Move NetEq headers to api/
This CL also introduces NetEqFactory and NetEqControllerFactory
interfaces, as well as several convenience classes for working with
them: DefaultNetEqFactory, DefaultNetEqControllerFactory and
CustomNetEqFactory.
Bug: webrtc:11005
Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29671}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 2cad71c..50d847d 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -48,6 +48,9 @@
"../../api:function_view",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
+ "../../api/neteq:custom_neteq_factory",
+ "../../api/neteq:default_neteq_controller_factory",
+ "../../api/neteq:neteq_api",
"../../common_audio",
"../../common_audio:common_audio_c",
"../../rtc_base:audio_format_to_string",
@@ -944,7 +947,6 @@
"neteq/decision_logic.h",
"neteq/decoder_database.cc",
"neteq/decoder_database.h",
- "neteq/defines.h",
"neteq/delay_manager.cc",
"neteq/delay_manager.h",
"neteq/delay_peak_detector.cc",
@@ -961,13 +963,10 @@
"neteq/expand_uma_logger.h",
"neteq/histogram.cc",
"neteq/histogram.h",
- "neteq/include/neteq.h",
"neteq/merge.cc",
"neteq/merge.h",
"neteq/nack_tracker.cc",
"neteq/nack_tracker.h",
- "neteq/neteq.cc",
- "neteq/neteq_controller.h",
"neteq/neteq_impl.cc",
"neteq/neteq_impl.h",
"neteq/normal.cc",
@@ -988,8 +987,6 @@
"neteq/statistics_calculator.h",
"neteq/sync_buffer.cc",
"neteq/sync_buffer.h",
- "neteq/tick_timer.cc",
- "neteq/tick_timer.h",
"neteq/time_stretch.cc",
"neteq/time_stretch.h",
"neteq/timestamp_scaler.cc",
@@ -1007,6 +1004,9 @@
"../../api:scoped_refptr",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
+ "../../api/neteq:neteq_api",
+ "../../api/neteq:neteq_controller_api",
+ "../../api/neteq:tick_timer",
"../../common_audio",
"../../common_audio:common_audio_c",
"../../rtc_base:audio_format_to_string",
@@ -1052,6 +1052,9 @@
"../../api:rtp_headers",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
+ "../../api/neteq:custom_neteq_factory",
+ "../../api/neteq:default_neteq_controller_factory",
+ "../../api/neteq:neteq_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
@@ -1467,6 +1470,7 @@
":neteq",
":neteq_test_tools",
"../../api/audio_codecs:builtin_audio_decoder_factory",
+ "../../api/neteq:neteq_api",
"../../rtc_base:rtc_base_approved",
"../../test:audio_codec_mocks",
"../../test:field_trial",
@@ -1558,9 +1562,11 @@
":neteq",
":neteq_test_tools",
":pcm16b",
+ "../../api:neteq_factory_with_codecs",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
+ "../../api/neteq:neteq_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
@@ -1581,6 +1587,9 @@
":neteq",
":neteq_test_tools",
"../../api/audio_codecs:builtin_audio_decoder_factory",
+ "../../api/neteq:custom_neteq_factory",
+ "../../api/neteq:default_neteq_controller_factory",
+ "../../api/neteq:neteq_api",
"../../rtc_base:checks",
"../../system_wrappers",
"../../test:fileutils",
@@ -1976,7 +1985,6 @@
"neteq/red_payload_splitter_unittest.cc",
"neteq/statistics_calculator_unittest.cc",
"neteq/sync_buffer_unittest.cc",
- "neteq/tick_timer_unittest.cc",
"neteq/time_stretch_unittest.cc",
"neteq/timestamp_scaler_unittest.cc",
"neteq/tools/input_audio_file_unittest.cc",
@@ -2007,6 +2015,7 @@
":webrtc_opus",
"..:module_api",
"..:module_api_public",
+ "../../api:neteq_factory_with_codecs",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -2015,6 +2024,12 @@
"../../api/audio_codecs/opus:audio_decoder_opus",
"../../api/audio_codecs/opus:audio_encoder_multiopus",
"../../api/audio_codecs/opus:audio_encoder_opus",
+ "../../api/neteq:custom_neteq_factory",
+ "../../api/neteq:default_neteq_controller_factory",
+ "../../api/neteq:neteq_api",
+ "../../api/neteq:neteq_controller_api",
+ "../../api/neteq:tick_timer",
+ "../../api/neteq:tick_timer_unittest",
"../../api/rtc_event_log",
"../../common_audio",
"../../common_audio:common_audio_c",
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 4019615..2723937 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -19,9 +19,11 @@
#include "absl/strings/match.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/custom_neteq_factory.h"
+#include "api/neteq/default_neteq_controller_factory.h"
+#include "api/neteq/neteq.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
-#include "modules/audio_coding/neteq/include/neteq.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -32,11 +34,24 @@
namespace acm2 {
+namespace {
+
+std::unique_ptr<NetEq> CreateNetEq(
+ const NetEq::Config& config,
+ Clock* clock,
+ const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+ CustomNetEqFactory neteq_factory(
+ decoder_factory, std::make_unique<DefaultNetEqControllerFactory>());
+ return neteq_factory.CreateNetEq(config, clock);
+}
+
+} // namespace
+
AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
: last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
- neteq_(NetEq::Create(config.neteq_config,
- config.clock,
- config.decoder_factory)),
+ neteq_(CreateNetEq(config.neteq_config,
+ config.clock,
+ config.decoder_factory)),
clock_(config.clock),
resampled_last_output_frame_(true) {
RTC_DCHECK(clock_);
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index da8ffb5..05d9380 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -20,8 +20,8 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/function_view.h"
+#include "api/neteq/neteq.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "modules/audio_coding/neteq/include/neteq.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/decision_logic.cc b/modules/audio_coding/neteq/decision_logic.cc
index 1c887b1..5c746ad 100644
--- a/modules/audio_coding/neteq/decision_logic.cc
+++ b/modules/audio_coding/neteq/decision_logic.cc
@@ -91,42 +91,43 @@
output_size_samples_ = output_size_samples;
}
-Operations DecisionLogic::GetDecision(const NetEqStatus& status,
- bool* reset_decoder) {
+NetEq::Operation DecisionLogic::GetDecision(const NetEqStatus& status,
+ bool* reset_decoder) {
// If last mode was CNG (or Expand, since this could be covering up for
// a lost CNG packet), remember that CNG is on. This is needed if comfort
// noise is interrupted by DTMF.
