chromium/third_party/webrtc/media/engine/payload_type_mapper.cc

/*
 *  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 "media/engine/payload_type_mapper.h"

#include <utility>

#include "absl/strings/ascii.h"
#include "api/audio_codecs/audio_format.h"
#include "media/base/codec.h"
#include "media/base/media_constants.h"

namespace cricket {

webrtc::SdpAudioFormat AudioCodecToSdpAudioFormat(const Codec& ac) {}

PayloadTypeMapper::PayloadTypeMapper()
    // RFC 3551 reserves payload type numbers in the range 96-127 exclusively
    // for dynamic assignment. Once those are used up, it is recommended that
    // payload types unassigned by the RFC are used for dynamic payload type
    // mapping, before any static payload ids. At this point, we only support
    // mapping within the exclusive range.
    :{}

PayloadTypeMapper::~PayloadTypeMapper() = default;

absl::optional<int> PayloadTypeMapper::GetMappingFor(
    const webrtc::SdpAudioFormat& format) {}

absl::optional<int> PayloadTypeMapper::FindMappingFor(
    const webrtc::SdpAudioFormat& format) const {}

absl::optional<Codec> PayloadTypeMapper::ToAudioCodec(
    const webrtc::SdpAudioFormat& format) {}

bool PayloadTypeMapper::SdpAudioFormatOrdering::operator()(
    const webrtc::SdpAudioFormat& a,
    const webrtc::SdpAudioFormat& b) const {}

}  // namespace cricket