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Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"

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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}
This commit is contained in:
Henrik Lundin
2018-06-21 12:36:28 +00:00
committed by Commit Bot
parent 07efe436c9
commit 1ff41eb784
23 changed files with 782 additions and 479 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
modules/audio_coding/BUILD.gn
View File

@ -1057,6 +1057,10 @@ rtc_static_library("neteq") {
"neteq/cross_correlation.h",
"neteq/decision_logic.cc",
"neteq/decision_logic.h",
"neteq/decision_logic_fax.cc",
"neteq/decision_logic_fax.h",
"neteq/decision_logic_normal.cc",
"neteq/decision_logic_normal.h",
"neteq/decoder_database.cc",
"neteq/decoder_database.h",
"neteq/defines.h",

4
modules/audio_coding/acm2/acm_receiver_unittest.cc
View File

@ -292,7 +292,7 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
protected:
AcmReceiverTestFaxModeOldApi() {
config_.neteq_config.for_test_no_time_stretching = true;
config_.neteq_config.playout_mode = kPlayoutFax;
}
void RunVerifyAudioFrame(RentACodec::CodecId codec_id) {
@ -301,7 +301,7 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
// timestamp increments predictable; in normal mode, NetEq may decide to do
// accelerate or pre-emptive expand operations after some time, offsetting
// the timestamp.
EXPECT_TRUE(config_.neteq_config.for_test_no_time_stretching);
EXPECT_EQ(kPlayoutFax, config_.neteq_config.playout_mode);
const RentACodec::CodecId kCodecId[] = {codec_id};
AddSetOfCodecs(kCodecId);