- if (status.last_mode == kModeRfc3389Cng) {
+ if (status.last_mode == NetEq::Mode::kRfc3389Cng) {
cng_state_ = kCngRfc3389On;
- } else if (status.last_mode == kModeCodecInternalCng) {
+ } else if (status.last_mode == NetEq::Mode::kCodecInternalCng) {
cng_state_ = kCngInternalOn;
}
size_t cur_size_samples = estimate_dtx_delay_
? status.packet_buffer_info.span_samples
: status.packet_buffer_info.num_samples;
-
prev_time_scale_ =
- prev_time_scale_ && (status.last_mode == kModeAccelerateSuccess ||
- status.last_mode == kModeAccelerateLowEnergy ||
- status.last_mode == kModePreemptiveExpandSuccess ||
- status.last_mode == kModePreemptiveExpandLowEnergy);
+ prev_time_scale_ &&
+ (status.last_mode == NetEq::Mode::kAccelerateSuccess ||
+ status.last_mode == NetEq::Mode::kAccelerateLowEnergy ||
+ status.last_mode == NetEq::Mode::kPreemptiveExpandSuccess ||
+ status.last_mode == NetEq::Mode::kPreemptiveExpandLowEnergy);
// Do not update buffer history if currently playing CNG since it will bias
// the filtered buffer level.
- if (status.last_mode != kModeRfc3389Cng &&
- status.last_mode != kModeCodecInternalCng &&
+ if (status.last_mode != NetEq::Mode::kRfc3389Cng &&
+ status.last_mode != NetEq::Mode::kCodecInternalCng &&
!(status.next_packet && status.next_packet->is_dtx &&
!estimate_dtx_delay_)) {
FilterBufferLevel(cur_size_samples);
}
// Guard for errors, to avoid getting stuck in error mode.
- if (status.last_mode == kModeError) {
+ if (status.last_mode == NetEq::Mode::kError) {
if (!status.next_packet) {
- return kExpand;
+ return NetEq::Operation::kExpand;
} else {
- return kUndefined; // Use kUndefined to flag for a reset.
+ // Use kUndefined to flag for a reset.
+ return NetEq::Operation::kUndefined;
}
}
@@ -145,7 +146,7 @@
// sender was restarted.
if (num_consecutive_expands_ > kReinitAfterExpands) {
*reset_decoder = true;
- return kNormal;
+ return NetEq::Operation::kNormal;
}
// Make sure we don't restart audio too soon after an expansion to avoid
@@ -158,13 +159,14 @@
const size_t current_span =
estimate_dtx_delay_ ? status.packet_buffer_info.span_samples
: status.packet_buffer_info.span_samples_no_dtx;
- if ((status.last_mode == kModeExpand || status.last_mode == kModeCodecPlc) &&
+ if ((status.last_mode == NetEq::Mode::kExpand ||
+ status.last_mode == NetEq::Mode::kCodecPlc) &&
status.expand_mutefactor < 16384 / 2 &&
current_span<static_cast<size_t>(delay_manager_->TargetLevel() *
packet_length_samples_ *
kPostponeDecodingLevel / 100)>> 8 &&
!status.packet_buffer_info.dtx_or_cng) {
- return kExpand;
+ return NetEq::Operation::kExpand;
}
const uint32_t five_seconds_samples = static_cast<uint32_t>(5 * sample_rate_);
@@ -182,12 +184,12 @@
} 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;
+ return NetEq::Operation::kUndefined;
}
}
-void DecisionLogic::ExpandDecision(Operations operation) {
- if (operation == kExpand) {
+void DecisionLogic::ExpandDecision(NetEq::Operation operation) {
+ if (operation == NetEq::Operation::kExpand) {
num_consecutive_expands_++;
} else {
num_consecutive_expands_ = 0;
@@ -241,10 +243,10 @@
time_stretched_cn_samples_ = 0;
}
-Operations DecisionLogic::CngOperation(Modes prev_mode,
- uint32_t target_timestamp,
- uint32_t available_timestamp,
- size_t generated_noise_samples) {
+NetEq::Operation DecisionLogic::CngOperation(NetEq::Mode 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) -
@@ -264,35 +266,36 @@
rtc::saturated_cast<int32_t>(timestamp_diff + excess_waiting_time_samp);
}
- if (timestamp_diff < 0 && prev_mode == kModeRfc3389Cng) {
+ if (timestamp_diff < 0 && prev_mode == NetEq::Mode::kRfc3389Cng) {
// Not time to play this packet yet. Wait another round before using this
// packet. Keep on playing CNG from previous CNG parameters.
- return kRfc3389CngNoPacket;
+ return NetEq::Operation::kRfc3389CngNoPacket;
} else {
// Otherwise, go for the CNG packet now.
noise_fast_forward_ = 0;
- return kRfc3389Cng;
+ return NetEq::Operation::kRfc3389Cng;
}
}
-Operations DecisionLogic::NoPacket(bool play_dtmf) {
+NetEq::Operation DecisionLogic::NoPacket(bool play_dtmf) {
if (cng_state_ == kCngRfc3389On) {
// Keep on playing comfort noise.
- return kRfc3389CngNoPacket;
+ return NetEq::Operation::kRfc3389CngNoPacket;
} else if (cng_state_ == kCngInternalOn) {
// Keep on playing codec internal comfort noise.
- return kCodecInternalCng;
+ return NetEq::Operation::kCodecInternalCng;
} else if (play_dtmf) {
- return kDtmf;
+ return NetEq::Operation::kDtmf;
} else {
// Nothing to play, do expand.
