Delay: " << delay;
   return ss.str();

modules/audio_processing/aec3/block_framer_unittest.cc

@@ -188,7 +188,7 @@ void RunWronglyInsertOrderTest(int sample_rate_hz,
 #endif
 
 std::string ProduceDebugText(int sample_rate_hz, size_t num_channels) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz;
   ss << ", number of channels: " << num_channels;
   return ss.Release();

modules/audio_processing/aec3/block_processor_unittest.cc

@@ -104,7 +104,7 @@ void RunCaptureNumBandsVerificationTest(const Environment& env,
 #endif
 
 std::string ProduceDebugText(int sample_rate_hz) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz;
   return ss.Release();
 }

modules/audio_processing/aec3/coarse_filter_update_gain_unittest.cc

@@ -118,13 +118,13 @@ void RunFilterUpdateTest(int num_blocks_to_process,
 }
 
 std::string ProduceDebugText(int filter_length_blocks) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Length: " << filter_length_blocks;
   return ss.Release();
 }
 
 std::string ProduceDebugText(size_t delay, int filter_length_blocks) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Delay: " << delay << ", ";
   ss << ProduceDebugText(filter_length_blocks);
   return ss.Release();

modules/audio_processing/aec3/decimator_unittest.cc

@@ -29,7 +29,7 @@ namespace webrtc {
 namespace {
 
 std::string ProduceDebugText(int sample_rate_hz) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz;
   return ss.Release();
 }

modules/audio_processing/aec3/echo_canceller3_unittest.cc

@@ -176,13 +176,13 @@ class RenderTransportVerificationProcessor : public BlockProcessor {
 };
 
 std::string ProduceDebugText(int sample_rate_hz) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz;
   return ss.Release();
 }
 
 std::string ProduceDebugText(int sample_rate_hz, int variant) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz << ", variant: " << variant;
   return ss.Release();
 }

modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc

@@ -26,7 +26,7 @@ namespace webrtc {
 namespace {
 
 std::string ProduceDebugText(size_t delay, size_t down_sampling_factor) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Delay: " << delay;
   ss << ", Down sampling factor: " << down_sampling_factor;
   return ss.Release();

modules/audio_processing/aec3/echo_remover_unittest.cc

@@ -29,13 +29,13 @@
 namespace webrtc {
 namespace {
 std::string ProduceDebugText(int sample_rate_hz) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz;
   return ss.Release();
 }
 
 std::string ProduceDebugText(int sample_rate_hz, int delay) {
-  rtc::StringBuilder ss(ProduceDebugText(sample_rate_hz));
+  StringBuilder ss(ProduceDebugText(sample_rate_hz));
   ss << ", Delay: " << delay;
   return ss.Release();
 }

modules/audio_processing/aec3/erl_estimator_unittest.cc

@@ -18,7 +18,7 @@ namespace webrtc {
 namespace {
 std::string ProduceDebugText(size_t num_render_channels,
                              size_t num_capture_channels) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Render channels: " << num_render_channels;
   ss << ", Capture channels: " << num_capture_channels;
   return ss.Release();

modules/audio_processing/aec3/frame_blocker_unittest.cc

@@ -260,7 +260,7 @@ void RunWrongExtractOrderTest(int sample_rate_hz,
 #endif
 
 std::string ProduceDebugText(int sample_rate_hz, size_t num_channels) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz;
   ss << ", number of channels: " << num_channels;
   return ss.Release();

modules/audio_processing/aec3/matched_filter_unittest.cc

@@ -35,7 +35,7 @@ namespace aec3 {
 namespace {
 
 std::string ProduceDebugText(size_t delay, size_t down_sampling_factor) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Delay: " << delay;
   ss << ", Down sampling factor: " << down_sampling_factor;
   return ss.Release();

modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc

@@ -219,13 +219,13 @@ void RunFilterUpdateTest(const Environment& env,
 }
 
 std::string ProduceDebugText(int filter_length_blocks) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Length: " << filter_length_blocks;
   return ss.Release();
 }
 
 std::string ProduceDebugText(size_t delay, int filter_length_blocks) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Delay: " << delay << ", ";
   ss << ProduceDebugText(filter_length_blocks);
   return ss.Release();

