| /* |
| * 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 "webrtc/audio/conversion.h" |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "webrtc/system_wrappers/include/tick_util.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" |
| |
| namespace webrtc { |
| std::string AudioReceiveStream::Config::Rtp::ToString() const { |
| std::stringstream ss; |
| ss << "{remote_ssrc: " << remote_ssrc; |
| ss << ", local_ssrc: " << local_ssrc; |
| 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 << ", receive_transport: " |
| << (receive_transport ? "(Transport)" : "nullptr"); |
| 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 << ", combined_audio_video_bwe: " |
| << (combined_audio_video_bwe ? "true" : "false"); |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| namespace internal { |
| AudioReceiveStream::AudioReceiveStream( |
| RemoteBitrateEstimator* remote_bitrate_estimator, |
| const webrtc::AudioReceiveStream::Config& config, |
| VoiceEngine* voice_engine) |
| : remote_bitrate_estimator_(remote_bitrate_estimator), |
| config_(config), |
| voice_engine_(voice_engine), |
| voe_base_(voice_engine), |
| rtp_header_parser_(RtpHeaderParser::Create()) { |
| LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); |
| RTC_DCHECK(config.voe_channel_id != -1); |
| RTC_DCHECK(remote_bitrate_estimator_ != nullptr); |
| RTC_DCHECK(voice_engine_ != nullptr); |
| RTC_DCHECK(rtp_header_parser_ != nullptr); |
| for (const auto& ext : config.rtp.extensions) { |
| // One-byte-extension local identifiers are in the range 1-14 inclusive. |
| RTC_DCHECK_GE(ext.id, 1); |
| RTC_DCHECK_LE(ext.id, 14); |
| if (ext.name == RtpExtension::kAudioLevel) { |
| RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionAudioLevel, ext.id)); |
| } else if (ext.name == RtpExtension::kAbsSendTime) { |
| RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, ext.id)); |
| } else if (ext.name == RtpExtension::kTransportSequenceNumber) { |
| RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransportSequenceNumber, ext.id)); |
| } else { |
| RTC_NOTREACHED() << "Unsupported RTP extension."; |
| } |
| } |
| } |
| |
| AudioReceiveStream::~AudioReceiveStream() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); |
| } |
| |
| 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_); |
| ScopedVoEInterface<VoENetEqStats> neteq(voice_engine_); |
| ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_); |
| ScopedVoEInterface<VoEVideoSync> sync(voice_engine_); |
| ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_); |
| unsigned int ssrc = 0; |
| webrtc::CallStatistics call_stats = {0}; |
| webrtc::CodecInst codec_inst = {0}; |
| // Only collect stats if we have seen some traffic with the SSRC. |
| if (rtp->GetRemoteSSRC(config_.voe_channel_id, ssrc) == -1 || |
| rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1 || |
| 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); |
| 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); |
| } |
| { |
| int jitter_buffer_delay_ms = 0; |
| int playout_buffer_delay_ms = 0; |
| sync->GetDelayEstimate(config_.voe_channel_id, &jitter_buffer_delay_ms, |
| &playout_buffer_delay_ms); |
| stats.delay_estimate_ms = |
| jitter_buffer_delay_ms + playout_buffer_delay_ms; |
| } |
| { |
| unsigned int level = 0; |
| if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level) |
| != -1) { |
| stats.audio_level = static_cast<int32_t>(level); |
| } |
| } |
| |
| webrtc::NetworkStatistics ns = {0}; |
| if (neteq->GetNetworkStatistics(config_.voe_channel_id, ns) != -1) { |
| // Get jitter buffer and total delay (alg + jitter + playout) stats. |
| 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); |
| } |
| |
| webrtc::AudioDecodingCallStats ds; |
| if (neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds) != -1) { |
| 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.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_; |
| |
| return stats; |
| } |
| |
| const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| return config_; |
| } |
| |
| void AudioReceiveStream::Start() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| } |
| |
| void AudioReceiveStream::Stop() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| } |
| |
| 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 false; |
| } |
| |
| 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 absolute sender time. RTP timestamps |
| // may have different rates for audio and video and shouldn't be mixed. |
| if (config_.combined_audio_video_bwe && |
| header.extension.hasAbsoluteSendTime) { |
| int64_t arrival_time_ms = TickTime::MillisecondTimestamp(); |
| 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, false); |
| } |
| return true; |
| } |
| } // namespace internal |
| } // namespace webrtc |