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/*
* 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 <utility>
#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
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