258
modules/audio_coding/neteq/decision_logic.cc
View File

@ -10,37 +10,47 @@
#include "modules/audio_coding/neteq/decision_logic.h"
#include <assert.h>
#include <algorithm>
#include <limits>
#include "modules/audio_coding/neteq/buffer_level_filter.h"
#include "modules/audio_coding/neteq/decoder_database.h"
#include "modules/audio_coding/neteq/decision_logic_fax.h"
#include "modules/audio_coding/neteq/decision_logic_normal.h"
#include "modules/audio_coding/neteq/delay_manager.h"
#include "modules/audio_coding/neteq/expand.h"
#include "modules/audio_coding/neteq/packet_buffer.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/include/module_common_types.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
DecisionLogic* DecisionLogic::Create(int fs_hz,
size_t output_size_samples,
bool disallow_time_stretching,
NetEqPlayoutMode playout_mode,
DecoderDatabase* decoder_database,
const PacketBuffer& packet_buffer,
DelayManager* delay_manager,
BufferLevelFilter* buffer_level_filter,
const TickTimer* tick_timer) {
return new DecisionLogic(fs_hz, output_size_samples, disallow_time_stretching,
decoder_database, packet_buffer, delay_manager,
buffer_level_filter, tick_timer);
switch (playout_mode) {
case kPlayoutOn:
case kPlayoutStreaming:
return new DecisionLogicNormal(
fs_hz, output_size_samples, playout_mode, decoder_database,
packet_buffer, delay_manager, buffer_level_filter, tick_timer);
case kPlayoutFax:
case kPlayoutOff:
return new DecisionLogicFax(
fs_hz, output_size_samples, playout_mode, decoder_database,
packet_buffer, delay_manager, buffer_level_filter, tick_timer);
}
// This line cannot be reached, but must be here to avoid compiler errors.
assert(false);
return NULL;
}
DecisionLogic::DecisionLogic(int fs_hz,
size_t output_size_samples,
bool disallow_time_stretching,
NetEqPlayoutMode playout_mode,
DecoderDatabase* decoder_database,
const PacketBuffer& packet_buffer,
DelayManager* delay_manager,
@ -55,13 +65,11 @@ DecisionLogic::DecisionLogic(int fs_hz,
packet_length_samples_(0),
sample_memory_(0),
prev_time_scale_(false),
disallow_time_stretching_(disallow_time_stretching),
timescale_countdown_(
tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1)),
num_consecutive_expands_(0),
postpone_decoding_after_expand_(field_trial::IsEnabled(
"WebRTC-Audio-NetEqPostponeDecodingAfterExpand")) {
delay_manager_->set_streaming_mode(false);
playout_mode_(playout_mode) {
delay_manager_->set_streaming_mode(playout_mode_ == kPlayoutStreaming);
SetSampleRate(fs_hz, output_size_samples);
}
@ -160,228 +168,4 @@ void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples,
}
}
Operations DecisionLogic::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) {
// Guard for errors, to avoid getting stuck in error mode.
if (prev_mode == kModeError) {
if (!next_packet) {
return kExpand;
} else {
return kUndefined; // Use kUndefined to flag for a reset.
}
}
uint32_t target_timestamp = sync_buffer.end_timestamp();
uint32_t available_timestamp = 0;
bool is_cng_packet = false;
if (next_packet) {
available_timestamp = next_packet->timestamp;
is_cng_packet =
decoder_database_->IsComfortNoise(next_packet->payload_type);
}
if (is_cng_packet) {
return CngOperation(prev_mode, target_timestamp, available_timestamp,
generated_noise_samples);
}
// Handle the case with no packet at all available (except maybe DTMF).
if (!next_packet) {
return NoPacket(play_dtmf);
}
// If the expand period was very long, reset NetEQ since it is likely that the
// sender was restarted.
if (num_consecutive_expands_ > kReinitAfterExpands) {
*reset_decoder = true;
return kNormal;
}
// Make sure we don't restart audio too soon after an expansion to avoid
// running out of data right away again. We should only wait if there are no
// DTX or CNG packets in the buffer (otherwise we should just play out what we
// have, since we cannot know the exact duration of DTX or CNG packets), and
// if the mute factor is low enough (otherwise the expansion was short enough
// to not be noticable).
// Note that the MuteFactor is in Q14, so a value of 16384 corresponds to 1.
if (postpone_decoding_after_expand_ && prev_mode == kModeExpand &&
!packet_buffer_.ContainsDtxOrCngPacket(decoder_database_) &&
cur_size_samples<static_cast<size_t>(delay_manager_->TargetLevel() *
packet_length_samples_)>> 8 &&
expand.MuteFactor(0) < 16384 / 2) {
return kExpand;
}
const uint32_t five_seconds_samples =
static_cast<uint32_t>(5 * 8000 * fs_mult_);
// Check if the required packet is available.
if (target_timestamp == available_timestamp) {
return ExpectedPacketAvailable(prev_mode, play_dtmf);
} else if (!PacketBuffer::IsObsoleteTimestamp(
available_timestamp, target_timestamp, five_seconds_samples)) {
return FuturePacketAvailable(
sync_buffer, expand, decoder_frame_length, prev_mode, target_timestamp,
available_timestamp, play_dtmf, generated_noise_samples);
} else {
// This implies that available_timestamp < target_timestamp, which can
// happen when a new stream or codec is received. Signal for a reset.
return kUndefined;
}
}
Operations DecisionLogic::CngOperation(Modes prev_mode,
uint32_t target_timestamp,
uint32_t available_timestamp,
size_t generated_noise_samples) {
// Signed difference between target and available timestamp.
int32_t timestamp_diff = static_cast<int32_t>(
static_cast<uint32_t>(generated_noise_samples + target_timestamp) -
available_timestamp);
int32_t optimal_level_samp = static_cast<int32_t>(
(delay_manager_->TargetLevel() * packet_length_samples_) >> 8);
const int64_t excess_waiting_time_samp =
-static_cast<int64_t>(timestamp_diff) - optimal_level_samp;
if (excess_waiting_time_samp > optimal_level_samp / 2) {
// The waiting time for this packet will be longer than 1.5
// times the wanted buffer delay. Apply fast-forward to cut the
// waiting time down to the optimal.
noise_fast_forward_ = rtc::dchecked_cast<size_t>(noise_fast_forward_ +
excess_waiting_time_samp);
timestamp_diff =
rtc::saturated_cast<int32_t>(timestamp_diff + excess_waiting_time_samp);
}
if (timestamp_diff < 0 && prev_mode == kModeRfc3389Cng) {
// Not time to play this packet yet. Wait another round before using this
// packet. Keep on playing CNG from previous CNG parameters.
return kRfc3389CngNoPacket;
} else {
// Otherwise, go for the CNG packet now.
noise_fast_forward_ = 0;
return kRfc3389Cng;
}
}
Operations DecisionLogic::NoPacket(bool play_dtmf) {
if (cng_state_ == kCngRfc3389On) {
// Keep on playing comfort noise.
return kRfc3389CngNoPacket;
} else if (cng_state_ == kCngInternalOn) {
// Keep on playing codec internal comfort noise.
return kCodecInternalCng;
} else if (play_dtmf) {
return kDtmf;
} else {
// Nothing to play, do expand.
return kExpand;
}
}
Operations DecisionLogic::ExpectedPacketAvailable(Modes prev_mode,
bool play_dtmf) {
if (!disallow_time_stretching_ && prev_mode != kModeExpand && !play_dtmf) {
// Check criterion for time-stretching.
int low_limit, high_limit;
delay_manager_->BufferLimits(&low_limit, &high_limit);
if (buffer_level_filter_->filtered_current_level() >= high_limit << 2)
return kFastAccelerate;
if (TimescaleAllowed()) {
if (buffer_level_filter_->filtered_current_level() >= high_limit)
return kAccelerate;
if (buffer_level_filter_->filtered_current_level() < low_limit)
return kPreemptiveExpand;
}
}
return kNormal;
}
Operations DecisionLogic::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) {
// Required packet is not available, but a future packet is.
// Check if we should continue with an ongoing expand because the new packet
// is too far into the future.
uint32_t timestamp_leap = available_timestamp - target_timestamp;
if ((prev_mode == kModeExpand) && !ReinitAfterExpands(timestamp_leap) &&
!MaxWaitForPacket() && PacketTooEarly(timestamp_leap) &&
UnderTargetLevel()) {
if (play_dtmf) {
// Still have DTMF to play, so do not do expand.
return kDtmf;
} else {
// Nothing to play.
return kExpand;
}
}
const size_t samples_left =
sync_buffer.FutureLength() - expand.overlap_length();
const size_t cur_size_samples =
samples_left + packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
// If previous was comfort noise, then no merge is needed.
if (prev_mode == kModeRfc3389Cng || prev_mode == kModeCodecInternalCng) {
// Keep the same delay as before the CNG, but make sure that the number of
// samples in buffer is no higher than 4 times the optimal level. (Note that
// TargetLevel() is in Q8.)
if (static_cast<uint32_t>(generated_noise_samples + target_timestamp) >=
available_timestamp ||
cur_size_samples >
((delay_manager_->TargetLevel() * packet_length_samples_) >> 8) *
4) {
// Time to play this new packet.
return kNormal;
} else {
// Too early to play this new packet; keep on playing comfort noise.
if (prev_mode == kModeRfc3389Cng) {
return kRfc3389CngNoPacket;
} else { // prevPlayMode == kModeCodecInternalCng.
return kCodecInternalCng;
}
}
}
// Do not merge unless we have done an expand before.
if (prev_mode == kModeExpand) {
return kMerge;
} else if (play_dtmf) {
// Play DTMF instead of expand.
return kDtmf;
} else {
return kExpand;
}
}
bool DecisionLogic::UnderTargetLevel() const {
return buffer_level_filter_->filtered_current_level() <=
delay_manager_->TargetLevel();
}
bool DecisionLogic::ReinitAfterExpands(uint32_t timestamp_leap) const {
return timestamp_leap >=
static_cast<uint32_t>(output_size_samples_ * kReinitAfterExpands);
}
bool DecisionLogic::PacketTooEarly(uint32_t timestamp_leap) const {
return timestamp_leap >
static_cast<uint32_t>(output_size_samples_ * num_consecutive_expands_);
}
bool DecisionLogic::MaxWaitForPacket() const {
return num_consecutive_expands_ >= kMaxWaitForPacket;
}
} // namespace webrtc

