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Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h | 81 |
1 files changed, 81 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h new file mode 100644 index 0000000000..6d9f0a8613 --- /dev/null +++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h @@ -0,0 +1,81 @@ +/* + * 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 AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_ +#define AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_ + +#include <memory> + +#include "api/frame_transformer_interface.h" +#include "api/sequence_checker.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "rtc_base/buffer.h" +#include "rtc_base/synchronization/mutex.h" +#include "rtc_base/task_queue.h" + +namespace webrtc { + +// Delegates calls to FrameTransformerInterface to transform frames, and to +// ChannelSend to send the transformed frames using `send_frame_callback_` on +// the `encoder_queue_`. +// OnTransformedFrame() can be called from any thread, the delegate ensures +// thread-safe access to the ChannelSend callback. +class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback { + public: + using SendFrameCallback = + std::function<int32_t(AudioFrameType frameType, + uint8_t payloadType, + uint32_t rtp_timestamp, + rtc::ArrayView<const uint8_t> payload, + int64_t absolute_capture_timestamp_ms)>; + ChannelSendFrameTransformerDelegate( + SendFrameCallback send_frame_callback, + rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, + rtc::TaskQueue* encoder_queue); + + // Registers `this` as callback for `frame_transformer_`, to get the + // transformed frames. + void Init(); + + // Unregisters and releases the `frame_transformer_` reference, and resets + // `send_frame_callback_` under lock. Called from ChannelSend destructor to + // prevent running the callback on a dangling channel. + void Reset(); + + // Delegates the call to FrameTransformerInterface::TransformFrame, to + // transform the frame asynchronously. + void Transform(AudioFrameType frame_type, + uint8_t payload_type, + uint32_t rtp_timestamp, + uint32_t rtp_start_timestamp, + const uint8_t* payload_data, + size_t payload_size, + int64_t absolute_capture_timestamp_ms, + uint32_t ssrc); + + // Implements TransformedFrameCallback. Can be called on any thread. + void OnTransformedFrame( + std::unique_ptr<TransformableFrameInterface> frame) override; + + // Delegates the call to ChannelSend::SendRtpAudio on the `encoder_queue_`, + // by calling `send_audio_callback_`. + void SendFrame(std::unique_ptr<TransformableFrameInterface> frame) const; + + protected: + ~ChannelSendFrameTransformerDelegate() override = default; + + private: + mutable Mutex send_lock_; + SendFrameCallback send_frame_callback_ RTC_GUARDED_BY(send_lock_); + rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_; + rtc::TaskQueue* encoder_queue_ RTC_GUARDED_BY(send_lock_); +}; +} // namespace webrtc +#endif // AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_ |