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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/voip/voip_codec.h | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/voip/voip_codec.h | 56 |
1 files changed, 56 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/voip/voip_codec.h b/third_party/libwebrtc/api/voip/voip_codec.h new file mode 100644 index 0000000000..46cddfad6c --- /dev/null +++ b/third_party/libwebrtc/api/voip/voip_codec.h @@ -0,0 +1,56 @@ +/* + * 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. + */ + +#ifndef API_VOIP_VOIP_CODEC_H_ +#define API_VOIP_VOIP_CODEC_H_ + +#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 { + public: + // 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; + + protected: + virtual ~VoipCodec() = default; +}; + +} // namespace webrtc + +#endif // API_VOIP_VOIP_CODEC_H_ |