80
modules/audio_coding/neteq/decision_logic.h
View File

@ -28,34 +28,32 @@ class PacketBuffer;
class SyncBuffer;
struct Packet;
// This is the class for the decision tree implementation.
// This is the base class for the decision tree implementations. Derived classes
// must implement the method GetDecisionSpecialized().
class DecisionLogic {
public:
// Static factory function which creates different types of objects depending
// on the |playout_mode|.
static DecisionLogic* Create(int fs_hz,
size_t output_size_samples,
bool disallow_time_stretching,
NetEqPlayoutMode playout_mode,
DecoderDatabase* decoder_database,
const PacketBuffer& packet_buffer,
DelayManager* delay_manager,
BufferLevelFilter* buffer_level_filter,
const TickTimer* tick_timer);
static const int kReinitAfterExpands = 100;
static const int kMaxWaitForPacket = 10;
// Constructor.
DecisionLogic(int fs_hz,
size_t output_size_samples,
bool disallow_time_stretching,
NetEqPlayoutMode playout_mode,
DecoderDatabase* decoder_database,
const PacketBuffer& packet_buffer,
DelayManager* delay_manager,
BufferLevelFilter* buffer_level_filter,
const TickTimer* tick_timer);
~DecisionLogic();
virtual ~DecisionLogic();
// Resets object to a clean state.
void Reset();
@ -96,7 +94,7 @@ class DecisionLogic {
// not. Note that this is necessary, since an expand decision can be changed
// to kNormal in NetEqImpl::GetDecision if there is still enough data in the
// sync buffer.
void ExpandDecision(Operations operation);
virtual void ExpandDecision(Operations operation);
// Adds |value| to |sample_memory_|.
void AddSampleMemory(int32_t value) { sample_memory_ += value; }
@ -109,17 +107,14 @@ class DecisionLogic {
packet_length_samples_ = value;
}
void set_prev_time_scale(bool value) { prev_time_scale_ = value; }
NetEqPlayoutMode playout_mode() const { return playout_mode_; }
private:
protected:
// The value 5 sets maximum time-stretch rate to about 100 ms/s.
static const int kMinTimescaleInterval = 5;
enum CngState { kCngOff, kCngRfc3389On, kCngInternalOn };
// Updates the |buffer_level_filter_| with the current buffer level
// |buffer_size_packets|.
void FilterBufferLevel(size_t buffer_size_packets, Modes prev_mode);
// Returns the operation that should be done next. |sync_buffer| and |expand|
// are provided for reference. |decoder_frame_length| is the number of samples
// obtained from the last decoded frame. If there is a packet available, it
@ -128,9 +123,8 @@ class DecisionLogic {
// |prev_mode|. If there is a DTMF event to play, |play_dtmf| should be set to
// true. The output variable |reset_decoder| will be set to true if a reset is
// required; otherwise it is left unchanged (i.e., it can remain true if it
// was true before the call).
// TODO(henrik.lundin) Fold this method into GetDecision.
Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
// was true before the call). Should be implemented by derived classes.
virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
const Expand& expand,
size_t decoder_frame_length,
const Packet* next_packet,
@ -138,53 +132,11 @@ class DecisionLogic {
bool play_dtmf,
bool* reset_decoder,
size_t generated_noise_samples,
size_t cur_size_samples);
size_t cur_size_samples) = 0;
// 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);
// Returns the operation given that no packets are available (except maybe
// a DTMF event, flagged by setting |play_dtmf| true).
Operations NoPacket(bool play_dtmf);
// Returns the operation to do given that the expected packet is available.
Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
// Returns the operation to do given that the expected packet is not
// available, but a packet further into the future is at hand.
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);
// 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;
// Updates the |buffer_level_filter_| with the current buffer level
// |buffer_size_packets|.
void FilterBufferLevel(size_t buffer_size_packets, Modes prev_mode);
DecoderDatabase* decoder_database_;
const PacketBuffer& packet_buffer_;
@ -199,11 +151,11 @@ class DecisionLogic {
size_t packet_length_samples_;
int sample_memory_;
bool prev_time_scale_;
bool disallow_time_stretching_;
std::unique_ptr<TickTimer::Countdown> timescale_countdown_;
int num_consecutive_expands_;
const bool postpone_decoding_after_expand_;
const NetEqPlayoutMode playout_mode_;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogic);
};

103
modules/audio_coding/neteq/decision_logic_fax.cc Normal file
View File

@ -0,0 +1,103 @@
/*
* 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.
*/
#include "modules/audio_coding/neteq/decision_logic_fax.h"
#include <assert.h>
#include <algorithm>
#include "modules/audio_coding/neteq/decoder_database.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
namespace webrtc {
Operations DecisionLogicFax::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*/) {
assert(playout_mode_ == kPlayoutFax || playout_mode_ == kPlayoutOff);
uint32_t target_timestamp = sync_buffer.end_timestamp();
uint32_t available_timestamp = 0;
int is_cng_packet = 0;
if (next_packet) {
available_timestamp = next_packet->timestamp;
is_cng_packet =
decoder_database_->IsComfortNoise(next_packet->payload_type);
}
if (is_cng_packet) {
if (static_cast<int32_t>((generated_noise_samples + target_timestamp) -
available_timestamp) >= 0) {
// Time to play this packet now.
return kRfc3389Cng;
} else {
// Wait before playing this packet.
return kRfc3389CngNoPacket;
}
}
if (!next_packet) {
// No packet. If in CNG mode, play as usual. Otherwise, use other method to
// generate data.
if (cng_state_ == kCngRfc3389On) {
// Continue playing comfort noise.
return kRfc3389CngNoPacket;
} else if (cng_state_ == kCngInternalOn) {
// Continue playing codec-internal comfort noise.
return kCodecInternalCng;
} else {
// Nothing to play. Generate some data to play out.
switch (playout_mode_) {
case kPlayoutOff:
return kAlternativePlc;
case kPlayoutFax:
return kAudioRepetition;
default:
assert(false);
return kUndefined;
}
}
} else if (target_timestamp == available_timestamp) {
return kNormal;
} else {
if (static_cast<int32_t>((generated_noise_samples + target_timestamp) -
available_timestamp) >= 0) {
return kNormal;
} else {
// If currently playing comfort noise, continue with that. Do not
// increase the timestamp counter since generated_noise_stopwatch_ in
// NetEqImpl will take care of the time-keeping.
if (cng_state_ == kCngRfc3389On) {
return kRfc3389CngNoPacket;
} else if (cng_state_ == kCngInternalOn) {
return kCodecInternalCng;
} else {
// Otherwise, do packet-loss concealment and increase the
// timestamp while waiting for the time to play this packet.
switch (playout_mode_) {
case kPlayoutOff:
return kAlternativePlcIncreaseTimestamp;
case kPlayoutFax:
return kAudioRepetitionIncreaseTimestamp;
default:
assert(0);
return kUndefined;
}
}
}
}
}
} // namespace webrtc