modules/audio_processing/aec3/render_delay_buffer_unittest.cc

@@ -25,7 +25,7 @@ namespace webrtc {
 namespace {
 
 std::string ProduceDebugText(int sample_rate_hz) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz;
   return ss.Release();
 }

modules/audio_processing/aec3/render_delay_controller_unittest.cc

@@ -29,7 +29,7 @@ namespace webrtc {
 namespace {
 
 std::string ProduceDebugText(int sample_rate_hz) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz;
   return ss.Release();
 }
@@ -38,7 +38,7 @@ std::string ProduceDebugText(int sample_rate_hz,
                              size_t delay,
                              size_t num_render_channels,
                              size_t num_capture_channels) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << ProduceDebugText(sample_rate_hz) << ", Delay: " << delay
      << ", Num render channels: " << num_render_channels
      << ", Num capture channels: " << num_capture_channels;

modules/audio_processing/aec3/render_signal_analyzer_unittest.cc

@@ -107,7 +107,7 @@ void RunNarrowBandDetectionTest(size_t num_channels) {
 }
 
 std::string ProduceDebugText(size_t num_channels) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "number of channels: " << num_channels;
   return ss.Release();
 }

modules/audio_processing/aec3/subtractor_unittest.cc

@@ -179,7 +179,7 @@ std::string ProduceDebugText(size_t num_render_channels,
                              size_t num_capture_channels,
                              size_t delay,
                              int filter_length_blocks) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "delay: " << delay << ", ";
   ss << "filter_length_blocks:" << filter_length_blocks << ", ";
   ss << "num_render_channels:" << num_render_channels << ", ";

modules/audio_processing/agc/agc_manager_direct_unittest.cc

@@ -152,7 +152,7 @@ constexpr char kMinMicLevelFieldTrial[] =
 
 std::string GetAgcMinMicLevelExperimentFieldTrial(const std::string& value) {
   char field_trial_buffer[64];
-  rtc::SimpleStringBuilder builder(field_trial_buffer);
+  SimpleStringBuilder builder(field_trial_buffer);
   builder << kMinMicLevelFieldTrial << "/" << value << "/";
   return builder.str();
 }
@@ -163,7 +163,7 @@ std::string GetAgcMinMicLevelExperimentFieldTrialEnabled(
   RTC_DCHECK_GE(enabled_value, 0);
   RTC_DCHECK_LE(enabled_value, 255);
   char field_trial_buffer[64];
-  rtc::SimpleStringBuilder builder(field_trial_buffer);
+  SimpleStringBuilder builder(field_trial_buffer);
   builder << kMinMicLevelFieldTrial << "/Enabled-" << enabled_value << suffix
           << "/";
   return builder.str();

modules/audio_processing/agc2/cpu_features.cc

@@ -18,7 +18,7 @@ namespace webrtc {
 
 std::string AvailableCpuFeatures::ToString() const {
   char buf[64];
-  rtc::SimpleStringBuilder builder(buf);
+  SimpleStringBuilder builder(buf);
   bool first = true;
   if (sse2) {
     builder << (first ? "SSE2" : "_SSE2");

modules/audio_processing/agc2/input_volume_stats_reporter.cc

@@ -52,7 +52,7 @@ constexpr absl::string_view MetricNamePrefix(
 
 metrics::Histogram* CreateVolumeHistogram(InputVolumeType input_volume_type) {
   char buffer[64];
-  rtc::SimpleStringBuilder builder(buffer);
+  SimpleStringBuilder builder(buffer);
   builder << MetricNamePrefix(input_volume_type) << "OnChange";
   return metrics::HistogramFactoryGetCountsLinear(/*name=*/builder.str(),
                                                   /*min=*/1,
@@ -63,7 +63,7 @@ metrics::Histogram* CreateVolumeHistogram(InputVolumeType input_volume_type) {
 metrics::Histogram* CreateRateHistogram(InputVolumeType input_volume_type,
                                         absl::string_view name) {
   char buffer[64];
-  rtc::SimpleStringBuilder builder(buffer);
+  SimpleStringBuilder builder(buffer);
   builder << MetricNamePrefix(input_volume_type) << name;
   return metrics::HistogramFactoryGetCountsLinear(/*name=*/builder.str(),
                                                   /*min=*/1,
@@ -74,7 +74,7 @@ metrics::Histogram* CreateRateHistogram(InputVolumeType input_volume_type,
 metrics::Histogram* CreateAverageHistogram(InputVolumeType input_volume_type,
                                            absl::string_view name) {
   char buffer[64];
-  rtc::SimpleStringBuilder builder(buffer);
+  SimpleStringBuilder builder(buffer);
   builder << MetricNamePrefix(input_volume_type) << name;
   return metrics::HistogramFactoryGetCountsLinear(/*name=*/builder.str(),
                                                   /*min=*/1,

