| /* | |
| * Copyright (c) 2025 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 "modules/audio_processing/capture_mixer/capture_mixer.h" | |
| #include <cstddef> | |
| #include <span> | |
| #include "modules/audio_processing/capture_mixer/channel_content_remixer.h" | |
| #include "rtc_base/checks.h" | |
| namespace webrtc { | |
| namespace { | |
| constexpr size_t kNumFramesForCrossfade = 20; | |
| } // namespace | |
| CaptureMixer::CaptureMixer(size_t num_samples_per_channel) | |
| : audio_content_analyzer_(num_samples_per_channel), | |
| channel_content_mixer_(num_samples_per_channel, kNumFramesForCrossfade), | |
| mixing_variant_(StereoMixingVariant::kUseAverage), | |
| remixing_logic_(num_samples_per_channel) {} | |
| void CaptureMixer::Mix(size_t num_output_channels, | |
| std::span<float> channel0, | |
| std::span<float> channel1) { | |
| RTC_DCHECK_GE(num_output_channels, 1); | |
| RTC_DCHECK_LE(num_output_channels, 2); | |
| const bool reliable_estimates = | |
| audio_content_analyzer_.Analyze(channel0, channel1); | |
| if (!reliable_estimates) { | |
| // Downmix to mono (fake-stereo content with the same channel content) in an | |
| // average manner until reliable estimates have been achieved. | |
| mixing_variant_ = StereoMixingVariant::kUseAverage; | |
| channel_content_mixer_.Mix(num_output_channels, mixing_variant_, channel0, | |
| channel1); | |
| return; | |
| } | |
| std::span<const float, 2> average_energies = | |
| audio_content_analyzer_.GetChannelEnergies(); | |
| std::span<const int, 2> num_frames_since_activity = | |
| audio_content_analyzer_.GetNumFramesSinceActivity(); | |
| std::span<const float, 2> saturation_factors = | |
| audio_content_analyzer_.GetSaturationFactors(); | |
| mixing_variant_ = remixing_logic_.SelectStereoChannelMixing( | |
| average_energies, num_frames_since_activity, saturation_factors); | |
| channel_content_mixer_.Mix(num_output_channels, mixing_variant_, channel0, | |
| channel1); | |
| } | |
| } // namespace webrtc |