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

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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}
This commit is contained in:
Taylor Brandstetter
2018-04-20 15:58:11 +00:00
committed by Commit Bot
parent 4049a25afd
commit bd7392829a
14 changed files with 107 additions and 34 deletions
Download Patch File Download Diff File Expand all files Collapse all files

33
api/audio_codecs/audio_format.cc
View File

@ -68,6 +68,20 @@ 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)
@ -94,4 +108,23 @@ 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

6
api/audio_codecs/audio_format.h
View File

@ -12,6 +12,7 @@
#define API_AUDIO_CODECS_AUDIO_FORMAT_H_
#include <map>
#include <ostream>
#include <string>
#include <utility>
@ -61,6 +62,7 @@ struct SdpAudioFormat {
};
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 @@ struct AudioCodecInfo {
// 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 @@ struct AudioCodecSpec {
AudioCodecInfo info;
};
std::ostream& operator<<(std::ostream& os, const AudioCodecSpec& acs);
} // namespace webrtc
#endif // API_AUDIO_CODECS_AUDIO_FORMAT_H_