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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_base.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_base.h | 114 |
1 files changed, 114 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/voip/voip_base.h b/third_party/libwebrtc/api/voip/voip_base.h new file mode 100644 index 0000000000..8df7bd0571 --- /dev/null +++ b/third_party/libwebrtc/api/voip/voip_base.h @@ -0,0 +1,114 @@ +/* + * 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_BASE_H_ +#define API_VOIP_VOIP_BASE_H_ + +#include "absl/base/attributes.h" +#include "absl/types/optional.h" + +namespace webrtc { + +class Transport; + +// VoipBase interface +// +// VoipBase provides a management interface on a media session using a +// concept called 'channel'. A channel represents an interface handle +// for application to request various media session operations. This +// notion of channel is used throughout other interfaces as well. +// +// Underneath the interface, a channel id is mapped into an audio session +// object that is capable of sending and receiving a single RTP stream with +// another media endpoint. It's possible to create and use multiple active +// channels simultaneously which would mean that particular application +// session has RTP streams with multiple remote endpoints. +// +// A typical example for the usage context is outlined in VoipEngine +// header file. + +enum class ChannelId : int {}; + +enum class ABSL_MUST_USE_RESULT VoipResult { + // kOk indicates the function was successfully invoked with no error. + kOk, + // kInvalidArgument indicates the caller specified an invalid argument, such + // as an invalid ChannelId. + kInvalidArgument, + // kFailedPrecondition indicates that the operation was failed due to not + // satisfying prerequisite such as not setting codec type before sending. + kFailedPrecondition, + // kInternal is used to indicate various internal failures that are not the + // caller's fault. Further detail is commented on each function that uses this + // return value. + kInternal, +}; + +class VoipBase { + public: + // Creates a channel. + // Each channel handle maps into one audio media session where each has + // its own separate module for send/receive rtp packet with one peer. + // Caller must set `transport`, webrtc::Transport callback pointer to + // receive rtp/rtcp packets from corresponding media session in VoIP engine. + // VoipEngine framework expects applications to handle network I/O directly + // and injection for incoming RTP from remote endpoint is handled via + // VoipNetwork interface. `local_ssrc` is optional and when local_ssrc is not + // set, some random value will be used by voip engine. + // Returns a ChannelId created for caller to handle subsequent Channel + // operations. + virtual ChannelId CreateChannel(Transport* transport, + absl::optional<uint32_t> local_ssrc) = 0; + + // Releases `channel_id` that no longer has any use. + // Returns following VoipResult; + // kOk - `channel_id` is released. + // kInvalidArgument - `channel_id` is invalid. + // kInternal - Fails to stop audio output device. + virtual VoipResult ReleaseChannel(ChannelId channel_id) = 0; + + // Starts sending on `channel_id`. This starts microphone if not started yet. + // Returns following VoipResult; + // kOk - Channel successfully started to send. + // kInvalidArgument - `channel_id` is invalid. + // kFailedPrecondition - Missing prerequisite on VoipCodec::SetSendCodec. + // kInternal - initialization has failed on selected microphone. + virtual VoipResult StartSend(ChannelId channel_id) = 0; + + // Stops sending on `channel_id`. If this is the last active channel, it will + // stop microphone input from underlying audio platform layer. + // Returns following VoipResult; + // kOk - Channel successfully stopped to send. + // kInvalidArgument - `channel_id` is invalid. + // kInternal - Failed to stop the active microphone device. + virtual VoipResult StopSend(ChannelId channel_id) = 0; + + // Starts playing on speaker device for `channel_id`. + // This will start underlying platform speaker device if not started. + // Returns following VoipResult; + // kOk - Channel successfully started to play out. + // kInvalidArgument - `channel_id` is invalid. + // kFailedPrecondition - Missing prerequisite on VoipCodec::SetReceiveCodecs. + // kInternal - Failed to initializate the selected speaker device. + virtual VoipResult StartPlayout(ChannelId channel_id) = 0; + + // Stops playing on speaker device for `channel_id`. + // Returns following VoipResult; + // kOk - Channel successfully stopped t play out. + // kInvalidArgument - `channel_id` is invalid. + virtual VoipResult StopPlayout(ChannelId channel_id) = 0; + + protected: + virtual ~VoipBase() = default; +}; + +} // namespace webrtc + +#endif // API_VOIP_VOIP_BASE_H_ |