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Telegram-Android/TMessagesProj/jni/voip/webrtc/audio/channel_receive.cc
DrKLO e8d88e56e4 Update to 7.8.0 (2359)
2021-06-25 03:43:10 +03:00

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/*
* 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 "audio/channel_receive.h"
#include <assert.h>
#include <algorithm>
#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "api/crypto/frame_decryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/sequence_checker.h"
#include "audio/audio_level.h"
#include "audio/channel_receive_frame_transformer_delegate.h"
#include "audio/channel_send.h"
#include "audio/utility/audio_frame_operations.h"
#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
#include "modules/rtp_rtcp/source/capture_clock_offset_updater.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/checks.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace voe {
namespace {
constexpr double kAudioSampleDurationSeconds = 0.01;
// Video Sync.
constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0;
constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
AudioCodingModule::Config AcmConfig(
NetEqFactory* neteq_factory,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout) {
AudioCodingModule::Config acm_config;
acm_config.neteq_factory = neteq_factory;
acm_config.decoder_factory = decoder_factory;
acm_config.neteq_config.codec_pair_id = codec_pair_id;
acm_config.neteq_config.max_packets_in_buffer = jitter_buffer_max_packets;
acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout;
acm_config.neteq_config.enable_muted_state = true;
return acm_config;
}
class ChannelReceive : public ChannelReceiveInterface {
public:
// Used for receive streams.
ChannelReceive(
Clock* clock,
ProcessThread* module_process_thread,
NetEqFactory* neteq_factory,
AudioDeviceModule* audio_device_module,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
~ChannelReceive() override;
void SetSink(AudioSinkInterface* sink) override;
void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
// API methods
void StartPlayout() override;
void StopPlayout() override;
// Codecs
absl::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec()
const override;
void ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
// RtpPacketSinkInterface.
void OnRtpPacket(const RtpPacketReceived& packet) override;
// Muting, Volume and Level.
void SetChannelOutputVolumeScaling(float scaling) override;
int GetSpeechOutputLevelFullRange() const override;
// See description of "totalAudioEnergy" in the WebRTC stats spec:
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
double GetTotalOutputEnergy() const override;
double GetTotalOutputDuration() const override;
// Stats.
NetworkStatistics GetNetworkStatistics(
bool get_and_clear_legacy_stats) const override;
AudioDecodingCallStats GetDecodingCallStatistics() const override;
// Audio+Video Sync.
uint32_t GetDelayEstimate() const override;
bool SetMinimumPlayoutDelay(int delayMs) override;
bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const override;
void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
int64_t time_ms) override;
absl::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
int64_t now_ms) const override;
// Audio quality.
bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
int GetBaseMinimumPlayoutDelayMs() const override;
// Produces the transport-related timestamps; current_delay_ms is left unset.
absl::optional<Syncable::Info> GetSyncInfo() const override;
void RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) override;
void ResetReceiverCongestionControlObjects() override;
CallReceiveStatistics GetRTCPStatistics() const override;
void SetNACKStatus(bool enable, int maxNumberOfPackets) override;
AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) override;
int PreferredSampleRate() const override;
void SetSourceTracker(SourceTracker* source_tracker) override;
// Associate to a send channel.
// Used for obtaining RTT for a receive-only channel.
void SetAssociatedSendChannel(const ChannelSendInterface* channel) override;
// Sets a frame transformer between the depacketizer and the decoder, to
// transform the received frames before decoding them.
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override;
private:
void ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header);
int ResendPackets(const uint16_t* sequence_numbers, int length);
void UpdatePlayoutTimestamp(bool rtcp, int64_t now_ms);
int GetRtpTimestampRateHz() const;
int64_t GetRTT() const;
void OnReceivedPayloadData(rtc::ArrayView<const uint8_t> payload,
const RTPHeader& rtpHeader);
void InitFrameTransformerDelegate(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer);
bool Playing() const {
MutexLock lock(&playing_lock_);
return playing_;
}
// Thread checkers document and lock usage of some methods to specific threads
// we know about. The goal is to eventually split up voe::ChannelReceive into
// parts with single-threaded semantics, and thereby reduce the need for
// locks.
