chromium/third_party/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc

/*
 *  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 "modules/audio_processing/aec_dump/aec_dump_impl.h"

#include <memory>
#include <utility>

#include "absl/base/nullability.h"
#include "absl/strings/string_view.h"
#include "api/task_queue/task_queue_base.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"

namespace webrtc {

namespace {
void CopyFromConfigToEvent(const webrtc::InternalAPMConfig& config,
                           webrtc::audioproc::Config* pb_cfg) {}

}  // namespace

AecDumpImpl::AecDumpImpl(FileWrapper debug_file,
                         int64_t max_log_size_bytes,
                         absl::Nonnull<TaskQueueBase*> worker_queue)
    :{}

AecDumpImpl::~AecDumpImpl() {}

void AecDumpImpl::WriteInitMessage(const ProcessingConfig& api_format,
                                   int64_t time_now_ms) {}

void AecDumpImpl::AddCaptureStreamInput(
    const AudioFrameView<const float>& src) {}

void AecDumpImpl::AddCaptureStreamOutput(
    const AudioFrameView<const float>& src) {}

void AecDumpImpl::AddCaptureStreamInput(const int16_t* const data,
                                        int num_channels,
                                        int samples_per_channel) {}

void AecDumpImpl::AddCaptureStreamOutput(const int16_t* const data,
                                         int num_channels,
                                         int samples_per_channel) {}

void AecDumpImpl::AddAudioProcessingState(const AudioProcessingState& state) {}

void AecDumpImpl::WriteCaptureStreamMessage() {}

void AecDumpImpl::WriteRenderStreamMessage(const int16_t* const data,
                                           int num_channels,
                                           int samples_per_channel) {}

void AecDumpImpl::WriteRenderStreamMessage(
    const AudioFrameView<const float>& src) {}

void AecDumpImpl::WriteConfig(const InternalAPMConfig& config) {}

void AecDumpImpl::WriteRuntimeSetting(
    const AudioProcessing::RuntimeSetting& runtime_setting) {}

void AecDumpImpl::PostWriteToFileTask(std::unique_ptr<audioproc::Event> event) {}

absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
    FileWrapper file,
    int64_t max_log_size_bytes,
    absl::Nonnull<TaskQueueBase*> worker_queue) {}

absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
    absl::string_view file_name,
    int64_t max_log_size_bytes,
    absl::Nonnull<TaskQueueBase*> worker_queue) {}

absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
    absl::Nonnull<FILE*> handle,
    int64_t max_log_size_bytes,
    absl::Nonnull<TaskQueueBase*> worker_queue) {}

}  // namespace webrtc