webrtc_m130/modules/audio_coding/neteq/decision_logic_normal.h

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Revert "NetEq: Deprecate playout modes Fax, Off and Streaming" This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1. 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}
2018-06-21 12:36:28 +00:00
/*
* 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 MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
#include "modules/audio_coding/neteq/decision_logic.h"
#include "rtc_base/constructormagic.h"
#include "system_wrappers/include/field_trial.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
// Implementation of the DecisionLogic class for playout modes kPlayoutOn and
// kPlayoutStreaming.
class DecisionLogicNormal : public DecisionLogic {
public:
// Constructor.
DecisionLogicNormal(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)
: DecisionLogic(fs_hz,
output_size_samples,
playout_mode,
decoder_database,
packet_buffer,
delay_manager,
buffer_level_filter,
tick_timer),
postpone_decoding_after_expand_(field_trial::IsEnabled(
"WebRTC-Audio-NetEqPostponeDecodingAfterExpand")) {}
protected:
static const int kReinitAfterExpands = 100;
static const int kMaxWaitForPacket = 10;
Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
const Expand& expand,
size_t decoder_frame_length,
const Packet* next_packet,
Modes prev_mode,
bool play_dtmf,
bool* reset_decoder,
size_t generated_noise_samples,
size_t cur_size_samples) override;
// Returns the operation to do given that the expected packet is not
// available, but a packet further into the future is at hand.
virtual Operations FuturePacketAvailable(const SyncBuffer& sync_buffer,
const Expand& expand,
size_t decoder_frame_length,
Modes prev_mode,
uint32_t target_timestamp,
uint32_t available_timestamp,
bool play_dtmf,
size_t generated_noise_samples);
// Returns the operation to do given that the expected packet is available.
virtual Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
// Returns the operation given that no packets are available (except maybe
// a DTMF event, flagged by setting |play_dtmf| true).
virtual Operations NoPacket(bool play_dtmf);
private:
// Returns the operation given that the next available packet is a comfort
// noise payload (RFC 3389 only, not codec-internal).
Operations CngOperation(Modes prev_mode,
uint32_t target_timestamp,
uint32_t available_timestamp,
size_t generated_noise_samples);
// Checks if enough time has elapsed since the last successful timescale
// operation was done (i.e., accelerate or preemptive expand).
bool TimescaleAllowed() const {
return !timescale_countdown_ || timescale_countdown_->Finished();
}
// Checks if the current (filtered) buffer level is under the target level.
bool UnderTargetLevel() const;
// Checks if |timestamp_leap| is so long into the future that a reset due
// to exceeding kReinitAfterExpands will be done.
bool ReinitAfterExpands(uint32_t timestamp_leap) const;
// Checks if we still have not done enough expands to cover the distance from
// the last decoded packet to the next available packet, the distance beeing
// conveyed in |timestamp_leap|.
bool PacketTooEarly(uint32_t timestamp_leap) const;
// Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
bool MaxWaitForPacket() const;
const bool postpone_decoding_after_expand_;
RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_