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Handle longer AudioSendStream::Config strings

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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}
This commit is contained in:
Evan Shrubsole 2021-02-09 17:47:52 +01:00 committed by Commit Bot
parent 7bad75b390
commit e44f24e199
1 changed files with 3 additions and 4 deletions
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7
call/audio_send_stream.cc
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@ -27,8 +27,7 @@ AudioSendStream::Config::Config(Transport* send_transport)
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 @@ std::string AudioSendStream::Config::ToString() const {
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;