Revert "Reland "Remove our stream << overloads from non-test build targets.""
This reverts commit d7ee72041f882c023c73e27a7436c626c4e43604.
Reason for revert: Broke downstream build which was using SdpAudioFormat operator<<
Original change's description:
> Reland "Remove our stream << overloads from non-test build targets."
>
> This is a reland of c841d18d257ba8e4ed7d77d105e3c46006bb1e7e
>
> Original change's description:
> > 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}
>
> TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org
>
> Bug: webrtc:8982
> Change-Id: Ibe08c6270e5e693eb661a6ce9e8f074b34ef8123
> Reviewed-on: https://webrtc-review.googlesource.com/71161
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22949}
TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org
Change-Id: I3c2b18ec2877d68a522ecbae7a2955c4eecf36df
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8982
Reviewed-on: https://webrtc-review.googlesource.com/71446
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22963}diff --git a/api/audio_codecs/audio_format.cc b/api/audio_codecs/audio_format.cc
index 9db5ce0..82c166f 100644
--- a/api/audio_codecs/audio_format.cc
+++ b/api/audio_codecs/audio_format.cc
@@ -68,6 +68,20 @@
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)
@@ -94,4 +108,23 @@
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 553ab8f..2a85c6f 100644
--- a/api/audio_codecs/audio_format.h
+++ b/api/audio_codecs/audio_format.h
@@ -12,6 +12,7 @@
#define API_AUDIO_CODECS_AUDIO_FORMAT_H_
#include <map>
+#include <ostream>
#include <string>
#include <utility>
@@ -61,6 +62,7 @@
};
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
@@ -119,6 +121,8 @@
// 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 {
@@ -132,6 +136,8 @@
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 55ac15e..5a8ba03 100644
--- a/api/rtcerror.cc
+++ b/api/rtcerror.cc
@@ -93,6 +93,10 @@
}
}
+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 c87ce91..d7dec29 100644
--- a/api/rtcerror.h
+++ b/api/rtcerror.h
@@ -11,9 +11,7 @@
#ifndef API_RTCERROR_H_
#define API_RTCERROR_H_
-#ifdef UNIT_TEST
#include <ostream>
-#endif // UNIT_TEST
#include <string>
#include <utility> // For std::move.
@@ -145,15 +143,9 @@
// error type.
//
// Only intended to be used for logging/disagnostics.
-std::string ToString(RTCErrorType error);
+std::ostream& operator<<(std::ostream& stream, 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
+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
diff --git a/api/rtptransceiverinterface.h b/api/rtptransceiverinterface.h
index 7d2a1df..7805579 100644
--- a/api/rtptransceiverinterface.h
+++ b/api/rtptransceiverinterface.h
@@ -30,6 +30,9 @@
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
diff --git a/common_types.h b/common_types.h
index 580f83a..ae47811 100644
--- a/common_types.h
+++ b/common_types.h
@@ -223,6 +223,16 @@
}
bool operator!=(const CodecInst& other) const { return !(*this == other); }
+
+ friend std::ostream& operator<<(std::ostream& os, const CodecInst& ci) {
+ os << "{pltype: " << ci.pltype;
+ os << ", plname: " << ci.plname;
+ os << ", plfreq: " << ci.plfreq;
+ os << ", pacsize: " << ci.pacsize;
+ os << ", channels: " << ci.channels;
+ os << ", rate: " << ci.rate << "}";
+ return os;
+ }
};
// RTP
diff --git a/media/base/codec.h b/media/base/codec.h
index 7bbb7db..30844c7 100644
--- a/media/base/codec.h
+++ b/media/base/codec.h
@@ -149,6 +149,28 @@
}
};
+inline std::ostream& operator<<(std::ostream& os, const AudioCodec& ac) {
+ os << "{id: " << ac.id;
+ os << ", name: " << ac.name;
+ os << ", clockrate: " << ac.clockrate;
+ os << ", bitrate: " << ac.bitrate;
+ os << ", channels: " << ac.channels;
+ os << ", params: {";
+ const char* sep = "";
+ for (const auto& kv : ac.params) {
+ os << sep << kv.first << ": " << kv.second;
+ sep = ", ";
+ }
+ os << "}, feedback_params: {";
+ sep = "";
+ for (const FeedbackParam& fp : ac.feedback_params.params()) {
+ os << sep << fp.id() << ": " << fp.param();
+ sep = ", ";
+ }
+ os << "}}";
+ return os;
+}
+
struct VideoCodec : public Codec {
// Creates a codec with the given parameters.
