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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

27
media/engine/webrtc_voice_engine.cc
View File

@ -1063,7 +1063,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options)
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
: call_(call), config_() {
RTC_DCHECK(call);
config_.rtp.remote_ssrc = remote_ssrc;
@ -1085,6 +1086,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
config_.codec_pair_id = codec_pair_id;
config_.frame_decryptor = frame_decryptor;
config_.crypto_options = crypto_options;
config_.frame_transformer = std::move(frame_transformer);
RecreateAudioReceiveStream();
}
@ -1215,6 +1217,13 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
return rtp_parameters;
}
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
config_.frame_transformer = std::move(frame_transformer);
ReconfigureAudioReceiveStream();
}
private:
void RecreateAudioReceiveStream() {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
@ -1868,7 +1877,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
engine()->audio_jitter_buffer_fast_accelerate_,
engine()->audio_jitter_buffer_min_delay_ms_,
engine()->audio_jitter_buffer_enable_rtx_handling_,
unsignaled_frame_decryptor_, crypto_options_)));
unsignaled_frame_decryptor_, crypto_options_, nullptr)));
recv_streams_[ssrc]->SetPlayout(playout_);
return true;
@ -2337,6 +2346,20 @@ void WebRtcVoiceMediaChannel::SetEncoderToPacketizerFrameTransformer(
std::move(frame_transformer));
}
void WebRtcVoiceMediaChannel::SetDepacketizerToDecoderFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
auto matching_stream = recv_streams_.find(ssrc);
if (matching_stream == recv_streams_.end()) {
RTC_LOG(LS_INFO) << "Attempting to set frame transformer for SSRC:" << ssrc
<< " which doesn't exist.";
return;
}
matching_stream->second->SetDepacketizerToDecoderFrameTransformer(
std::move(frame_transformer));
}
bool WebRtcVoiceMediaChannel::MaybeDeregisterUnsignaledRecvStream(
uint32_t ssrc) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());