SequenceChecker worker_thread_checker_;
// Methods accessed from audio and video threads are checked for sequential-
// only access. We don't necessarily own and control these threads, so thread
// checkers cannot be used. E.g. Chromium may transfer "ownership" from one
// audio thread to another, but access is still sequential.
rtc::RaceChecker audio_thread_race_checker_;
rtc::RaceChecker video_capture_thread_race_checker_;
Mutex callback_mutex_;
Mutex volume_settings_mutex_;
mutable Mutex playing_lock_;
bool playing_ RTC_GUARDED_BY(&playing_lock_) = false;
RtcEventLog* const event_log_;
// Indexed by payload type.
std::map<uint8_t, int> payload_type_frequencies_;
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
const uint32_t remote_ssrc_;
SourceTracker* source_tracker_ = nullptr;
// Info for GetSyncInfo is updated on network or worker thread, and queried on
// the worker thread.
mutable Mutex sync_info_lock_;
absl::optional<uint32_t> last_received_rtp_timestamp_
RTC_GUARDED_BY(&sync_info_lock_);
absl::optional<int64_t> last_received_rtp_system_time_ms_
RTC_GUARDED_BY(&sync_info_lock_);
// The AcmReceiver is thread safe, using its own lock.
acm2::AcmReceiver acm_receiver_;
AudioSinkInterface* audio_sink_ = nullptr;
AudioLevel _outputAudioLevel;
Clock* const clock_;
RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
// Timestamp of the audio pulled from NetEq.
absl::optional<uint32_t> jitter_buffer_playout_timestamp_;
mutable Mutex video_sync_lock_;
uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_);
absl::optional<int64_t> playout_timestamp_rtp_time_ms_
RTC_GUARDED_BY(video_sync_lock_);
uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_);
absl::optional<int64_t> playout_timestamp_ntp_
RTC_GUARDED_BY(video_sync_lock_);
absl::optional<int64_t> playout_timestamp_ntp_time_ms_
RTC_GUARDED_BY(video_sync_lock_);
mutable Mutex ts_stats_lock_;
std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
// The rtp timestamp of the first played out audio frame.
int64_t capture_start_rtp_time_stamp_;
// The capture ntp time (in local timebase) of the first played out audio
// frame.
int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_);
ProcessThread* const module_process_thread_;
AudioDeviceModule* _audioDeviceModulePtr;
float _outputGain RTC_GUARDED_BY(volume_settings_mutex_);
const ChannelSendInterface* associated_send_channel_
RTC_GUARDED_BY(worker_thread_checker_);
PacketRouter* packet_router_ = nullptr;
SequenceChecker construction_thread_;
// E2EE Audio Frame Decryption
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
webrtc::CryptoOptions crypto_options_;
webrtc::AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_;
webrtc::CaptureClockOffsetUpdater capture_clock_offset_updater_;
rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate>
frame_transformer_delegate_;
};
void ChannelReceive::OnReceivedPayloadData(
rtc::ArrayView<const uint8_t> payload,
const RTPHeader& rtpHeader) {
if (!Playing()) {
// Avoid inserting into NetEQ when we are not playing. Count the
// packet as discarded.
// If we have a source_tracker_, tell it that the frame has been
// "delivered". Normally, this happens in AudioReceiveStream when audio
// frames are pulled out, but when playout is muted, nothing is pulling
// frames. The downside of this approach is that frames delivered this way
// won't be delayed for playout, and therefore will be unsynchronized with
// (a) audio delay when playing and (b) any audio/video synchronization. But
// the alternative is that muting playout also stops the SourceTracker from
// updating RtpSource information.
if (source_tracker_) {
RtpPacketInfos::vector_type packet_vector = {
RtpPacketInfo(rtpHeader, clock_->CurrentTime())};
source_tracker_->OnFrameDelivered(RtpPacketInfos(packet_vector));
}
return;
}
// Push the incoming payload (parsed and ready for decoding) into the ACM
if (acm_receiver_.InsertPacket(rtpHeader, payload) != 0) {
RTC_DLOG(LS_ERROR) << "ChannelReceive::OnReceivedPayloadData() unable to "
"push data to the ACM";
return;
}
int64_t round_trip_time = 0;
rtp_rtcp_->RTT(remote_ssrc_, &round_trip_time, NULL, NULL, NULL);
std::vector<uint16_t> nack_list = acm_receiver_.GetNackList(round_trip_time);
if (!nack_list.empty()) {
// Can't use nack_list.data() since it's not supported by all
// compilers.
