blob: d12edf37c3dfa2709914228adefa5ecaf7425b47 [file] [log] [blame]
/*
* Copyright (c) 2015 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/audio_receive_stream.h"
#include <string>
#include <utility>
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_format.h"
#include "api/call/audio_sink.h"
#include "api/rtp_parameters.h"
#include "api/sequence_checker.h"
#include "audio/audio_send_stream.h"
#include "audio/audio_state.h"
#include "audio/channel_receive.h"
#include "audio/conversion.h"
#include "call/rtp_config.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/time_utils.h"
namespace webrtc {
std::string AudioReceiveStream::Config::Rtp::ToString() const {
char ss_buf[1024];
rtc::SimpleStringBuilder ss(ss_buf);
ss << "{remote_ssrc: " << remote_ssrc;
ss << ", local_ssrc: " << local_ssrc;
ss << ", transport_cc: " << (transport_cc ? "on" : "off");
ss << ", nack: " << nack.ToString();
ss << ", extensions: [";
for (size_t i = 0; i < extensions.size(); ++i) {
ss << extensions[i].ToString();
if (i != extensions.size() - 1) {
ss << ", ";
}
}
ss << ']';
ss << '}';
return ss.str();
}
std::string AudioReceiveStream::Config::ToString() const {
char ss_buf[1024];
rtc::SimpleStringBuilder ss(ss_buf);
ss << "{rtp: " << rtp.ToString();
ss << ", rtcp_send_transport: "
<< (rtcp_send_transport ? "(Transport)" : "null");
if (!sync_group.empty()) {
ss << ", sync_group: " << sync_group;
}
ss << '}';
return ss.str();
}
namespace internal {
namespace {
std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
Clock* clock,
webrtc::AudioState* audio_state,
NetEqFactory* neteq_factory,
const webrtc::AudioReceiveStream::Config& config,
RtcEventLog* event_log) {
RTC_DCHECK(audio_state);
internal::AudioState* internal_audio_state =
static_cast<internal::AudioState*>(audio_state);
return voe::CreateChannelReceive(
clock, neteq_factory, internal_audio_state->audio_device_module(),
config.rtcp_send_transport, event_log, config.rtp.local_ssrc,
config.rtp.remote_ssrc, config.jitter_buffer_max_packets,
config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
config.jitter_buffer_enable_rtx_handling, config.enable_non_sender_rtt,
config.decoder_factory, config.codec_pair_id,
std::move(config.frame_decryptor), config.crypto_options,
std::move(config.frame_transformer));
}
} // namespace
AudioReceiveStream::AudioReceiveStream(
Clock* clock,
PacketRouter* packet_router,
NetEqFactory* neteq_factory,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log)
: AudioReceiveStream(clock,
packet_router,
config,
audio_state,
event_log,
CreateChannelReceive(clock,
audio_state.get(),
neteq_factory,
config,
event_log)) {}
AudioReceiveStream::AudioReceiveStream(
Clock* clock,
PacketRouter* packet_router,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log,
std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)
: config_(config),
audio_state_(audio_state),
source_tracker_(clock),
channel_receive_(std::move(channel_receive)) {
RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc;
RTC_DCHECK(config.decoder_factory);
RTC_DCHECK(config.rtcp_send_transport);
RTC_DCHECK(audio_state_);
RTC_DCHECK(channel_receive_);
packet_sequence_checker_.Detach();
RTC_DCHECK(packet_router);
// Configure bandwidth estimation.
channel_receive_->RegisterReceiverCongestionControlObjects(packet_router);
// When output is muted, ChannelReceive will directly notify the source
// tracker of "delivered" frames, so RtpReceiver information will continue to
// be updated.
channel_receive_->SetSourceTracker(&source_tracker_);
// Complete configuration.
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
channel_receive_->SetNACKStatus(config.rtp.nack.rtp_history_ms != 0,
config.rtp.nack.rtp_history_ms / 20);
channel_receive_->SetReceiveCodecs(config.decoder_map);
// `frame_transformer` and `frame_decryptor` have been given to
// `channel_receive_` already.
