| Index: webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
|
| diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
|
| index 31eff216890882ecc0b2496914d45b312a5cd831..95ed4d61c3024c28e9930c112ee736f081f493d0 100644
|
| --- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
|
| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
|
| @@ -81,12 +81,28 @@ std::unique_ptr<FrameLengthController> CreateFrameLengthController(
|
| RTC_CHECK(config.has_fl_20ms_to_60ms_bandwidth_bps());
|
| RTC_CHECK(config.has_fl_60ms_to_20ms_bandwidth_bps());
|
|
|
| + std::map<FrameLengthController::Config::FrameLengthChange, int>
|
| + fl_changing_bandwidths_bps = {
|
| + {FrameLengthController::Config::FrameLengthChange(20, 60),
|
| + config.fl_20ms_to_60ms_bandwidth_bps()},
|
| + {FrameLengthController::Config::FrameLengthChange(60, 20),
|
| + config.fl_60ms_to_20ms_bandwidth_bps()}};
|
| +
|
| + if (config.has_fl_60ms_to_120ms_bandwidth_bps() &&
|
| + config.has_fl_120ms_to_60ms_bandwidth_bps()) {
|
| + fl_changing_bandwidths_bps.insert(std::make_pair(
|
| + FrameLengthController::Config::FrameLengthChange(60, 120),
|
| + config.fl_60ms_to_120ms_bandwidth_bps()));
|
| + fl_changing_bandwidths_bps.insert(std::make_pair(
|
| + FrameLengthController::Config::FrameLengthChange(120, 60),
|
| + config.fl_120ms_to_60ms_bandwidth_bps()));
|
| + }
|
| +
|
| FrameLengthController::Config ctor_config(
|
| std::vector<int>(), initial_frame_length_ms,
|
| config.fl_increasing_packet_loss_fraction(),
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| config.fl_decreasing_packet_loss_fraction(),
|
| - config.fl_20ms_to_60ms_bandwidth_bps(),
|
| - config.fl_60ms_to_20ms_bandwidth_bps());
|
| + std::move(fl_changing_bandwidths_bps));
|
|
|
| for (auto frame_length : encoder_frame_lengths_ms)
|
| ctor_config.encoder_frame_lengths_ms.push_back(frame_length);
|
|
|