58
modules/audio_coding/neteq/decision_logic_fax.h Normal file
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@ -0,0 +1,58 @@
/*
* 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_FAX_H_
#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
#include "modules/audio_coding/neteq/decision_logic.h"
#include "rtc_base/constructormagic.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
// Implementation of the DecisionLogic class for playout modes kPlayoutFax and
// kPlayoutOff.
class DecisionLogicFax : public DecisionLogic {
public:
// Constructor.
DecisionLogicFax(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) {}
protected:
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;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicFax);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_

253
modules/audio_coding/neteq/decision_logic_normal.cc Normal file
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@ -0,0 +1,253 @@
/*
* 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.
*/
#include "modules/audio_coding/neteq/decision_logic_normal.h"
#include <assert.h>
#include <algorithm>
#include <limits>
#include "modules/audio_coding/neteq/buffer_level_filter.h"
#include "modules/audio_coding/neteq/decoder_database.h"
#include "modules/audio_coding/neteq/delay_manager.h"
#include "modules/audio_coding/neteq/expand.h"
#include "modules/audio_coding/neteq/packet_buffer.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
namespace webrtc {
Operations DecisionLogicNormal::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) {
assert(playout_mode_ == kPlayoutOn || playout_mode_ == kPlayoutStreaming);
// Guard for errors, to avoid getting stuck in error mode.
if (prev_mode == kModeError) {
if (!next_packet) {
return kExpand;
} else {
return kUndefined; // Use kUndefined to flag for a reset.
}
}
uint32_t target_timestamp = sync_buffer.end_timestamp();
uint32_t available_timestamp = 0;
bool is_cng_packet = false;
if (next_packet) {
available_timestamp = next_packet->timestamp;
is_cng_packet =
decoder_database_->IsComfortNoise(next_packet->payload_type);
}
if (is_cng_packet) {
return CngOperation(prev_mode, target_timestamp, available_timestamp,
generated_noise_samples);
}
// Handle the case with no packet at all available (except maybe DTMF).
if (!next_packet) {
return NoPacket(play_dtmf);
}
// If the expand period was very long, reset NetEQ since it is likely that the
// sender was restarted.
if (num_consecutive_expands_ > kReinitAfterExpands) {
*reset_decoder = true;
return kNormal;
}
// Make sure we don't restart audio too soon after an expansion to avoid
// running out of data right away again. We should only wait if there are no
// DTX or CNG packets in the buffer (otherwise we should just play out what we
// have, since we cannot know the exact duration of DTX or CNG packets), and
// if the mute factor is low enough (otherwise the expansion was short enough
// to not be noticable).
// Note that the MuteFactor is in Q14, so a value of 16384 corresponds to 1.
if (postpone_decoding_after_expand_ && prev_mode == kModeExpand &&
!packet_buffer_.ContainsDtxOrCngPacket(decoder_database_) &&
cur_size_samples<static_cast<size_t>(delay_manager_->TargetLevel() *
packet_length_samples_)>> 8 &&
expand.MuteFactor(0) < 16384 / 2) {
return kExpand;
}
const uint32_t five_seconds_samples =
static_cast<uint32_t>(5 * 8000 * fs_mult_);
// Check if the required packet is available.
if (target_timestamp == available_timestamp) {
return ExpectedPacketAvailable(prev_mode, play_dtmf);
} else if (!PacketBuffer::IsObsoleteTimestamp(
available_timestamp, target_timestamp, five_seconds_samples)) {
return FuturePacketAvailable(
sync_buffer, expand, decoder_frame_length, prev_mode, target_timestamp,
available_timestamp, play_dtmf, generated_noise_samples);
} else {
// This implies that available_timestamp < target_timestamp, which can
// happen when a new stream or codec is received. Signal for a reset.
return kUndefined;
}
}
Operations DecisionLogicNormal::CngOperation(Modes prev_mode,
uint32_t target_timestamp,
uint32_t available_timestamp,
size_t generated_noise_samples) {
// Signed difference between target and available timestamp.
int32_t timestamp_diff = static_cast<int32_t>(
static_cast<uint32_t>(generated_noise_samples + target_timestamp) -
available_timestamp);
int32_t optimal_level_samp = static_cast<int32_t>(
(delay_manager_->TargetLevel() * packet_length_samples_) >> 8);
const int64_t excess_waiting_time_samp =
-static_cast<int64_t>(timestamp_diff) - optimal_level_samp;
if (excess_waiting_time_samp > optimal_level_samp / 2) {
// The waiting time for this packet will be longer than 1.5
// times the wanted buffer delay. Apply fast-forward to cut the
// waiting time down to the optimal.
noise_fast_forward_ = rtc::dchecked_cast<size_t>(noise_fast_forward_ +
excess_waiting_time_samp);
timestamp_diff =
rtc::saturated_cast<int32_t>(timestamp_diff + excess_waiting_time_samp);
}
if (timestamp_diff < 0 && prev_mode == kModeRfc3389Cng) {
// Not time to play this packet yet. Wait another round before using this
// packet. Keep on playing CNG from previous CNG parameters.
return kRfc3389CngNoPacket;
} else {
// Otherwise, go for the CNG packet now.
noise_fast_forward_ = 0;
return kRfc3389Cng;
}
}
Operations DecisionLogicNormal::NoPacket(bool play_dtmf) {
if (cng_state_ == kCngRfc3389On) {
// Keep on playing comfort noise.
return kRfc3389CngNoPacket;
} else if (cng_state_ == kCngInternalOn) {
// Keep on playing codec internal comfort noise.
return kCodecInternalCng;
} else if (play_dtmf) {
return kDtmf;
} else {
// Nothing to play, do expand.
return kExpand;
}
}
Operations DecisionLogicNormal::ExpectedPacketAvailable(Modes prev_mode,
bool play_dtmf) {
if (prev_mode != kModeExpand && !play_dtmf) {
// Check criterion for time-stretching.
int low_limit, high_limit;
delay_manager_->BufferLimits(&low_limit, &high_limit);
if (buffer_level_filter_->filtered_current_level() >= high_limit << 2)
return kFastAccelerate;
if (TimescaleAllowed()) {
if (buffer_level_filter_->filtered_current_level() >= high_limit)
return kAccelerate;
if (buffer_level_filter_->filtered_current_level() < low_limit)
return kPreemptiveExpand;
}
}
return kNormal;
}
Operations DecisionLogicNormal::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) {
// Required packet is not available, but a future packet is.
// Check if we should continue with an ongoing expand because the new packet
// is too far into the future.
uint32_t timestamp_leap = available_timestamp - target_timestamp;
if ((prev_mode == kModeExpand) && !ReinitAfterExpands(timestamp_leap) &&
!MaxWaitForPacket() && PacketTooEarly(timestamp_leap) &&
UnderTargetLevel()) {
if (play_dtmf) {
// Still have DTMF to play, so do not do expand.
return kDtmf;
} else {
// Nothing to play.
return kExpand;
}
}
const size_t samples_left =
sync_buffer.FutureLength() - expand.overlap_length();
const size_t cur_size_samples =
samples_left + packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
// If previous was comfort noise, then no merge is needed.
if (prev_mode == kModeRfc3389Cng || prev_mode == kModeCodecInternalCng) {
// Keep the same delay as before the CNG, but make sure that the number of
// samples in buffer is no higher than 4 times the optimal level. (Note that
// TargetLevel() is in Q8.)
if (static_cast<uint32_t>(generated_noise_samples + target_timestamp) >=
available_timestamp ||
cur_size_samples >
((delay_manager_->TargetLevel() * packet_length_samples_) >> 8) *
4) {
// Time to play this new packet.
return kNormal;
} else {
// Too early to play this new packet; keep on playing comfort noise.
if (prev_mode == kModeRfc3389Cng) {
return kRfc3389CngNoPacket;
} else { // prevPlayMode == kModeCodecInternalCng.
return kCodecInternalCng;
}
}
}
// Do not merge unless we have done an expand before.
if (prev_mode == kModeExpand) {
return kMerge;
} else if (play_dtmf) {
// Play DTMF instead of expand.
return kDtmf;
} else {
return kExpand;
}
}
bool DecisionLogicNormal::UnderTargetLevel() const {
return buffer_level_filter_->filtered_current_level() <=
delay_manager_->TargetLevel();
}
bool DecisionLogicNormal::ReinitAfterExpands(uint32_t timestamp_leap) const {
return timestamp_leap >=
static_cast<uint32_t>(output_size_samples_ * kReinitAfterExpands);
}
bool DecisionLogicNormal::PacketTooEarly(uint32_t timestamp_leap) const {
return timestamp_leap >
static_cast<uint32_t>(output_size_samples_ * num_consecutive_expands_);
}
bool DecisionLogicNormal::MaxWaitForPacket() const {
return num_consecutive_expands_ >= kMaxWaitForPacket;
}
} // namespace webrtc

