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/*
* 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 TransformableAudioFrame : public TransformableFrameInterface {
public:
TransformableAudioFrame(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) {}
~TransformableAudioFrame() 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 { return payload_type_; }
int64_t GetAbsoluteCaptureTimestampMs() const {
return absolute_capture_timestamp_ms_;
}
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<TransformableAudioFrame>(
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_);
if (!send_frame_callback_)
return;
auto* transformed_frame = static_cast<TransformableAudioFrame*>(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
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