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/*
* 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/neteq/defines.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Forward declarations.
class BufferLevelFilter;
class DecoderDatabase;
class DelayManager;
class Expand;
class PacketBuffer;
class SyncBuffer;
struct RTPHeader;
// This is the base class for the decision tree implementations. Derived classes
// must implement the method GetDecisionSpecialized().
class DecisionLogic {
public:
// Static factory function which creates different types of objects depending
// on the |playout_mode|.
static 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);
// Constructor.
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);
// Destructor.
virtual ~DecisionLogic() {}
// Resets object to a clean state.
void Reset();
// Resets parts of the state. Typically done when switching codecs.
void SoftReset();
// Sets the sample rate and the output block size.
void SetSampleRate(int fs_hz, size_t output_size_samples);
// Returns the operation that should be done next. |sync_buffer| and |expand|
// are provided for reference. |decoder_frame_length| is the number of samples
// obtained from the last decoded frame. If there is a packet available, the
// packet header should be supplied in |packet_header|; otherwise it should
// be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
// supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
// should be set to true. The output variable |reset_decoder| will be set to
// true if a reset is required; otherwise it is left unchanged (i.e., it can
// remain true if it was true before the call).
// This method end with calling GetDecisionSpecialized to get the actual
// return value.
Operations 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);
// These methods test the |cng_state_| for different conditions.
bool CngRfc3389On() const { return cng_state_ == kCngRfc3389On; }
bool CngOff() const { return cng_state_ == kCngOff; }
// Resets the |cng_state_| to kCngOff.
void SetCngOff() { cng_state_ = kCngOff; }
// Reports back to DecisionLogic whether the decision to do expand remains or
// not. Note that this is necessary, since an expand decision can be changed
// to kNormal in NetEqImpl::GetDecision if there is still enough data in the
// sync buffer.
virtual void ExpandDecision(Operations operation);
// Adds |value| to |sample_memory_|.
void AddSampleMemory(int32_t value) {
sample_memory_ += value;
}
// Accessors and mutators.
void set_sample_memory(int32_t value) { sample_memory_ = value; }
size_t generated_noise_samples() const { return generated_noise_samples_; }
void set_generated_noise_samples(size_t value) {
generated_noise_samples_ = value;
}
size_t packet_length_samples() const { return packet_length_samples_; }
void set_packet_length_samples(size_t value) {
packet_length_samples_ = value;
}
void set_prev_time_scale(bool value) { prev_time_scale_ = value; }
NetEqPlayoutMode playout_mode() const { return playout_mode_; }
protected:
// The value 6 sets maximum time-stretch rate to about 100 ms/s.
static const int kMinTimescaleInterval = 6;
enum CngState {
kCngOff,
kCngRfc3389On,
kCngInternalOn
};
// Returns the operation that should be done next. |sync_buffer| and |expand|
// are provided for reference. |decoder_frame_length| is the number of samples
// obtained from the last decoded frame. If there is a packet available, the
// packet header should be supplied in |packet_header|; otherwise it should
// be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
// supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
// should be set to true. The output variable |reset_decoder| will be set to
// true if a reset is required; otherwise it is left unchanged (i.e., it can
// remain true if it was true before the call).
// Should be implemented by derived classes.
virtual Operations GetDecisionSpecialized(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) = 0;
// Updates the |buffer_level_filter_| with the current buffer level
// |buffer_size_packets|.
void FilterBufferLevel(size_t buffer_size_packets, Modes prev_mode);
DecoderDatabase* decoder_database_;
const PacketBuffer& packet_buffer_;
DelayManager* delay_manager_;
BufferLevelFilter* buffer_level_filter_;
int fs_mult_;
size_t output_size_samples_;
CngState cng_state_; // Remember if comfort noise is interrupted by other
// event (e.g., DTMF).
size_t generated_noise_samples_;
size_t packet_length_samples_;
int sample_memory_;
bool prev_time_scale_;
int timescale_hold_off_;
int num_consecutive_expands_;
const NetEqPlayoutMode playout_mode_;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogic);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
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