- return kExpand;
+ return NetEq::Operation::kExpand;
}
}
-Operations DecisionLogic::ExpectedPacketAvailable(Modes prev_mode,
- bool play_dtmf) {
- if (!disallow_time_stretching_ && prev_mode != kModeExpand && !play_dtmf) {
+NetEq::Operation DecisionLogic::ExpectedPacketAvailable(NetEq::Mode prev_mode,
+ bool play_dtmf) {
+ if (!disallow_time_stretching_ && prev_mode != NetEq::Mode::kExpand &&
+ !play_dtmf) {
// Check criterion for time-stretching. The values are in number of packets
// in Q8.
int low_limit, high_limit;
@@ -304,20 +307,20 @@
packet_length_samples_;
}
if (buffer_level_packets >= high_limit << 2)
- return kFastAccelerate;
+ return NetEq::Operation::kFastAccelerate;
if (TimescaleAllowed()) {
if (buffer_level_packets >= high_limit)
- return kAccelerate;
+ return NetEq::Operation::kAccelerate;
if (buffer_level_packets < low_limit)
- return kPreemptiveExpand;
+ return NetEq::Operation::kPreemptiveExpand;
}
}
- return kNormal;
+ return NetEq::Operation::kNormal;
}
-Operations DecisionLogic::FuturePacketAvailable(
+NetEq::Operation DecisionLogic::FuturePacketAvailable(
size_t decoder_frame_length,
- Modes prev_mode,
+ NetEq::Mode prev_mode,
uint32_t target_timestamp,
uint32_t available_timestamp,
bool play_dtmf,
@@ -328,24 +331,26 @@
// 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 || prev_mode == kModeCodecPlc) &&
+ if ((prev_mode == NetEq::Mode::kExpand ||
+ prev_mode == NetEq::Mode::kCodecPlc) &&
!ReinitAfterExpands(timestamp_leap) && !MaxWaitForPacket() &&
PacketTooEarly(timestamp_leap) && UnderTargetLevel()) {
if (play_dtmf) {
// Still have DTMF to play, so do not do expand.
- return kDtmf;
+ return NetEq::Operation::kDtmf;
} else {
// Nothing to play.
- return kExpand;
+ return NetEq::Operation::kExpand;
}
}
- if (prev_mode == kModeCodecPlc) {
- return kNormal;
+ if (prev_mode == NetEq::Mode::kCodecPlc) {
+ return NetEq::Operation::kNormal;
}
// If previous was comfort noise, then no merge is needed.
- if (prev_mode == kModeRfc3389Cng || prev_mode == kModeCodecInternalCng) {
+ if (prev_mode == NetEq::Mode::kRfc3389Cng ||
+ prev_mode == NetEq::Mode::kCodecInternalCng) {
size_t cur_size_samples =
estimate_dtx_delay_
? cur_size_samples = span_samples_in_packet_buffer
@@ -370,7 +375,7 @@
if ((generated_enough_noise && !below_target_window) ||
above_target_window) {
time_stretched_cn_samples_ = timestamp_leap - generated_noise_samples;
- return kNormal;
+ return NetEq::Operation::kNormal;
}
} else {
// Keep the same delay as before the CNG, but make sure that the number of
@@ -378,26 +383,26 @@
if (generated_enough_noise ||
cur_size_samples > target_level_samples * 4) {
// Time to play this new packet.
- return kNormal;
+ return NetEq::Operation::kNormal;
}
}
// Too early to play this new packet; keep on playing comfort noise.
- if (prev_mode == kModeRfc3389Cng) {
- return kRfc3389CngNoPacket;
+ if (prev_mode == NetEq::Mode::kRfc3389Cng) {
+ return NetEq::Operation::kRfc3389CngNoPacket;
}
// prevPlayMode == kModeCodecInternalCng.
- return kCodecInternalCng;
+ return NetEq::Operation::kCodecInternalCng;
}
// Do not merge unless we have done an expand before.
- if (prev_mode == kModeExpand) {
- return kMerge;
+ if (prev_mode == NetEq::Mode::kExpand) {
+ return NetEq::Operation::kMerge;
} else if (play_dtmf) {
// Play DTMF instead of expand.
- return kDtmf;
+ return NetEq::Operation::kDtmf;
} else {
- return kExpand;
+ return NetEq::Operation::kExpand;
}
}
diff --git a/modules/audio_coding/neteq/decision_logic.h b/modules/audio_coding/neteq/decision_logic.h
index f0d7a93..b8dbefe 100644
--- a/modules/audio_coding/neteq/decision_logic.h
+++ b/modules/audio_coding/neteq/decision_logic.h
@@ -11,12 +11,12 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "api/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/buffer_level_filter.h"
-#include "modules/audio_coding/neteq/defines.h"
#include "modules/audio_coding/neteq/delay_manager.h"
#include "modules/audio_coding/neteq/delay_peak_detector.h"
-#include "modules/audio_coding/neteq/neteq_controller.h"
-#include "modules/audio_coding/neteq/tick_timer.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/experiments/field_trial_parser.h"
@@ -52,8 +52,8 @@
// 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).
- Operations GetDecision(const NetEqStatus& status,
- bool* reset_decoder) override;
+ NetEq::Operation GetDecision(const NetEqController::NetEqStatus& status,
+ bool* reset_decoder) override;
// These methods test the |cng_state_| for different conditions.
bool CngRfc3389On() const override { return cng_state_ == kCngRfc3389On; }
@@ -66,7 +66,7 @@
// 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) override;
+ void ExpandDecision(NetEq::Operation operation) override;
// Adds |value| to |sample_memory_|.
void AddSampleMemory(int32_t value) override { sample_memory_ += value; }
@@ -126,28 +126,29 @@
// 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);
+ NetEq::Operation CngOperation(NetEq::Mode 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);
+ NetEq::Operation NoPacket(bool play_dtmf);
// Returns the operation to do given that the expected packet is available.
- Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
+ NetEq::Operation ExpectedPacketAvailable(NetEq::Mode 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(size_t decoder_frame_length,
- Modes prev_mode,
- uint32_t target_timestamp,
- uint32_t available_timestamp,
- bool play_dtmf,
- size_t generated_noise_samples,
- size_t span_samples_in_packet_buffer,
- size_t num_packets_in_packet_buffer);
+ NetEq::Operation FuturePacketAvailable(size_t decoder_frame_length,
+ NetEq::Mode prev_mode,
+ uint32_t target_timestamp,
+ uint32_t available_timestamp,
+ bool play_dtmf,
+ size_t generated_noise_samples,
+ size_t span_samples_in_packet_buffer,
+ size_t num_packets_in_packet_buffer);
// Checks if enough time has elapsed since the last successful timescale
// operation was done (i.e., accelerate or preemptive expand).
