blob: a1233f2910ab1003886046ff9c3540d16c74cd84 [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 "api/call/audio_sink.h"
#include "audio/audio_send_stream.h"
#include "audio/audio_state.h"
#include "audio/conversion.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/include/rtp_receiver.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/timeutils.h"
#include "voice_engine/channel_proxy.h"
#include "voice_engine/include/voe_base.h"
#include "voice_engine/voice_engine_impl.h"
namespace webrtc {
std::string AudioReceiveStream::Config::Rtp::ToString() const {
std::stringstream ss;
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 {
std::stringstream ss;
ss << "{rtp: " << rtp.ToString();
ss << ", rtcp_send_transport: "
<< (rtcp_send_transport ? "(Transport)" : "null");
ss << ", voe_channel_id: " << voe_channel_id;
if (!sync_group.empty()) {
ss << ", sync_group: " << sync_group;
}
ss << '}';
return ss.str();
}
namespace internal {
AudioReceiveStream::AudioReceiveStream(
RtpStreamReceiverControllerInterface* receiver_controller,
PacketRouter* packet_router,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log)
: config_(config), audio_state_(audio_state) {
RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
RTC_DCHECK_NE(config_.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
RTC_DCHECK(packet_router);
module_process_thread_checker_.DetachFromThread();
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
channel_proxy_->SetRtcEventLog(event_log);
channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
config_.rtp.nack.rtp_history_ms / 20);
// TODO(ossu): This is where we'd like to set the decoder factory to
// use. However, since it needs to be included when constructing Channel, we
// cannot do that until we're able to move Channel ownership into the
// Audio{Send,Receive}Streams. The best we can do is check that we're not
// trying to use two different factories using the different interfaces.
RTC_CHECK(config.decoder_factory);
RTC_CHECK_EQ(config.decoder_factory,
channel_proxy_->GetAudioDecoderFactory());
channel_proxy_->RegisterTransport(config.rtcp_send_transport);
channel_proxy_->SetReceiveCodecs(config.decoder_map);
for (const auto& extension : config.rtp.extensions) {
if (extension.uri == RtpExtension::kAudioLevelUri) {
channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
} else {
RTC_NOTREACHED() << "Unsupported RTP extension.";
}
}
// Configure bandwidth estimation.
channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router);
// Register with transport.
rtp_stream_receiver_ =
receiver_controller->CreateReceiver(config_.rtp.remote_ssrc,
channel_proxy_.get());
}
AudioReceiveStream::~AudioReceiveStream() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
if (playing_) {
Stop();
}
channel_proxy_->DisassociateSendChannel();
channel_proxy_->RegisterTransport(nullptr);
channel_proxy_->ResetReceiverCongestionControlObjects();
channel_proxy_->SetRtcEventLog(nullptr);
}
void AudioReceiveStream::Start() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (playing_) {
return;
}
int error = SetVoiceEnginePlayout(true);
if (error != 0) {
RTC_LOG(LS_ERROR) << "AudioReceiveStream::Start failed with error: "
<< error;
return;
}
if (!audio_state()->mixer()->AddSource(this)) {
RTC_LOG(LS_ERROR) << "Failed to add source to mixer.";
SetVoiceEnginePlayout(false);
return;
}
playing_ = true;
}
void AudioReceiveStream::Stop() {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (!playing_) {
return;
}
playing_ = false;
audio_state()->mixer()->RemoveSource(this);
SetVoiceEnginePlayout(false);
}
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
webrtc::AudioReceiveStream::Stats stats;
stats.remote_ssrc = config_.rtp.remote_ssrc;
webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
// TODO(solenberg): Don't return here if we can't get the codec - return the
// stats we *can* get.
