Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 153 insertions, 0 deletions

diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
new file mode 100644
index 0000000000..f54c81ec47
--- /dev/null
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -0,0 +1,153 @@
+/*
+ *  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/voice_engine/channel_proxy.h"
+
+#include <utility>
+
+#include "webrtc/audio/audio_sink.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/voice_engine/channel.h"
+
+namespace webrtc {
+namespace voe {
+ChannelProxy::ChannelProxy() : channel_owner_(nullptr) {}
+
+ChannelProxy::ChannelProxy(const ChannelOwner& channel_owner) :
+    channel_owner_(channel_owner) {
+  RTC_CHECK(channel_owner_.channel());
+}
+
+ChannelProxy::~ChannelProxy() {}
+
+void ChannelProxy::SetRTCPStatus(bool enable) {
+  channel()->SetRTCPStatus(enable);
+}
+
+void ChannelProxy::SetLocalSSRC(uint32_t ssrc) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->SetLocalSSRC(ssrc);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::SetRTCP_CNAME(const std::string& c_name) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  // Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array.
+  std::string c_name_limited = c_name.substr(0, 255);
+  int error = channel()->SetRTCP_CNAME(c_name_limited.c_str());
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::SetSendAbsoluteSenderTimeStatus(bool enable, int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->SetSendAbsoluteSenderTimeStatus(enable, id);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->SetSendAudioLevelIndicationStatus(enable, id);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::EnableSendTransportSequenceNumber(int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->EnableSendTransportSequenceNumber(id);
+}
+
+void ChannelProxy::SetReceiveAbsoluteSenderTimeStatus(bool enable, int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->SetReceiveAbsoluteSenderTimeStatus(enable, id);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->SetReceiveAudioLevelIndicationStatus(enable, id);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::SetCongestionControlObjects(
+    RtpPacketSender* rtp_packet_sender,
+    TransportFeedbackObserver* transport_feedback_observer,
+    PacketRouter* packet_router) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->SetCongestionControlObjects(
+      rtp_packet_sender, transport_feedback_observer, packet_router);
+}
+
+CallStatistics ChannelProxy::GetRTCPStatistics() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  CallStatistics stats = {0};
+  int error = channel()->GetRTPStatistics(stats);
+  RTC_DCHECK_EQ(0, error);
+  return stats;
+}
+
+std::vector<ReportBlock> ChannelProxy::GetRemoteRTCPReportBlocks() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  std::vector<webrtc::ReportBlock> blocks;
+  int error = channel()->GetRemoteRTCPReportBlocks(&blocks);
+  RTC_DCHECK_EQ(0, error);
+  return blocks;
+}
+
+NetworkStatistics ChannelProxy::GetNetworkStatistics() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  NetworkStatistics stats = {0};
+  int error = channel()->GetNetworkStatistics(stats);
+  RTC_DCHECK_EQ(0, error);
+  return stats;
+}
+
+AudioDecodingCallStats ChannelProxy::GetDecodingCallStatistics() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  AudioDecodingCallStats stats;
+  channel()->GetDecodingCallStatistics(&stats);
+  return stats;
+}
+
+int32_t ChannelProxy::GetSpeechOutputLevelFullRange() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  uint32_t level = 0;
+  int error = channel()->GetSpeechOutputLevelFullRange(level);
+  RTC_DCHECK_EQ(0, error);
+  return static_cast<int32_t>(level);
+}
+
+uint32_t ChannelProxy::GetDelayEstimate() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return channel()->GetDelayEstimate();
+}
+
+bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return channel()->SetSendTelephoneEventPayloadType(payload_type) == 0;
+}
+
+bool ChannelProxy::SendTelephoneEventOutband(uint8_t event,
+                                             uint32_t duration_ms) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return
+      channel()->SendTelephoneEventOutband(event, duration_ms, 10, false) == 0;
+}
+
+void ChannelProxy::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->SetSink(std::move(sink));
+}
+
+Channel* ChannelProxy::channel() const {
+  RTC_DCHECK(channel_owner_.channel());
+  return channel_owner_.channel();
+}
+
+}  // namespace voe
+}  // namespace webrtc