audio_coding_module.cc | Explore in Territory

/*
 *  Copyright (c) 2012 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_coding/include/audio_coding_module.h"

#include <algorithm>
#include <cstdint>

#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "modules/audio_coding/acm2/acm_remixing.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/include/module_common_types.h"
#include "modules/include/module_common_types_public.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/metrics.h"

namespace webrtc {

namespace {

// Initial size for the buffer in InputBuffer. This matches 6 channels of 10 ms
// 48 kHz data.
constexpr size_t kInitialInputDataBufferSize = …;

constexpr int32_t kMaxInputSampleRateHz = …;

class AudioCodingModuleImpl final : public AudioCodingModule { … };

// Adds a codec usage sample to the histogram.
void UpdateCodecTypeHistogram(size_t codec_type) { … }

void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) { … }

AudioCodingModuleImpl::AudioCodingModuleImpl()
    : … { … }

AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;

int32_t AudioCodingModuleImpl::Encode(
    const InputData& input_data,
    absl::optional<int64_t> absolute_capture_timestamp_ms) { … }

/////////////////////////////////////////
//   Sender
//

void AudioCodingModuleImpl::ModifyEncoder(
    rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) { … }

// Register a transport callback which will be called to deliver
// the encoded buffers.
int AudioCodingModuleImpl::RegisterTransportCallback(
    AudioPacketizationCallback* transport) { … }

// Add 10MS of raw (PCM) audio data to the encoder.
int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) { … }

int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
                                               InputData* input_data) { … }

// Perform a resampling and down-mix if required. We down-mix only if
// encoder is mono and input is stereo. In case of dual-streaming, both
// encoders has to be mono for down-mix to take place.
// |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing
// is required, |*ptr_out| points to `in_frame`.
// TODO(yujo): Make this more efficient for muted frames.
int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
                                               const AudioFrame** ptr_out) { … }

/////////////////////////////////////////
//   (FEC) Forward Error Correction (codec internal)
//

int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) { … }

/////////////////////////////////////////
//   Statistics
//

bool AudioCodingModuleImpl::HaveValidEncoder(
    absl::string_view caller_name) const { … }

ANAStats AudioCodingModuleImpl::GetANAStats() const { … }

int AudioCodingModuleImpl::GetTargetBitrate() const { … }

}  // namespace

std::unique_ptr<AudioCodingModule> AudioCodingModule::Create() { … }

}  // namespace webrtc
chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
