Add ToString() methods to classes with << operators, preparing for deprecations.

Bug: webrtc:8982
Change-Id: I9b8792a229539dd9848f4d9936fe343f4bf9ad49
Reviewed-on: https://webrtc-review.googlesource.com/63200
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22705}

PRESUBMIT.py

@@ -432,7 +432,7 @@ def CheckNoStreamUsageIsAdded(input_api, output_api,
   include_re = input_api.re.compile(r'#include <(i|o|s)stream>')
   usage_re = input_api.re.compile(r'std::(w|i|o|io|wi|wo|wio)(string)*stream')
   no_presubmit_re = input_api.re.compile(
-      r'  // no-presubmit-check TODO\(webrtc:8982\)')
+      r'// no-presubmit-check TODO\(webrtc:8982\)')
   for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
     if f.LocalPath() == 'PRESUBMIT.py':
       continue

api/rtcerror.cc

@@ -94,8 +94,13 @@ void RTCError::set_message(std::string&& message) {
 }
 
 std::ostream& operator<<(std::ostream& stream, RTCErrorType error) {
+  return stream << ToString(error);
+}
+
+// TODO(jonasolsson): Change to use absl::string_view when it's available.
+std::string ToString(RTCErrorType error) {
   int index = static_cast<int>(error);
-  return stream << kRTCErrorTypeNames[index];
+  return std::string(kRTCErrorTypeNames[index]);
 }
 
 }  // namespace webrtc

api/rtcerror.h

@@ -145,6 +145,8 @@ class RTCError {
 // Only intended to be used for logging/disagnostics.
 std::ostream& operator<<(std::ostream& stream, RTCErrorType error);
 
+std::string ToString(RTCErrorType error);
+
 // Helper macro that can be used by implementations to create an error with a
 // message and log it. |message| should be a string literal or movable
 // std::string.

modules/congestion_controller/network_control/network_units.cc

@@ -50,5 +50,4 @@ DataSize operator*(const TimeDelta& duration, const DataRate& rate) {
   return rate * duration;
 }
 
-
 }  // namespace webrtc

modules/video_coding/codecs/h264/include/h264_globals.h

@@ -14,6 +14,8 @@
 #ifndef MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
 #define MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
 
+#include <string>
+
 namespace webrtc {
 
 // The packetization types that we support: single, aggregated, and fragmented.
@@ -40,17 +42,20 @@ enum class H264PacketizationMode {
 // This function is declared inline because it is not clear which
 // .cc file it should belong to.
 // TODO(hta): Refactor. https://bugs.webrtc.org/6842
+// TODO(jonasolsson): Use absl::string_view instead when that's available.
+inline std::string ToString(H264PacketizationMode mode) {
+  if (mode == H264PacketizationMode::NonInterleaved) {
+    return "NonInterleaved";
+  } else if (mode == H264PacketizationMode::SingleNalUnit) {
+    return "SingleNalUnit";
+  }
+  RTC_NOTREACHED();
+  return "";
+}
+
 inline std::ostream& operator<<(std::ostream& stream,
                                 H264PacketizationMode mode) {
-  switch (mode) {
-    case H264PacketizationMode::NonInterleaved:
-      stream << "NonInterleaved";
-      break;
-    case H264PacketizationMode::SingleNalUnit:
-      stream << "SingleNalUnit";
-      break;
-  }
-  return stream;
+  return stream << ToString(mode);
 }
 
 struct NaluInfo {

rtc_base/BUILD.gn

@@ -334,6 +334,17 @@ rtc_source_set("stringutils") {
   ]
 }
 
+rtc_source_set("audio_format_to_string") {
+  sources = [
+    "strings/audio_format_to_string.cc",
+    "strings/audio_format_to_string.h",
+  ]
+  deps = [
+    ":stringutils",
+    "../api/audio_codecs:audio_codecs_api",
+  ]
+}
+
 rtc_source_set("type_traits") {
   sources = [
     "type_traits.h",

rtc_base/ipaddress.cc

@@ -217,12 +217,16 @@ const InterfaceAddress& InterfaceAddress::operator=(
 }
 
 std::ostream& operator<<(std::ostream& os, const InterfaceAddress& ip) {
-  os << static_cast<const IPAddress&>(ip);
+  return os << ip.ToString();
+}
 
-  if (ip.family() == AF_INET6)
-    os << "|flags:0x" << std::hex << ip.ipv6_flags();
+std::string InterfaceAddress::ToString() const {
+  std::string result = IPAddress::ToString();
 
-  return os;
+  if (family() == AF_INET6)
+    result += "|flags:0x" + rtc::ToHex(ipv6_flags());
+
+  return result;
 }
 
 static bool IPIsPrivateNetworkV4(const IPAddress& ip) {

rtc_base/ipaddress.h

@@ -144,6 +144,8 @@ class InterfaceAddress : public IPAddress {
   friend std::ostream& operator<<(std::ostream& os,
                                   const InterfaceAddress& addr);
 