112
modules/audio_coding/neteq/decision_logic_normal.h Normal file
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@ -0,0 +1,112 @@
/*
* 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_

14
modules/audio_coding/neteq/decision_logic_unittest.cc
View File

@ -33,7 +33,19 @@ TEST(DecisionLogic, CreateAndDestroy) {
DelayManager delay_manager(240, &delay_peak_detector, &tick_timer);
BufferLevelFilter buffer_level_filter;
DecisionLogic* logic = DecisionLogic::Create(
fs_hz, output_size_samples, false, &decoder_database, packet_buffer,
fs_hz, output_size_samples, kPlayoutOn, &decoder_database, packet_buffer,
&delay_manager, &buffer_level_filter, &tick_timer);
delete logic;
logic = DecisionLogic::Create(
fs_hz, output_size_samples, kPlayoutStreaming, &decoder_database,
packet_buffer, &delay_manager, &buffer_level_filter, &tick_timer);
delete logic;
logic = DecisionLogic::Create(
fs_hz, output_size_samples, kPlayoutFax, &decoder_database, packet_buffer,
&delay_manager, &buffer_level_filter, &tick_timer);
delete logic;
logic = DecisionLogic::Create(
fs_hz, output_size_samples, kPlayoutOff, &decoder_database, packet_buffer,
&delay_manager, &buffer_level_filter, &tick_timer);
delete logic;
}

4
modules/audio_coding/neteq/defines.h
View File

@ -24,6 +24,10 @@ enum Operations {
kRfc3389CngNoPacket,
kCodecInternalCng,
kDtmf,
kAlternativePlc,
kAlternativePlcIncreaseTimestamp,
kAudioRepetition,
kAudioRepetitionIncreaseTimestamp,
kUndefined = -1
};

19
modules/audio_coding/neteq/include/neteq.h
View File

@ -74,6 +74,13 @@ struct NetEqLifetimeStatistics {
uint64_t voice_concealed_samples = 0;
};
enum NetEqPlayoutMode {
kPlayoutOn,
kPlayoutOff,
kPlayoutFax,
kPlayoutStreaming
};
// This is the interface class for NetEq.
class NetEq {
public:
@ -91,10 +98,10 @@ class NetEq {
bool enable_post_decode_vad = false;
size_t max_packets_in_buffer = 50;
int max_delay_ms = 2000;
NetEqPlayoutMode playout_mode = kPlayoutOn;
bool enable_fast_accelerate = false;
bool enable_muted_state = false;
absl::optional<AudioCodecPairId> codec_pair_id;
bool for_test_no_time_stretching = false; // Use only for testing.
};
enum ReturnCodes { kOK = 0, kFail = -1, kNotImplemented = -2 };
@ -202,6 +209,16 @@ class NetEq {
// The packet buffer part of the delay is not updated during DTX/CNG periods.
virtual int FilteredCurrentDelayMs() const = 0;
// Sets the playout mode to |mode|.
// Deprecated. Set the mode in the Config struct passed to the constructor.
// TODO(henrik.lundin) Delete.
virtual void SetPlayoutMode(NetEqPlayoutMode mode) = 0;
// Returns the current playout mode.
// Deprecated.
// TODO(henrik.lundin) Delete.
virtual NetEqPlayoutMode PlayoutMode() const = 0;
// Writes the current network statistics to |stats|. The statistics are reset
// after the call.
virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;