modules/audio_processing/agc2/input_volume_stats_reporter_unittest.cc

@@ -32,37 +32,33 @@ class InputVolumeStatsReporterTest
  protected:
   InputVolumeType InputVolumeType() const { return GetParam(); }
   std::string VolumeLabel() const {
-    return (rtc::StringBuilder(kLabelPrefix) << VolumeTypeLabel() << "OnChange")
+    return (StringBuilder(kLabelPrefix) << VolumeTypeLabel() << "OnChange")
         .str();
   }
   std::string DecreaseRateLabel() const {
-    return (rtc::StringBuilder(kLabelPrefix)
+    return (StringBuilder(kLabelPrefix) << VolumeTypeLabel() << "DecreaseRate")
-            << VolumeTypeLabel() << "DecreaseRate")
         .str();
   }
   std::string DecreaseAverageLabel() const {
-    return (rtc::StringBuilder(kLabelPrefix)
+    return (StringBuilder(kLabelPrefix)
             << VolumeTypeLabel() << "DecreaseAverage")
         .str();
   }
   std::string IncreaseRateLabel() const {
-    return (rtc::StringBuilder(kLabelPrefix)
+    return (StringBuilder(kLabelPrefix) << VolumeTypeLabel() << "IncreaseRate")
-            << VolumeTypeLabel() << "IncreaseRate")
         .str();
   }
   std::string IncreaseAverageLabel() const {
-    return (rtc::StringBuilder(kLabelPrefix)
+    return (StringBuilder(kLabelPrefix)
             << VolumeTypeLabel() << "IncreaseAverage")
         .str();
   }
   std::string UpdateRateLabel() const {
-    return (rtc::StringBuilder(kLabelPrefix)
+    return (StringBuilder(kLabelPrefix) << VolumeTypeLabel() << "UpdateRate")
-            << VolumeTypeLabel() << "UpdateRate")
         .str();
   }
   std::string UpdateAverageLabel() const {
-    return (rtc::StringBuilder(kLabelPrefix)
+    return (StringBuilder(kLabelPrefix) << VolumeTypeLabel() << "UpdateAverage")
-            << VolumeTypeLabel() << "UpdateAverage")
         .str();
   }
 

modules/audio_processing/agc2/interpolated_gain_curve.cc

@@ -34,16 +34,16 @@ InterpolatedGainCurve::InterpolatedGainCurve(
     ApmDataDumper* apm_data_dumper,
     absl::string_view histogram_name_prefix)
     : region_logger_(
-          (rtc::StringBuilder("WebRTC.Audio.")
+          (StringBuilder("WebRTC.Audio.")
            << histogram_name_prefix << ".FixedDigitalGainCurveRegion.Identity")
               .str(),
-          (rtc::StringBuilder("WebRTC.Audio.")
+          (StringBuilder("WebRTC.Audio.")
            << histogram_name_prefix << ".FixedDigitalGainCurveRegion.Knee")
               .str(),
-          (rtc::StringBuilder("WebRTC.Audio.")
+          (StringBuilder("WebRTC.Audio.")
            << histogram_name_prefix << ".FixedDigitalGainCurveRegion.Limiter")
               .str(),
-          (rtc::StringBuilder("WebRTC.Audio.")
+          (StringBuilder("WebRTC.Audio.")
            << histogram_name_prefix
            << ".FixedDigitalGainCurveRegion.Saturation")
               .str()),

modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc

@@ -33,7 +33,7 @@ constexpr float kTestPitchStrengthHigh = 0.75f;
 template <class T>
 std::string PrintTestIndexAndCpuFeatures(
     const ::testing::TestParamInfo<T>& info) {
-  rtc::StringBuilder builder;
+  StringBuilder builder;
   builder << info.index << "_" << info.param.cpu_features.ToString();
   return builder.str();
 }

