blob: 9ddf81e41e4eeed9eb3f98478ae42e0e68db640e [file] [log] [blame]
* Copyright (c) 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 <memory>
#include <vector>
#include "api/array_view.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/agc2/limiter.h"
namespace webrtc {
class ApmDataDumper;
class FrameCombiner {
enum class LimiterType { kNoLimiter, kApmAgcLimiter, kApmAgc2Limiter };
explicit FrameCombiner(bool use_limiter);
// Combine several frames into one. Assumes sample_rate,
// samples_per_channel of the input frames match the parameters. The
// parameters 'number_of_channels' and 'sample_rate' are needed
// because 'mix_list' can be empty. The parameter
// 'number_of_streams' is used for determining whether to pass the
// data through a limiter.
void Combine(rtc::ArrayView<AudioFrame* const> mix_list,
size_t number_of_channels,
int sample_rate,
size_t number_of_streams,
AudioFrame* audio_frame_for_mixing);
// Stereo, 48 kHz, 10 ms.
static constexpr size_t kMaximumNumberOfChannels = 8;
static constexpr size_t kMaximumChannelSize = 48 * 10;
using MixingBuffer = std::array<std::array<float, kMaximumChannelSize>,
void LogMixingStats(rtc::ArrayView<const AudioFrame* const> mix_list,
int sample_rate,
size_t number_of_streams) const;
std::unique_ptr<ApmDataDumper> data_dumper_;
std::unique_ptr<MixingBuffer> mixing_buffer_;
Limiter limiter_;
const bool use_limiter_;
mutable int uma_logging_counter_ = 0;
} // namespace webrtc