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