diff --git a/modules/audio_coding/neteq/decision_logic_unittest.cc b/modules/audio_coding/neteq/decision_logic_unittest.cc
index 9ba3b9d..d1dd123 100644
--- a/modules/audio_coding/neteq/decision_logic_unittest.cc
+++ b/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -12,14 +12,14 @@
#include "modules/audio_coding/neteq/decision_logic.h"
+#include "api/neteq/neteq_controller.h"
+#include "api/neteq/tick_timer.h"
#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/delay_peak_detector.h"
-#include "modules/audio_coding/neteq/neteq_controller.h"
#include "modules/audio_coding/neteq/packet_buffer.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
-#include "modules/audio_coding/neteq/tick_timer.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder_factory.h"
diff --git a/modules/audio_coding/neteq/defines.h b/modules/audio_coding/neteq/defines.h
deleted file mode 100644
index 46926fa..0000000
--- a/modules/audio_coding/neteq/defines.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2012 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_DEFINES_H_
-#define MODULES_AUDIO_CODING_NETEQ_DEFINES_H_
-
-namespace webrtc {
-
-enum Operations {
- kNormal = 0,
- kMerge,
- kExpand,
- kAccelerate,
- kFastAccelerate,
- kPreemptiveExpand,
- kRfc3389Cng,
- kRfc3389CngNoPacket,
- kCodecInternalCng,
- kDtmf,
- kUndefined = -1
-};
-
-enum Modes {
- kModeNormal = 0,
- kModeExpand,
- kModeMerge,
- kModeAccelerateSuccess,
- kModeAccelerateLowEnergy,
- kModeAccelerateFail,
- kModePreemptiveExpandSuccess,
- kModePreemptiveExpandLowEnergy,
- kModePreemptiveExpandFail,
- kModeRfc3389Cng,
- kModeCodecInternalCng,
- kModeCodecPlc,
- kModeDtmf,
- kModeError,
- kModeUndefined = -1
-};
-
-} // namespace webrtc
-#endif // MODULES_AUDIO_CODING_NETEQ_DEFINES_H_
diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index f1f24ac..c440447 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -17,8 +17,8 @@
#include <memory>
#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/histogram.h"
-#include "modules/audio_coding/neteq/tick_timer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/delay_peak_detector.h b/modules/audio_coding/neteq/delay_peak_detector.h
index 15db189..dce8bb4 100644
--- a/modules/audio_coding/neteq/delay_peak_detector.h
+++ b/modules/audio_coding/neteq/delay_peak_detector.h
@@ -17,7 +17,7 @@
#include <list>
#include <memory>
-#include "modules/audio_coding/neteq/tick_timer.h"
+#include "api/neteq/tick_timer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/expand_uma_logger.h b/modules/audio_coding/neteq/expand_uma_logger.h
index 1139bb6..246aaff 100644
--- a/modules/audio_coding/neteq/expand_uma_logger.h
+++ b/modules/audio_coding/neteq/expand_uma_logger.h
@@ -16,7 +16,7 @@
#include <string>
#include "absl/types/optional.h"
-#include "modules/audio_coding/neteq/tick_timer.h"
+#include "api/neteq/tick_timer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
deleted file mode 100644
index b53b5ad..0000000
--- a/modules/audio_coding/neteq/include/neteq.h
+++ /dev/null
@@ -1,311 +0,0 @@
-/*
- * Copyright (c) 2012 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_INCLUDE_NETEQ_H_
-#define MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
-
-#include <string.h> // Provide access to size_t.
-
-#include <map>
-#include <string>
-#include <vector>
-
-#include "absl/types/optional.h"
-#include "api/audio_codecs/audio_codec_pair_id.h"
-#include "api/audio_codecs/audio_decoder.h"
-#include "api/audio_codecs/audio_format.h"
-#include "api/rtp_headers.h"
-#include "api/scoped_refptr.h"
-#include "modules/audio_coding/neteq/defines.h"
-#include "rtc_base/constructor_magic.h"
-
-namespace webrtc {
-
-// Forward declarations.
-class AudioFrame;
-class AudioDecoderFactory;
-class Clock;
-
-struct NetEqNetworkStatistics {
- uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
- uint16_t preferred_buffer_size_ms; // Target buffer size in ms.
- uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky
- // jitter; 0 otherwise.
- uint16_t packet_loss_rate; // Loss rate (network + late) in Q14.
- uint16_t expand_rate; // Fraction (of original stream) of synthesized
- // audio inserted through expansion (in Q14).
- uint16_t speech_expand_rate; // Fraction (of original stream) of synthesized
- // speech inserted through expansion (in Q14).
- uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
- // expansion (in Q14).
- uint16_t accelerate_rate; // Fraction of data removed through acceleration
- // (in Q14).
- uint16_t secondary_decoded_rate; // Fraction of data coming from FEC/RED
- // decoding (in Q14).
- uint16_t secondary_discarded_rate; // Fraction of discarded FEC/RED data (in
- // Q14).
- size_t added_zero_samples; // Number of zero samples added in "off" mode.
- // Statistics for packet waiting times, i.e., the time between a packet
- // arrives until it is decoded.
- int mean_waiting_time_ms;
- int median_waiting_time_ms;
- int min_waiting_time_ms;
- int max_waiting_time_ms;
-};
-
-// NetEq statistics that persist over the lifetime of the class.
-// These metrics are never reset.
-struct NetEqLifetimeStatistics {
- // Stats below correspond to similarly-named fields in the WebRTC stats spec.
- // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
- uint64_t total_samples_received = 0;
- uint64_t concealed_samples = 0;
- uint64_t concealment_events = 0;
- uint64_t jitter_buffer_delay_ms = 0;
- uint64_t jitter_buffer_emitted_count = 0;
- uint64_t inserted_samples_for_deceleration = 0;
- uint64_t removed_samples_for_acceleration = 0;
- uint64_t silent_concealed_samples = 0;
- uint64_t fec_packets_received = 0;
- uint64_t fec_packets_discarded = 0;
- // Below stats are not part of the spec.
- uint64_t delayed_packet_outage_samples = 0;
- // This is sum of relative packet arrival delays of received packets so far.
- // Since end-to-end delay of a packet is difficult to measure and is not
- // necessarily useful for measuring jitter buffer performance, we report a
- // relative packet arrival delay. The relative packet arrival delay of a
- // packet is defined as the arrival delay compared to the first packet
- // received, given that it had zero delay. To avoid clock drift, the "first"
- // packet can be made dynamic.
- uint64_t relative_packet_arrival_delay_ms = 0;
- uint64_t jitter_buffer_packets_received = 0;
- // An interruption is a loss-concealment event lasting at least 150 ms. The
- // two stats below count the number os such events and the total duration of
- // these events.
- int32_t interruption_count = 0;
- int32_t total_interruption_duration_ms = 0;
-};
-
-// Metrics that describe the operations performed in NetEq, and the internal
-// state.
-struct NetEqOperationsAndState {
- // These sample counters are cumulative, and don't reset. As a reference, the
- // total number of output samples can be found in
- // NetEqLifetimeStatistics::total_samples_received.
