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Insert audio frame transformer between depacketizer and decoder.

Browse Source 
The frame transformer is passed from RTPReceiverInterface through the
library to be eventually set in ChannelReceive, where the frame
transformation will occur in the follow-up CL.

Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I5af06d1431047ef50d00e304cf95e92a832b4220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171872
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30956}
This commit is contained in:
Marina Ciocea
2020-04-01 07:46:16 +02:00
committed by Commit Bot
parent 784630f0e6
commit 3e9af7fe05
10 changed files with 126 additions and 24 deletions
Download Patch File Download Diff File Expand all files Collapse all files

24
pc/audio_rtp_receiver.cc
View File

@ -225,6 +225,20 @@ std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
RTC_FROM_HERE, [&] { return media_channel_->GetSources(*ssrc_); });
}
void AudioRtpReceiver::SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
if (media_channel_ && ssrc_.has_value() && !stopped_) {
worker_thread_->Invoke<void>(
RTC_FROM_HERE,
[this, frame_transformer = std::move(frame_transformer)] {
RTC_DCHECK_RUN_ON(worker_thread_);
frame_transformer_ = frame_transformer;
media_channel_->SetDepacketizerToDecoderFrameTransformer(
*ssrc_, frame_transformer);
});
}
}
void AudioRtpReceiver::Reconfigure() {
if (!media_channel_ || stopped_) {
RTC_LOG(LS_ERROR)
@ -237,6 +251,16 @@ void AudioRtpReceiver::Reconfigure() {
// Reattach the frame decryptor if we were reconfigured.
MaybeAttachFrameDecryptorToMediaChannel(
ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
if (media_channel_ && ssrc_.has_value() && !stopped_) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!frame_transformer_)
return;
media_channel_->SetDepacketizerToDecoderFrameTransformer(
*ssrc_, frame_transformer_);
});
}
}
void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {

5
pc/audio_rtp_receiver.h
View File

@ -104,6 +104,9 @@ class AudioRtpReceiver : public ObserverInterface,
std::vector<RtpSource> GetSources() const override;
int AttachmentId() const override { return attachment_id_; }
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override;
private:
void RestartMediaChannel(absl::optional<uint32_t> ssrc);
@ -128,6 +131,8 @@ class AudioRtpReceiver : public ObserverInterface,
// Allows to thread safely change playout delay. Handles caching cases if
// |SetJitterBufferMinimumDelay| is called before start.
rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
RTC_GUARDED_BY(worker_thread_);
};
} // namespace webrtc