Files
platform-external-webrtc/webrtc/modules/audio_coding/neteq/decision_logic.cc
ossu 61a208b1b8 Added a ParsePayload method to AudioDecoder.
It allows the decoder to split the input up into usable chunks before
they are put into NetEq's PacketBuffer. Eventually, all packet splitting
will move into ParsePayload.

There's currently a base implementation of ParsePayload. It will
generate a single Frame that calls the underlying AudioDecoder for
getting Duration() and to Decode.

BUG=webrtc:5805
BUG=chromium:428099

Review-Url: https://codereview.webrtc.org/2326953003
Cr-Commit-Position: refs/heads/master@{#14300}
2016-09-20 08:38:09 +00:00

172 lines
6.6 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/include/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,
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 RTPHeader* packet_header,
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,
packet_header, prev_mode, play_dtmf,
reset_decoder, generated_noise_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