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/*
* 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