
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}
113 lines
4.8 KiB
C++
113 lines
4.8 KiB
C++
/*
|
|
* 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_
|