| /* |
| * 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 "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 |