}
AudioReceiveStream::~AudioReceiveStream() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << remote_ssrc();
Stop();
channel_receive_->SetAssociatedSendChannel(nullptr);
channel_receive_->ResetReceiverCongestionControlObjects();
}
void AudioReceiveStream::RegisterWithTransport(
RtpStreamReceiverControllerInterface* receiver_controller) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK(!rtp_stream_receiver_);
rtp_stream_receiver_ = receiver_controller->CreateReceiver(
remote_ssrc(), channel_receive_.get());
}
void AudioReceiveStream::UnregisterFromTransport() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_stream_receiver_.reset();
}
void AudioReceiveStream::ReconfigureForTesting(
const webrtc::AudioReceiveStream::Config& config) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
// SSRC can't be changed mid-stream.
RTC_DCHECK_EQ(remote_ssrc(), config.rtp.remote_ssrc);
RTC_DCHECK_EQ(local_ssrc(), config.rtp.local_ssrc);
// Configuration parameters which cannot be changed.
RTC_DCHECK_EQ(config_.rtcp_send_transport, config.rtcp_send_transport);
// Decoder factory cannot be changed because it is configured at
// voe::Channel construction time.
RTC_DCHECK_EQ(config_.decoder_factory, config.decoder_factory);
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
RTC_DCHECK_EQ(config_.rtp.nack.rtp_history_ms, config.rtp.nack.rtp_history_ms)
<< "Use SetUseTransportCcAndNackHistory";
RTC_DCHECK(config_.decoder_map == config.decoder_map) << "Use SetDecoderMap";
RTC_DCHECK_EQ(config_.frame_transformer, config.frame_transformer)
<< "Use SetDepacketizerToDecoderFrameTransformer";
config_ = config;
}
void AudioReceiveStream::Start() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (playing_) {
return;
}
channel_receive_->StartPlayout();
playing_ = true;
audio_state()->AddReceivingStream(this);
}
void AudioReceiveStream::Stop() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (!playing_) {
return;
}
channel_receive_->StopPlayout();
playing_ = false;
audio_state()->RemoveReceivingStream(this);
}
bool AudioReceiveStream::transport_cc() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return config_.rtp.transport_cc;
}
bool AudioReceiveStream::IsRunning() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return playing_;
}
void AudioReceiveStream::SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_receive_->SetDepacketizerToDecoderFrameTransformer(
std::move(frame_transformer));
}
void AudioReceiveStream::SetDecoderMap(
std::map<int, SdpAudioFormat> decoder_map) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
config_.decoder_map = std::move(decoder_map);
channel_receive_->SetReceiveCodecs(config_.decoder_map);
}
void AudioReceiveStream::SetUseTransportCcAndNackHistory(bool use_transport_cc,
int history_ms) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_GE(history_ms, 0);
config_.rtp.transport_cc = use_transport_cc;
if (config_.rtp.nack.rtp_history_ms != history_ms) {
config_.rtp.nack.rtp_history_ms = history_ms;
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
channel_receive_->SetNACKStatus(history_ms != 0, history_ms / 20);
}
}
void AudioReceiveStream::SetNonSenderRttMeasurement(bool enabled) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
config_.enable_non_sender_rtt = enabled;
channel_receive_->SetNonSenderRttMeasurement(enabled);
}
void AudioReceiveStream::SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
// TODO(bugs.webrtc.org/11993): This is called via WebRtcAudioReceiveStream,
// expect to be called on the network thread.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_receive_->SetFrameDecryptor(std::move(frame_decryptor));
}
void AudioReceiveStream::SetRtpExtensions(
std::vector<RtpExtension> extensions) {
// TODO(bugs.webrtc.org/11993): This is called via WebRtcAudioReceiveStream,
// expect to be called on the network thread.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
config_.rtp.extensions = std::move(extensions);
}
const std::vector<RtpExtension>& AudioReceiveStream::GetRtpExtensions() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return config_.rtp.extensions;
}
RtpHeaderExtensionMap AudioReceiveStream::GetRtpExtensionMap() const {
return RtpHeaderExtensionMap(config_.rtp.extensions);
}
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats(
bool get_and_clear_legacy_stats) const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
webrtc::AudioReceiveStream::Stats stats;
stats.remote_ssrc = remote_ssrc();
webrtc::CallReceiveStatistics call_stats =
channel_receive_->GetRTCPStatistics();
// TODO(solenberg): Don't return here if we can't get the codec - return the
// stats we *can* get.