- uint64_t preemptive_samples = 0;
- uint64_t accelerate_samples = 0;
- // Count of the number of buffer flushes.
- uint64_t packet_buffer_flushes = 0;
- // The number of primary packets that were discarded.
- uint64_t discarded_primary_packets = 0;
- // The statistics below are not cumulative.
- // The waiting time of the last decoded packet.
- uint64_t last_waiting_time_ms = 0;
- // The sum of the packet and jitter buffer size in ms.
- uint64_t current_buffer_size_ms = 0;
- // The current frame size in ms.
- uint64_t current_frame_size_ms = 0;
- // Flag to indicate that the next packet is available.
- bool next_packet_available = false;
-};
-
-// This is the interface class for NetEq.
-class NetEq {
- public:
- struct Config {
- Config();
- Config(const Config&);
- Config(Config&&);
- ~Config();
- Config& operator=(const Config&);
- Config& operator=(Config&&);
-
- std::string ToString() const;
-
- int sample_rate_hz = 16000; // Initial value. Will change with input data.
- bool enable_post_decode_vad = false;
- size_t max_packets_in_buffer = 200;
- int max_delay_ms = 0;
- int min_delay_ms = 0;
- bool enable_fast_accelerate = false;
- bool enable_muted_state = false;
- bool enable_rtx_handling = false;
- absl::optional<AudioCodecPairId> codec_pair_id;
- bool for_test_no_time_stretching = false; // Use only for testing.
- };
-
- enum ReturnCodes { kOK = 0, kFail = -1 };
-
- // Return type for GetDecoderFormat.
- struct DecoderFormat {
- int sample_rate_hz;
- int num_channels;
- SdpAudioFormat sdp_format;
- };
-
- // Creates a new NetEq object, with parameters set in |config|. The |config|
- // object will only have to be valid for the duration of the call to this
- // method.
- static NetEq* Create(
- const NetEq::Config& config,
- Clock* clock,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
-
- virtual ~NetEq() {}
-
- // Inserts a new packet into NetEq.
- // Returns 0 on success, -1 on failure.
- virtual int InsertPacket(const RTPHeader& rtp_header,
- rtc::ArrayView<const uint8_t> payload) = 0;
-
- // Deprecated. Use the version without the `receive_timestamp` argument.
- int InsertPacket(const RTPHeader& rtp_header,
- rtc::ArrayView<const uint8_t> payload,
- uint32_t /*receive_timestamp*/) {
- return InsertPacket(rtp_header, payload);
- }
-
- // Lets NetEq know that a packet arrived with an empty payload. This typically
- // happens when empty packets are used for probing the network channel, and
- // these packets use RTP sequence numbers from the same series as the actual
- // audio packets.
- virtual void InsertEmptyPacket(const RTPHeader& rtp_header) = 0;
-
- // Instructs NetEq to deliver 10 ms of audio data. The data is written to
- // |audio_frame|. All data in |audio_frame| is wiped; |data_|, |speech_type_|,
- // |num_channels_|, |sample_rate_hz_|, |samples_per_channel_|, and
- // |vad_activity_| are updated upon success. If an error is returned, some
- // fields may not have been updated, or may contain inconsistent values.
- // If muted state is enabled (through Config::enable_muted_state), |muted|
- // may be set to true after a prolonged expand period. When this happens, the
- // |data_| in |audio_frame| is not written, but should be interpreted as being
- // all zeros. For testing purposes, an override can be supplied in the
- // |action_override| argument, which will cause NetEq to take this action
- // next, instead of the action it would normally choose.
- // Returns kOK on success, or kFail in case of an error.
- virtual int GetAudio(
- AudioFrame* audio_frame,
- bool* muted,
- absl::optional<Operations> action_override = absl::nullopt) = 0;
-
- // Replaces the current set of decoders with the given one.
- virtual void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) = 0;
-
- // Associates |rtp_payload_type| with the given codec, which NetEq will
- // instantiate when it needs it. Returns true iff successful.
- virtual bool RegisterPayloadType(int rtp_payload_type,
- const SdpAudioFormat& audio_format) = 0;
-
- // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
- // -1 on failure. Removing a payload type that is not registered is ok and
- // will not result in an error.
- virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
-
- // Removes all payload types from the codec database.
- virtual void RemoveAllPayloadTypes() = 0;
-
- // Sets a minimum delay in millisecond for packet buffer. The minimum is
- // maintained unless a higher latency is dictated by channel condition.
- // Returns true if the minimum is successfully applied, otherwise false is
- // returned.
- virtual bool SetMinimumDelay(int delay_ms) = 0;
-
- // Sets a maximum delay in milliseconds for packet buffer. The latency will
- // not exceed the given value, even required delay (given the channel
- // conditions) is higher. Calling this method has the same effect as setting
- // the |max_delay_ms| value in the NetEq::Config struct.
- virtual bool SetMaximumDelay(int delay_ms) = 0;
-
- // Sets a base minimum delay in milliseconds for packet buffer. The minimum
- // delay which is set via |SetMinimumDelay| can't be lower than base minimum
- // delay. Calling this method is similar to setting the |min_delay_ms| value
- // in the NetEq::Config struct. Returns true if the base minimum is
- // successfully applied, otherwise false is returned.
- virtual bool SetBaseMinimumDelayMs(int delay_ms) = 0;
-
- // Returns current value of base minimum delay in milliseconds.
- virtual int GetBaseMinimumDelayMs() const = 0;
-
- // Returns the current target delay in ms. This includes any extra delay
- // requested through SetMinimumDelay.
- virtual int TargetDelayMs() const = 0;
-
- // Returns the current total delay (packet buffer and sync buffer) in ms,
- // with smoothing applied to even out short-time fluctuations due to jitter.
- // The packet buffer part of the delay is not updated during DTX/CNG periods.
- virtual int FilteredCurrentDelayMs() const = 0;
-
- // Writes the current network statistics to |stats|. The statistics are reset
- // after the call.
- virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
-
- // Returns a copy of this class's lifetime statistics. These statistics are
- // never reset.
- virtual NetEqLifetimeStatistics GetLifetimeStatistics() const = 0;
-
- // Returns statistics about the performed operations and internal state. These
- // statistics are never reset.
- virtual NetEqOperationsAndState GetOperationsAndState() const = 0;
-
- // Enables post-decode VAD. When enabled, GetAudio() will return
- // kOutputVADPassive when the signal contains no speech.
- virtual void EnableVad() = 0;
-
- // Disables post-decode VAD.
