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In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
242 lines
8.5 KiB
C++
242 lines
8.5 KiB
C++
/*
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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 "modules/audio_coding/neteq/decision_logic_normal.h"
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#include <assert.h>
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#include <algorithm>
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#include "modules/audio_coding/neteq/buffer_level_filter.h"
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#include "modules/audio_coding/neteq/decoder_database.h"
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#include "modules/audio_coding/neteq/delay_manager.h"
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#include "modules/audio_coding/neteq/expand.h"
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#include "modules/audio_coding/neteq/packet_buffer.h"
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#include "modules/audio_coding/neteq/sync_buffer.h"
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#include "modules/include/module_common_types.h"
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namespace webrtc {
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Operations DecisionLogicNormal::GetDecisionSpecialized(
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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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bool* reset_decoder,
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size_t generated_noise_samples) {
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assert(playout_mode_ == kPlayoutOn || playout_mode_ == kPlayoutStreaming);
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// Guard for errors, to avoid getting stuck in error mode.
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if (prev_mode == kModeError) {
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if (!next_packet) {
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return kExpand;
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} else {
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return kUndefined; // Use kUndefined to flag for a reset.
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}
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}
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uint32_t target_timestamp = sync_buffer.end_timestamp();
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uint32_t available_timestamp = 0;
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bool is_cng_packet = false;
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if (next_packet) {
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available_timestamp = next_packet->timestamp;
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is_cng_packet =
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decoder_database_->IsComfortNoise(next_packet->payload_type);
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}
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if (is_cng_packet) {
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return CngOperation(prev_mode, target_timestamp, available_timestamp,
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generated_noise_samples);
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}
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// Handle the case with no packet at all available (except maybe DTMF).
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if (!next_packet) {
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return NoPacket(play_dtmf);
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}
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// If the expand period was very long, reset NetEQ since it is likely that the
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// sender was restarted.
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if (num_consecutive_expands_ > kReinitAfterExpands) {
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*reset_decoder = true;
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return kNormal;
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}
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const uint32_t five_seconds_samples =
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static_cast<uint32_t>(5 * 8000 * fs_mult_);
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// Check if the required packet is available.
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if (target_timestamp == available_timestamp) {
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return ExpectedPacketAvailable(prev_mode, play_dtmf);
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} else if (!PacketBuffer::IsObsoleteTimestamp(
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available_timestamp, target_timestamp, five_seconds_samples)) {
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return FuturePacketAvailable(sync_buffer, expand, decoder_frame_length,
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prev_mode, target_timestamp,
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available_timestamp, play_dtmf,
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generated_noise_samples);
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} else {
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// This implies that available_timestamp < target_timestamp, which can
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// happen when a new stream or codec is received. Signal for a reset.
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return kUndefined;
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}
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}
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Operations DecisionLogicNormal::CngOperation(Modes prev_mode,
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uint32_t target_timestamp,
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uint32_t available_timestamp,
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size_t generated_noise_samples) {
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// Signed difference between target and available timestamp.
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int32_t timestamp_diff = static_cast<int32_t>(
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static_cast<uint32_t>(generated_noise_samples + target_timestamp) -
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available_timestamp);
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int32_t optimal_level_samp = static_cast<int32_t>(
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(delay_manager_->TargetLevel() * packet_length_samples_) >> 8);
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const int64_t excess_waiting_time_samp =
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-static_cast<int64_t>(timestamp_diff) - optimal_level_samp;
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if (excess_waiting_time_samp > optimal_level_samp / 2) {
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// The waiting time for this packet will be longer than 1.5
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// times the wanted buffer delay. Apply fast-forward to cut the
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// waiting time down to the optimal.
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noise_fast_forward_ = rtc::dchecked_cast<size_t>(noise_fast_forward_ +
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excess_waiting_time_samp);
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timestamp_diff =
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rtc::saturated_cast<int32_t>(timestamp_diff + excess_waiting_time_samp);
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}
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if (timestamp_diff < 0 && prev_mode == kModeRfc3389Cng) {
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// Not time to play this packet yet. Wait another round before using this
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// packet. Keep on playing CNG from previous CNG parameters.
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return kRfc3389CngNoPacket;
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} else {
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// Otherwise, go for the CNG packet now.