+  std::string ToString() const;
+
  private:
   int ipv6_flags_;
 };

rtc_base/strings/audio_format_to_string.cc

@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_base/strings/audio_format_to_string.h"
+
+#include "rtc_base/strings/string_builder.h"
+
+namespace rtc {
+  std::string ToString(const webrtc::SdpAudioFormat& saf) {
+    char sb_buf[1024];
+    rtc::SimpleStringBuilder sb(sb_buf);
+    sb << "{name: " << saf.name;
+    sb << ", clockrate_hz: " << saf.clockrate_hz;
+    sb << ", num_channels: " << saf.num_channels;
+    sb << ", parameters: {";
+    const char* sep = "";
+    for (const auto& kv : saf.parameters) {
+      sb << sep << kv.first << ": " << kv.second;
+      sep = ", ";
+    }
+    sb << "}}";
+    return sb.str();
+  }
+  std::string ToString(const webrtc::AudioCodecInfo& aci) {
+    char sb_buf[1024];
+    rtc::SimpleStringBuilder sb(sb_buf);
+    sb << "{sample_rate_hz: " << aci.sample_rate_hz;
+    sb << ", num_channels: " << aci.num_channels;
+    sb << ", default_bitrate_bps: " << aci.default_bitrate_bps;
+    sb << ", min_bitrate_bps: " << aci.min_bitrate_bps;
+    sb << ", max_bitrate_bps: " << aci.max_bitrate_bps;
+    sb << ", allow_comfort_noise: " << aci.allow_comfort_noise;
+    sb << ", supports_network_adaption: " << aci.supports_network_adaption;
+    sb << "}";
+    return sb.str();
+  }
+  std::string ToString(const webrtc::AudioCodecSpec& acs) {
+    char sb_buf[1024];
+    rtc::SimpleStringBuilder sb(sb_buf);
+    sb << "{format: " << ToString(acs.format);
+    sb << ", info: " << ToString(acs.info);
+    sb << "}";
+    return sb.str();
+  }
+}  // namespace rtc

rtc_base/strings/audio_format_to_string.h

@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_BASE_STRINGS_AUDIO_FORMAT_TO_STRING_H_
+#define RTC_BASE_STRINGS_AUDIO_FORMAT_TO_STRING_H_
+
+#include <string>
+
+#include "api/audio_codecs/audio_format.h"
+
+namespace rtc {
+  std::string ToString(const webrtc::SdpAudioFormat& saf);
+  std::string ToString(const webrtc::AudioCodecInfo& saf);
+  std::string ToString(const webrtc::AudioCodecSpec& acs);
+}  // namespace rtc
+
+#endif  // RTC_BASE_STRINGS_AUDIO_FORMAT_TO_STRING_H_

rtc_base/stringutils.cc

@@ -7,6 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#include <algorithm>
+#include <cstdio>
 
 #include "rtc_base/stringutils.h"
 #include "rtc_base/checks.h"
@@ -130,4 +132,11 @@ std::string string_trim(const std::string& s) {
   return s.substr(first, last - first + 1);
 }
 
+std::string ToHex(const int i) {
+  char buffer[50];
+  snprintf(buffer, sizeof(buffer), "%x", i);
+
+  return std::string(buffer);
+}
+
 }  // namespace rtc

rtc_base/stringutils.h

@@ -91,9 +91,9 @@ const size_t SIZE_UNKNOWN = static_cast<size_t>(-1);
 template<class CTYPE>
 struct Traits {
   // STL string type
-  //typedef XXX string;
+  // typedef XXX string;
   // Null-terminated string
-  //inline static const CTYPE* empty_str();
+  // inline static const CTYPE* empty_str();
 };
 
 ///////////////////////////////////////////////////////////////////////////////
@@ -119,7 +119,7 @@ const CTYPE* strchr(const CTYPE* str, const CTYPE* chs) {
 
 template<class CTYPE>
 const CTYPE* strchrn(const CTYPE* str, size_t slen, CTYPE ch) {
-  for (size_t i=0; i<slen && str[i]; ++i) {
+  for (size_t i=0; i < slen && str[i]; ++i) {
     if (str[i] == ch) {
       return str + i;
     }
@@ -312,6 +312,7 @@ bool ends_with(const char *s1, const char *s2);
 // Remove leading and trailing whitespaces.
 std::string string_trim(const std::string& s);
 
+std::string ToHex(const int i);
 }  // namespace rtc
 
 #endif  // RTC_BASE_STRINGUTILS_H_

rtc_base/stringutils_unittest.cc

@@ -16,10 +16,10 @@ namespace rtc {
 // Tests for string_match().
 
 TEST(string_matchTest, Matches) {
-  EXPECT_TRUE( string_match("A.B.C.D", "a.b.c.d"));
-  EXPECT_TRUE( string_match("www.TEST.GOOGLE.COM", "www.*.com"));
-  EXPECT_TRUE( string_match("127.0.0.1",  "12*.0.*1"));
-  EXPECT_TRUE( string_match("127.1.0.21", "12*.0.*1"));
+  EXPECT_TRUE(string_match("A.B.C.D", "a.b.c.d"));
+  EXPECT_TRUE(string_match("www.TEST.GOOGLE.COM", "www.*.com"));
+  EXPECT_TRUE(string_match("127.0.0.1",  "12*.0.*1"));
+  EXPECT_TRUE(string_match("127.1.0.21", "12*.0.*1"));
   EXPECT_FALSE(string_match("127.0.0.0",  "12*.0.*1"));
   EXPECT_FALSE(string_match("127.0.0.0",  "12*.0.*1"));
   EXPECT_FALSE(string_match("127.1.1.21", "12*.0.*1"));
@@ -105,4 +105,10 @@ TEST(string_endsTest, EndsWith) {
   EXPECT_FALSE(ends_with("", "f"));
 }
 
+TEST(string_toHexTest, ToHex) {
+  EXPECT_EQ(ToHex(0), "0");
+  EXPECT_EQ(ToHex(0X1243E), "1243e");
+  EXPECT_EQ(ToHex(-20), "ffffffec");
+}
+
 }  // namespace rtc