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Remove our stream << overloads from non-test build targets.

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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}
This commit is contained in:
Jonas Olsson
2018-04-18 10:11:07 +02:00
committed by Commit Bot
parent b02644f2b8
commit c841d18d25
14 changed files with 34 additions and 107 deletions
Download Patch File Download Diff File Expand all files Collapse all files

33
api/audio_codecs/audio_format.cc
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@ -68,20 +68,6 @@ void swap(SdpAudioFormat& a, SdpAudioFormat& b) {
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 @@ AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
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