| /* |
| * Copyright (c) 2016 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 "webrtc/api/audio_codecs/audio_format.h" |
| |
| #include "webrtc/common_types.h" |
| |
| namespace webrtc { |
| |
| SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default; |
| SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default; |
| |
| SdpAudioFormat::SdpAudioFormat(const char* name, |
| int clockrate_hz, |
| size_t num_channels) |
| : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {} |
| |
| SdpAudioFormat::SdpAudioFormat(const std::string& name, |
| int clockrate_hz, |
| size_t num_channels) |
| : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {} |
| |
| SdpAudioFormat::SdpAudioFormat(const char* name, |
| int clockrate_hz, |
| size_t num_channels, |
| const Parameters& param) |
| : name(name), |
| clockrate_hz(clockrate_hz), |
| num_channels(num_channels), |
| parameters(param) {} |
| |
| SdpAudioFormat::SdpAudioFormat(const std::string& name, |
| int clockrate_hz, |
| size_t num_channels, |
| const Parameters& param) |
| : name(name), |
| clockrate_hz(clockrate_hz), |
| num_channels(num_channels), |
| parameters(param) {} |
| |
| bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const { |
| return STR_CASE_CMP(name.c_str(), o.name.c_str()) == 0 && |
| clockrate_hz == o.clockrate_hz && num_channels == o.num_channels; |
| } |
| |
| SdpAudioFormat::~SdpAudioFormat() = default; |
| SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default; |
| SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default; |
| |
| bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) { |
| return STR_CASE_CMP(a.name.c_str(), b.name.c_str()) == 0 && |
| a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels && |
| a.parameters == b.parameters; |
| } |
| |
| void swap(SdpAudioFormat& a, SdpAudioFormat& b) { |
| using std::swap; |
| swap(a.name, b.name); |
| swap(a.clockrate_hz, b.clockrate_hz); |
| swap(a.num_channels, b.num_channels); |
| 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) |
| : AudioCodecInfo(sample_rate_hz, |
| num_channels, |
| bitrate_bps, |
| bitrate_bps, |
| bitrate_bps) {} |
| |
| AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, |
| size_t num_channels, |
| int default_bitrate_bps, |
| int min_bitrate_bps, |
| int max_bitrate_bps) |
| : sample_rate_hz(sample_rate_hz), |
| num_channels(num_channels), |
| default_bitrate_bps(default_bitrate_bps), |
| min_bitrate_bps(min_bitrate_bps), |
| max_bitrate_bps(max_bitrate_bps) { |
| RTC_DCHECK_GT(sample_rate_hz, 0); |
| RTC_DCHECK_GT(num_channels, 0); |
| RTC_DCHECK_GE(min_bitrate_bps, 0); |
| RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps); |
| 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 |