auto receive_codec = channel_receive_->GetReceiveCodec();
if (!receive_codec) {
return stats;
}
stats.payload_bytes_rcvd = call_stats.payload_bytes_rcvd;
stats.header_and_padding_bytes_rcvd =
call_stats.header_and_padding_bytes_rcvd;
stats.packets_rcvd = call_stats.packetsReceived;
stats.packets_lost = call_stats.cumulativeLost;
stats.nacks_sent = call_stats.nacks_sent;
stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
stats.last_packet_received_timestamp_ms =
call_stats.last_packet_received_timestamp_ms;
stats.codec_name = receive_codec->second.name;
stats.codec_payload_type = receive_codec->first;
int clockrate_khz = receive_codec->second.clockrate_hz / 1000;
if (clockrate_khz > 0) {
stats.jitter_ms = call_stats.jitterSamples / clockrate_khz;
}
stats.delay_estimate_ms = channel_receive_->GetDelayEstimate();
stats.audio_level = channel_receive_->GetSpeechOutputLevelFullRange();
stats.total_output_energy = channel_receive_->GetTotalOutputEnergy();
stats.total_output_duration = channel_receive_->GetTotalOutputDuration();
stats.estimated_playout_ntp_timestamp_ms =
channel_receive_->GetCurrentEstimatedPlayoutNtpTimestampMs(
rtc::TimeMillis());
// Get jitter buffer and total delay (alg + jitter + playout) stats.
auto ns = channel_receive_->GetNetworkStatistics(get_and_clear_legacy_stats);
stats.packets_discarded = ns.packetsDiscarded;
stats.fec_packets_received = ns.fecPacketsReceived;
stats.fec_packets_discarded = ns.fecPacketsDiscarded;
stats.jitter_buffer_ms = ns.currentBufferSize;
stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
stats.total_samples_received = ns.totalSamplesReceived;
stats.concealed_samples = ns.concealedSamples;
stats.silent_concealed_samples = ns.silentConcealedSamples;
stats.concealment_events = ns.concealmentEvents;
stats.jitter_buffer_delay_seconds =
static_cast<double>(ns.jitterBufferDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec);
stats.jitter_buffer_emitted_count = ns.jitterBufferEmittedCount;
stats.jitter_buffer_target_delay_seconds =
static_cast<double>(ns.jitterBufferTargetDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec);
stats.inserted_samples_for_deceleration = ns.insertedSamplesForDeceleration;
stats.removed_samples_for_acceleration = ns.removedSamplesForAcceleration;
stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
stats.secondary_discarded_rate = Q14ToFloat(ns.currentSecondaryDiscardedRate);
stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
stats.jitter_buffer_flushes = ns.packetBufferFlushes;
stats.delayed_packet_outage_samples = ns.delayedPacketOutageSamples;
stats.relative_packet_arrival_delay_seconds =
static_cast<double>(ns.relativePacketArrivalDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec);
stats.interruption_count = ns.interruptionCount;
stats.total_interruption_duration_ms = ns.totalInterruptionDurationMs;
auto ds = channel_receive_->GetDecodingCallStatistics();
stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
stats.decoding_calls_to_neteq = ds.calls_to_neteq;
stats.decoding_normal = ds.decoded_normal;
stats.decoding_plc = ds.decoded_neteq_plc;
stats.decoding_codec_plc = ds.decoded_codec_plc;
stats.decoding_cng = ds.decoded_cng;
stats.decoding_plc_cng = ds.decoded_plc_cng;
stats.decoding_muted_output = ds.decoded_muted_output;
stats.last_sender_report_timestamp_ms =
call_stats.last_sender_report_timestamp_ms;
stats.last_sender_report_remote_timestamp_ms =
call_stats.last_sender_report_remote_timestamp_ms;
stats.sender_reports_packets_sent = call_stats.sender_reports_packets_sent;
stats.sender_reports_bytes_sent = call_stats.sender_reports_bytes_sent;
stats.sender_reports_reports_count = call_stats.sender_reports_reports_count;
stats.round_trip_time = call_stats.round_trip_time;
stats.round_trip_time_measurements = call_stats.round_trip_time_measurements;
stats.total_round_trip_time = call_stats.total_round_trip_time;
return stats;
}
