Split webrtc/video into webrtc/{audio,call,video}.

webrtc/video/audio_receive_stream.cc

-/*
- *  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/video/audio_receive_stream.h"
-
-#include <string>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-
-namespace webrtc {
-std::string AudioReceiveStream::Config::Rtp::ToString() const {
-  std::stringstream ss;
-  ss << "{remote_ssrc: " << remote_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 << ", voe_channel_id: " << voe_channel_id;
-  if (!sync_group.empty())
-    ss << ", sync_group: " << sync_group;
-  ss << '}';
-  return ss.str();
-}
-
-namespace internal {
-AudioReceiveStream::AudioReceiveStream(
-      RemoteBitrateEstimator* remote_bitrate_estimator,
-      const webrtc::AudioReceiveStream::Config& config)
-    : remote_bitrate_estimator_(remote_bitrate_estimator),
-      config_(config),
-      rtp_header_parser_(RtpHeaderParser::Create()) {
-  RTC_DCHECK(config.voe_channel_id != -1);
-  RTC_DCHECK(remote_bitrate_estimator_ != 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.";
-    }
-  }
-}
-
-webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
-  return webrtc::AudioReceiveStream::Stats();
-}
-
-void AudioReceiveStream::Start() {
-}
-
-void AudioReceiveStream::Stop() {
-}
-
-void AudioReceiveStream::SignalNetworkState(NetworkState state) {
-}
-
-bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  return false;
-}
-
-bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
-                                    size_t length,
-                                    const PacketTime& packet_time) {
-  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