From 0a713b63eded9b8b21339e3c9c4654b69760be00 Mon Sep 17 00:00:00 2001 From: Jonas Olsson Date: Wed, 4 Apr 2018 15:49:32 +0200 Subject: replace stringstream in call/ Bug: webrtc:8982 Change-Id: Ib4149bd421afa9018dcd76c60d0a6acfc3b764ff Reviewed-on: https://webrtc-review.googlesource.com/64881 Commit-Queue: Oleh Prypin Reviewed-by: Fredrik Solenberg Cr-Commit-Position: refs/heads/master@{#22737} --- call/audio_send_stream.cc | 10 +++++++--- call/call.cc | 4 +++- call/flexfec_receive_stream_impl.cc | 7 +++++-- call/rampup_tests.cc | 9 +++------ call/rtp_config.cc | 9 +++++---- call/video_config.cc | 8 +++++--- call/video_receive_stream.cc | 13 +++++++++---- call/video_send_stream.cc | 22 +++++++++++++++------- 8 files changed, 52 insertions(+), 30 deletions(-) (limited to 'call') diff --git a/call/audio_send_stream.cc b/call/audio_send_stream.cc index 1c08c2eeed..e2cf20cf84 100644 --- a/call/audio_send_stream.cc +++ b/call/audio_send_stream.cc @@ -11,6 +11,7 @@ #include "call/audio_send_stream.h" #include "rtc_base/stringencode.h" #include "rtc_base/strings/audio_format_to_string.h" +#include "rtc_base/strings/string_builder.h" namespace webrtc { @@ -23,7 +24,8 @@ AudioSendStream::Config::Config(Transport* send_transport) AudioSendStream::Config::~Config() = default; std::string AudioSendStream::Config::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{rtp: " << rtp.ToString(); ss << ", send_transport: " << (send_transport ? "(Transport)" : "null"); ss << ", min_bitrate_bps: " << min_bitrate_bps; @@ -39,7 +41,8 @@ AudioSendStream::Config::Rtp::Rtp() = default; AudioSendStream::Config::Rtp::~Rtp() = default; std::string AudioSendStream::Config::Rtp::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{ssrc: " << ssrc; ss << ", extensions: ["; for (size_t i = 0; i < extensions.size(); ++i) { @@ -62,7 +65,8 @@ AudioSendStream::Config::SendCodecSpec::SendCodecSpec( AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default; std::string AudioSendStream::Config::SendCodecSpec::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{nack_enabled: " << (nack_enabled ? "true" : "false"); ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false"); ss << ", cng_payload_type: " diff --git a/call/call.cc b/call/call.cc index 54784eb2ce..aee43aa7a1 100644 --- a/call/call.cc +++ b/call/call.cc @@ -55,6 +55,7 @@ #include "rtc_base/rate_limiter.h" #include "rtc_base/sequenced_task_checker.h" #include "rtc_base/synchronization/rw_lock_wrapper.h" +#include "rtc_base/strings/string_builder.h" #include "rtc_base/task_queue.h" #include "rtc_base/thread_annotations.h" #include "rtc_base/trace_event.h" @@ -379,7 +380,8 @@ class Call : public webrtc::Call, } // namespace internal std::string Call::Stats::ToString(int64_t time_ms) const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "Call stats: " << time_ms << ", {"; ss << "send_bw_bps: " << send_bandwidth_bps << ", "; ss << "recv_bw_bps: " << recv_bandwidth_bps << ", "; diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc index 038c1f6a1b..973df66164 100644 --- a/call/flexfec_receive_stream_impl.cc +++ b/call/flexfec_receive_stream_impl.cc @@ -21,19 +21,22 @@ #include "rtc_base/checks.h" #include "rtc_base/location.h" #include "rtc_base/logging.h" +#include "rtc_base/strings/string_builder.h" #include "system_wrappers/include/clock.h" namespace webrtc { std::string FlexfecReceiveStream::Stats::ToString(int64_t time_ms) const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "FlexfecReceiveStream stats: " << time_ms << ", {flexfec_bitrate_bps: " << flexfec_bitrate_bps << "}"; return ss.str(); } std::string