/* * 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 "api/audio_codecs/audio_format.h" #include "common_types.h" // NOLINT(build/include) 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