Adding ChannelController to audio network adaptor.

webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc

Index: webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
new file mode 100644
index 0000000000000000000000000000000000000000..785bfd0e27d79e28b199d0df3a1e455ed4b2d038
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2016 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 <algorithm>
+
+#include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+ChannelController::Config::Config(size_t num_encoder_channels,
+                                  size_t intial_channels_to_encode,
+                                  int channel_1_to_2_bandwidth_bps,
+                                  int channel_2_to_1_bandwidth_bps)
+    : num_encoder_channels(num_encoder_channels),
+      intial_channels_to_encode(intial_channels_to_encode),
+      channel_1_to_2_bandwidth_bps(channel_1_to_2_bandwidth_bps),
+      channel_2_to_1_bandwidth_bps(channel_2_to_1_bandwidth_bps) {}
+
+ChannelController::Config::~Config() = default;
+
+ChannelController::ChannelController(const Config& config)
+    : config_(config), channels_to_encode_(config_.intial_channels_to_encode) {

[minyue-webrtc, 2016/09/08 13:22:03]

this is a minor change. it is just a style thingy.

+  RTC_DCHECK_GT(config_.intial_channels_to_encode, 0lu);
+  // Currently, we require |intial_channels_to_encode| to be <= 2.
+  RTC_DCHECK_LE(config_.intial_channels_to_encode, 2lu);
+  RTC_DCHECK_GE(config_.num_encoder_channels,
+                config_.intial_channels_to_encode);
+}
+
+void ChannelController::MakeDecision(
+    const NetworkMetrics& metrics,
+    AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
+  // Decision on |num_channels| should not have been made.
+  RTC_DCHECK(!config->num_channels);
+
+  if (metrics.uplink_bandwidth_bps) {
+    if (channels_to_encode_ == 2 &&
+        *metrics.uplink_bandwidth_bps <= config_.channel_2_to_1_bandwidth_bps) {
+      channels_to_encode_ = 1;
+    } else if (channels_to_encode_ == 1 &&
+               *metrics.uplink_bandwidth_bps >=
+                   config_.channel_1_to_2_bandwidth_bps) {
+      channels_to_encode_ = std::min(static_cast<size_t>(2),
+                                     config_.num_encoder_channels);
+    }
+  }
+  config->num_channels = rtc::Optional<size_t>(channels_to_encode_);
+}
+
+}  // namespace webrtc