blob: 46cddfad6c8e00c908f97834ebcc4bd2f18e76d4 [file] [log] [blame]
* Copyright (c) 2020 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 <map>
#include "api/audio_codecs/audio_format.h"
#include "api/voip/voip_base.h"
namespace webrtc {
// VoipCodec interface currently provides any codec related interface
// such as setting encoder and decoder types that are negotiated with
// remote endpoint. Typically after SDP offer and answer exchange,
// the local endpoint understands what are the codec payload types that
// are used with negotiated codecs. This interface is subject to expand
// as needed in future.
// This interface requires a channel id created via VoipBase interface.
class VoipCodec {
// Set encoder type here along with its payload type to use.
// Returns following VoipResult;
// kOk - sending codec is set as provided.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult SetSendCodec(ChannelId channel_id,
int payload_type,
const SdpAudioFormat& encoder_spec) = 0;
// Set decoder payload type here. In typical offer and answer model,
// this should be called after payload type has been agreed in media
// session. Note that payload type can differ with same codec in each
// direction.
// Returns following VoipResult;
// kOk - receiving codecs are set as provided.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult SetReceiveCodecs(
ChannelId channel_id,
const std::map<int, SdpAudioFormat>& decoder_specs) = 0;
virtual ~VoipCodec() = default;
} // namespace webrtc