ResendPackets(&(nack_list[0]), static_cast<int>(nack_list.size()));
}
}
void ChannelReceive::InitFrameTransformerDelegate(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
RTC_DCHECK(frame_transformer);
RTC_DCHECK(!frame_transformer_delegate_);
// Pass a callback to ChannelReceive::OnReceivedPayloadData, to be called by
// the delegate to receive transformed audio.
ChannelReceiveFrameTransformerDelegate::ReceiveFrameCallback
receive_audio_callback = [this](rtc::ArrayView<const uint8_t> packet,
const RTPHeader& header) {
OnReceivedPayloadData(packet, header);
};
frame_transformer_delegate_ =
rtc::make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
std::move(receive_audio_callback), std::move(frame_transformer),
rtc::Thread::Current());
frame_transformer_delegate_->Init();
}
AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) {
RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
audio_frame->sample_rate_hz_ = sample_rate_hz;
event_log_->Log(std::make_unique<RtcEventAudioPlayout>(remote_ssrc_));
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
bool muted;
if (acm_receiver_.GetAudio(audio_frame->sample_rate_hz_, audio_frame,
&muted) == -1) {
RTC_DLOG(LS_ERROR)
<< "ChannelReceive::GetAudioFrame() PlayoutData10Ms() failed!";
// In all likelihood, the audio in this frame is garbage. We return an
// error so that the audio mixer module doesn't add it to the mix. As
// a result, it won't be played out and the actions skipped here are
// irrelevant.
return AudioMixer::Source::AudioFrameInfo::kError;
}
if (muted) {
// TODO(henrik.lundin): We should be able to do better than this. But we
// will have to go through all the cases below where the audio samples may
// be used, and handle the muted case in some way.
AudioFrameOperations::Mute(audio_frame);
}
{
// Pass the audio buffers to an optional sink callback, before applying
// scaling/panning, as that applies to the mix operation.
// External recipients of the audio (e.g. via AudioTrack), will do their
// own mixing/dynamic processing.
MutexLock lock(&callback_mutex_);
if (audio_sink_) {
AudioSinkInterface::Data data(
audio_frame->data(), audio_frame->samples_per_channel_,
audio_frame->sample_rate_hz_, audio_frame->num_channels_,
audio_frame->timestamp_);
audio_sink_->OnData(data);
}
}
float output_gain = 1.0f;
{
MutexLock lock(&volume_settings_mutex_);
output_gain = _outputGain;
}
// Output volume scaling
if (output_gain < 0.99f || output_gain > 1.01f) {
// TODO(solenberg): Combine with mute state - this can cause clicks!
AudioFrameOperations::ScaleWithSat(output_gain, audio_frame);
}
// Measure audio level (0-9)
// TODO(henrik.lundin) Use the |muted| information here too.
// TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| (see
// https://crbug.com/webrtc/7517).
_outputAudioLevel.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds);
if (capture_start_rtp_time_stamp_ < 0 && audio_frame->timestamp_ != 0) {
// The first frame with a valid rtp timestamp.
capture_start_rtp_time_stamp_ = audio_frame->timestamp_;
}
if (capture_start_rtp_time_stamp_ >= 0) {
// audio_frame.timestamp_ should be valid from now on.
// Compute elapsed time.
int64_t unwrap_timestamp =
rtp_ts_wraparound_handler_->Unwrap(audio_frame->timestamp_);
audio_frame->elapsed_time_ms_ =
(unwrap_timestamp - capture_start_rtp_time_stamp_) /
(GetRtpTimestampRateHz() / 1000);
{
MutexLock lock(&ts_stats_lock_);
// Compute ntp time.
audio_frame->ntp_time_ms_ =
ntp_estimator_.Estimate(audio_frame->timestamp_);
// |ntp_time_ms_| won't be valid until at least 2 RTCP SRs are received.
if (audio_frame->ntp_time_ms_ > 0) {
// Compute |capture_start_ntp_time_ms_| so that
// |capture_start_ntp_time_ms_| + |elapsed_time_ms_| == |ntp_time_ms_|
capture_start_ntp_time_ms_ =
audio_frame->ntp_time_ms_ - audio_frame->elapsed_time_ms_;
}
}
}
// Fill in local capture clock offset in |audio_frame->packet_infos_|.
