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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
130 lines
4.6 KiB
C++
130 lines
4.6 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/api/audio_codecs/audio_format.h"
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#include "webrtc/common_types.h"
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namespace webrtc {
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SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
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SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
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SdpAudioFormat::SdpAudioFormat(const char* name,
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int clockrate_hz,
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size_t num_channels)
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: name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
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SdpAudioFormat::SdpAudioFormat(const std::string& name,
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int clockrate_hz,
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size_t num_channels)
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: name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
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SdpAudioFormat::SdpAudioFormat(const char* name,
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int clockrate_hz,
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size_t num_channels,
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const Parameters& param)
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: name(name),
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clockrate_hz(clockrate_hz),
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num_channels(num_channels),
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parameters(param) {}
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SdpAudioFormat::SdpAudioFormat(const std::string& name,
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int clockrate_hz,
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size_t num_channels,
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const Parameters& param)
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: name(name),
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clockrate_hz(clockrate_hz),
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num_channels(num_channels),
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parameters(param) {}
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bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
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return STR_CASE_CMP(name.c_str(), o.name.c_str()) == 0 &&
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clockrate_hz == o.clockrate_hz && num_channels == o.num_channels;
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}
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SdpAudioFormat::~SdpAudioFormat() = default;
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SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
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SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
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bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
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return STR_CASE_CMP(a.name.c_str(), b.name.c_str()) == 0 &&
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a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
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a.parameters == b.parameters;
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}
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void swap(SdpAudioFormat& a, SdpAudioFormat& b) {
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using std::swap;
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swap(a.name, b.name);
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swap(a.clockrate_hz, b.clockrate_hz);
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swap(a.num_channels, b.num_channels);
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swap(a.parameters, b.parameters);
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}
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std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) {
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os << "{name: " << saf.name;
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os << ", clockrate_hz: " << saf.clockrate_hz;
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os << ", num_channels: " << saf.num_channels;
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os << ", parameters: {";
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const char* sep = "";
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for (const auto& kv : saf.parameters) {
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os << sep << kv.first << ": " << kv.second;
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sep = ", ";
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}
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os << "}}";
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return os;
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}
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AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
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size_t num_channels,
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int bitrate_bps)
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: AudioCodecInfo(sample_rate_hz,
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num_channels,
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bitrate_bps,
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bitrate_bps,
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bitrate_bps) {}
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AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
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size_t num_channels,
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int default_bitrate_bps,
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int min_bitrate_bps,
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int max_bitrate_bps)
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: sample_rate_hz(sample_rate_hz),
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num_channels(num_channels),
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default_bitrate_bps(default_bitrate_bps),
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min_bitrate_bps(min_bitrate_bps),
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max_bitrate_bps(max_bitrate_bps) {
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RTC_DCHECK_GT(sample_rate_hz, 0);
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RTC_DCHECK_GT(num_channels, 0);
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RTC_DCHECK_GE(min_bitrate_bps, 0);
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RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps);
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RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
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}
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std::ostream& operator<<(std::ostream& os, const AudioCodecInfo& aci) {
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os << "{sample_rate_hz: " << aci.sample_rate_hz;
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os << ", num_channels: " << aci.num_channels;
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os << ", default_bitrate_bps: " << aci.default_bitrate_bps;
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os << ", min_bitrate_bps: " << aci.min_bitrate_bps;
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os << ", max_bitrate_bps: " << aci.max_bitrate_bps;
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os << ", allow_comfort_noise: " << aci.allow_comfort_noise;
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os << ", supports_network_adaption: " << aci.supports_network_adaption;
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os << "}";
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return os;
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}
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std::ostream& operator<<(std::ostream& os, const AudioCodecSpec& acs) {
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os << "{format: " << acs.format;
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os << ", info: " << acs.info;
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os << "}";
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return os;
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}
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} // namespace webrtc
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