chromium/third_party/webrtc/call/audio_send_stream.cc

/*
 *  Copyright (c) 2015 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 "call/audio_send_stream.h"

#include <stddef.h>

#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/strings/string_builder.h"

namespace webrtc {

AudioSendStream::Stats::Stats() = default;
AudioSendStream::Stats::~Stats() = default;

AudioSendStream::Config::Config(Transport* send_transport)
    : … { … }

AudioSendStream::Config::~Config() = default;

std::string AudioSendStream::Config::ToString() const { … }

AudioSendStream::Config::Rtp::Rtp() = default;

AudioSendStream::Config::Rtp::~Rtp() = default;

std::string AudioSendStream::Config::Rtp::ToString() const { … }

AudioSendStream::Config::SendCodecSpec::SendCodecSpec(
    int payload_type,
    const SdpAudioFormat& format)
    : … { … }
AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default;

std::string AudioSendStream::Config::SendCodecSpec::ToString() const { … }

bool AudioSendStream::Config::SendCodecSpec::operator==(
    const AudioSendStream::Config::SendCodecSpec& rhs) const { … }
}  // namespace webrtc