blob: 92bca162afed1c1019d11e0795ddd06036d1de61 [file] [log] [blame]
/*
* Copyright 2017 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 "api/mediastreaminterface.h"
namespace webrtc {
const char MediaStreamTrackInterface::kVideoKind[] = "video";
const char MediaStreamTrackInterface::kAudioKind[] = "audio";
// TODO(ivoc): Remove this when the function becomes pure virtual.
AudioProcessorInterface::AudioProcessorStatistics
AudioProcessorInterface::GetStats(bool /*has_remote_tracks*/) {
AudioProcessorStats stats;
GetStats(&stats);
AudioProcessorStatistics new_stats;
new_stats.aec_divergent_filter_fraction =
rtc::Optional<double>(stats.aec_divergent_filter_fraction);
new_stats.aec_quality_min = rtc::Optional<double>(stats.aec_quality_min);
new_stats.echo_delay_median_ms =
rtc::Optional<int32_t>(stats.echo_delay_median_ms);
new_stats.echo_delay_std_ms = rtc::Optional<int32_t>(stats.echo_delay_std_ms);
new_stats.echo_return_loss = rtc::Optional<double>(stats.echo_return_loss);
new_stats.echo_return_loss_enhancement =
rtc::Optional<double>(stats.echo_return_loss_enhancement);
new_stats.residual_echo_likelihood =
rtc::Optional<double>(stats.residual_echo_likelihood);
new_stats.residual_echo_likelihood_recent_max =
rtc::Optional<double>(stats.residual_echo_likelihood_recent_max);
return new_stats;
}
} // namespace webrtc