VideoCodec(int id, const std::string& name);
diff --git a/modules/video_coding/codecs/h264/include/h264_globals.h b/modules/video_coding/codecs/h264/include/h264_globals.h
index bfeba67..365dbca 100644
--- a/modules/video_coding/codecs/h264/include/h264_globals.h
+++ b/modules/video_coding/codecs/h264/include/h264_globals.h
@@ -53,6 +53,11 @@
return "";
}
+inline std::ostream& operator<<(std::ostream& stream,
+ H264PacketizationMode mode) {
+ return stream << ToString(mode);
+}
+
struct NaluInfo {
uint8_t type;
int sps_id;
diff --git a/pc/rtpmediautils.h b/pc/rtpmediautils.h
index 5f02d89..6de6f8f 100644
--- a/pc/rtpmediautils.h
+++ b/pc/rtpmediautils.h
@@ -44,14 +44,6 @@
// Returns an unspecified string representation of the given direction.
const char* RtpTransceiverDirectionToString(RtpTransceiverDirection direction);
-#ifdef UNIT_TEST
-inline std::ostream& operator<<( // no-presubmit-check TODO(webrtc:8982)
- std::ostream& os, // no-presubmit-check TODO(webrtc:8982)
- RtpTransceiverDirection direction) {
- return os << RtpTransceiverDirectionToString(direction);
-}
-#endif // UNIT_TEST
-
} // namespace webrtc
#endif // PC_RTPMEDIAUTILS_H_
diff --git a/pc/rtptransceiver.cc b/pc/rtptransceiver.cc
index 37770b5..41ee487 100644
--- a/pc/rtptransceiver.cc
+++ b/pc/rtptransceiver.cc
@@ -16,6 +16,10 @@
namespace webrtc {
+std::ostream& operator<<(std::ostream& os, RtpTransceiverDirection direction) {
+ return os << RtpTransceiverDirectionToString(direction);
+}
+
RtpTransceiver::RtpTransceiver(cricket::MediaType media_type)
: unified_plan_(false), media_type_(media_type) {
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
diff --git a/rtc_base/ipaddress.cc b/rtc_base/ipaddress.cc
index e6a02c4..3f7698a 100644
--- a/rtc_base/ipaddress.cc
+++ b/rtc_base/ipaddress.cc
@@ -120,6 +120,11 @@
return false;
}
+std::ostream& operator<<(std::ostream& os, const IPAddress& ip) {
+ os << ip.ToString();
+ return os;
+}
+
in6_addr IPAddress::ipv6_address() const {
return u_.ip6;
}
@@ -211,6 +216,10 @@
return *this;
}
+std::ostream& operator<<(std::ostream& os, const InterfaceAddress& ip) {
+ return os << ip.ToString();
+}
+
std::string InterfaceAddress::ToString() const {
std::string result = IPAddress::ToString();
diff --git a/rtc_base/ipaddress.h b/rtc_base/ipaddress.h
index c965cf1..aa8bb1a 100644
--- a/rtc_base/ipaddress.h
+++ b/rtc_base/ipaddress.h
@@ -83,13 +83,7 @@
bool operator!=(const IPAddress& other) const;
bool operator <(const IPAddress& other) const;
bool operator >(const IPAddress& other) const;
-
-#ifdef UNIT_TEST
- inline std::ostream& operator<<( // no-presubmit-check TODO(webrtc:8982)
- std::ostream& os) { // no-presubmit-check TODO(webrtc:8982)
- return os << ToString();
- }
-#endif // UNIT_TEST
+ friend std::ostream& operator<<(std::ostream& os, const IPAddress& addr);
int family() const { return family_; }
in_addr ipv4_address() const;
@@ -147,6 +141,8 @@
bool operator!=(const InterfaceAddress& other) const;
int ipv6_flags() const { return ipv6_flags_; }
+ friend std::ostream& operator<<(std::ostream& os,
+ const InterfaceAddress& addr);
std::string ToString() const;
diff --git a/rtc_base/socketaddress.cc b/rtc_base/socketaddress.cc
index a58c001..bddb7bc 100644
--- a/rtc_base/socketaddress.cc
+++ b/rtc_base/socketaddress.cc
@@ -199,6 +199,11 @@
return true;
}
+std::ostream& operator<<(std::ostream& os, const SocketAddress& addr) {
+ os << addr.HostAsURIString() << ":" << addr.port();
+ return os;
+}
+
bool SocketAddress::IsAnyIP() const {
return IPIsAny(ip_);
}
diff --git a/rtc_base/socketaddress.h b/rtc_base/socketaddress.h
index 90919c2..d58eed8 100644
--- a/rtc_base/socketaddress.h
+++ b/rtc_base/socketaddress.h
@@ -13,9 +13,6 @@
#include <iosfwd>
#include <string>
-#ifdef UNIT_TEST
-#include <ostream> // no-presubmit-check TODO(webrtc:8982)
-#endif // UNIT_TEST
#include <vector>
#include "rtc_base/basictypes.h"
#include "rtc_base/ipaddress.h"
@@ -129,12 +126,7 @@
// Parses hostname:port and [hostname]:port.
bool FromString(const std::string& str);
-#ifdef UNIT_TEST
- inline std::ostream& operator<<( // no-presubmit-check TODO(webrtc:8982)
- std::ostream& os) { // no-presubmit-check TODO(webrtc:8982)
- return os << HostAsURIString() << ":" << port();
- }
-#endif // UNIT_TEST
+ friend std::ostream& operator<<(std::ostream& os, const SocketAddress& addr);
// Determines whether this represents a missing / any IP address.
// That is, 0.0.0.0 or ::.