modules/audio_processing/audio_processing_unittest.cc

@@ -172,7 +172,7 @@ void OpenFileAndWriteMessage(absl::string_view filename,
 }
 
 std::string ResourceFilePath(absl::string_view name, int sample_rate_hz) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   // Resource files are all stereo.
   ss << name << sample_rate_hz / 1000 << "_stereo";
   return test::ResourcePath(ss.str(), "pcm");
@@ -193,7 +193,7 @@ std::string OutputFilePath(absl::string_view name,
                            size_t num_reverse_input_channels,
                            size_t num_reverse_output_channels,
                            StreamDirection file_direction) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << name << "_i" << num_input_channels << "_" << input_rate / 1000 << "_ir"
      << num_reverse_input_channels << "_" << reverse_input_rate / 1000 << "_";
   if (num_output_channels == 1) {
@@ -2341,7 +2341,7 @@ std::string ProduceDebugText(int render_input_sample_rate_hz,
                              size_t render_output_num_channels,
                              size_t capture_input_num_channels,
                              size_t capture_output_num_channels) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rates:"
         "\n Render input: "
      << render_input_sample_rate_hz

modules/audio_processing/ns/noise_suppressor_unittest.cc

@@ -26,7 +26,7 @@ namespace {
 std::string ProduceDebugText(int sample_rate_hz,
                              size_t num_channels,
                              NsConfig::SuppressionLevel level) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "Sample rate: " << sample_rate_hz << ", num_channels: " << num_channels
      << ", level: " << static_cast<int>(level);
   return ss.Release();

modules/audio_processing/test/audio_processing_simulator.cc

@@ -42,7 +42,7 @@ namespace {
 
 std::string GetIndexedOutputWavFilename(absl::string_view wav_name,
                                         int counter) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << wav_name.substr(0, wav_name.size() - 4) << "_" << counter
      << wav_name.substr(wav_name.size() - 4);
   return ss.Release();

modules/audio_processing/test/echo_canceller3_config_json.cc

@@ -430,7 +430,7 @@ void Aec3ConfigFromJsonString(absl::string_view json_string,
 }
 
 std::string Aec3ConfigToJsonString(const EchoCanceller3Config& config) {
-  rtc::StringBuilder ost;
+  StringBuilder ost;
   ost << "{";
   ost << "\"aec3\": {";
   ost << "\"buffering\": {";

modules/audio_processing/test/fake_recording_device_unittest.cc

@@ -98,13 +98,13 @@ void CheckSameSign(const ChannelBuffer<float>* src,
 }
 
 std::string FakeRecordingDeviceKindToString(int fake_rec_device_kind) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "fake recording device: " << fake_rec_device_kind;
   return ss.Release();
 }
 
 std::string AnalogLevelToString(int level) {
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "analog level: " << level;
   return ss.Release();
 }

modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc

@@ -38,7 +38,7 @@ class LossBasedBweV2Test : public ::testing::TestWithParam<bool> {
  protected:
   std::string Config(bool enabled, bool valid) {
     char buffer[1024];
-    rtc::SimpleStringBuilder config_string(buffer);
+    SimpleStringBuilder config_string(buffer);
 
     config_string << "WebRTC-Bwe-LossBasedBweV2/";
 
@@ -79,7 +79,7 @@ class LossBasedBweV2Test : public ::testing::TestWithParam<bool> {
 
   std::string ShortObservationConfig(std::string custom_config) {
     char buffer[1024];
-    rtc::SimpleStringBuilder config_string(buffer);
+    SimpleStringBuilder config_string(buffer);
 
     config_string << "WebRTC-Bwe-LossBasedBweV2/"
                      "MinNumObservations:1,ObservationWindowSize:2,";

modules/remote_bitrate_estimator/tools/rtp_to_text.cc

@@ -47,7 +47,7 @@ int main(int argc, char* argv[]) {
         toffset != 0)
       ++non_zero_ts_offsets;
     if (arrival_time_only) {
-      rtc::StringBuilder ss;
+      webrtc::StringBuilder ss;
       ss << static_cast<int64_t>(packet.time_ms) * 1000000;
       fprintf(stdout, "%s\n", ss.str().c_str());
     } else {

