| /* |
| * 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 "audio/channel_receive_frame_transformer_delegate.h" |
| |
| #include <cstdint> |
| #include <memory> |
| #include <optional> |
| #include <string> |
| #include <utility> |
| |
| #include "api/array_view.h" |
| #include "api/frame_transformer_interface.h" |
| #include "api/rtp_headers.h" |
| #include "api/scoped_refptr.h" |
| #include "api/sequence_checker.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "api/units/timestamp.h" |
| #include "rtc_base/buffer.h" |
| #include "rtc_base/string_encode.h" |
| |
| namespace webrtc { |
| |
| class TransformableIncomingAudioFrame |
| : public TransformableAudioFrameInterface { |
| public: |
| TransformableIncomingAudioFrame(rtc::ArrayView<const uint8_t> payload, |
| const RTPHeader& header, |
| uint32_t ssrc, |
| const std::string& codec_mime_type, |
| Timestamp receive_time) |
| : TransformableAudioFrameInterface(Passkey()), |
| payload_(payload.data(), payload.size()), |
| header_(header), |
| ssrc_(ssrc), |
| codec_mime_type_(codec_mime_type), |
| receive_time_(receive_time) {} |
| ~TransformableIncomingAudioFrame() override = default; |
| rtc::ArrayView<const uint8_t> GetData() const override { return payload_; } |
| |
| void SetData(rtc::ArrayView<const uint8_t> data) override { |
| payload_.SetData(data.data(), data.size()); |
| } |
| |
| void SetRTPTimestamp(uint32_t timestamp) override { |
| header_.timestamp = timestamp; |
| } |
| |
| uint8_t GetPayloadType() const override { return header_.payloadType; } |
| uint32_t GetSsrc() const override { return ssrc_; } |
| uint32_t GetTimestamp() const override { return header_.timestamp; } |
| rtc::ArrayView<const uint32_t> GetContributingSources() const override { |
| return rtc::ArrayView<const uint32_t>(header_.arrOfCSRCs, header_.numCSRCs); |
| } |
| Direction GetDirection() const override { return Direction::kReceiver; } |
| |
| std::string GetMimeType() const override { return codec_mime_type_; } |
| const std::optional<uint16_t> SequenceNumber() const override { |
| return header_.sequenceNumber; |
| } |
| |
| std::optional<uint64_t> AbsoluteCaptureTimestamp() const override { |
| // This could be extracted from received header extensions + extrapolation, |
| // if required in future, eg for being able to re-send received frames. |
| return std::nullopt; |
| } |
| const RTPHeader& Header() const { return header_; } |
| |
| FrameType Type() const override { |
| if (!header_.extension.audio_level()) { |
| // Audio level extension not set. |
| return FrameType::kAudioFrameCN; |
| } |
| return header_.extension.audio_level()->voice_activity() |
| ? FrameType::kAudioFrameSpeech |
| : FrameType::kAudioFrameCN; |
| } |
| |
| std::optional<uint8_t> AudioLevel() const override { |
| if (header_.extension.audio_level()) { |
| return header_.extension.audio_level()->level(); |
| } |
| return std::nullopt; |
| } |
| |
| std::optional<Timestamp> ReceiveTime() const override { |
| return receive_time_ == Timestamp::MinusInfinity() |
| ? std::nullopt |
| : std::optional<Timestamp>(receive_time_); |
| } |
| |
| private: |
| rtc::Buffer payload_; |
| RTPHeader header_; |
| uint32_t ssrc_; |
| std::string codec_mime_type_; |
| Timestamp receive_time_; |
| }; |
| |
| ChannelReceiveFrameTransformerDelegate::ChannelReceiveFrameTransformerDelegate( |
| ReceiveFrameCallback receive_frame_callback, |
| rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, |
| TaskQueueBase* channel_receive_thread) |
| : receive_frame_callback_(receive_frame_callback), |
| frame_transformer_(std::move(frame_transformer)), |
| channel_receive_thread_(channel_receive_thread) {} |
| |
| void ChannelReceiveFrameTransformerDelegate::Init() { |
| RTC_DCHECK_RUN_ON(&sequence_checker_); |
| frame_transformer_->RegisterTransformedFrameCallback( |
