commit e44f24e1992330ffbdf71950a1a31f61183a408b
author Evan Shrubsole <eshr@google.com> Tue Feb 09 16:47:52 2021
committer Commit Bot <commit-bot@chromium.org> Wed Feb 10 10:53:29 2021
tree 9d40feceb0cea08d371d45cc9b67f658c057043c
parent 7bad75b3906ae78b67b2a8cec095d877deb58215

Handle longer AudioSendStream::Config strings

Switch to using StringBuilder which suports a variable sized
buffer.

Bug: webrtc:12455
Change-Id: I956d2385e6a26ce6fbb73869506d9d79de786a2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206473
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33215}

call/audio_send_stream.cc

diff --git a/call/audio_send_stream.cc b/call/audio_send_stream.cc
index 76480f2..9d25b77 100644
--- a/call/audio_send_stream.cc
+++ b/call/audio_send_stream.cc
@@ -27,8 +27,7 @@
 AudioSendStream::Config::~Config() = default;
 
 std::string AudioSendStream::Config::ToString() const {
-  char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
+  rtc::StringBuilder ss;
   ss << "{rtp: " << rtp.ToString();
   ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms;
   ss << ", send_transport: " << (send_transport ? "(Transport)" : "null");
@@ -39,8 +38,8 @@
   ss << ", has_dscp: " << (has_dscp ? "true" : "false");
   ss << ", send_codec_spec: "
      << (send_codec_spec ? send_codec_spec->ToString() : "<unset>");
-  ss << '}';
-  return ss.str();
+  ss << "}";
+  return ss.Release();
 }
 
 AudioSendStream::Config::Rtp::Rtp() = default;