2
modules/audio_coding/neteq/neteq.cc
View File

@ -30,7 +30,7 @@ std::string NetEq::Config::ToString() const {
ss << "sample_rate_hz=" << sample_rate_hz << ", enable_post_decode_vad="
<< (enable_post_decode_vad ? "true" : "false")
<< ", max_packets_in_buffer=" << max_packets_in_buffer
<< ", enable_fast_accelerate="
<< ", playout_mode=" << playout_mode << ", enable_fast_accelerate="
<< (enable_fast_accelerate ? " true" : "false")
<< ", enable_muted_state=" << (enable_muted_state ? " true" : "false");
return ss.str();

82
modules/audio_coding/neteq/neteq_impl.cc
View File

@ -101,6 +101,7 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
reset_decoder_(false),
ssrc_(0),
first_packet_(true),
playout_mode_(config.playout_mode),
enable_fast_accelerate_(config.enable_fast_accelerate),
nack_enabled_(false),
enable_muted_state_(config.enable_muted_state),
@ -109,8 +110,7 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
tick_timer_.get()),
speech_expand_uma_logger_("WebRTC.Audio.SpeechExpandRatePercent",
10, // Report once every 10 s.
tick_timer_.get()),
no_time_stretching_(config.for_test_no_time_stretching) {
tick_timer_.get()) {
RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
int fs = config.sample_rate_hz;
if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
@ -358,6 +358,23 @@ int NetEqImpl::FilteredCurrentDelayMs() const {
return static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000);
}
// Deprecated.
// TODO(henrik.lundin) Delete.
void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
rtc::CritScope lock(&crit_sect_);
if (mode != playout_mode_) {
playout_mode_ = mode;
CreateDecisionLogic();
}
}
// Deprecated.
// TODO(henrik.lundin) Delete.
NetEqPlayoutMode NetEqImpl::PlayoutMode() const {
rtc::CritScope lock(&crit_sect_);
return playout_mode_;
}
int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
rtc::CritScope lock(&crit_sect_);
assert(decoder_database_.get());
@ -920,6 +937,33 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
return_value = DoDtmf(dtmf_event, &play_dtmf);
break;
}
case kAlternativePlc: {
// TODO(hlundin): Write test for this.
DoAlternativePlc(false);
break;
}
case kAlternativePlcIncreaseTimestamp: {
// TODO(hlundin): Write test for this.
DoAlternativePlc(true);
break;
}
case kAudioRepetitionIncreaseTimestamp: {
// TODO(hlundin): Write test for this.
sync_buffer_->IncreaseEndTimestamp(
static_cast<uint32_t>(output_size_samples_));
// Skipping break on purpose. Execution should move on into the
// next case.
RTC_FALLTHROUGH();
}
case kAudioRepetition: {
// TODO(hlundin): Write test for this.
// Copy last |output_size_samples_| from |sync_buffer_| to
// |algorithm_buffer|.
algorithm_buffer_->PushBackFromIndex(
*sync_buffer_, sync_buffer_->Size() - output_size_samples_);
expand_->Reset();
break;
}
case kUndefined: {
RTC_LOG(LS_ERROR) << "Invalid operation kUndefined.";
assert(false); // This should not happen.
@ -1247,7 +1291,10 @@ int NetEqImpl::GetDecision(Operations* operation,
// Get packets from buffer.
int extracted_samples = 0;
if (packet) {
if (packet && *operation != kAlternativePlc &&
*operation != kAlternativePlcIncreaseTimestamp &&
*operation != kAudioRepetition &&
*operation != kAudioRepetitionIncreaseTimestamp) {
sync_buffer_->IncreaseEndTimestamp(packet->timestamp - end_timestamp);
if (decision_logic_->CngOff()) {
// Adjustment of timestamp only corresponds to an actual packet loss
@ -1836,6 +1883,29 @@ int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
return 0;
}
void NetEqImpl::DoAlternativePlc(bool increase_timestamp) {
AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
size_t length;
if (decoder && decoder->HasDecodePlc()) {
// Use the decoder's packet-loss concealment.
// TODO(hlundin): Will probably need a longer buffer for multi-channel.
int16_t decoded_buffer[kMaxFrameSize];
length = decoder->DecodePlc(1, decoded_buffer);
if (length > 0)
algorithm_buffer_->PushBackInterleaved(decoded_buffer, length);
} else {
// Do simple zero-stuffing.
length = output_size_samples_;
algorithm_buffer_->Zeros(length);
// By not advancing the timestamp, NetEq inserts samples.
stats_.AddZeros(length);
}
if (increase_timestamp) {
sync_buffer_->IncreaseEndTimestamp(static_cast<uint32_t>(length));
}
expand_->Reset();
}
int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event,
size_t num_channels,
int16_t* output) const {
@ -2060,8 +2130,8 @@ NetEqImpl::OutputType NetEqImpl::LastOutputType() {
void NetEqImpl::CreateDecisionLogic() {
decision_logic_.reset(DecisionLogic::Create(
fs_hz_, output_size_samples_, no_time_stretching_,
decoder_database_.get(), *packet_buffer_.get(), delay_manager_.get(),
buffer_level_filter_.get(), tick_timer_.get()));
fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
*packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
tick_timer_.get()));
}
} // namespace webrtc

18
modules/audio_coding/neteq/neteq_impl.h
View File

@ -168,6 +168,16 @@ class NetEqImpl : public webrtc::NetEq {
int FilteredCurrentDelayMs() const override;
// Sets the playout mode to |mode|.
// Deprecated.
// TODO(henrik.lundin) Delete.
void SetPlayoutMode(NetEqPlayoutMode mode) override;
// Returns the current playout mode.
// Deprecated.
// TODO(henrik.lundin) Delete.
NetEqPlayoutMode PlayoutMode() const override;
// Writes the current network statistics to |stats|. The statistics are reset
// after the call.
int NetworkStatistics(NetEqNetworkStatistics* stats) override;
@ -326,6 +336,12 @@ class NetEqImpl : public webrtc::NetEq {
int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Produces packet-loss concealment using alternative methods. If the codec
// has an internal PLC, it is called to generate samples. Otherwise, the
// method performs zero-stuffing.
void DoAlternativePlc(bool increase_timestamp)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Overdub DTMF on top of |output|.
int DtmfOverdub(const DtmfEvent& dtmf_event,
size_t num_channels,
@ -413,6 +429,7 @@ class NetEqImpl : public webrtc::NetEq {
RTC_GUARDED_BY(crit_sect_);
uint32_t ssrc_ RTC_GUARDED_BY(crit_sect_);
bool first_packet_ RTC_GUARDED_BY(crit_sect_);
NetEqPlayoutMode playout_mode_ RTC_GUARDED_BY(crit_sect_);
bool enable_fast_accelerate_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(crit_sect_);
bool nack_enabled_ RTC_GUARDED_BY(crit_sect_);
@ -424,7 +441,6 @@ class NetEqImpl : public webrtc::NetEq {
std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
bool no_time_stretching_ RTC_GUARDED_BY(crit_sect_); // Only used for test.
private:
RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);