- virtual void DisableVad() = 0;
-
- // Returns the RTP timestamp for the last sample delivered by GetAudio().
- // The return value will be empty if no valid timestamp is available.
- virtual absl::optional<uint32_t> GetPlayoutTimestamp() const = 0;
-
- // Returns the sample rate in Hz of the audio produced in the last GetAudio
- // call. If GetAudio has not been called yet, the configured sample rate
- // (Config::sample_rate_hz) is returned.
- virtual int last_output_sample_rate_hz() const = 0;
-
- // Returns the decoder info for the given payload type. Returns empty if no
- // such payload type was registered.
- virtual absl::optional<DecoderFormat> GetDecoderFormat(
- int payload_type) const = 0;
-
- // Flushes both the packet buffer and the sync buffer.
- virtual void FlushBuffers() = 0;
-
- // Enables NACK and sets the maximum size of the NACK list, which should be
- // positive and no larger than Nack::kNackListSizeLimit. If NACK is already
- // enabled then the maximum NACK list size is modified accordingly.
- virtual void EnableNack(size_t max_nack_list_size) = 0;
-
- virtual void DisableNack() = 0;
-
- // Returns a list of RTP sequence numbers corresponding to packets to be
- // retransmitted, given an estimate of the round-trip time in milliseconds.
- virtual std::vector<uint16_t> GetNackList(
- int64_t round_trip_time_ms) const = 0;
-
- // Returns a vector containing the timestamps of the packets that were decoded
- // in the last GetAudio call. If no packets were decoded in the last call, the
- // vector is empty.
- // Mainly intended for testing.
- virtual std::vector<uint32_t> LastDecodedTimestamps() const = 0;
-
- // Returns the length of the audio yet to play in the sync buffer.
- // Mainly intended for testing.
- virtual int SyncBufferSizeMs() const = 0;
-
- protected:
- NetEq() {}
-
- private:
- RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
-};
-
-} // namespace webrtc
-#endif // MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
diff --git a/modules/audio_coding/neteq/mock/mock_neteq_controller.h b/modules/audio_coding/neteq/mock/mock_neteq_controller.h
index 38aa3e7..d1008c8 100644
--- a/modules/audio_coding/neteq/mock/mock_neteq_controller.h
+++ b/modules/audio_coding/neteq/mock/mock_neteq_controller.h
@@ -11,7 +11,7 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_NETEQ_CONTROLLER_H_
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_NETEQ_CONTROLLER_H_
-#include "modules/audio_coding/neteq/neteq_controller.h"
+#include "api/neteq/neteq_controller.h"
#include "test/gmock.h"
namespace webrtc {
@@ -24,8 +24,8 @@
MOCK_METHOD0(Reset, void());
MOCK_METHOD0(SoftReset, void());
MOCK_METHOD2(GetDecision,
- Operations(const NetEqStatus& neteq_status,
- bool* reset_decoder));
+ NetEq::Operation(const NetEqStatus& neteq_status,
+ bool* reset_decoder));
MOCK_METHOD6(Update,
void(uint16_t sequence_number,
uint32_t timestamp,
@@ -42,7 +42,7 @@
MOCK_CONST_METHOD0(CngRfc3389On, bool());
MOCK_CONST_METHOD0(CngOff, bool());
MOCK_METHOD0(SetCngOff, void());
- MOCK_METHOD1(ExpandDecision, void(Operations operation));
+ MOCK_METHOD1(ExpandDecision, void(NetEq::Operation operation));
MOCK_METHOD1(AddSampleMemory, void(int32_t value));
MOCK_METHOD0(TargetLevelMs, int());
MOCK_METHOD6(PacketArrived,
diff --git a/modules/audio_coding/neteq/neteq.cc b/modules/audio_coding/neteq/neteq.cc
deleted file mode 100644
index 0a36cb2..0000000
--- a/modules/audio_coding/neteq/neteq.cc
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2012 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/include/neteq.h"
-
-#include "modules/audio_coding/neteq/neteq_impl.h"
-#include "rtc_base/strings/string_builder.h"
-
-namespace webrtc {
-
-NetEq::Config::Config() = default;
-NetEq::Config::Config(const Config&) = default;
-NetEq::Config::Config(Config&&) = default;
-NetEq::Config::~Config() = default;
-NetEq::Config& NetEq::Config::operator=(const Config&) = default;
-NetEq::Config& NetEq::Config::operator=(Config&&) = default;
-
-std::string NetEq::Config::ToString() const {
- char buf[1024];
- rtc::SimpleStringBuilder ss(buf);
- 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
- << ", min_delay_ms=" << min_delay_ms << ", enable_fast_accelerate="
- << (enable_fast_accelerate ? "true" : "false")
- << ", enable_muted_state=" << (enable_muted_state ? "true" : "false")
- << ", enable_rtx_handling=" << (enable_rtx_handling ? "true" : "false");
- return ss.str();
-}
-
-// Creates all classes needed and inject them into a new NetEqImpl object.
-// Return the new object.
-NetEq* NetEq::Create(
- const NetEq::Config& config,
- Clock* clock,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
- return new NetEqImpl(config,
- NetEqImpl::Dependencies(config, clock, decoder_factory));
-}
-
-} // namespace webrtc
diff --git a/modules/audio_coding/neteq/neteq_controller.h b/modules/audio_coding/neteq/neteq_controller.h
deleted file mode 100644
index 16e3a8e..0000000
--- a/modules/audio_coding/neteq/neteq_controller.h
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- * Copyright (c) 2019 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_NETEQ_CONTROLLER_H_
-#define MODULES_AUDIO_CODING_NETEQ_NETEQ_CONTROLLER_H_
-
-#include <cstddef>
-#include <cstdint>
-
-#include <functional>
-#include <memory>
-
-#include "absl/types/optional.h"
-#include "modules/audio_coding/neteq/defines.h"
-#include "modules/audio_coding/neteq/tick_timer.h"
-
-namespace webrtc {
-
-// Decides the actions that NetEq should take. This affects the behavior of the
-// jitter buffer, and how it reacts to network conditions.
-// This class will undergo substantial refactoring in the near future, and the
-// API is expected to undergo significant changes. A target API is given below:
-//
-// class NetEqController {
-// public:
-// // Resets object to a clean state.
-// void Reset();
-// // Given NetEq status, make a decision.
-// Operation GetDecision(NetEqStatus neteq_status);
-// // Register every packet received.
-// void RegisterPacket(PacketInfo packet_info);
-// // Register empty packet.
-// void RegisterEmptyPacket();
-// // Register a codec switching.