| rtc::scoped_refptr<TransformedFrameCallback>(this)); |
| } |
| |
| void ChannelReceiveFrameTransformerDelegate::Reset() { |
| RTC_DCHECK_RUN_ON(&sequence_checker_); |
| frame_transformer_->UnregisterTransformedFrameCallback(); |
| frame_transformer_ = nullptr; |
| receive_frame_callback_ = ReceiveFrameCallback(); |
| } |
| |
| void ChannelReceiveFrameTransformerDelegate::Transform( |
| rtc::ArrayView<const uint8_t> packet, |
| const RTPHeader& header, |
| uint32_t ssrc, |
| const std::string& codec_mime_type, |
| Timestamp receive_time) { |
| RTC_DCHECK_RUN_ON(&sequence_checker_); |
| if (short_circuit_) { |
| receive_frame_callback_(packet, header, receive_time); |
| } else { |
| frame_transformer_->Transform( |
| std::make_unique<TransformableIncomingAudioFrame>( |
| packet, header, ssrc, codec_mime_type, receive_time)); |
| } |
| } |
| |
| void ChannelReceiveFrameTransformerDelegate::OnTransformedFrame( |
| std::unique_ptr<TransformableFrameInterface> frame) { |
| rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate(this); |
| channel_receive_thread_->PostTask( |
| [delegate = std::move(delegate), frame = std::move(frame)]() mutable { |
| delegate->ReceiveFrame(std::move(frame)); |
| }); |
| } |
| |
| void ChannelReceiveFrameTransformerDelegate::StartShortCircuiting() { |
| rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate(this); |
| channel_receive_thread_->PostTask([delegate = std::move(delegate)]() mutable { |
| RTC_DCHECK_RUN_ON(&delegate->sequence_checker_); |
| delegate->short_circuit_ = true; |
| }); |
| } |
| |
| void ChannelReceiveFrameTransformerDelegate::ReceiveFrame( |
| std::unique_ptr<TransformableFrameInterface> frame) const { |
| RTC_DCHECK_RUN_ON(&sequence_checker_); |
| if (!receive_frame_callback_) |
| return; |
| |
| auto* transformed_frame = |
| static_cast<TransformableAudioFrameInterface*>(frame.get()); |
| Timestamp receive_time = |
| transformed_frame->ReceiveTime().value_or(Timestamp::MinusInfinity()); |
| RTPHeader header; |
| if (frame->GetDirection() == |
| TransformableFrameInterface::Direction::kSender) { |
| header.payloadType = transformed_frame->GetPayloadType(); |
| header.timestamp = transformed_frame->GetTimestamp(); |
| header.ssrc = transformed_frame->GetSsrc(); |
| if (transformed_frame->AbsoluteCaptureTimestamp().has_value()) { |
| header.extension.absolute_capture_time = AbsoluteCaptureTime(); |
| header.extension.absolute_capture_time->absolute_capture_timestamp = |
| transformed_frame->AbsoluteCaptureTimestamp().value(); |
| } |
| } else { |
| auto* transformed_incoming_frame = |
| static_cast<TransformableIncomingAudioFrame*>(frame.get()); |
| header = transformed_incoming_frame->Header(); |
| } |
| |
| // TODO(crbug.com/1464860): Take an explicit struct with the required |
| // information rather than the RTPHeader to make it easier to |
| // construct the required information when injecting transformed frames not |
| // originally from this receiver. |
| receive_frame_callback_(frame->GetData(), header, receive_time); |
| } |
| |
| rtc::scoped_refptr<FrameTransformerInterface> |
| ChannelReceiveFrameTransformerDelegate::FrameTransformer() { |
| RTC_DCHECK_RUN_ON(&sequence_checker_); |
| return frame_transformer_; |
| } |
| |
| std::unique_ptr<TransformableAudioFrameInterface> CloneReceiverAudioFrame( |
| TransformableAudioFrameInterface* original) { |
| RTC_CHECK(original->GetDirection() == |
| TransformableFrameInterface::Direction::kReceiver); |
| |
| auto* original_incoming_frame = |
| static_cast<TransformableIncomingAudioFrame*>(original); |
| return std::make_unique<TransformableIncomingAudioFrame>( |
| original->GetData(), original_incoming_frame->Header(), |
| original->GetSsrc(), original->GetMimeType(), |
| original->ReceiveTime().value_or(Timestamp::MinusInfinity())); |
| } |
| } // namespace webrtc |