d94659dc27
This is the first public upload of the new NetEq, version 4. It has been through extensive internal review during the course of the project. TEST=trybots Review URL: https://webrtc-codereview.appspot.com/1073005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
107 lines
4.8 KiB
C++
107 lines
4.8 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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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_NORMAL_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_NORMAL_H_
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#include "webrtc/modules/audio_coding/neteq4/decision_logic.h"
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#include "webrtc/system_wrappers/interface/constructor_magic.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Implementation of the DecisionLogic class for playout modes kPlayoutOn and
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// kPlayoutStreaming.
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class DecisionLogicNormal : public DecisionLogic {
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public:
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// Constructor.
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DecisionLogicNormal(int fs_hz,
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int output_size_samples,
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NetEqPlayoutMode playout_mode,
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DecoderDatabase* decoder_database,
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const PacketBuffer& packet_buffer,
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DelayManager* delay_manager,
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BufferLevelFilter* buffer_level_filter)
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: DecisionLogic(fs_hz, output_size_samples, playout_mode,
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decoder_database, packet_buffer, delay_manager,
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buffer_level_filter) {
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}
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// Destructor.
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virtual ~DecisionLogicNormal() {}
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protected:
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// Returns the operation that should be done next. |sync_buffer| and |expand|
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// are provided for reference. |decoder_frame_length| is the number of samples
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// obtained from the last decoded frame. If there is a packet available, the
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// packet header should be supplied in |packet_header|; otherwise it should
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// be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
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// supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
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// should be set to true. The output variable |reset_decoder| will be set to
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// true if a reset is required; otherwise it is left unchanged (i.e., it can
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// remain true if it was true before the call).
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virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
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const Expand& expand,
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int decoder_frame_length,
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const RTPHeader* packet_header,
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Modes prev_mode, bool play_dtmf,
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bool* reset_decoder);
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private:
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static const int kAllowMergeWithoutExpandMs = 20; // 20 ms.
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static const int kReinitAfterExpands = 100;
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static const int kMaxWaitForPacket = 10;
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// Returns the operation given that the next available packet is a comfort
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// noise payload (RFC 3389 only, not codec-internal).
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Operations CngOperation(Modes prev_mode, uint32_t target_timestamp,
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uint32_t available_timestamp);
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// Returns the operation given that no packets are available (except maybe
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// a DTMF event, flagged by setting |play_dtmf| true).
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Operations NoPacket(bool play_dtmf);
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// Returns the operation to do given that the expected packet is available.
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Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
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// Returns the operation to do given that the expected packet is not
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// available, but a packet further into the future is at hand.
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Operations FuturePacketAvailable(const SyncBuffer& sync_buffer,
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const Expand& expand,
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int decoder_frame_length, 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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// Checks if enough time has elapsed since the last successful timescale
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// operation was done (i.e., accelerate or preemptive expand).
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bool TimescaleAllowed() const { return timescale_hold_off_ == 0; }
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// Checks if the current (filtered) buffer level is under the target level.
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bool UnderTargetLevel() const;
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// Checks if |timestamp_leap| is so long into the future that a reset due
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// to exceeding kReinitAfterExpands will be done.
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bool ReinitAfterExpands(uint32_t timestamp_leap) const;
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// Checks if we still have not done enough expands to cover the distance from
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// the last decoded packet to the next available packet, the distance beeing
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// conveyed in |timestamp_leap|.
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bool PacketTooEarly(uint32_t timestamp_leap) const;
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// Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
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bool MaxWaitForPacket() const;
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DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_NORMAL_H_
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