Commit 27aab564 authored by Sam Zackrisson's avatar Sam Zackrisson Committed by Commit Bot

Delete deprecated audio processing stats getters

Related deprecation PSA:
https://groups.google.com/d/msg/discuss-webrtc/NgqEPvkNuDE/7HtwnMmADgAJ

Bug: webrtc:9947
Change-Id: I392a9c27641a1b37170c27c9f42721bdae353615
Reviewed-on: https://chromium-review.googlesource.com/c/1309785Reviewed-by: default avatarMax Morin <maxmorin@chromium.org>
Commit-Queue: Sam Zackrisson <saza@chromium.org>
Cr-Commit-Position: refs/heads/master@{#604252}
parent 1e26651d
...@@ -102,12 +102,6 @@ void AudioServiceAudioProcessorProxy::SetControls( ...@@ -102,12 +102,6 @@ void AudioServiceAudioProcessorProxy::SetControls(
aec_dump_message_filter_->AddDelegate(this); aec_dump_message_filter_->AddDelegate(this);
} }
void AudioServiceAudioProcessorProxy::GetStats(AudioProcessorStats* out) {
// This is the old GetStats interface from webrtc::AudioProcessorInterface.
// It should not be in use by Chrome any longer.
NOTREACHED();
}
webrtc::AudioProcessorInterface::AudioProcessorStatistics webrtc::AudioProcessorInterface::AudioProcessorStatistics
AudioServiceAudioProcessorProxy::GetStats(bool has_remote_tracks) { AudioServiceAudioProcessorProxy::GetStats(bool has_remote_tracks) {
base::AutoLock lock(stats_lock_); base::AutoLock lock(stats_lock_);
......
...@@ -41,10 +41,6 @@ class CONTENT_EXPORT AudioServiceAudioProcessorProxy ...@@ -41,10 +41,6 @@ class CONTENT_EXPORT AudioServiceAudioProcessorProxy
// this method. // this method.
void Stop(); void Stop();
// webrtc::AudioProcessorInterface implementation.
// This method is called on the libjingle thread.
void GetStats(AudioProcessorStats* stats) override;
// This method is called on the libjingle thread. // This method is called on the libjingle thread.
AudioProcessorStatistics GetStats(bool has_remote_tracks) override; AudioProcessorStatistics GetStats(bool has_remote_tracks) override;
......
...@@ -524,12 +524,6 @@ void MediaStreamAudioProcessor::OnRenderThreadChanged() { ...@@ -524,12 +524,6 @@ void MediaStreamAudioProcessor::OnRenderThreadChanged() {
DCHECK(render_thread_checker_.CalledOnValidThread()); DCHECK(render_thread_checker_.CalledOnValidThread());
} }
void MediaStreamAudioProcessor::GetStats(AudioProcessorStats* stats) {
// This is the old GetStats interface from webrtc::AudioProcessorInterface.
// It should not be in use by Chrome any longer.
NOTREACHED();
}
webrtc::AudioProcessorInterface::AudioProcessorStatistics webrtc::AudioProcessorInterface::AudioProcessorStatistics
MediaStreamAudioProcessor::GetStats(bool has_remote_tracks) { MediaStreamAudioProcessor::GetStats(bool has_remote_tracks) {
AudioProcessorStatistics stats; AudioProcessorStatistics stats;
......
...@@ -134,10 +134,6 @@ class CONTENT_EXPORT MediaStreamAudioProcessor ...@@ -134,10 +134,6 @@ class CONTENT_EXPORT MediaStreamAudioProcessor
void OnPlayoutDataSourceChanged() override; void OnPlayoutDataSourceChanged() override;
void OnRenderThreadChanged() override; void OnRenderThreadChanged() override;
// webrtc::AudioProcessorInterface implementation.
// This method is called on the libjingle thread.
void GetStats(AudioProcessorStats* stats) override;
// This method is called on the libjingle thread. // This method is called on the libjingle thread.
AudioProcessorStatistics GetStats(bool has_remote_tracks) override; AudioProcessorStatistics GetStats(bool has_remote_tracks) override;
......
...@@ -217,22 +217,4 @@ void ConfigPreAmplifier(webrtc::AudioProcessing::Config* apm_config, ...@@ -217,22 +217,4 @@ void ConfigPreAmplifier(webrtc::AudioProcessing::Config* apm_config,
} }
} }
void GetAudioProcessingStats(
AudioProcessing* audio_processing,
webrtc::AudioProcessorInterface::AudioProcessorStats* stats) {
// TODO(ivoc): Change the APM stats to use optional instead of default values.
auto apm_stats = audio_processing->GetStatistics();
stats->echo_return_loss = apm_stats.echo_return_loss.instant();
stats->echo_return_loss_enhancement =
apm_stats.echo_return_loss_enhancement.instant();
stats->aec_divergent_filter_fraction = apm_stats.divergent_filter_fraction;
stats->echo_delay_median_ms = apm_stats.delay_median;
stats->echo_delay_std_ms = apm_stats.delay_standard_deviation;
stats->residual_echo_likelihood = apm_stats.residual_echo_likelihood;
stats->residual_echo_likelihood_recent_max =
apm_stats.residual_echo_likelihood_recent_max;
}
} // namespace content } // namespace content
...@@ -131,10 +131,6 @@ void EnableAutomaticGainControl(AudioProcessing* audio_processing, ...@@ -131,10 +131,6 @@ void EnableAutomaticGainControl(AudioProcessing* audio_processing,
void ConfigPreAmplifier(webrtc::AudioProcessing::Config* apm_config, void ConfigPreAmplifier(webrtc::AudioProcessing::Config* apm_config,
base::Optional<double> fixed_gain_factor); base::Optional<double> fixed_gain_factor);
void GetAudioProcessingStats(
AudioProcessing* audio_processing,
webrtc::AudioProcessorInterface::AudioProcessorStats* stats);
} // namespace content } // namespace content
#endif // CONTENT_RENDERER_MEDIA_STREAM_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_ #endif // CONTENT_RENDERER_MEDIA_STREAM_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_
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