126 lines
4.5 KiB
C++
126 lines
4.5 KiB
C++
/*
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* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio/channel_send_frame_transformer_delegate.h"
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#include <utility>
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namespace webrtc {
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namespace {
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class TransformableAudioFrame : public TransformableFrameInterface {
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public:
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TransformableAudioFrame(AudioFrameType frame_type,
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uint8_t payload_type,
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uint32_t rtp_timestamp,
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uint32_t rtp_start_timestamp,
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const uint8_t* payload_data,
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size_t payload_size,
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int64_t absolute_capture_timestamp_ms,
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uint32_t ssrc)
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: frame_type_(frame_type),
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payload_type_(payload_type),
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rtp_timestamp_(rtp_timestamp),
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rtp_start_timestamp_(rtp_start_timestamp),
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payload_(payload_data, payload_size),
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absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms),
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ssrc_(ssrc) {}
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~TransformableAudioFrame() override = default;
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rtc::ArrayView<const uint8_t> GetData() const override { return payload_; }
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void SetData(rtc::ArrayView<const uint8_t> data) override {
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payload_.SetData(data.data(), data.size());
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}
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uint32_t GetTimestamp() const override {
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return rtp_timestamp_ + rtp_start_timestamp_;
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}
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uint32_t GetStartTimestamp() const { return rtp_start_timestamp_; }
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uint32_t GetSsrc() const override { return ssrc_; }
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AudioFrameType GetFrameType() const { return frame_type_; }
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uint8_t GetPayloadType() const { return payload_type_; }
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int64_t GetAbsoluteCaptureTimestampMs() const {
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return absolute_capture_timestamp_ms_;
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}
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private:
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AudioFrameType frame_type_;
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uint8_t payload_type_;
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uint32_t rtp_timestamp_;
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uint32_t rtp_start_timestamp_;
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rtc::Buffer payload_;
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int64_t absolute_capture_timestamp_ms_;
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uint32_t ssrc_;
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};
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} // namespace
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ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate(
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SendFrameCallback send_frame_callback,
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
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rtc::TaskQueue* encoder_queue)
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: send_frame_callback_(send_frame_callback),
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frame_transformer_(std::move(frame_transformer)),
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encoder_queue_(encoder_queue) {}
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void ChannelSendFrameTransformerDelegate::Init() {
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frame_transformer_->RegisterTransformedFrameCallback(
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rtc::scoped_refptr<TransformedFrameCallback>(this));
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}
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void ChannelSendFrameTransformerDelegate::Reset() {
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frame_transformer_->UnregisterTransformedFrameCallback();
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frame_transformer_ = nullptr;
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MutexLock lock(&send_lock_);
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send_frame_callback_ = SendFrameCallback();
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}
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void ChannelSendFrameTransformerDelegate::Transform(
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AudioFrameType frame_type,
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uint8_t payload_type,
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uint32_t rtp_timestamp,
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uint32_t rtp_start_timestamp,
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const uint8_t* payload_data,
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size_t payload_size,
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int64_t absolute_capture_timestamp_ms,
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uint32_t ssrc) {
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frame_transformer_->Transform(std::make_unique<TransformableAudioFrame>(
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frame_type, payload_type, rtp_timestamp, rtp_start_timestamp,
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payload_data, payload_size, absolute_capture_timestamp_ms, ssrc));
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}
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void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
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std::unique_ptr<TransformableFrameInterface> frame) {
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MutexLock lock(&send_lock_);
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if (!send_frame_callback_)
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return;
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rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate = this;
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encoder_queue_->PostTask(
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[delegate = std::move(delegate), frame = std::move(frame)]() mutable {
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delegate->SendFrame(std::move(frame));
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});
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}
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void ChannelSendFrameTransformerDelegate::SendFrame(
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std::unique_ptr<TransformableFrameInterface> frame) const {
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MutexLock lock(&send_lock_);
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RTC_DCHECK_RUN_ON(encoder_queue_);
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if (!send_frame_callback_)
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return;
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auto* transformed_frame = static_cast<TransformableAudioFrame*>(frame.get());
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send_frame_callback_(transformed_frame->GetFrameType(),
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transformed_frame->GetPayloadType(),
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transformed_frame->GetTimestamp() -
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transformed_frame->GetStartTimestamp(),
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transformed_frame->GetData(),
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transformed_frame->GetAbsoluteCaptureTimestampMs());
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}
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} // namespace webrtc
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