blob: 6f08a0caf1c1b1234ac51779f13d5cada99c474c [file] [log] [blame] [edit]
/*
* 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"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
const char MediaStreamTrackInterface::kVideoKind[] = "video";
const char MediaStreamTrackInterface::kAudioKind[] = "audio";
void AudioProcessorInterface::GetStats(AudioProcessorStats* /*stats*/) {
RTC_NOTREACHED() << "Old-style GetStats() is called but it has no "
<< "implementation.";
RTC_LOG(LS_ERROR) << "Old-style GetStats() is called but it has no "
<< "implementation.";
}
// 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.apm_statistics.divergent_filter_fraction =
stats.aec_divergent_filter_fraction;
new_stats.apm_statistics.delay_median_ms = stats.echo_delay_median_ms;
new_stats.apm_statistics.delay_standard_deviation_ms =
stats.echo_delay_std_ms;
new_stats.apm_statistics.echo_return_loss = stats.echo_return_loss;
new_stats.apm_statistics.echo_return_loss_enhancement =
stats.echo_return_loss_enhancement;
new_stats.apm_statistics.residual_echo_likelihood =
stats.residual_echo_likelihood;
new_stats.apm_statistics.residual_echo_likelihood_recent_max =
stats.residual_echo_likelihood_recent_max;
return new_stats;
}
VideoTrackInterface::ContentHint VideoTrackInterface::content_hint() const {
return ContentHint::kNone;
}
bool AudioTrackInterface::GetSignalLevel(int* level) {
return false;
}
rtc::scoped_refptr<AudioProcessorInterface>
AudioTrackInterface::GetAudioProcessor() {
return nullptr;
}
} // namespace webrtc