modules/rtp_rtcp/source/rtcp_packet/sdes_unittest.cc

@@ -75,7 +75,7 @@ TEST(RtcpPacketSdesTest, CreateWithTooManyChunks) {
   Sdes sdes;
   for (size_t i = 0; i < kMaxChunks; ++i) {
     uint32_t ssrc = kSenderSsrc + i;
-    rtc::StringBuilder oss;
+    StringBuilder oss;
     oss << "cname" << i;
     EXPECT_TRUE(sdes.AddCName(ssrc, oss.str()));
   }

modules/rtp_rtcp/source/rtp_packet.cc

@@ -693,7 +693,7 @@ bool RtpPacket::RemoveExtension(ExtensionType type) {
 }
 
 std::string RtpPacket::ToString() const {
-  rtc::StringBuilder result;
+  StringBuilder result;
   result << "{payload_type=" << payload_type_ << ", marker=" << marker_
          << ", sequence_number=" << sequence_number_
          << ", padding_size=" << padding_size_ << ", timestamp=" << timestamp_

modules/video_coding/codecs/av1/av1_svc_config.cc

@@ -42,7 +42,7 @@ int GetLimitedNumSpatialLayers(int width, int height) {
 std::optional<ScalabilityMode> BuildScalabilityMode(int num_temporal_layers,
                                                     int num_spatial_layers) {
   char name[20];
-  rtc::SimpleStringBuilder ss(name);
+  SimpleStringBuilder ss(name);
   ss << "L" << num_spatial_layers << "T" << num_temporal_layers;
   if (num_spatial_layers > 1) {
     ss << "_KEY";

modules/video_coding/codecs/test/video_codec_test.cc

@@ -172,7 +172,7 @@ std::string TestName() {
 
 std::string TestOutputPath() {
   std::string output_path =
-      (rtc::StringBuilder() << OutputPath() << TestName()).str();
+      (StringBuilder() << OutputPath() << TestName()).str();
   std::string output_dir = DirName(output_path);
   bool result = CreateDir(output_dir);
   RTC_CHECK(result) << "Cannot create " << output_dir;
@@ -618,7 +618,7 @@ TEST(VideoCodecTest, DISABLED_EncodeDecode) {
   for (int sidx = 0; sidx < num_spatial_layers; ++sidx) {
     for (int tidx = 0; tidx < num_temporal_layers; ++tidx) {
       std::string metric_name_prefix =
-          (rtc::StringBuilder() << "s" << sidx << "t" << tidx << "_").str();
+          (StringBuilder() << "s" << sidx << "t" << tidx << "_").str();
       stream = stats->Aggregate(
           {.layer_id = {{.spatial_idx = sidx, .temporal_idx = tidx}}});
       stream.LogMetrics(GetGlobalMetricsLogger(), TestName(),
@@ -628,10 +628,9 @@ TEST(VideoCodecTest, DISABLED_EncodeDecode) {
   }
 
   if (absl::GetFlag(FLAGS_write_csv)) {
-    stats->LogMetrics(
+    stats->LogMetrics((StringBuilder() << TestOutputPath() << ".csv").str(),
-        (rtc::StringBuilder() << TestOutputPath() << ".csv").str(),
+                      stats->Slice(Filter{}, /*merge=*/false), /*metadata=*/
-        stats->Slice(Filter{}, /*merge=*/false), /*metadata=*/
+                      {{"test_name", TestName()}});
-        {{"test_name", TestName()}});
   }
 }
 

modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc

@@ -120,7 +120,7 @@ void ConfigureSvc(VideoCodec* codec_settings) {
 
 std::string CodecSpecificToString(const VideoCodec& codec) {
   char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  SimpleStringBuilder ss(buf);
   switch (codec.codecType) {
     case kVideoCodecVP8:
       ss << "\nnum_temporal_layers: "
@@ -309,7 +309,7 @@ size_t VideoCodecTestFixtureImpl::Config::NumberOfSimulcastStreams() const {
 
 std::string VideoCodecTestFixtureImpl::Config::ToString() const {
   std::string codec_type = CodecTypeToPayloadString(codec_settings.codecType);
-  rtc::StringBuilder ss;
+  StringBuilder ss;
   ss << "test_name: " << test_name;
   ss << "\nfilename: " << filename;
   ss << "\nnum_frames: " << num_frames;
@@ -563,7 +563,7 @@ void VideoCodecTestFixtureImpl::AnalyzeAllFrames(
 