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noise_fast_forward_ = 0;
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return kRfc3389Cng;
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}
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}
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Operations DecisionLogicNormal::NoPacket(bool play_dtmf) {
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if (cng_state_ == kCngRfc3389On) {
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// Keep on playing comfort noise.
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return kRfc3389CngNoPacket;
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} else if (cng_state_ == kCngInternalOn) {
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// Keep on playing codec internal comfort noise.
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return kCodecInternalCng;
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} else if (play_dtmf) {
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return kDtmf;
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} else {
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// Nothing to play, do expand.
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return kExpand;
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}
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}
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Operations DecisionLogicNormal::ExpectedPacketAvailable(Modes prev_mode,
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bool play_dtmf) {
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if (prev_mode != kModeExpand && !play_dtmf) {
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// Check criterion for time-stretching.
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int low_limit, high_limit;
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delay_manager_->BufferLimits(&low_limit, &high_limit);
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if (buffer_level_filter_->filtered_current_level() >= high_limit << 2)
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return kFastAccelerate;
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if (TimescaleAllowed()) {
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if (buffer_level_filter_->filtered_current_level() >= high_limit)
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return kAccelerate;
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if (buffer_level_filter_->filtered_current_level() < low_limit)
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return kPreemptiveExpand;
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}
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}
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return kNormal;
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}
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Operations DecisionLogicNormal::FuturePacketAvailable(
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const SyncBuffer& sync_buffer,
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const Expand& expand,
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size_t decoder_frame_length,
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Modes prev_mode,
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uint32_t target_timestamp,
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uint32_t available_timestamp,
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bool play_dtmf,
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size_t generated_noise_samples) {
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// Required packet is not available, but a future packet is.
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// Check if we should continue with an ongoing expand because the new packet
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// is too far into the future.
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uint32_t timestamp_leap = available_timestamp - target_timestamp;
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if ((prev_mode == kModeExpand) &&
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!ReinitAfterExpands(timestamp_leap) &&
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!MaxWaitForPacket() &&
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PacketTooEarly(timestamp_leap) &&
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UnderTargetLevel()) {
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if (play_dtmf) {
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// Still have DTMF to play, so do not do expand.
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return kDtmf;
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} else {
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// Nothing to play.
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return kExpand;
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}
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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 = samples_left +
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packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
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// If previous was comfort noise, then no merge is needed.
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if (prev_mode == kModeRfc3389Cng ||
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prev_mode == kModeCodecInternalCng) {
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// Keep the same delay as before the CNG (or maximum 70 ms in buffer as
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// safety precaution), but make sure that the number of samples in buffer
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// is no higher than 4 times the optimal level. (Note that TargetLevel()
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// is in Q8.)
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if (static_cast<uint32_t>(generated_noise_samples + target_timestamp) >=
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available_timestamp ||
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cur_size_samples >
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((delay_manager_->TargetLevel() * packet_length_samples_) >> 8) *
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4) {
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// Time to play this new packet.
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return kNormal;
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} else {
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// Too early to play this new packet; keep on playing comfort noise.
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if (prev_mode == kModeRfc3389Cng) {
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return kRfc3389CngNoPacket;
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} else { // prevPlayMode == kModeCodecInternalCng.
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return kCodecInternalCng;
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}
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}
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}
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// Do not merge unless we have done an expand before.
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if (prev_mode == kModeExpand) {
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return kMerge;
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} else if (play_dtmf) {
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// Play DTMF instead of expand.
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return kDtmf;
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} else {
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return kExpand;
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}
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}
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bool DecisionLogicNormal::UnderTargetLevel() const {
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return buffer_level_filter_->filtered_current_level() <=
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delay_manager_->TargetLevel();
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}
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bool DecisionLogicNormal::ReinitAfterExpands(uint32_t timestamp_leap) const {
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return timestamp_leap >=
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static_cast<uint32_t>(output_size_samples_ * kReinitAfterExpands);
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}
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bool DecisionLogicNormal::PacketTooEarly(uint32_t timestamp_leap) const {
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return timestamp_leap >
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static_cast<uint32_t>(output_size_samples_ * num_consecutive_expands_);
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}
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bool DecisionLogicNormal::MaxWaitForPacket() const {
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return num_consecutive_expands_ >= kMaxWaitForPacket;
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}
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} // namespace webrtc
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