RtpPacketInfos::vector_type packet_infos;
for (auto& packet_info : audio_frame->packet_infos_) {
absl::optional<int64_t> local_capture_clock_offset;
if (packet_info.absolute_capture_time().has_value()) {
local_capture_clock_offset =
capture_clock_offset_updater_.AdjustEstimatedCaptureClockOffset(
packet_info.absolute_capture_time()
->estimated_capture_clock_offset);
}
RtpPacketInfo new_packet_info(packet_info);
new_packet_info.set_local_capture_clock_offset(local_capture_clock_offset);
packet_infos.push_back(std::move(new_packet_info));
}
audio_frame->packet_infos_ = RtpPacketInfos(packet_infos);
{
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.TargetJitterBufferDelayMs",
acm_receiver_.TargetDelayMs());
const int jitter_buffer_delay = acm_receiver_.FilteredCurrentDelayMs();
MutexLock lock(&video_sync_lock_);
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDelayEstimateMs",
jitter_buffer_delay + playout_delay_ms_);
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverJitterBufferDelayMs",
jitter_buffer_delay);
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDeviceDelayMs",
playout_delay_ms_);
}
return muted ? AudioMixer::Source::AudioFrameInfo::kMuted
: AudioMixer::Source::AudioFrameInfo::kNormal;
}
int ChannelReceive::PreferredSampleRate() const {
RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
// Return the bigger of playout and receive frequency in the ACM.
return std::max(acm_receiver_.last_packet_sample_rate_hz().value_or(0),
acm_receiver_.last_output_sample_rate_hz());
}
void ChannelReceive::SetSourceTracker(SourceTracker* source_tracker) {
source_tracker_ = source_tracker;
}
ChannelReceive::ChannelReceive(
Clock* clock,
ProcessThread* module_process_thread,
NetEqFactory* neteq_factory,
AudioDeviceModule* audio_device_module,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer)
: event_log_(rtc_event_log),
rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
remote_ssrc_(remote_ssrc),
acm_receiver_(AcmConfig(neteq_factory,
decoder_factory,
codec_pair_id,
jitter_buffer_max_packets,
jitter_buffer_fast_playout)),
_outputAudioLevel(),
clock_(clock),
ntp_estimator_(clock),
playout_timestamp_rtp_(0),
playout_delay_ms_(0),
rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
capture_start_rtp_time_stamp_(-1),
capture_start_ntp_time_ms_(-1),
module_process_thread_(module_process_thread),
_audioDeviceModulePtr(audio_device_module),
_outputGain(1.0f),
associated_send_channel_(nullptr),
frame_decryptor_(frame_decryptor),
crypto_options_(crypto_options),
absolute_capture_time_interpolator_(clock) {
RTC_DCHECK(module_process_thread_);
RTC_DCHECK(audio_device_module);
acm_receiver_.ResetInitialDelay();
acm_receiver_.SetMinimumDelay(0);
acm_receiver_.SetMaximumDelay(0);
acm_receiver_.FlushBuffers();
_outputAudioLevel.ResetLevelFullRange();
rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc_, true);
RtpRtcpInterface::Configuration configuration;
configuration.clock = clock;
configuration.audio = true;
configuration.receiver_only = true;
configuration.outgoing_transport = rtcp_send_transport;
configuration.receive_statistics = rtp_receive_statistics_.get();
configuration.event_log = event_log_;
configuration.local_media_ssrc = local_ssrc;
if (frame_transformer)
InitFrameTransformerDelegate(std::move(frame_transformer));
rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration);
rtp_rtcp_->SetSendingMediaStatus(false);
rtp_rtcp_->SetRemoteSSRC(remote_ssrc_);
// Ensure that RTCP is enabled for the created channel.
rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
// TODO(tommi): This should be an implementation detail of ModuleRtpRtcpImpl2
// and the pointer to the process thread should be there (which also localizes
// the problem of getting rid of that dependency).
module_process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
}
ChannelReceive::~ChannelReceive() {
RTC_DCHECK(construction_thread_.IsCurrent());
// Unregister the module before stopping playout etc, to match the order
// things were set up in the ctor.
module_process_thread_->DeRegisterModule(rtp_rtcp_.get());
// Resets the delegate's callback to ChannelReceive::OnReceivedPayloadData.