       // For perf dashboard.
       char modifier_buf[256];
-      rtc::SimpleStringBuilder modifier(modifier_buf);
+      SimpleStringBuilder modifier(modifier_buf);
       modifier << "_r" << rate_profile_idx << "_sl" << layer_stat.spatial_idx;
 
       auto PrintResultHelper = [&modifier, this](

modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc

@@ -128,7 +128,7 @@ std::unique_ptr<ScalableVideoController> CreateVp9ScalabilityStructure(
   }
 
   char name[20];
-  rtc::SimpleStringBuilder ss(name);
+  SimpleStringBuilder ss(name);
   if (codec.mode == VideoCodecMode::kScreensharing) {
     // TODO(bugs.webrtc.org/11999): Compose names of the structures when they
     // are implemented.

modules/video_coding/svc/scalability_structure_unittest.cc

@@ -45,7 +45,7 @@ using ::testing::TestWithParam;
 using ::testing::Values;
 
 std::string FrameDependencyTemplateToString(const FrameDependencyTemplate& t) {
-  rtc::StringBuilder sb;
+  StringBuilder sb;
   sb << "S" << t.spatial_id << "T" << t.temporal_id;
   sb << ": dtis = ";
   for (const auto dtis : t.decode_target_indications) {

modules/video_coding/utility/vp9_uncompressed_header_parser.cc

@@ -252,7 +252,7 @@ const Vp9InterpolationFilter kLiteralToType[4] = {
 
 std::string Vp9UncompressedHeader::ToString() const {
   char buf[1024];
-  rtc::SimpleStringBuilder oss(buf);
+  SimpleStringBuilder oss(buf);
 
   oss << "Vp9UncompressedHeader { " << "profile = " << profile;
 

net/dcsctp/packet/chunk/chunk.cc

@@ -40,7 +40,7 @@ namespace dcsctp {
 template <class Chunk>
 bool ParseAndPrint(uint8_t chunk_type,
                    rtc::ArrayView<const uint8_t> data,
-                   rtc::StringBuilder& sb) {
+                   webrtc::StringBuilder& sb) {
   if (chunk_type == Chunk::kType) {
     std::optional<Chunk> c = Chunk::Parse(data);
     if (c.has_value()) {
@@ -54,7 +54,7 @@ bool ParseAndPrint(uint8_t chunk_type,
 }
 
 std::string DebugConvertChunkToString(rtc::ArrayView<const uint8_t> data) {
-  rtc::StringBuilder sb;
+  webrtc::StringBuilder sb;
 
   if (data.empty()) {
     sb << "Failed to parse chunk due to empty data";

net/dcsctp/packet/chunk/data_chunk.cc

@@ -86,7 +86,7 @@ void DataChunk::SerializeTo(std::vector<uint8_t>& out) const {
 }
 
 std::string DataChunk::ToString() const {
-  rtc::StringBuilder sb;
+  webrtc::StringBuilder sb;
   sb << "DATA, type=" << (options().is_unordered ? "unordered" : "ordered")
      << "::"
      << (*options().is_beginning && *options().is_end ? "complete"

net/dcsctp/packet/chunk/forward_tsn_chunk.cc

@@ -85,7 +85,7 @@ void ForwardTsnChunk::SerializeTo(std::vector<uint8_t>& out) const {
 }
 
 std::string ForwardTsnChunk::ToString() const {
-  rtc::StringBuilder sb;
+  webrtc::StringBuilder sb;
   sb << "FORWARD-TSN, new_cumulative_tsn=" << *new_cumulative_tsn();
   for (const auto& skipped : skipped_streams()) {
     sb << ", skip " << skipped.stream_id.value() << ":" << *skipped.ssn;

net/dcsctp/packet/chunk/idata_chunk.cc

@@ -89,7 +89,7 @@ void IDataChunk::SerializeTo(std::vector<uint8_t>& out) const {
 }
 
 std::string IDataChunk::ToString() const {
-  rtc::StringBuilder sb;
+  webrtc::StringBuilder sb;
   sb << "I-DATA, type=" << (options().is_unordered ? "unordered" : "ordered")
      << "::"
      << (*options().is_beginning && *options().is_end ? "complete"