rhubarb-lip-sync/lib/webrtc-8d2248ff/webrtc/audio/audio_receive_stream.cc

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2016-06-21 20:13:05 +00:00
/*
* 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 "webrtc/audio/audio_receive_stream.h"
#include <string>
#include <utility>
#include "webrtc/audio_sink.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_neteq_stats.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
namespace {
bool UseSendSideBwe(const webrtc::AudioReceiveStream::Config& config) {
if (!config.rtp.transport_cc) {
return false;
}
for (const auto& extension : config.rtp.extensions) {
if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
return true;
}
}
return false;
}
} // namespace
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)" : "nullptr");
ss << ", voe_channel_id: " << voe_channel_id;
if (!sync_group.empty()) {
ss << ", sync_group: " << sync_group;
}
ss << '}';
return ss.str();
}
namespace internal {
AudioReceiveStream::AudioReceiveStream(
CongestionController* congestion_controller,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
: config_(config),
audio_state_(audio_state),
rtp_header_parser_(RtpHeaderParser::Create()) {
LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
RTC_DCHECK_NE(config_.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
RTC_DCHECK(congestion_controller);
RTC_DCHECK(rtp_header_parser_);
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
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_->RegisterExternalTransport(config.rtcp_send_transport);
for (const auto& extension : config.rtp.extensions) {
if (extension.uri == RtpExtension::kAudioLevelUri) {
channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionAudioLevel, extension.id);
RTC_DCHECK(registered);
} else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id);
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, extension.id);
RTC_DCHECK(registered);
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber, extension.id);
RTC_DCHECK(registered);
} else {
RTC_NOTREACHED() << "Unsupported RTP extension.";
}
}
// Configure bandwidth estimation.
channel_proxy_->RegisterReceiverCongestionControlObjects(
congestion_controller->packet_router());
if (UseSendSideBwe(config)) {
remote_bitrate_estimator_ =
congestion_controller->GetRemoteBitrateEstimator(true);
}
}
AudioReceiveStream::~AudioReceiveStream() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->ResetCongestionControlObjects();
if (remote_bitrate_estimator_) {
remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
}
}
void AudioReceiveStream::Start() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
void AudioReceiveStream::Stop() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
webrtc::AudioReceiveStream::Stats stats;
stats.remote_ssrc = config_.rtp.remote_ssrc;
ScopedVoEInterface<VoECodec> codec(voice_engine());
webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
webrtc::CodecInst codec_inst = {0};
if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
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.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();
// 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.expand_rate = Q14ToFloat(ns.currentExpandRate);
stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
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;
return stats;
}
void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel_proxy_->SetSink(std::move(sink));
}
void AudioReceiveStream::SetGain(float gain) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel_proxy_->SetChannelOutputVolumeScaling(gain);
}
const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return config_;
}
void AudioReceiveStream::SignalNetworkState(NetworkState state) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
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);
}
bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
// 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());
RTPHeader header;
if (!rtp_header_parser_->Parse(packet, length, &header)) {
return false;
}
// Only forward if the parsed header has one of the headers necessary for
// bandwidth estimation. RTP timestamps has different rates for audio and
// video and shouldn't be mixed.
if (remote_bitrate_estimator_ &&
header.extension.hasTransportSequenceNumber) {
int64_t arrival_time_ms = rtc::TimeMillis();
if (packet_time.timestamp >= 0)
arrival_time_ms = (packet_time.timestamp + 500) / 1000;
size_t payload_size = length - header.headerLength;
remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
header);
}
return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
}
VoiceEngine* AudioReceiveStream::voice_engine() const {
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
VoiceEngine* voice_engine = audio_state->voice_engine();
RTC_DCHECK(voice_engine);
return voice_engine;
}
} // namespace internal
} // namespace webrtc