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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>
#include <vector>
namespace webrtc {
namespace {
using IfaceFrameType = TransformableAudioFrameInterface::FrameType;
IfaceFrameType InternalFrameTypeToInterfaceFrameType(
const AudioFrameType frame_type) {
switch (frame_type) {
case AudioFrameType::kEmptyFrame:
return IfaceFrameType::kEmptyFrame;
case AudioFrameType::kAudioFrameSpeech:
return IfaceFrameType::kAudioFrameSpeech;
case AudioFrameType::kAudioFrameCN:
return IfaceFrameType::kAudioFrameCN;
}
RTC_DCHECK_NOTREACHED();
return IfaceFrameType::kEmptyFrame;
}
AudioFrameType InterfaceFrameTypeToInternalFrameType(
const IfaceFrameType frame_type) {
switch (frame_type) {
case IfaceFrameType::kEmptyFrame:
return AudioFrameType::kEmptyFrame;
case IfaceFrameType::kAudioFrameSpeech:
return AudioFrameType::kAudioFrameSpeech;
case IfaceFrameType::kAudioFrameCN:
return AudioFrameType::kAudioFrameCN;
}
RTC_DCHECK_NOTREACHED();
return AudioFrameType::kEmptyFrame;
}
class TransformableOutgoingAudioFrame
: public TransformableAudioFrameInterface {
public:
TransformableOutgoingAudioFrame(
AudioFrameType frame_type,
uint8_t payload_type,
uint32_t rtp_timestamp_with_offset,
const uint8_t* payload_data,
size_t payload_size,
absl::optional<uint64_t> absolute_capture_timestamp_ms,
uint32_t ssrc,
std::vector<uint32_t> csrcs,
const std::string& codec_mime_type)
: frame_type_(frame_type),
payload_type_(payload_type),
rtp_timestamp_with_offset_(rtp_timestamp_with_offset),
payload_(payload_data, payload_size),
absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms),
ssrc_(ssrc),
csrcs_(std::move(csrcs)),
codec_mime_type_(codec_mime_type) {}
~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_with_offset_; }
uint32_t GetSsrc() const override { return ssrc_; }
IfaceFrameType Type() const override {
return InternalFrameTypeToInterfaceFrameType(frame_type_);
}
uint8_t GetPayloadType() const override { return payload_type_; }
Direction GetDirection() const override { return Direction::kSender; }
std::string GetMimeType() const override { return codec_mime_type_; }
rtc::ArrayView<const uint32_t> GetContributingSources() const override {
return csrcs_;
}
const absl::optional<uint16_t> SequenceNumber() const override {
return absl::nullopt;
}
void SetRTPTimestamp(uint32_t rtp_timestamp_with_offset) override {
rtp_timestamp_with_offset_ = rtp_timestamp_with_offset;
}
absl::optional<uint64_t> AbsoluteCaptureTimestamp() const override {
return absolute_capture_timestamp_ms_;
}
private:
AudioFrameType frame_type_;
uint8_t payload_type_;
uint32_t rtp_timestamp_with_offset_;
rtc::Buffer payload_;
absl::optional<uint64_t> absolute_capture_timestamp_ms_;
uint32_t ssrc_;
std::vector<uint32_t> csrcs_;
std::string codec_mime_type_;
};
} // 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,
const uint8_t* payload_data,
size_t payload_size,
int64_t absolute_capture_timestamp_ms,
uint32_t ssrc,
const std::string& codec_mimetype) {
{
MutexLock lock(&send_lock_);
if (short_circuit_) {
send_frame_callback_(
frame_type, payload_type, rtp_timestamp,
rtc::ArrayView<const uint8_t>(payload_data, payload_size),
absolute_capture_timestamp_ms, /*csrcs=*/{});
return;
}
}
frame_transformer_->Transform(
std::make_unique<TransformableOutgoingAudioFrame>(
frame_type, payload_type, rtp_timestamp, payload_data, payload_size,
absolute_capture_timestamp_ms, ssrc,
/*csrcs=*/std::vector<uint32_t>(), codec_mimetype));
}
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::StartShortCircuiting() {
MutexLock lock(&send_lock_);
short_circuit_ = true;
}
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<TransformableAudioFrameInterface*>(frame.get());
send_frame_callback_(
InterfaceFrameTypeToInternalFrameType(transformed_frame->Type()),
transformed_frame->GetPayloadType(), transformed_frame->GetTimestamp(),
transformed_frame->GetData(),
transformed_frame->AbsoluteCaptureTimestamp()
? *transformed_frame->AbsoluteCaptureTimestamp()
: 0,
transformed_frame->GetContributingSources());
}
std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
TransformableAudioFrameInterface* original) {
std::vector<uint32_t> csrcs;
csrcs.assign(original->GetContributingSources().begin(),
original->GetContributingSources().end());
return std::make_unique<TransformableOutgoingAudioFrame>(
InterfaceFrameTypeToInternalFrameType(original->Type()),
original->GetPayloadType(), original->GetTimestamp(),
original->GetData().data(), original->GetData().size(),
original->AbsoluteCaptureTimestamp(), original->GetSsrc(),
std::move(csrcs), original->GetMimeType());
}
} // namespace webrtc
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