if (frame_transformer_delegate_)
frame_transformer_delegate_->Reset();
StopPlayout();
}
void ChannelReceive::SetSink(AudioSinkInterface* sink) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
MutexLock lock(&callback_mutex_);
audio_sink_ = sink;
}
void ChannelReceive::StartPlayout() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
MutexLock lock(&playing_lock_);
playing_ = true;
}
void ChannelReceive::StopPlayout() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
MutexLock lock(&playing_lock_);
playing_ = false;
_outputAudioLevel.ResetLevelFullRange();
}
absl::optional<std::pair<int, SdpAudioFormat>> ChannelReceive::GetReceiveCodec()
const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return acm_receiver_.LastDecoder();
}
void ChannelReceive::SetReceiveCodecs(
const std::map<int, SdpAudioFormat>& codecs) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
for (const auto& kv : codecs) {
RTC_DCHECK_GE(kv.second.clockrate_hz, 1000);
payload_type_frequencies_[kv.first] = kv.second.clockrate_hz;
}
acm_receiver_.SetCodecs(codecs);
}
void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// TODO(bugs.webrtc.org/11993): Expect to be called exclusively on the
// network thread. Once that's done, the same applies to
// UpdatePlayoutTimestamp and
int64_t now_ms = rtc::TimeMillis();
{
MutexLock lock(&sync_info_lock_);
last_received_rtp_timestamp_ = packet.Timestamp();
last_received_rtp_system_time_ms_ = now_ms;
}
// Store playout timestamp for the received RTP packet
UpdatePlayoutTimestamp(false, now_ms);
const auto& it = payload_type_frequencies_.find(packet.PayloadType());
if (it == payload_type_frequencies_.end())
return;
// TODO(nisse): Set payload_type_frequency earlier, when packet is parsed.
RtpPacketReceived packet_copy(packet);
packet_copy.set_payload_type_frequency(it->second);
rtp_receive_statistics_->OnRtpPacket(packet_copy);
RTPHeader header;
packet_copy.GetHeader(&header);
// Interpolates absolute capture timestamp RTP header extension.
header.extension.absolute_capture_time =
absolute_capture_time_interpolator_.OnReceivePacket(
AbsoluteCaptureTimeInterpolator::GetSource(header.ssrc,
header.arrOfCSRCs),
header.timestamp,
rtc::saturated_cast<uint32_t>(packet_copy.payload_type_frequency()),
header.extension.absolute_capture_time);
ReceivePacket(packet_copy.data(), packet_copy.size(), header);
}
void ChannelReceive::ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header) {
const uint8_t* payload = packet + header.headerLength;
assert(packet_length >= header.headerLength);
size_t payload_length = packet_length - header.headerLength;
size_t payload_data_length = payload_length - header.paddingLength;
// E2EE Custom Audio Frame Decryption (This is optional).
// Keep this buffer around for the lifetime of the OnReceivedPayloadData call.
rtc::Buffer decrypted_audio_payload;
if (frame_decryptor_ != nullptr) {
const size_t max_plaintext_size = frame_decryptor_->GetMaxPlaintextByteSize(
cricket::MEDIA_TYPE_AUDIO, payload_length);
decrypted_audio_payload.SetSize(max_plaintext_size);
const std::vector<uint32_t> csrcs(header.arrOfCSRCs,
header.arrOfCSRCs + header.numCSRCs);
const FrameDecryptorInterface::Result decrypt_result =
frame_decryptor_->Decrypt(
cricket::MEDIA_TYPE_AUDIO, csrcs,
/*additional_data=*/nullptr,
rtc::ArrayView<const uint8_t>(payload, payload_data_length),
decrypted_audio_payload);
if (decrypt_result.IsOk()) {
decrypted_audio_payload.SetSize(decrypt_result.bytes_written);
} else {
// Interpret failures as a silent frame.
decrypted_audio_payload.SetSize(0);
}
payload = decrypted_audio_payload.data();
payload_data_length = decrypted_audio_payload.size();
} else if (crypto_options_.sframe.require_frame_encryption) {
RTC_DLOG(LS_ERROR)
<< "FrameDecryptor required but not set, dropping packet";
payload_data_length = 0;
}
rtc::ArrayView<const uint8_t> payload_data(payload, payload_data_length);
if (frame_transformer_delegate_) {
// Asynchronously transform the received payload. After the payload is
// transformed, the delegate will call OnReceivedPayloadData to handle it.
frame_transformer_delegate_->Transform(payload_data, header, remote_ssrc_);
} else {
OnReceivedPayloadData(payload_data, header);
}
}
void ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// TODO(bugs.webrtc.org/11993): Expect to be called exclusively on the
// network thread.