webrtc::CodecInst codec_inst = {0};
if (!channel_proxy_->GetRecCodec(&codec_inst)) {
return stats;
}
stats.bytes_rcvd = call_stats.bytesReceived;
stats.packets_rcvd = call_stats.packetsReceived;
stats.packets_lost = call_stats.cumulativeLost;
stats.fraction_lost = Q8ToFloat(call_stats.fractionLost);
stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
if (codec_inst.pltype != -1) {
stats.codec_name = codec_inst.plname;
stats.codec_payload_type = codec_inst.pltype;
}
stats.ext_seqnum = call_stats.extendedMax;
if (codec_inst.plfreq / 1000 > 0) {
stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000);
}
stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate();
stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange();
stats.total_output_energy = channel_proxy_->GetTotalOutputEnergy();
stats.total_output_duration = channel_proxy_->GetTotalOutputDuration();
// Get jitter buffer and total delay (alg + jitter + playout) stats.
auto ns = channel_proxy_->GetNetworkStatistics();
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.concealment_events = ns.concealmentEvents;
stats.jitter_buffer_delay_seconds =
static_cast<double>(ns.jitterBufferDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec);
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);
auto ds = channel_proxy_->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_plc;
stats.decoding_cng = ds.decoded_cng;
stats.decoding_plc_cng = ds.decoded_plc_cng;
stats.decoding_muted_output = ds.decoded_muted_output;
return stats;
}
int AudioReceiveStream::GetOutputLevel() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return channel_proxy_->GetSpeechOutputLevel();
}
void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_proxy_->SetSink(std::move(sink));
}
void AudioReceiveStream::SetGain(float gain) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_proxy_->SetChannelOutputVolumeScaling(gain);
}
std::vector<RtpSource> AudioReceiveStream::GetSources() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return channel_proxy_->GetSources();
}
AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) {
return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
}
int AudioReceiveStream::Ssrc() const {
return config_.rtp.remote_ssrc;
}
int AudioReceiveStream::PreferredSampleRate() const {
return channel_proxy_->PreferredSampleRate();
}
int AudioReceiveStream::id() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return config_.rtp.remote_ssrc;
}
rtc::Optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
Syncable::Info info;
RtpRtcp* rtp_rtcp = nullptr;
RtpReceiver* rtp_receiver = nullptr;
channel_proxy_->GetRtpRtcp(&rtp_rtcp, &rtp_receiver);
RTC_DCHECK(rtp_rtcp);
RTC_DCHECK(rtp_receiver);
if (!rtp_receiver->GetLatestTimestamps(
&info.latest_received_capture_timestamp,
&info.latest_receive_time_ms)) {
return rtc::nullopt;
}
if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs,
&info.capture_time_ntp_frac,
nullptr,
nullptr,
&info.capture_time_source_clock) != 0) {
return rtc::nullopt;
}
info.current_delay_ms = channel_proxy_->GetDelayEstimate();
return info;
}
uint32_t AudioReceiveStream::GetPlayoutTimestamp() const {
// Called on video capture thread.
return channel_proxy_->GetPlayoutTimestamp();
}
void AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
return channel_proxy_->SetMinimumPlayoutDelay(delay_ms);
}
void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (send_stream) {
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
std::unique_ptr<voe::ChannelProxy> send_channel_proxy =
voe_impl->GetChannelProxy(send_stream->GetConfig().voe_channel_id);
channel_proxy_->AssociateSendChannel(*send_channel_proxy.get());
} else {
channel_proxy_->DisassociateSendChannel();
}
}
void AudioReceiveStream::SignalNetworkState(NetworkState state) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
}
bool 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_.CalledOnValidThread());
return channel_proxy_->ReceivedRTCPPacket(packet, length);
}
void AudioReceiveStream::OnRtpPacket(const RtpPacketReceived& packet) {
// 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_.CalledOnValidThread());
channel_proxy_->OnRtpPacket(packet);
}
const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return config_;
}
VoiceEngine* AudioReceiveStream::voice_engine() const {
auto* voice_engine = audio_state()->voice_engine();
RTC_DCHECK(voice_engine);
return voice_engine;
}
internal::AudioState* AudioReceiveStream::audio_state() const {
auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
RTC_DCHECK(audio_state);
return audio_state;
}
int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
ScopedVoEInterface<VoEBase> base(voice_engine());
if (playout) {
return base->StartPlayout(config_.voe_channel_id);
} else {
return base->StopPlayout(config_.voe_channel_id);
}
}
} // namespace internal
} // namespace webrtc