26
modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -1377,6 +1377,32 @@ class NetEqImplTest120ms : public NetEqImplTest {
uint16_t sequence_number_ = 1;
};
TEST_F(NetEqImplTest120ms, AudioRepetition) {
config_.playout_mode = kPlayoutFax;
CreateInstanceNoMocks();
Register120msCodec(AudioDecoder::kSpeech);
InsertPacket(first_timestamp());
GetFirstPacket();
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
EXPECT_EQ(kAudioRepetition, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, AlternativePlc) {
config_.playout_mode = kPlayoutOff;
CreateInstanceNoMocks();
Register120msCodec(AudioDecoder::kSpeech);
InsertPacket(first_timestamp());
GetFirstPacket();
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
EXPECT_EQ(kAlternativePlc, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, CodecInternalCng) {
CreateInstanceNoMocks();
Register120msCodec(AudioDecoder::kComfortNoise);

37
modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -25,8 +25,6 @@
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/messagedigest.h"
@ -557,7 +555,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
protected:
NetEqDecodingTestFaxMode() : NetEqDecodingTest() {
config_.for_test_no_time_stretching = true;
config_.playout_mode = kPlayoutFax;
}
void TestJitterBufferDelay(bool apply_packet_loss);
};
@ -1725,37 +1723,4 @@ TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithLoss) {
TestJitterBufferDelay(true);
}
namespace test {
// TODO(henrik.lundin) NetEqRtpDumpInput requires protobuf support. It shouldn't
// need it, but because it is bundled with NetEqEventLogInput, it is neded.
// This should be refactored.
#if WEBRTC_ENABLE_PROTOBUF
TEST(NetEqNoTimeStretchingMode, RunTest) {
NetEq::Config config;
config.for_test_no_time_stretching = true;
auto codecs = NetEqTest::StandardDecoderMap();
NetEqTest::ExtDecoderMap ext_codecs;
NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
{1, kRtpExtensionAudioLevel},
{3, kRtpExtensionAbsoluteSendTime},
{5, kRtpExtensionTransportSequenceNumber},
{7, kRtpExtensionVideoContentType},
{8, kRtpExtensionVideoTiming}};
std::unique_ptr<NetEqInput> input(new NetEqRtpDumpInput(
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"),
rtp_ext_map));
std::unique_ptr<TimeLimitedNetEqInput> input_time_limit(
new TimeLimitedNetEqInput(std::move(input), 20000));
std::unique_ptr<AudioSink> output(new VoidAudioSink);
NetEqTest::Callbacks callbacks;
NetEqTest test(config, codecs, ext_codecs, std::move(input_time_limit),
std::move(output), callbacks);
test.Run();
const auto stats = test.SimulationStats();
EXPECT_EQ(0, stats.accelerate_rate);
EXPECT_EQ(0, stats.preemptive_rate);
}
#endif
} // namespace test
} // namespace webrtc

45
modules/audio_coding/neteq/tools/neteq_input.cc
View File

@ -28,50 +28,5 @@ std::string NetEqInput::PacketData::ToString() const {
return ss.str();
}
TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
int64_t duration_ms)
: input_(std::move(input)),
start_time_ms_(input_->NextEventTime()),
duration_ms_(duration_ms) {}
rtc::Optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
return ended_ ? rtc::Optional<int64_t>() : input_->NextPacketTime();
}
rtc::Optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
return ended_ ? rtc::Optional<int64_t>() : input_->NextOutputEventTime();
}
std::unique_ptr<NetEqInput::PacketData> TimeLimitedNetEqInput::PopPacket() {
if (ended_) {
return std::unique_ptr<PacketData>();
}
auto packet = input_->PopPacket();
MaybeSetEnded();
return packet;
}
void TimeLimitedNetEqInput::AdvanceOutputEvent() {
if (!ended_) {
input_->AdvanceOutputEvent();
MaybeSetEnded();
}
}
bool TimeLimitedNetEqInput::ended() const {
return ended_ || input_->ended();
}
rtc::Optional<RTPHeader> TimeLimitedNetEqInput::NextHeader() const {
return ended_ ? rtc::Optional<RTPHeader>() : input_->NextHeader();
}
void TimeLimitedNetEqInput::MaybeSetEnded() {
if (NextEventTime() && start_time_ms_ &&
*NextEventTime() - *start_time_ms_ > duration_ms_) {
ended_ = true;
}
}
} // namespace test
} // namespace webrtc

22
modules/audio_coding/neteq/tools/neteq_input.h
View File

@ -78,28 +78,6 @@ class NetEqInput {
virtual absl::optional<RTPHeader> NextHeader() const = 0;
};
// Wrapper class to impose a time limit on a NetEqInput object, typically
// another time limit than what the object itself provides. For example, an
// input taken from a file can be cut shorter by wrapping it in this class.
class TimeLimitedNetEqInput : public NetEqInput {
public:
TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
rtc::Optional<int64_t> NextPacketTime() const override;
rtc::Optional<int64_t> NextOutputEventTime() const override;
std::unique_ptr<PacketData> PopPacket() override;
void AdvanceOutputEvent() override;
bool ended() const override;
rtc::Optional<RTPHeader> NextHeader() const override;
private:
void MaybeSetEnded();
std::unique_ptr<NetEqInput> input_;
const rtc::Optional<int64_t> start_time_ms_;
const int64_t duration_ms_;
bool ended_ = false;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_

