
This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1. Reason for revert: Breaks downstream tests. Original change's description: > NetEq: Deprecate playout modes Fax, Off and Streaming > > The playout modes other than Normal have not been reachable for a long > time, other than through tests. It is time to deprecate them. > > The only meaningful use was that Fax mode was sometimes set from > tests, in order to avoid time-stretching operations (accelerate and > pre-emptive expand) from messing with the test results. With this CL, > a new config is added instead, which lets the user specify exactly > this: don't do time-stretching. > > As a result of Fax and Off modes being removed, the following code > clean-up was done: > - Fold DecisionLogicNormal into DecisionLogic. > - Remove AudioRepetition and AlternativePlc operations, since they can > no longer be reached. > > Bug: webrtc:9421 > Change-Id: I651458e9c1931a99f3b07e242817d303bac119df > Reviewed-on: https://webrtc-review.googlesource.com/84123 > Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Minyue Li <minyue@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23704} TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9421 Reviewed-on: https://webrtc-review.googlesource.com/84680 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23706}
172 lines
6.5 KiB
C++
172 lines
6.5 KiB
C++
/*
|
|
* Copyright (c) 2013 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/neteq/decision_logic.h"
|
|
|
|
#include <algorithm>
|
|
|
|
#include "modules/audio_coding/neteq/buffer_level_filter.h"
|
|
#include "modules/audio_coding/neteq/decision_logic_fax.h"
|
|
#include "modules/audio_coding/neteq/decision_logic_normal.h"
|
|
#include "modules/audio_coding/neteq/delay_manager.h"
|
|
#include "modules/audio_coding/neteq/expand.h"
|
|
#include "modules/audio_coding/neteq/packet_buffer.h"
|
|
#include "modules/audio_coding/neteq/sync_buffer.h"
|
|
#include "modules/include/module_common_types.h"
|
|
|
|
namespace webrtc {
|
|
|
|
DecisionLogic* DecisionLogic::Create(int fs_hz,
|
|
size_t output_size_samples,
|
|
NetEqPlayoutMode playout_mode,
|
|
DecoderDatabase* decoder_database,
|
|
const PacketBuffer& packet_buffer,
|
|
DelayManager* delay_manager,
|
|
BufferLevelFilter* buffer_level_filter,
|
|
const TickTimer* tick_timer) {
|
|
switch (playout_mode) {
|
|
case kPlayoutOn:
|
|
case kPlayoutStreaming:
|
|
return new DecisionLogicNormal(
|
|
fs_hz, output_size_samples, playout_mode, decoder_database,
|
|
packet_buffer, delay_manager, buffer_level_filter, tick_timer);
|
|
case kPlayoutFax:
|
|
case kPlayoutOff:
|
|
return new DecisionLogicFax(
|
|
fs_hz, output_size_samples, playout_mode, decoder_database,
|
|
packet_buffer, delay_manager, buffer_level_filter, tick_timer);
|
|
}
|
|
// This line cannot be reached, but must be here to avoid compiler errors.
|
|
assert(false);
|
|
return NULL;
|
|
}
|
|
|
|
DecisionLogic::DecisionLogic(int fs_hz,
|
|
size_t output_size_samples,
|
|
NetEqPlayoutMode playout_mode,
|
|
DecoderDatabase* decoder_database,
|
|
const PacketBuffer& packet_buffer,
|
|
DelayManager* delay_manager,
|
|
BufferLevelFilter* buffer_level_filter,
|
|
const TickTimer* tick_timer)
|
|
: decoder_database_(decoder_database),
|
|
packet_buffer_(packet_buffer),
|
|
delay_manager_(delay_manager),
|
|
buffer_level_filter_(buffer_level_filter),
|
|
tick_timer_(tick_timer),
|
|
cng_state_(kCngOff),
|
|
packet_length_samples_(0),
|
|
sample_memory_(0),
|
|
prev_time_scale_(false),
|
|
timescale_countdown_(
|
|
tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1)),
|
|
num_consecutive_expands_(0),
|
|
playout_mode_(playout_mode) {
|
|
delay_manager_->set_streaming_mode(playout_mode_ == kPlayoutStreaming);
|
|
SetSampleRate(fs_hz, output_size_samples);
|
|
}
|
|
|
|
DecisionLogic::~DecisionLogic() = default;
|
|
|
|
void DecisionLogic::Reset() {
|
|
cng_state_ = kCngOff;
|
|
noise_fast_forward_ = 0;
|
|
packet_length_samples_ = 0;
|
|
sample_memory_ = 0;
|
|
prev_time_scale_ = false;
|
|
timescale_countdown_.reset();
|
|
num_consecutive_expands_ = 0;
|
|
}
|
|
|
|
void DecisionLogic::SoftReset() {
|
|
packet_length_samples_ = 0;
|
|
sample_memory_ = 0;
|
|
prev_time_scale_ = false;
|
|
timescale_countdown_ =
|
|
tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1);
|
|
}
|
|
|
|
void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
|
|
// TODO(hlundin): Change to an enumerator and skip assert.
