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This reverts commit 80c4cca491
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Reason for revert: Breaks downstream tests.
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org
Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9421
Reviewed-on: https://webrtc-review.googlesource.com/84680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23706}
112 lines
4.8 KiB
C++
112 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 MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
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#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
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#include "modules/audio_coding/neteq/decision_logic.h"
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#include "rtc_base/constructormagic.h"
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#include "system_wrappers/include/field_trial.h"
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#include "typedefs.h" // NOLINT(build/include)
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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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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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postpone_decoding_after_expand_(field_trial::IsEnabled(
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"WebRTC-Audio-NetEqPostponeDecodingAfterExpand")) {}
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protected:
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static const int kReinitAfterExpands = 100;
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static const int kMaxWaitForPacket = 10;
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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 Packet* next_packet,
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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,
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size_t cur_size_samples) override;
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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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virtual Operations FuturePacketAvailable(const SyncBuffer& sync_buffer,
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const Expand& expand,
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size_t decoder_frame_length,
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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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size_t generated_noise_samples);
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// Returns the operation to do given that the expected packet is available.
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virtual Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
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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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virtual Operations NoPacket(bool play_dtmf);
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private:
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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,
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uint32_t target_timestamp,
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uint32_t available_timestamp,
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size_t generated_noise_samples);
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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 {
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return !timescale_countdown_ || timescale_countdown_->Finished();
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}
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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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const bool postpone_decoding_after_expand_;
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RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
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