47b17dc59c
The timescale_holdoff_ is a counter in the DecisionLogic class. The purpose is to enforce a minimum number of GetAudio calls between (successfull) time-scaling operations (i.e., Accelerate and Pre-emptive Expand operations). With this change, the counter is replaced with a Countdown timer obtained from a TickTimer object. BUG=webrtc:5608 R=tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1945863002 . Cr-Commit-Position: refs/heads/master@{#12670}
67 lines
2.8 KiB
C++
67 lines
2.8 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_NETEQ_DECISION_LOGIC_FAX_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/audio_coding/neteq/decision_logic.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 kPlayoutFax and
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// kPlayoutOff.
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class DecisionLogicFax : public DecisionLogic {
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public:
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// Constructor.
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DecisionLogicFax(int fs_hz,
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size_t 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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const TickTimer* tick_timer)
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: DecisionLogic(fs_hz,
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output_size_samples,
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playout_mode,
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decoder_database,
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packet_buffer,
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delay_manager,
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buffer_level_filter,
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tick_timer) {}
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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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Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
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const Expand& expand,
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size_t decoder_frame_length,
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const RTPHeader* packet_header,
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Modes prev_mode,
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bool play_dtmf,
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bool* reset_decoder,
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size_t generated_noise_samples) override;
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicFax);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
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