-// void CodecSwithed();
-// // Sets the sample rate.
-// void SetSampleRate(int fs_hz);
-// // Sets the packet length in samples.
-// void SetPacketLengthSamples();
-// // Sets maximum delay.
-// void SetMaximumDelay(int delay_ms);
-// // Sets mininum delay.
-// void SetMinimumDelay(int delay_ms);
-// // Sets base mininum delay.
-// void SetBaseMinimumDelay(int delay_ms);
-// // Gets target buffer level.
-// int GetTargetBufferLevelMs() const;
-// // Gets filtered buffer level.
-// int GetFilteredBufferLevel() const;
-// // Gets base minimum delay.
-// int GetBaseMinimumDelay() const;
-// }
-
-class NetEqController {
- public:
- // This struct is used to create a NetEqController.
- struct Config {
- bool allow_time_stretching;
- bool enable_rtx_handling;
- int max_packets_in_buffer;
- int base_min_delay_ms;
- TickTimer* tick_timer;
- };
-
- struct PacketInfo {
- uint32_t timestamp;
- bool is_dtx;
- bool is_cng;
- };
-
- struct PacketBufferInfo {
- bool dtx_or_cng;
- size_t num_samples;
- size_t span_samples;
- size_t span_samples_no_dtx;
- size_t num_packets;
- };
-
- struct NetEqStatus {
- uint32_t target_timestamp;
- int16_t expand_mutefactor;
- size_t last_packet_samples;
- absl::optional<PacketInfo> next_packet;
- Modes last_mode;
- bool play_dtmf;
- size_t generated_noise_samples;
- PacketBufferInfo packet_buffer_info;
- };
-
- virtual ~NetEqController() = default;
-
- // Resets object to a clean state.
- virtual void Reset() = 0;
-
- // Resets parts of the state. Typically done when switching codecs.
- virtual void SoftReset() = 0;
-
- // Given info about the latest received packet, and current jitter buffer
- // status, returns the operation. |target_timestamp| and |expand_mutefactor|
- // are provided for reference. |last_packet_samples| is the number of samples
- // obtained from the last decoded frame. If there is a packet available, it
- // should be supplied in |packet|. The mode resulting from the last call to
- // NetEqImpl::GetAudio is supplied in |last_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).
- virtual Operations GetDecision(const NetEqStatus& status,
- bool* reset_decoder) = 0;
-
- // Inform NetEqController that an empty packet has arrived.
- virtual void RegisterEmptyPacket() = 0;
-
- // Sets the sample rate and the output block size.
- virtual void SetSampleRate(int fs_hz, size_t output_size_samples) = 0;
-
- // Sets a minimum or maximum delay in millisecond.
- // Returns true if the delay bound is successfully applied, otherwise false.
- virtual bool SetMaximumDelay(int delay_ms) = 0;
- virtual bool SetMinimumDelay(int delay_ms) = 0;
-
- // Sets a base minimum delay in milliseconds for packet buffer. The effective
- // minimum delay can't be lower than base minimum delay, even if a lower value
- // is set using SetMinimumDelay.
- // Returns true if the base minimum is successfully applied, otherwise false.
- virtual bool SetBaseMinimumDelay(int delay_ms) = 0;
- virtual int GetBaseMinimumDelay() const = 0;
-
- // These methods test the |cng_state_| for different conditions.
- virtual bool CngRfc3389On() const = 0;
- virtual bool CngOff() const = 0;
-
- // Resets the |cng_state_| to kCngOff.
- virtual void SetCngOff() = 0;
-
- // Reports back to DecisionLogic whether the decision to do expand remains or
- // 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.
- virtual void ExpandDecision(Operations operation) = 0;
-
- // Adds |value| to |sample_memory_|.
- virtual void AddSampleMemory(int32_t value) = 0;
-
- // Returns the target buffer level in ms.
- virtual int TargetLevelMs() = 0;
-
- // Notify the NetEqController that a packet has arrived. Returns the relative
- // arrival delay, if it can be computed.
- virtual absl::optional<int> PacketArrived(bool last_cng_or_dtmf,
- size_t packet_length_samples,
- bool should_update_stats,
- uint16_t main_sequence_number,
- uint32_t main_timestamp,
- int fs_hz) = 0;
-
- // Returns true if a peak was found.
- virtual bool PeakFound() const = 0;
-
- // Get the filtered buffer level in samples.
- virtual int GetFilteredBufferLevel() const = 0;
-
- // Accessors and mutators.
- virtual void set_sample_memory(int32_t value) = 0;
- virtual size_t noise_fast_forward() const = 0;
- virtual size_t packet_length_samples() const = 0;
- virtual void set_packet_length_samples(size_t value) = 0;
- virtual void set_prev_time_scale(bool value) = 0;
-};
-
-} // namespace webrtc
-#endif // MODULES_AUDIO_CODING_NETEQ_NETEQ_CONTROLLER_H_
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 7290e93..b6f5971 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -21,6 +21,7 @@
#include <vector>
#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/tick_timer.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
#include "modules/audio_coding/neteq/accelerate.h"
@@ -28,7 +29,6 @@
#include "modules/audio_coding/neteq/comfort_noise.h"
#include "modules/audio_coding/neteq/decision_logic.h"
#include "modules/audio_coding/neteq/decoder_database.h"
-#include "modules/audio_coding/neteq/defines.h"
#include "modules/audio_coding/neteq/dtmf_buffer.h"
#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
#include "modules/audio_coding/neteq/expand.h"
@@ -42,7 +42,6 @@
#include "modules/audio_coding/neteq/red_payload_splitter.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
-#include "modules/audio_coding/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/time_stretch.h"
#include "modules/audio_coding/neteq/timestamp_scaler.h"
#include "rtc_base/checks.h"
@@ -57,6 +56,7 @@
namespace {
std::unique_ptr<NetEqController> CreateNetEqController(
+ const NetEqControllerFactory& controller_factory,
int base_min_delay,
int max_packets_in_buffer,
bool enable_rtx_handling,
@@ -68,7 +68,7 @@
config.enable_rtx_handling = enable_rtx_handling;
config.allow_time_stretching = allow_time_stretching;
config.tick_timer = tick_timer;
- return std::make_unique<DecisionLogic>(std::move(config));
+ return controller_factory.CreateNetEqController(config);
}
} // namespace
@@ -76,7 +76,8 @@
NetEqImpl::Dependencies::Dependencies(
const NetEq::Config& config,
Clock* clock,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
+ const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+ const NetEqControllerFactory& controller_factory)
: clock(clock),
tick_timer(new TickTimer),
stats(new StatisticsCalculator),
@@ -87,7 +88,8 @@
packet_buffer(
new PacketBuffer(config.max_packets_in_buffer, tick_timer.get())),
neteq_controller(
- CreateNetEqController(config.min_delay_ms,
+ CreateNetEqController(controller_factory,
+ config.min_delay_ms,
config.max_packets_in_buffer,
config.enable_rtx_handling,
!config.for_test_no_time_stretching,
@@ -117,7 +119,7 @@
preemptive_expand_factory_(std::move(deps.preemptive_expand_factory)),
stats_(std::move(deps.stats)),
controller_(std::move(deps.neteq_controller)),
- last_mode_(kModeNormal),
+ last_mode_(Mode::kNormal),
decoded_buffer_length_(kMaxFrameSize),
decoded_buffer_(new int16_t[decoded_buffer_length_]),
playout_timestamp_(0),
@@ -230,7 +232,7 @@
int NetEqImpl::GetAudio(AudioFrame* audio_frame,
bool* muted,
- absl::optional<Operations> action_override) {
+ absl::optional<Operation> action_override) {
TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
rtc::CritScope lock(&crit_sect_);
if (GetAudioInternal(audio_frame, muted, action_override) != 0) {
@@ -381,8 +383,8 @@
absl::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
rtc::CritScope lock(&crit_sect_);
- if (first_packet_ || last_mode_ == kModeRfc3389Cng ||
- last_mode_ == kModeCodecInternalCng) {
+ if (first_packet_ || last_mode_ == Mode::kRfc3389Cng ||
+ last_mode_ == Mode::kCodecInternalCng) {
// We don't have a valid RTP timestamp until we have decoded our first
// RTP packet. Also, the RTP timestamp is not accurate while playing CNG,
// which is indicated by returning an empty value.
