mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00

Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead. The old fields are preserved for compatibility with downstream projects, but will be removed in the future. Bug: webrtc:15788 Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41947}
172 lines
6.2 KiB
C++
172 lines
6.2 KiB
C++
/*
|
|
* 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 <string>
|
|
#include <utility>
|
|
|
|
#include "rtc_base/buffer.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
class TransformableIncomingAudioFrame
|
|
: public TransformableAudioFrameInterface {
|
|
public:
|
|
TransformableIncomingAudioFrame(rtc::ArrayView<const uint8_t> payload,
|
|
const RTPHeader& header,
|
|
uint32_t ssrc,
|
|
const std::string& codec_mime_type)
|
|
: payload_(payload.data(), payload.size()),
|
|
header_(header),
|
|
ssrc_(ssrc),
|
|
codec_mime_type_(codec_mime_type) {}
|
|
~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 absl::optional<uint16_t> SequenceNumber() const override {
|
|
return header_.sequenceNumber;
|
|
}
|
|
|
|
absl::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 absl::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;
|
|
}
|
|
|
|
private:
|
|
rtc::Buffer payload_;
|
|
RTPHeader header_;
|
|
uint32_t ssrc_;
|
|
std::string codec_mime_type_;
|
|
};
|
|
} // namespace
|
|
|
|
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) {
|
|
RTC_DCHECK_RUN_ON(&sequence_checker_);
|
|
if (short_circuit_) {
|
|
receive_frame_callback_(packet, header);
|
|
} else {
|
|
frame_transformer_->Transform(
|
|
std::make_unique<TransformableIncomingAudioFrame>(packet, header, ssrc,
|
|
codec_mime_type));
|
|
}
|
|
}
|
|
|
|
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;
|
|
|
|
RTPHeader header;
|
|
if (frame->GetDirection() ==
|
|
TransformableFrameInterface::Direction::kSender) {
|
|
auto* transformed_frame =
|
|
static_cast<TransformableAudioFrameInterface*>(frame.get());
|
|
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_frame =
|
|
static_cast<TransformableIncomingAudioFrame*>(frame.get());
|
|
header = transformed_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);
|
|
}
|
|
|
|
rtc::scoped_refptr<FrameTransformerInterface>
|
|
ChannelReceiveFrameTransformerDelegate::FrameTransformer() {
|
|
RTC_DCHECK_RUN_ON(&sequence_checker_);
|
|
return frame_transformer_;
|
|
}
|
|
|
|
} // namespace webrtc
|