a2f30deea3
BUG= R=solenberg@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1400333002 Cr-Commit-Position: refs/heads/master@{#10286}
124 lines
4.1 KiB
C++
124 lines
4.1 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/audio/audio_receive_stream.h"
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#include <string>
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
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#include "webrtc/system_wrappers/interface/tick_util.h"
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namespace webrtc {
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std::string AudioReceiveStream::Config::Rtp::ToString() const {
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std::stringstream ss;
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ss << "{remote_ssrc: " << remote_ssrc;
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ss << ", extensions: [";
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for (size_t i = 0; i < extensions.size(); ++i) {
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ss << extensions[i].ToString();
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if (i != extensions.size() - 1)
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ss << ", ";
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}
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ss << ']';
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ss << '}';
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return ss.str();
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}
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std::string AudioReceiveStream::Config::ToString() const {
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std::stringstream ss;
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ss << "{rtp: " << rtp.ToString();
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ss << ", voe_channel_id: " << voe_channel_id;
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if (!sync_group.empty())
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ss << ", sync_group: " << sync_group;
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ss << '}';
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return ss.str();
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}
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namespace internal {
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AudioReceiveStream::AudioReceiveStream(
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RemoteBitrateEstimator* remote_bitrate_estimator,
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const webrtc::AudioReceiveStream::Config& config)
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: remote_bitrate_estimator_(remote_bitrate_estimator),
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config_(config),
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rtp_header_parser_(RtpHeaderParser::Create()) {
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LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
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RTC_DCHECK(config.voe_channel_id != -1);
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RTC_DCHECK(remote_bitrate_estimator_ != nullptr);
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RTC_DCHECK(rtp_header_parser_ != nullptr);
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for (const auto& ext : config.rtp.extensions) {
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// One-byte-extension local identifiers are in the range 1-14 inclusive.
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RTC_DCHECK_GE(ext.id, 1);
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RTC_DCHECK_LE(ext.id, 14);
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if (ext.name == RtpExtension::kAudioLevel) {
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RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
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kRtpExtensionAudioLevel, ext.id));
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} else if (ext.name == RtpExtension::kAbsSendTime) {
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RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
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kRtpExtensionAbsoluteSendTime, ext.id));
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} else if (ext.name == RtpExtension::kTransportSequenceNumber) {
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RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
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kRtpExtensionTransportSequenceNumber, ext.id));
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} else {
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RTC_NOTREACHED() << "Unsupported RTP extension.";
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}
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}
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}
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AudioReceiveStream::~AudioReceiveStream() {
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LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
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}
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webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
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return webrtc::AudioReceiveStream::Stats();
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}
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const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
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return config_;
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}
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void AudioReceiveStream::Start() {
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}
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void AudioReceiveStream::Stop() {
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}
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void AudioReceiveStream::SignalNetworkState(NetworkState state) {
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}
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bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
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return false;
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}
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bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
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size_t length,
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const PacketTime& packet_time) {
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RTPHeader header;
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if (!rtp_header_parser_->Parse(packet, length, &header)) {
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return false;
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}
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// Only forward if the parsed header has absolute sender time. RTP timestamps
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// may have different rates for audio and video and shouldn't be mixed.
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if (config_.combined_audio_video_bwe &&
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header.extension.hasAbsoluteSendTime) {
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int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
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if (packet_time.timestamp >= 0)
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arrival_time_ms = (packet_time.timestamp + 500) / 1000;
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size_t payload_size = length - header.headerLength;
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remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
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header, false);
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}
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return true;
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}
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} // namespace internal
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} // namespace webrtc
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