Files
platform-external-webrtc/webrtc/modules/audio_coding/neteq/decision_logic.cc
Peter Kasting dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00

187 lines
7.2 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 "webrtc/modules/audio_coding/neteq/decision_logic.h"
#include <algorithm>
#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
#include "webrtc/modules/audio_coding/neteq/decision_logic_fax.h"
#include "webrtc/modules/audio_coding/neteq/decision_logic_normal.h"
#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
#include "webrtc/system_wrappers/interface/logging.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) {
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);
case kPlayoutFax:
case kPlayoutOff:
return new DecisionLogicFax(fs_hz,
output_size_samples,
playout_mode,
decoder_database,
packet_buffer,
delay_manager,
buffer_level_filter);
}
// 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)
: decoder_database_(decoder_database),
packet_buffer_(packet_buffer),
delay_manager_(delay_manager),
buffer_level_filter_(buffer_level_filter),
cng_state_(kCngOff),
generated_noise_samples_(0),
packet_length_samples_(0),
sample_memory_(0),
prev_time_scale_(false),
timescale_hold_off_(kMinTimescaleInterval),
num_consecutive_expands_(0),
playout_mode_(playout_mode) {
delay_manager_->set_streaming_mode(playout_mode_ == kPlayoutStreaming);
SetSampleRate(fs_hz, output_size_samples);
}
void DecisionLogic::Reset() {
cng_state_ = kCngOff;
generated_noise_samples_ = 0;
packet_length_samples_ = 0;
sample_memory_ = 0;
prev_time_scale_ = false;
timescale_hold_off_ = 0;
num_consecutive_expands_ = 0;
}
void DecisionLogic::SoftReset() {
packet_length_samples_ = 0;
sample_memory_ = 0;
prev_time_scale_ = false;
timescale_hold_off_ = kMinTimescaleInterval;
}
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 RTPHeader* packet_header,
Modes prev_mode,
bool play_dtmf, bool* reset_decoder) {
if (prev_mode == kModeRfc3389Cng ||
prev_mode == kModeCodecInternalCng ||
prev_mode == kModeExpand) {
// If last mode was CNG (or Expand, since this could be covering up for
// a lost CNG packet), increase the |generated_noise_samples_| counter.
generated_noise_samples_ += output_size_samples_;
// 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_database_,
decoder_frame_length);
LOG(LS_VERBOSE) << "Buffers: " << packet_buffer_.NumPacketsInBuffer() <<
" packets * " << decoder_frame_length << " samples/packet + " <<
samples_left << " samples in sync buffer = " << cur_size_samples;
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,
packet_header, prev_mode, play_dtmf,
reset_decoder);
}
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) {
const int elapsed_time_ms =
static_cast<int>(output_size_samples_ / (8 * fs_mult_));
delay_manager_->UpdateCounters(elapsed_time_ms);
// 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_hold_off_ = kMinTimescaleInterval;
}
buffer_level_filter_->Update(buffer_size_packets, sample_memory_local,
packet_length_samples_);
prev_time_scale_ = false;
}
timescale_hold_off_ = std::max(timescale_hold_off_ - 1, 0);
}
} // namespace webrtc