// Store playout timestamp for the received RTCP packet
UpdatePlayoutTimestamp(true, rtc::TimeMillis());
// Deliver RTCP packet to RTP/RTCP module for parsing
rtp_rtcp_->IncomingRtcpPacket(data, length);
int64_t rtt = GetRTT();
if (rtt == 0) {
// Waiting for valid RTT.
return;
}
uint32_t ntp_secs = 0;
uint32_t ntp_frac = 0;
uint32_t rtp_timestamp = 0;
if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac,
/*rtcp_arrival_time_secs=*/nullptr,
/*rtcp_arrival_time_frac=*/nullptr,
&rtp_timestamp) != 0) {
// Waiting for RTCP.
return;
}
{
MutexLock lock(&ts_stats_lock_);
ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
absl::optional<int64_t> remote_to_local_clock_offset_ms =
ntp_estimator_.EstimateRemoteToLocalClockOffsetMs();
if (remote_to_local_clock_offset_ms.has_value()) {
capture_clock_offset_updater_.SetRemoteToLocalClockOffset(
Int64MsToQ32x32(*remote_to_local_clock_offset_ms));
}
}
}
int ChannelReceive::GetSpeechOutputLevelFullRange() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return _outputAudioLevel.LevelFullRange();
}
double ChannelReceive::GetTotalOutputEnergy() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return _outputAudioLevel.TotalEnergy();
}
double ChannelReceive::GetTotalOutputDuration() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return _outputAudioLevel.TotalDuration();
}
void ChannelReceive::SetChannelOutputVolumeScaling(float scaling) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
MutexLock lock(&volume_settings_mutex_);
_outputGain = scaling;
}
void ChannelReceive::RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK(packet_router);
RTC_DCHECK(!packet_router_);
constexpr bool remb_candidate = false;
packet_router->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
packet_router_ = packet_router;
}
void ChannelReceive::ResetReceiverCongestionControlObjects() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK(packet_router_);
packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get());
packet_router_ = nullptr;
}
CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
CallReceiveStatistics stats;
// The jitter statistics is updated for each received RTP packet and is based
// on received packets.
RtpReceiveStats rtp_stats;
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(remote_ssrc_);
if (statistician) {
rtp_stats = statistician->GetStats();
}
stats.cumulativeLost = rtp_stats.packets_lost;
stats.jitterSamples = rtp_stats.jitter;
stats.rttMs = GetRTT();
// Data counters.
if (statistician) {
stats.payload_bytes_rcvd = rtp_stats.packet_counter.payload_bytes;
stats.header_and_padding_bytes_rcvd =
rtp_stats.packet_counter.header_bytes +
rtp_stats.packet_counter.padding_bytes;
stats.packetsReceived = rtp_stats.packet_counter.packets;
stats.last_packet_received_timestamp_ms =
rtp_stats.last_packet_received_timestamp_ms;
} else {
stats.payload_bytes_rcvd = 0;
stats.header_and_padding_bytes_rcvd = 0;
stats.packetsReceived = 0;
stats.last_packet_received_timestamp_ms = absl::nullopt;
}
// Timestamps.
{
MutexLock lock(&ts_stats_lock_);
stats.capture_start_ntp_time_ms_ = capture_start_ntp_time_ms_;
}
absl::optional<RtpRtcpInterface::SenderReportStats> rtcp_sr_stats =
rtp_rtcp_->GetSenderReportStats();
if (rtcp_sr_stats.has_value()) {
// Number of seconds since 1900 January 1 00:00 GMT (see
// https://tools.ietf.org/html/rfc868).
constexpr int64_t kNtpJan1970Millisecs =
2208988800 * rtc::kNumMillisecsPerSec;
stats.last_sender_report_timestamp_ms =
rtcp_sr_stats->last_arrival_timestamp.ToMs() - kNtpJan1970Millisecs;
stats.last_sender_report_remote_timestamp_ms =
rtcp_sr_stats->last_remote_timestamp.ToMs() - kNtpJan1970Millisecs;
stats.sender_reports_packets_sent = rtcp_sr_stats->packets_sent;
stats.sender_reports_bytes_sent = rtcp_sr_stats->bytes_sent;
stats.sender_reports_reports_count = rtcp_sr_stats->reports_count;
}
return stats;
}
void ChannelReceive::SetNACKStatus(bool enable, int max_packets) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// None of these functions can fail.
