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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/audio/channel_send_frame_transformer_delegate.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.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 'third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc')
-rw-r--r-- | third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc | 130 |
1 files changed, 130 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc new file mode 100644 index 0000000000..29bb0b81d8 --- /dev/null +++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc @@ -0,0 +1,130 @@ +/* + * 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_send_frame_transformer_delegate.h" + +#include <utility> + +namespace webrtc { +namespace { + +class TransformableOutgoingAudioFrame : public TransformableFrameInterface { + public: + TransformableOutgoingAudioFrame(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) + : frame_type_(frame_type), + payload_type_(payload_type), + rtp_timestamp_(rtp_timestamp), + rtp_start_timestamp_(rtp_start_timestamp), + payload_(payload_data, payload_size), + absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms), + ssrc_(ssrc) {} + ~TransformableOutgoingAudioFrame() 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()); + } + uint32_t GetTimestamp() const override { + return rtp_timestamp_ + rtp_start_timestamp_; + } + uint32_t GetStartTimestamp() const { return rtp_start_timestamp_; } + uint32_t GetSsrc() const override { return ssrc_; } + + AudioFrameType GetFrameType() const { return frame_type_; } + uint8_t GetPayloadType() const override { return payload_type_; } + int64_t GetAbsoluteCaptureTimestampMs() const { + return absolute_capture_timestamp_ms_; + } + Direction GetDirection() const override { return Direction::kSender; } + + private: + AudioFrameType frame_type_; + uint8_t payload_type_; + uint32_t rtp_timestamp_; + uint32_t rtp_start_timestamp_; + rtc::Buffer payload_; + int64_t absolute_capture_timestamp_ms_; + uint32_t ssrc_; +}; +} // namespace + +ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate( + SendFrameCallback send_frame_callback, + rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, + rtc::TaskQueue* encoder_queue) + : send_frame_callback_(send_frame_callback), + frame_transformer_(std::move(frame_transformer)), + encoder_queue_(encoder_queue) {} + +void ChannelSendFrameTransformerDelegate::Init() { + frame_transformer_->RegisterTransformedFrameCallback( + rtc::scoped_refptr<TransformedFrameCallback>(this)); +} + +void ChannelSendFrameTransformerDelegate::Reset() { + frame_transformer_->UnregisterTransformedFrameCallback(); + frame_transformer_ = nullptr; + + MutexLock lock(&send_lock_); + send_frame_callback_ = SendFrameCallback(); +} + +void ChannelSendFrameTransformerDelegate::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) { + frame_transformer_->Transform( + std::make_unique<TransformableOutgoingAudioFrame>( + frame_type, payload_type, rtp_timestamp, rtp_start_timestamp, + payload_data, payload_size, absolute_capture_timestamp_ms, ssrc)); +} + +void ChannelSendFrameTransformerDelegate::OnTransformedFrame( + std::unique_ptr<TransformableFrameInterface> frame) { + MutexLock lock(&send_lock_); + if (!send_frame_callback_) + return; + rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate(this); + encoder_queue_->PostTask( + [delegate = std::move(delegate), frame = std::move(frame)]() mutable { + delegate->SendFrame(std::move(frame)); + }); +} + +void ChannelSendFrameTransformerDelegate::SendFrame( + std::unique_ptr<TransformableFrameInterface> frame) const { + MutexLock lock(&send_lock_); + RTC_DCHECK_RUN_ON(encoder_queue_); + RTC_CHECK_EQ(frame->GetDirection(), + TransformableFrameInterface::Direction::kSender); + if (!send_frame_callback_) + return; + auto* transformed_frame = + static_cast<TransformableOutgoingAudioFrame*>(frame.get()); + send_frame_callback_(transformed_frame->GetFrameType(), + transformed_frame->GetPayloadType(), + transformed_frame->GetTimestamp() - + transformed_frame->GetStartTimestamp(), + transformed_frame->GetData(), + transformed_frame->GetAbsoluteCaptureTimestampMs()); +} + +} // namespace webrtc |