Allow an external audio processing module to be used in WebRTC
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index f89da98..f8ee3ab 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -279,8 +279,9 @@
stats.audio_level = base->transmit_mixer()->AudioLevelFullRange();
RTC_DCHECK_LE(0, stats.audio_level);
- RTC_DCHECK(base->audio_processing());
- auto audio_processing_stats = base->audio_processing()->GetStatistics();
+ RTC_DCHECK(audio_state_->audio_processing());
+ auto audio_processing_stats =
+ audio_state_->audio_processing()->GetStatistics();
stats.echo_delay_median_ms = audio_processing_stats.delay_median;
stats.echo_delay_std_ms = audio_processing_stats.delay_standard_deviation;
stats.echo_return_loss = audio_processing_stats.echo_return_loss.instant();