if (enable) {
rtp_receive_statistics_->SetMaxReorderingThreshold(max_packets);
acm_receiver_.EnableNack(max_packets);
} else {
rtp_receive_statistics_->SetMaxReorderingThreshold(
kDefaultMaxReorderingThreshold);
acm_receiver_.DisableNack();
}
}
// Called when we are missing one or more packets.
int ChannelReceive::ResendPackets(const uint16_t* sequence_numbers,
int length) {
return rtp_rtcp_->SendNACK(sequence_numbers, length);
}
void ChannelReceive::SetAssociatedSendChannel(
const ChannelSendInterface* channel) {
// TODO(bugs.webrtc.org/11993): Expect to be called on the network thread.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
associated_send_channel_ = channel;
}
void ChannelReceive::SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// Depending on when the channel is created, the transformer might be set
// twice. Don't replace the delegate if it was already initialized.
if (!frame_transformer || frame_transformer_delegate_)
return;
InitFrameTransformerDelegate(std::move(frame_transformer));
}
NetworkStatistics ChannelReceive::GetNetworkStatistics(
bool get_and_clear_legacy_stats) const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
NetworkStatistics stats;
acm_receiver_.GetNetworkStatistics(&stats, get_and_clear_legacy_stats);
return stats;
}
AudioDecodingCallStats ChannelReceive::GetDecodingCallStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
AudioDecodingCallStats stats;
acm_receiver_.GetDecodingCallStatistics(&stats);
return stats;
}
uint32_t ChannelReceive::GetDelayEstimate() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
uint32_t playout_delay;
{
MutexLock lock(&video_sync_lock_);
playout_delay = playout_delay_ms_;
}
// Return the current jitter buffer delay + playout delay.
return acm_receiver_.FilteredCurrentDelayMs() + playout_delay;
}
bool ChannelReceive::SetMinimumPlayoutDelay(int delay_ms) {
// TODO(bugs.webrtc.org/11993): This should run on the network thread.
// We get here via RtpStreamsSynchronizer. Once that's done, many (all?) of
// these locks aren't needed.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// Limit to range accepted by both VoE and ACM, so we're at least getting as
// close as possible, instead of failing.
delay_ms = rtc::SafeClamp(delay_ms, kVoiceEngineMinMinPlayoutDelayMs,
kVoiceEngineMaxMinPlayoutDelayMs);
if (acm_receiver_.SetMinimumDelay(delay_ms) != 0) {
RTC_DLOG(LS_ERROR)
<< "SetMinimumPlayoutDelay() failed to set min playout delay";
return false;
}
return true;
}
bool ChannelReceive::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const {
RTC_DCHECK_RUNS_SERIALIZED(&video_capture_thread_race_checker_);
{
MutexLock lock(&video_sync_lock_);
if (!playout_timestamp_rtp_time_ms_)
return false;
*rtp_timestamp = playout_timestamp_rtp_;
*time_ms = playout_timestamp_rtp_time_ms_.value();
return true;
}
}
void ChannelReceive::SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
int64_t time_ms) {
RTC_DCHECK_RUNS_SERIALIZED(&video_capture_thread_race_checker_);
MutexLock lock(&video_sync_lock_);
playout_timestamp_ntp_ = ntp_timestamp_ms;
playout_timestamp_ntp_time_ms_ = time_ms;
}
absl::optional<int64_t>
ChannelReceive::GetCurrentEstimatedPlayoutNtpTimestampMs(int64_t now_ms) const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
MutexLock lock(&video_sync_lock_);
if (!playout_timestamp_ntp_ || !playout_timestamp_ntp_time_ms_)
return absl::nullopt;
int64_t elapsed_ms = now_ms - *playout_timestamp_ntp_time_ms_;
return *playout_timestamp_ntp_ + elapsed_ms;
}
bool ChannelReceive::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
return acm_receiver_.SetBaseMinimumDelayMs(delay_ms);
}
int ChannelReceive::GetBaseMinimumPlayoutDelayMs() const {
return acm_receiver_.GetBaseMinimumDelayMs();
}
absl::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const {
// TODO(bugs.webrtc.org/11993): This should run on the network thread.
