| /* |
| * 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 "modules/audio_coding/audio_network_adaptor/bitrate_controller.h" |
| |
| #include <algorithm> |
| |
| #include "rtc_base/checks.h" |
| #include "system_wrappers/include/field_trial.h" |
| |
| namespace webrtc { |
| namespace audio_network_adaptor { |
| |
| BitrateController::Config::Config(int initial_bitrate_bps, |
| int initial_frame_length_ms, |
| int fl_increase_overhead_offset, |
| int fl_decrease_overhead_offset) |
| : initial_bitrate_bps(initial_bitrate_bps), |
| initial_frame_length_ms(initial_frame_length_ms), |
| fl_increase_overhead_offset(fl_increase_overhead_offset), |
| fl_decrease_overhead_offset(fl_decrease_overhead_offset) {} |
| |
| BitrateController::Config::~Config() = default; |
| |
| BitrateController::BitrateController(const Config& config) |
| : config_(config), |
| bitrate_bps_(config_.initial_bitrate_bps), |
| frame_length_ms_(config_.initial_frame_length_ms) { |
| RTC_DCHECK_GT(bitrate_bps_, 0); |
| RTC_DCHECK_GT(frame_length_ms_, 0); |
| } |
| |
| BitrateController::~BitrateController() = default; |
| |
| void BitrateController::UpdateNetworkMetrics( |
| const NetworkMetrics& network_metrics) { |
| if (network_metrics.target_audio_bitrate_bps) |
| target_audio_bitrate_bps_ = network_metrics.target_audio_bitrate_bps; |
| if (network_metrics.overhead_bytes_per_packet) { |
| RTC_DCHECK_GT(*network_metrics.overhead_bytes_per_packet, 0); |
| overhead_bytes_per_packet_ = network_metrics.overhead_bytes_per_packet; |
| } |
| } |
| |
| void BitrateController::MakeDecision(AudioEncoderRuntimeConfig* config) { |
| // Decision on |bitrate_bps| should not have been made. |
| RTC_DCHECK(!config->bitrate_bps); |
| if (target_audio_bitrate_bps_ && overhead_bytes_per_packet_) { |
| if (config->frame_length_ms) |
| frame_length_ms_ = *config->frame_length_ms; |
| int offset = config->last_fl_change_increase |
| ? config_.fl_increase_overhead_offset |
| : config_.fl_decrease_overhead_offset; |
| // Check that |
| // -(*overhead_bytes_per_packet_) <= offset <= (*overhead_bytes_per_packet_) |
| RTC_DCHECK_GE(*overhead_bytes_per_packet_, -offset); |
| RTC_DCHECK_LE(offset, *overhead_bytes_per_packet_); |
| int overhead_rate_bps = static_cast<int>( |
| (*overhead_bytes_per_packet_ + offset) * 8 * 1000 / frame_length_ms_); |
| bitrate_bps_ = std::max(0, *target_audio_bitrate_bps_ - overhead_rate_bps); |
| } |
| config->bitrate_bps = bitrate_bps_; |
| } |
| |
| } // namespace audio_network_adaptor |
| } // namespace webrtc |