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c0df5fc25b
This is one last CL that includes the rest of VoIP API implementation. Bug: webrtc:11251 Change-Id: I3f1b0bf2fd48be864ffc73482105f9514f75f9e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173860 Commit-Queue: Tim Na <natim@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31168}
50 lines
1.7 KiB
C++
50 lines
1.7 KiB
C++
/*
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* Copyright (c) 2020 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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#ifndef API_VOIP_VOIP_CODEC_H_
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#define API_VOIP_VOIP_CODEC_H_
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#include <map>
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#include "api/audio_codecs/audio_format.h"
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#include "api/voip/voip_base.h"
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namespace webrtc {
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// VoipCodec interface currently provides any codec related interface
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// such as setting encoder and decoder types that are negotiated with
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// remote endpoint. Typically after SDP offer and answer exchange,
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// the local endpoint understands what are the codec payload types that
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// are used with negotiated codecs. This interface is subject to expand
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// as needed in future.
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//
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// This interface requires a channel id created via VoipBase interface.
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class VoipCodec {
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public:
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// Set encoder type here along with its payload type to use.
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virtual void SetSendCodec(ChannelId channel_id,
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int payload_type,
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const SdpAudioFormat& encoder_spec) = 0;
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// Set decoder payload type here. In typical offer and answer model,
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// this should be called after payload type has been agreed in media
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// session. Note that payload type can differ with same codec in each
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// direction.
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virtual void SetReceiveCodecs(
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ChannelId channel_id,
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const std::map<int, SdpAudioFormat>& decoder_specs) = 0;
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protected:
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virtual ~VoipCodec() = default;
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};
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} // namespace webrtc
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#endif // API_VOIP_VOIP_CODEC_H_
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