Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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modules/audio_coding/neteq/decision_logic.cc
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170
modules/audio_coding/neteq/decision_logic.cc
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/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
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#include <algorithm>
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#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
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#include "webrtc/modules/audio_coding/neteq/decision_logic_fax.h"
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#include "webrtc/modules/audio_coding/neteq/decision_logic_normal.h"
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#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
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#include "webrtc/modules/audio_coding/neteq/expand.h"
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#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
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#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
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namespace webrtc {
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DecisionLogic* DecisionLogic::Create(int fs_hz,
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size_t output_size_samples,
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NetEqPlayoutMode playout_mode,
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DecoderDatabase* decoder_database,
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const PacketBuffer& packet_buffer,
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DelayManager* delay_manager,
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BufferLevelFilter* buffer_level_filter,
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const TickTimer* tick_timer) {
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switch (playout_mode) {
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case kPlayoutOn:
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case kPlayoutStreaming:
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return new DecisionLogicNormal(
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fs_hz, output_size_samples, playout_mode, decoder_database,
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packet_buffer, delay_manager, buffer_level_filter, tick_timer);
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case kPlayoutFax:
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case kPlayoutOff:
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return new DecisionLogicFax(
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fs_hz, output_size_samples, playout_mode, decoder_database,
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packet_buffer, delay_manager, buffer_level_filter, tick_timer);
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}
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// This line cannot be reached, but must be here to avoid compiler errors.
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assert(false);
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return NULL;
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}
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DecisionLogic::DecisionLogic(int fs_hz,
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size_t output_size_samples,
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NetEqPlayoutMode playout_mode,
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DecoderDatabase* decoder_database,
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const PacketBuffer& packet_buffer,
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DelayManager* delay_manager,
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BufferLevelFilter* buffer_level_filter,
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const TickTimer* tick_timer)
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: decoder_database_(decoder_database),
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packet_buffer_(packet_buffer),
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delay_manager_(delay_manager),
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buffer_level_filter_(buffer_level_filter),
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tick_timer_(tick_timer),
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cng_state_(kCngOff),
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packet_length_samples_(0),
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sample_memory_(0),
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prev_time_scale_(false),
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timescale_countdown_(
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tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1)),
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num_consecutive_expands_(0),
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playout_mode_(playout_mode) {
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delay_manager_->set_streaming_mode(playout_mode_ == kPlayoutStreaming);
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SetSampleRate(fs_hz, output_size_samples);
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}
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DecisionLogic::~DecisionLogic() = default;
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void DecisionLogic::Reset() {
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cng_state_ = kCngOff;
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noise_fast_forward_ = 0;
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packet_length_samples_ = 0;
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sample_memory_ = 0;
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prev_time_scale_ = false;
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timescale_countdown_.reset();
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num_consecutive_expands_ = 0;
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}
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void DecisionLogic::SoftReset() {
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packet_length_samples_ = 0;
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sample_memory_ = 0;
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prev_time_scale_ = false;
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timescale_countdown_ =
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tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1);
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}
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void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
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// TODO(hlundin): Change to an enumerator and skip assert.
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assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000);
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fs_mult_ = fs_hz / 8000;
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output_size_samples_ = output_size_samples;
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}
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Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
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const Expand& expand,
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size_t decoder_frame_length,
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const Packet* next_packet,
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Modes prev_mode,
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bool play_dtmf,
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size_t generated_noise_samples,
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bool* reset_decoder) {
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// If last mode was CNG (or Expand, since this could be covering up for
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// a lost CNG packet), remember that CNG is on. This is needed if comfort
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// noise is interrupted by DTMF.
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if (prev_mode == kModeRfc3389Cng) {
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cng_state_ = kCngRfc3389On;
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} else if (prev_mode == kModeCodecInternalCng) {
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cng_state_ = kCngInternalOn;
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}
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const size_t samples_left =
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sync_buffer.FutureLength() - expand.overlap_length();
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const size_t cur_size_samples =
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samples_left + packet_buffer_.NumSamplesInBuffer(decoder_frame_length);
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prev_time_scale_ = prev_time_scale_ &&
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(prev_mode == kModeAccelerateSuccess ||
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prev_mode == kModeAccelerateLowEnergy ||
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prev_mode == kModePreemptiveExpandSuccess ||
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prev_mode == kModePreemptiveExpandLowEnergy);
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FilterBufferLevel(cur_size_samples, prev_mode);
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return GetDecisionSpecialized(sync_buffer, expand, decoder_frame_length,
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next_packet, prev_mode, play_dtmf,
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reset_decoder, generated_noise_samples);
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}
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void DecisionLogic::ExpandDecision(Operations operation) {
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if (operation == kExpand) {
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num_consecutive_expands_++;
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} else {
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num_consecutive_expands_ = 0;
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}
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}
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void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples,
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Modes prev_mode) {
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// Do not update buffer history if currently playing CNG since it will bias
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// the filtered buffer level.
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if ((prev_mode != kModeRfc3389Cng) && (prev_mode != kModeCodecInternalCng)) {
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buffer_level_filter_->SetTargetBufferLevel(
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delay_manager_->base_target_level());
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size_t buffer_size_packets = 0;
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if (packet_length_samples_ > 0) {
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// Calculate size in packets.
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buffer_size_packets = buffer_size_samples / packet_length_samples_;
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}
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int sample_memory_local = 0;
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if (prev_time_scale_) {
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sample_memory_local = sample_memory_;
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timescale_countdown_ =
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tick_timer_->GetNewCountdown(kMinTimescaleInterval);
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}
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buffer_level_filter_->Update(buffer_size_packets, sample_memory_local,
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packet_length_samples_);
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prev_time_scale_ = false;
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}
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}
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} // namespace webrtc
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