Remove our stream << overloads from non-test build targets.
Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and
SocketAddress are kept behind gtest's #ifdef UNIT_TEST.
Bug: webrtc:8982
Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70
Reviewed-on: https://webrtc-review.googlesource.com/64143
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22916}diff --git a/api/audio_codecs/audio_format.cc b/api/audio_codecs/audio_format.cc
index 82c166f..9db5ce0 100644
--- a/api/audio_codecs/audio_format.cc
+++ b/api/audio_codecs/audio_format.cc
@@ -68,20 +68,6 @@
swap(a.parameters, b.parameters);
}
-std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) {
- os << "{name: " << saf.name;
- os << ", clockrate_hz: " << saf.clockrate_hz;
- os << ", num_channels: " << saf.num_channels;
- os << ", parameters: {";
- const char* sep = "";
- for (const auto& kv : saf.parameters) {
- os << sep << kv.first << ": " << kv.second;
- sep = ", ";
- }
- os << "}}";
- return os;
-}
-
AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
size_t num_channels,
int bitrate_bps)
@@ -108,23 +94,4 @@
RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
}
-std::ostream& operator<<(std::ostream& os, const AudioCodecInfo& aci) {
- os << "{sample_rate_hz: " << aci.sample_rate_hz;
- os << ", num_channels: " << aci.num_channels;
- os << ", default_bitrate_bps: " << aci.default_bitrate_bps;
- os << ", min_bitrate_bps: " << aci.min_bitrate_bps;
- os << ", max_bitrate_bps: " << aci.max_bitrate_bps;
- os << ", allow_comfort_noise: " << aci.allow_comfort_noise;
- os << ", supports_network_adaption: " << aci.supports_network_adaption;
- os << "}";
- return os;
-}
-
-std::ostream& operator<<(std::ostream& os, const AudioCodecSpec& acs) {
- os << "{format: " << acs.format;
- os << ", info: " << acs.info;
- os << "}";
- return os;
-}
-
} // namespace webrtc
diff --git a/api/audio_codecs/audio_format.h b/api/audio_codecs/audio_format.h
index 2a85c6f..553ab8f 100644
--- a/api/audio_codecs/audio_format.h
+++ b/api/audio_codecs/audio_format.h
@@ -12,7 +12,6 @@
#define API_AUDIO_CODECS_AUDIO_FORMAT_H_
#include <map>
-#include <ostream>
#include <string>
#include <utility>
@@ -62,7 +61,6 @@
};
void swap(SdpAudioFormat& a, SdpAudioFormat& b);
-std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf);
// Information about how an audio format is treated by the codec implementation.
// Contains basic information, such as sample rate and number of channels, which
@@ -121,8 +119,6 @@
// network conditions.
};
-std::ostream& operator<<(std::ostream& os, const AudioCodecInfo& aci);
-
// AudioCodecSpec ties an audio format to specific information about the codec
// and its implementation.
struct AudioCodecSpec {
@@ -136,8 +132,6 @@
AudioCodecInfo info;
};
-std::ostream& operator<<(std::ostream& os, const AudioCodecSpec& acs);
-
} // namespace webrtc
#endif // API_AUDIO_CODECS_AUDIO_FORMAT_H_
diff --git a/api/rtcerror.cc b/api/rtcerror.cc
index 5a8ba03..55ac15e 100644
--- a/api/rtcerror.cc
+++ b/api/rtcerror.cc
@@ -93,10 +93,6 @@
}
}
-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);
diff --git a/api/rtcerror.h b/api/rtcerror.h
index d7dec29..c87ce91 100644
--- a/api/rtcerror.h
+++ b/api/rtcerror.h
@@ -11,7 +11,9 @@
#ifndef API_RTCERROR_H_
#define API_RTCERROR_H_
+#ifdef UNIT_TEST
#include <ostream>
+#endif // UNIT_TEST
#include <string>
#include <utility> // For std::move.
@@ -143,10 +145,16 @@
// error type.
//
// Only intended to be used for logging/disagnostics.
-std::ostream& operator<<(std::ostream& stream, RTCErrorType error);
-
std::string ToString(RTCErrorType error);
+#ifdef UNIT_TEST
+inline std::ostream& operator<<( // no-presubmit-check TODO(webrtc:8982)
+ std::ostream& stream, // no-presubmit-check TODO(webrtc:8982)
+ RTCErrorType error) {
+ return stream << ToString(error);
+}
+#endif // UNIT_TEST
+
// 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.
diff --git a/api/rtptransceiverinterface.h b/api/rtptransceiverinterface.h
index 7805579..7d2a1df 100644
--- a/api/rtptransceiverinterface.h
+++ b/api/rtptransceiverinterface.h
@@ -30,9 +30,6 @@
kInactive
};
-// This is provided as a debugging aid. The format of the output is unspecified.
-std::ostream& operator<<(std::ostream& os, RtpTransceiverDirection direction);
-
// Structure for initializing an RtpTransceiver in a call to
// PeerConnectionInterface::AddTransceiver.
// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit