acm_receiver.cc | Explore in Territory

/*
 *  Copyright (c) 2013 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/acm2/acm_receiver.h"

#include <stdlib.h>
#include <string.h>

#include <cstdint>
#include <vector>

#include "absl/strings/match.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/neteq/neteq.h"
#include "api/units/timestamp.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/strings/audio_format_to_string.h"
#include "system_wrappers/include/clock.h"

namespace webrtc {

namespace acm2 {

namespace {

std::unique_ptr<NetEq> CreateNetEq(
    NetEqFactory* neteq_factory,
    const NetEq::Config& config,
    const Environment& env,
    scoped_refptr<AudioDecoderFactory> decoder_factory) { … }

}  // namespace

AcmReceiver::Config::Config(
    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
    : … { … }

AcmReceiver::Config::Config(const Config&) = default;
AcmReceiver::Config::~Config() = default;

AcmReceiver::AcmReceiver(const Environment& env, Config config)
    : … { … }

AcmReceiver::~AcmReceiver() = default;

int AcmReceiver::SetMinimumDelay(int delay_ms) { … }

int AcmReceiver::SetMaximumDelay(int delay_ms) { … }

bool AcmReceiver::SetBaseMinimumDelayMs(int delay_ms) { … }

int AcmReceiver::GetBaseMinimumDelayMs() const { … }

absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const { … }

int AcmReceiver::last_output_sample_rate_hz() const { … }

int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
                              rtc::ArrayView<const uint8_t> incoming_payload,
                              Timestamp receive_time) { … }

int AcmReceiver::GetAudio(int desired_freq_hz,
                          AudioFrame* audio_frame,
                          bool* muted) { … }

void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) { … }

void AcmReceiver::FlushBuffers() { … }

absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() { … }

int AcmReceiver::FilteredCurrentDelayMs() const { … }

int AcmReceiver::TargetDelayMs() const { … }

absl::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder()
    const { … }

void AcmReceiver::GetNetworkStatistics(
    NetworkStatistics* acm_stat,
    bool get_and_clear_legacy_stats /* = true */) const { … }

int AcmReceiver::EnableNack(size_t max_nack_list_size) { … }

void AcmReceiver::DisableNack() { … }

std::vector<uint16_t> AcmReceiver::GetNackList(
    int64_t round_trip_time_ms) const { … }

void AcmReceiver::ResetInitialDelay() { … }

uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const { … }

void AcmReceiver::GetDecodingCallStatistics(
    AudioDecodingCallStats* stats) const { … }

}  // namespace acm2

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