FlexfecReceiveStream::Config::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{payload_type: " << payload_type; ss << ", remote_ssrc: " << remote_ssrc; ss << ", local_ssrc: " << local_ssrc; diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc index a93dd7957c..c5920e2f3d 100644 --- a/call/rampup_tests.cc +++ b/call/rampup_tests.cc @@ -13,6 +13,7 @@ #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/platform_thread.h" +#include "rtc_base/stringencode.h" #include "test/encoder_settings.h" #include "test/gtest.h" #include "test/testsupport/perf_test.h" @@ -447,17 +448,13 @@ Call::Config RampUpDownUpTester::GetReceiverCallConfig() { std::string RampUpDownUpTester::GetModifierString() const { std::string str("_"); if (num_video_streams_ > 0) { - std::ostringstream s; - s << num_video_streams_; - str += s.str(); + str += rtc::ToString(num_video_streams_); str += "stream"; str += (num_video_streams_ > 1 ? "s" : ""); str += "_"; } if (num_audio_streams_ > 0) { - std::ostringstream s; - s << num_audio_streams_; - str += s.str(); + str += rtc::ToString(num_audio_streams_); str += "stream"; str += (num_audio_streams_ > 1 ? "s" : ""); str += "_"; diff --git a/call/rtp_config.cc b/call/rtp_config.cc index 3621f72890..71322f9940 100644 --- a/call/rtp_config.cc +++ b/call/rtp_config.cc @@ -9,20 +9,21 @@ */ #include "call/rtp_config.h" - -#include +#include "rtc_base/strings/string_builder.h" namespace webrtc { std::string NackConfig::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{rtp_history_ms: " << rtp_history_ms; ss << '}'; return ss.str(); } std::string UlpfecConfig::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::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; diff --git a/call/video_config.cc b/call/video_config.cc index eb32ed583e..3010cdf581 100644 --- a/call/video_config.cc +++ b/call/video_config.cc @@ -10,10 +10,10 @@ #include "call/video_config.h" #include -#include #include #include "rtc_base/checks.h" +#include "rtc_base/strings/string_builder.h" namespace webrtc { VideoStream::VideoStream() @@ -30,7 +30,8 @@ VideoStream::VideoStream(const VideoStream& other) = default; VideoStream::~VideoStream() = default; std::string VideoStream::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{width: " << width; ss << ", height: " << height; ss << ", max_framerate: " << max_framerate; @@ -59,7 +60,8 @@ VideoEncoderConfig::VideoEncoderConfig(VideoEncoderConfig&&) = default; VideoEncoderConfig::~VideoEncoderConfig() = default; std::string VideoEncoderConfig::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{codec_type: "; ss << CodecTypeToPayloadString(codec_type); ss << ", content_type: "; diff --git a/call/video_receive_stream.cc b/call/video_receive_stream.cc index f338805746..b42c9f9dcf 100644 --- a/call/video_receive_stream.cc +++ b/call/video_receive_stream.cc @@ -9,6 +9,7 @@ */ #include "call/video_receive_stream.h" +#include "rtc_base/strings/string_builder.h" namespace webrtc { @@ -17,7 +18,8 @@ VideoReceiveStream::Decoder::Decoder(const Decoder&) = default; VideoReceiveStream::Decoder::~Decoder() = default; std::string VideoReceiveStream::Decoder::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{decoder: " << (decoder ? "(VideoDecoder)" : "nullptr"); ss << ", payload_type: " << payload_type; ss << ", payload_name: " << payload_name; @@ -34,7 +36,8 @@ VideoReceiveStream::Stats::Stats() = default; VideoReceiveStream::Stats::~Stats() = default; std::string VideoReceiveStream::Stats::ToString(int64_t time_ms) const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "VideoReceiveStream stats: " << time_ms << ", {ssrc: " << ssrc << ", "; ss << "total_bps: " << total_bitrate_bps << ", "; ss << "width: " << width << ", "; @@ -71,7 +74,8 @@ VideoReceiveStream::Config& VideoReceiveStream::Config::operator=(Config&&) = VideoReceiveStream::Config::Config::~Config() = default; std::string VideoReceiveStream::Config::ToString() const { - std::stringstream ss; + char buf[4 * 1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{decoders: ["; for (size_t i = 0; i < decoders.size(); ++i) { ss << decoders[i].ToString(); @@ -97,7 +101,8 @@ VideoReceiveStream::Config::Rtp::Rtp(const Rtp&) = default; VideoReceiveStream::Config::Rtp::~Rtp() = default; std::string VideoReceiveStream::Config::Rtp::ToString() const { - std::stringstream ss; + char buf[2 * 1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{remote_ssrc: " << remote_ssrc; ss << ", local_ssrc: " << local_ssrc; ss << ", rtcp_mode: " diff --git a/call/video_send_stream.cc b/call/video_send_stream.cc index 80f538d67f..5bf965c6ba 100644 --- a/call/video_send_stream.cc +++ b/call/video_send_stream.cc @@ -9,6 +9,7 @@ */ #include "call/video_send_stream.h" +#include "rtc_base/strings/string_builder.h" namespace webrtc { @@ -16,7 +17,8 @@ VideoSendStream::StreamStats::StreamStats() = default; VideoSendStream::StreamStats::~StreamStats() = default; std::string VideoSendStream::StreamStats::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "width: " << width << ", "; ss << "height: " << height << ", "; ss << "key: " << frame_counts.key_frames << ", "; @@ -37,7 +39,8 @@ VideoSendStream::Stats::Stats() = default; VideoSendStream::Stats::~Stats() = default; std::string VideoSendStream::Stats::ToString(int64_t time_ms) const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "VideoSendStream stats: " << time_ms << ", {"; ss << "input_fps: " << input_frame_rate << ", "; ss << "encode_fps: " << encode_frame_rate << ", "; @@ -68,7 +71,8 @@ VideoSendStream::Config& VideoSendStream::Config::operator=(Config&&) = default; VideoSendStream::Config::Config::~Config() = default; std::string VideoSendStream::Config::ToString() const { - std::stringstream ss; + char buf[2 * 1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{encoder_settings: " << encoder_settings.ToString(); ss << ", rtp: " << rtp.ToString(); ss << ", rtcp: " << rtcp.ToString(); @@ -85,7 +89,8 @@ std::string VideoSendStream::Config::ToString() const { } std::string VideoSendStream::Config::EncoderSettings::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{payload_name: " << payload_name; ss << ", payload_type: " << payload_type; ss << ", encoder_factory: " @@ -104,7 +109,8 @@ VideoSendStream::Config::Rtp::Flexfec::Flexfec(const Flexfec&) = default; VideoSendStream::Config::Rtp::Flexfec::~Flexfec() = default; std::string VideoSendStream::Config::Rtp::ToString() const { - std::stringstream ss; + char buf[2 * 1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{ssrcs: ["; for (size_t i = 0; i < ssrcs.size(); ++i) { ss << ssrcs[i]; @@ -148,7 +154,8 @@ VideoSendStream::Config::Rtp::Rtx::Rtx(const Rtx&) = default; VideoSendStream::Config::Rtp::Rtx::~Rtx() = default; std::string VideoSendStream::Config::Rtp::Rtx::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{ssrcs: ["; for (size_t i = 0; i < ssrcs.size(); ++i) { ss << ssrcs[i]; @@ -167,7 +174,8 @@ VideoSendStream::Config::Rtcp::Rtcp(const Rtcp&) = default; VideoSendStream::Config::Rtcp::~Rtcp() = default; std::string VideoSendStream::Config::Rtcp::ToString() const { - std::stringstream ss; + char buf[1024]; + rtc::SimpleStringBuilder ss(buf); ss << "{video_report_interval_ms: " << video_report_interval_ms; ss << ", audio_report_interval_ms: " << audio_report_interval_ms; ss << '}'; -- cgit v1.2.3