// We get here via RtpStreamsSynchronizer. Once that's done, many of
// these locks aren't needed.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
Syncable::Info info;
if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs,
&info.capture_time_ntp_frac,
/*rtcp_arrival_time_secs=*/nullptr,
/*rtcp_arrival_time_frac=*/nullptr,
&info.capture_time_source_clock) != 0) {
return absl::nullopt;
}
{
MutexLock lock(&sync_info_lock_);
if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
return absl::nullopt;
}
info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
}
int jitter_buffer_delay = acm_receiver_.FilteredCurrentDelayMs();
{
MutexLock lock(&video_sync_lock_);
info.current_delay_ms = jitter_buffer_delay + playout_delay_ms_;
}
return info;
}
void ChannelReceive::UpdatePlayoutTimestamp(bool rtcp, int64_t now_ms) {
// TODO(bugs.webrtc.org/11993): Expect to be called exclusively on the
// network thread. Once that's done, we won't need video_sync_lock_.
jitter_buffer_playout_timestamp_ = acm_receiver_.GetPlayoutTimestamp();
if (!jitter_buffer_playout_timestamp_) {
// This can happen if this channel has not received any RTP packets. In
// this case, NetEq is not capable of computing a playout timestamp.
return;
}
uint16_t delay_ms = 0;
if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
RTC_DLOG(LS_WARNING)
<< "ChannelReceive::UpdatePlayoutTimestamp() failed to read"
" playout delay from the ADM";
return;
}
RTC_DCHECK(jitter_buffer_playout_timestamp_);
uint32_t playout_timestamp = *jitter_buffer_playout_timestamp_;
// Remove the playout delay.
playout_timestamp -= (delay_ms * (GetRtpTimestampRateHz() / 1000));
{
MutexLock lock(&video_sync_lock_);
if (!rtcp && playout_timestamp != playout_timestamp_rtp_) {
playout_timestamp_rtp_ = playout_timestamp;
playout_timestamp_rtp_time_ms_ = now_ms;
}
playout_delay_ms_ = delay_ms;
}
}
int ChannelReceive::GetRtpTimestampRateHz() const {
const auto decoder = acm_receiver_.LastDecoder();
// Default to the playout frequency if we've not gotten any packets yet.
// TODO(ossu): Zero clockrate can only happen if we've added an external
// decoder for a format we don't support internally. Remove once that way of
// adding decoders is gone!
// TODO(kwiberg): `decoder->second.clockrate_hz` is an RTP clockrate as it
// should, but `acm_receiver_.last_output_sample_rate_hz()` is a codec sample
// rate, which is not always the same thing.
return (decoder && decoder->second.clockrate_hz != 0)
? decoder->second.clockrate_hz
: acm_receiver_.last_output_sample_rate_hz();
}
int64_t ChannelReceive::GetRTT() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
std::vector<ReportBlockData> report_blocks =
rtp_rtcp_->GetLatestReportBlockData();
if (report_blocks.empty()) {
// Try fall back on an RTT from an associated channel.
if (!associated_send_channel_) {
return 0;
}
return associated_send_channel_->GetRTT();
}
// TODO(nisse): This method computes RTT based on sender reports, even though
// a receive stream is not supposed to do that.
for (const ReportBlockData& data : report_blocks) {
if (data.report_block().sender_ssrc == remote_ssrc_) {
return data.last_rtt_ms();
}
}
return 0;
}
} // namespace
std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
Clock* clock,
ProcessThread* module_process_thread,
NetEqFactory* neteq_factory,
AudioDeviceModule* audio_device_module,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
return std::make_unique<ChannelReceive>(
clock, module_process_thread, neteq_factory, audio_device_module,
rtcp_send_transport, rtc_event_log, local_ssrc, remote_ssrc,
jitter_buffer_max_packets, jitter_buffer_fast_playout,
jitter_buffer_min_delay_ms, jitter_buffer_enable_rtx_handling,
decoder_factory, codec_pair_id, frame_decryptor, crypto_options,
std::move(frame_transformer));
}
} // namespace voe
} // namespace webrtc
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