void AudioReceiveStream::SetSink(AudioSinkInterface* sink) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_receive_->SetSink(sink);
}
void AudioReceiveStream::SetGain(float gain) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_receive_->SetChannelOutputVolumeScaling(gain);
}
bool AudioReceiveStream::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return channel_receive_->SetBaseMinimumPlayoutDelayMs(delay_ms);
}
int AudioReceiveStream::GetBaseMinimumPlayoutDelayMs() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return channel_receive_->GetBaseMinimumPlayoutDelayMs();
}
std::vector<RtpSource> AudioReceiveStream::GetSources() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return source_tracker_.GetSources();
}
AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) {
AudioMixer::Source::AudioFrameInfo audio_frame_info =
channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
if (audio_frame_info != AudioMixer::Source::AudioFrameInfo::kError) {
source_tracker_.OnFrameDelivered(audio_frame->packet_infos_);
}
return audio_frame_info;
}
int AudioReceiveStream::Ssrc() const {
return remote_ssrc();
}
int AudioReceiveStream::PreferredSampleRate() const {
return channel_receive_->PreferredSampleRate();
}
uint32_t AudioReceiveStream::id() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return remote_ssrc();
}
absl::optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
// TODO(bugs.webrtc.org/11993): This is called via RtpStreamsSynchronizer,
// expect to be called on the network thread.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return channel_receive_->GetSyncInfo();
}
bool AudioReceiveStream::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const {
// Called on video capture thread.
return channel_receive_->GetPlayoutRtpTimestamp(rtp_timestamp, time_ms);
}
void AudioReceiveStream::SetEstimatedPlayoutNtpTimestampMs(
int64_t ntp_timestamp_ms,
int64_t time_ms) {
// Called on video capture thread.
channel_receive_->SetEstimatedPlayoutNtpTimestampMs(ntp_timestamp_ms,
time_ms);
}
bool AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
// TODO(bugs.webrtc.org/11993): This is called via RtpStreamsSynchronizer,
// expect to be called on the network thread.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return channel_receive_->SetMinimumPlayoutDelay(delay_ms);
}
void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
channel_receive_->SetAssociatedSendChannel(
send_stream ? send_stream->GetChannel() : nullptr);
associated_send_stream_ = send_stream;
}
void AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
// TODO(solenberg): Tests call this function on a network thread, libjingle
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
// RTC_DCHECK(!thread_checker_.IsCurrent());
channel_receive_->ReceivedRTCPPacket(packet, length);
}
void AudioReceiveStream::SetSyncGroup(absl::string_view sync_group) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
config_.sync_group = std::string(sync_group);
}
void AudioReceiveStream::SetLocalSsrc(uint32_t local_ssrc) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
// TODO(tommi): Consider storing local_ssrc in one place.
config_.rtp.local_ssrc = local_ssrc;
channel_receive_->OnLocalSsrcChange(local_ssrc);
}
uint32_t AudioReceiveStream::local_ssrc() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK_EQ(config_.rtp.local_ssrc, channel_receive_->GetLocalSsrc());
return config_.rtp.local_ssrc;
}
const std::string& AudioReceiveStream::sync_group() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return config_.sync_group;
}
const AudioSendStream* AudioReceiveStream::GetAssociatedSendStreamForTesting()
const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return associated_send_stream_;
}
internal::AudioState* AudioReceiveStream::audio_state() const {
auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
RTC_DCHECK(audio_state);
return audio_state;
}
} // namespace internal
} // namespace webrtc