28
modules/audio_coding/neteq/tools/neteq_test.cc
View File

@ -115,34 +115,6 @@ NetEqLifetimeStatistics NetEqTest::LifetimeStats() const {
return neteq_->GetLifetimeStatistics();
}
NetEqTest::DecoderMap NetEqTest::StandardDecoderMap() {
DecoderMap codecs = {
{0, std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu")},
{8, std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma")},
{102, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
{103, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
#if !defined(WEBRTC_ANDROID)
{104, std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb")},
#endif
{111, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
{93, std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb")},
{94, std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb")},
{95, std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32")},
{96, std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48")},
{9, std::make_pair(NetEqDecoder::kDecoderG722, "g722")},
{106, std::make_pair(NetEqDecoder::kDecoderAVT, "avt")},
{114, std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16")},
{115, std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32")},
{116, std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48")},
{117, std::make_pair(NetEqDecoder::kDecoderRED, "red")},
{13, std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb")},
{98, std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb")},
{99, std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32")},
{100, std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48")}
};
return codecs;
}
void NetEqTest::RegisterDecoders(const DecoderMap& codecs) {
for (const auto& c : codecs) {
RTC_CHECK_EQ(

2
modules/audio_coding/neteq/tools/neteq_test.h
View File

@ -91,8 +91,6 @@ class NetEqTest {
NetEqNetworkStatistics SimulationStats();
NetEqLifetimeStatistics LifetimeStats() const;
static DecoderMap StandardDecoderMap();
private:
void RegisterDecoders(const DecoderMap& codecs);
void RegisterExternalDecoders(const ExtDecoderMap& codecs);

5
modules/audio_coding/test/TwoWayCommunication.cc
View File

@ -40,9 +40,8 @@ TwoWayCommunication::TwoWayCommunication(int testMode)
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
_testMode(testMode) {
AudioCodingModule::Config config;
// The clicks will be more obvious if time-stretching is not allowed.
// TODO(henrik.lundin) Really?
config.neteq_config.for_test_no_time_stretching = true;
// The clicks will be more obvious in FAX mode. TODO(henrik.lundin) Really?
config.neteq_config.playout_mode = kPlayoutFax;
config.decoder_factory = CreateBuiltinAudioDecoderFactory();
_acmB.reset(AudioCodingModule::Create(config));
_acmRefB.reset(AudioCodingModule::Create(config));

30
test/fuzzers/neteq_rtp_fuzzer.cc
View File

@ -133,13 +133,29 @@ void FuzzOneInputTest(const uint8_t* data, size_t size) {
std::unique_ptr<AudioChecksum> output(new AudioChecksum);
NetEqTest::Callbacks callbacks;
NetEq::Config config;
auto codecs = NetEqTest::StandardDecoderMap();
// kPayloadType is the payload type that will be used for encoding. Verify
// that it is included in the standard decoder map, and that it points to the
// expected decoder type.
RTC_CHECK_EQ(codecs.count(kPayloadType), 1);
RTC_CHECK(codecs[kPayloadType].first == NetEqDecoder::kDecoderPCM16Bswb32kHz);
NetEqTest::DecoderMap codecs;
codecs[0] = std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu");
codecs[8] = std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma");
codecs[103] = std::make_pair(NetEqDecoder::kDecoderISAC, "isac");
codecs[104] = std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb");
codecs[111] = std::make_pair(NetEqDecoder::kDecoderOpus, "opus");
codecs[93] = std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb");
codecs[94] = std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb");
codecs[96] =
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48");
codecs[9] = std::make_pair(NetEqDecoder::kDecoderG722, "g722");
codecs[106] = std::make_pair(NetEqDecoder::kDecoderAVT, "avt");
codecs[114] = std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16");
codecs[115] = std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32");
codecs[116] = std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48");
codecs[117] = std::make_pair(NetEqDecoder::kDecoderRED, "red");
codecs[13] = std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb");
codecs[98] = std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb");
codecs[99] = std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32");
codecs[100] = std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48");
// This is the payload type that will be used for encoding.
codecs[kPayloadType] =
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32");
NetEqTest::ExtDecoderMap ext_codecs;
NetEqTest test(config, codecs, ext_codecs, std::move(input),

41
test/fuzzers/neteq_signal_fuzzer.cc
View File

@ -167,22 +167,31 @@ void FuzzOneInputTest(const uint8_t* data, size_t size) {
NetEq::Config config;
config.enable_post_decode_vad = true;
config.enable_fast_accelerate = true;
auto codecs = NetEqTest::StandardDecoderMap();
// rate_types contains the payload types that will be used for encoding.
// Verify that they all are included in the standard decoder map, and that
// they point to the expected decoder types.
RTC_CHECK_EQ(codecs.count(rate_types[0].second), 1);
RTC_CHECK(codecs[rate_types[0].second].first == NetEqDecoder::kDecoderPCM16B);
RTC_CHECK_EQ(codecs.count(rate_types[1].second), 1);
RTC_CHECK(codecs[rate_types[1].second].first ==
NetEqDecoder::kDecoderPCM16Bwb);
RTC_CHECK_EQ(codecs.count(rate_types[2].second), 1);
RTC_CHECK(codecs[rate_types[2].second].first ==
NetEqDecoder::kDecoderPCM16Bswb32kHz);
RTC_CHECK_EQ(codecs.count(rate_types[3].second), 1);
RTC_CHECK(codecs[rate_types[3].second].first ==
NetEqDecoder::kDecoderPCM16Bswb48kHz);
NetEqTest::DecoderMap codecs;
codecs[0] = std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu");
codecs[8] = std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma");
codecs[103] = std::make_pair(NetEqDecoder::kDecoderISAC, "isac");
codecs[104] = std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb");
codecs[111] = std::make_pair(NetEqDecoder::kDecoderOpus, "opus");
codecs[9] = std::make_pair(NetEqDecoder::kDecoderG722, "g722");
codecs[106] = std::make_pair(NetEqDecoder::kDecoderAVT, "avt");
codecs[114] = std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16");
codecs[115] = std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32");
codecs[116] = std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48");
codecs[117] = std::make_pair(NetEqDecoder::kDecoderRED, "red");
codecs[13] = std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb");
codecs[98] = std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb");
codecs[99] = std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32");
codecs[100] = std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48");
// One of these payload types will be used for encoding.
codecs[rate_types[0].second] =
std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb");
codecs[rate_types[1].second] =
std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb");
codecs[rate_types[2].second] =
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32");
codecs[rate_types[3].second] =
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48");
NetEqTest::ExtDecoderMap ext_codecs;
NetEqTest test(config, codecs, ext_codecs, std::move(input),