|
|
assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000);
|
|
fs_mult_ = fs_hz / 8000;
|
|
output_size_samples_ = output_size_samples;
|
|
}
|
|
|
|
Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
|
|
const Expand& expand,
|
|
size_t decoder_frame_length,
|
|
const Packet* next_packet,
|
|
Modes prev_mode,
|
|
bool play_dtmf,
|
|
size_t generated_noise_samples,
|
|
bool* reset_decoder) {
|
|
// If last mode was CNG (or Expand, since this could be covering up for
|
|
// a lost CNG packet), remember that CNG is on. This is needed if comfort
|
|
// noise is interrupted by DTMF.
|
|
if (prev_mode == kModeRfc3389Cng) {
|
|
cng_state_ = kCngRfc3389On;
|
|
} else if (prev_mode == kModeCodecInternalCng) {
|
|
cng_state_ = kCngInternalOn;
|
|
}
|
|
|
|
const size_t samples_left =
|
|
sync_buffer.FutureLength() - expand.overlap_length();
|
|
const size_t cur_size_samples =
|
|
samples_left + packet_buffer_.NumSamplesInBuffer(decoder_frame_length);
|
|
|
|
prev_time_scale_ =
|
|
prev_time_scale_ && (prev_mode == kModeAccelerateSuccess ||
|
|
prev_mode == kModeAccelerateLowEnergy ||
|
|
prev_mode == kModePreemptiveExpandSuccess ||
|
|
prev_mode == kModePreemptiveExpandLowEnergy);
|
|
|
|
FilterBufferLevel(cur_size_samples, prev_mode);
|
|
|
|
return GetDecisionSpecialized(
|
|
sync_buffer, expand, decoder_frame_length, next_packet, prev_mode,
|
|
play_dtmf, reset_decoder, generated_noise_samples, cur_size_samples);
|
|
}
|
|
|
|
void DecisionLogic::ExpandDecision(Operations operation) {
|
|
if (operation == kExpand) {
|
|
num_consecutive_expands_++;
|
|
} else {
|
|
num_consecutive_expands_ = 0;
|
|
}
|
|
}
|
|
|
|
void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples,
|
|
Modes prev_mode) {
|
|
// Do not update buffer history if currently playing CNG since it will bias
|
|
// the filtered buffer level.
|
|
if ((prev_mode != kModeRfc3389Cng) && (prev_mode != kModeCodecInternalCng)) {
|
|
buffer_level_filter_->SetTargetBufferLevel(
|
|
delay_manager_->base_target_level());
|
|
|
|
size_t buffer_size_packets = 0;
|
|
if (packet_length_samples_ > 0) {
|
|
// Calculate size in packets.
|
|
buffer_size_packets = buffer_size_samples / packet_length_samples_;
|
|
}
|
|
int sample_memory_local = 0;
|
|
if (prev_time_scale_) {
|
|
sample_memory_local = sample_memory_;
|
|
timescale_countdown_ =
|
|
tick_timer_->GetNewCountdown(kMinTimescaleInterval);
|
|
}
|
|
buffer_level_filter_->Update(buffer_size_packets, sample_memory_local,
|
|
packet_length_samples_);
|
|
prev_time_scale_ = false;
|
|
}
|
|
}
|
|
|
|
} // namespace webrtc
|