@@ -470,7 +472,7 @@
return sync_buffer_.get();
}
-Operations NetEqImpl::last_operation_for_test() const {
+NetEq::Operation NetEqImpl::last_operation_for_test() const {
rtc::CritScope lock(&crit_sect_);
return last_operation_;
}
@@ -743,10 +745,10 @@
int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
bool* muted,
- absl::optional<Operations> action_override) {
+ absl::optional<Operation> action_override) {
PacketList packet_list;
DtmfEvent dtmf_event;
- Operations operation;
+ Operation operation;
bool play_dtmf;
*muted = false;
last_decoded_timestamps_.clear();
@@ -763,7 +765,7 @@
// Check for muted state.
if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) {
- RTC_DCHECK_EQ(last_mode_, kModeExpand);
+ RTC_DCHECK_EQ(last_mode_, Mode::kExpand);
audio_frame->Reset();
RTC_DCHECK(audio_frame->muted()); // Reset() should mute the frame.
playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
@@ -782,7 +784,7 @@
int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
&play_dtmf, action_override);
if (return_value != 0) {
- last_mode_ = kModeError;
+ last_mode_ = Mode::kError;
return return_value;
}
@@ -793,7 +795,8 @@
Decode(&packet_list, &operation, &length, &speech_type);
assert(vad_.get());
- bool sid_frame_available = (operation == kRfc3389Cng && !packet_list.empty());
+ bool sid_frame_available =
+ (operation == Operation::kRfc3389Cng && !packet_list.empty());
vad_->Update(decoded_buffer_.get(), static_cast<size_t>(length), speech_type,
sid_frame_available, fs_hz_);
@@ -810,18 +813,18 @@
algorithm_buffer_->Clear();
switch (operation) {
- case kNormal: {
+ case Operation::kNormal: {
DoNormal(decoded_buffer_.get(), length, speech_type, play_dtmf);
if (length > 0) {
stats_->DecodedOutputPlayed();
}
break;
}
- case kMerge: {
+ case Operation::kMerge: {
DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf);
break;
}
- case kExpand: {
+ case Operation::kExpand: {
RTC_DCHECK_EQ(return_value, 0);
if (!current_rtp_payload_type_ || !DoCodecPlc()) {
return_value = DoExpand(play_dtmf);
@@ -830,40 +833,40 @@
output_size_samples_);
break;
}
- case kAccelerate:
- case kFastAccelerate: {
+ case Operation::kAccelerate:
+ case Operation::kFastAccelerate: {
const bool fast_accelerate =
- enable_fast_accelerate_ && (operation == kFastAccelerate);
+ enable_fast_accelerate_ && (operation == Operation::kFastAccelerate);
return_value = DoAccelerate(decoded_buffer_.get(), length, speech_type,
play_dtmf, fast_accelerate);
break;
}
- case kPreemptiveExpand: {
+ case Operation::kPreemptiveExpand: {
return_value = DoPreemptiveExpand(decoded_buffer_.get(), length,
speech_type, play_dtmf);
break;
}
- case kRfc3389Cng:
- case kRfc3389CngNoPacket: {
+ case Operation::kRfc3389Cng:
+ case Operation::kRfc3389CngNoPacket: {
return_value = DoRfc3389Cng(&packet_list, play_dtmf);
break;
}
- case kCodecInternalCng: {
+ case Operation::kCodecInternalCng: {
// This handles the case when there is no transmission and the decoder
// should produce internal comfort noise.
// TODO(hlundin): Write test for codec-internal CNG.
DoCodecInternalCng(decoded_buffer_.get(), length);
break;
}
- case kDtmf: {
+ case Operation::kDtmf: {
// TODO(hlundin): Write test for this.
return_value = DoDtmf(dtmf_event, &play_dtmf);
break;
}
- case kUndefined: {
+ case Operation::kUndefined: {
RTC_LOG(LS_ERROR) << "Invalid operation kUndefined.";
assert(false); // This should not happen.
- last_mode_ = kModeError;
+ last_mode_ = Mode::kError;
return kInvalidOperation;
}
} // End of switch.
@@ -872,7 +875,7 @@
return return_value;
}
- if (last_mode_ != kModeRfc3389Cng) {
+ if (last_mode_ != Mode::kRfc3389Cng) {
comfort_noise_->Reset();
}
@@ -941,20 +944,20 @@
// Update the background noise parameters if last operation wrote data
// straight from the decoder to the |sync_buffer_|. That is, none of the
// operations that modify the signal can be followed by a parameter update.
- if ((last_mode_ == kModeNormal) || (last_mode_ == kModeAccelerateFail) ||
- (last_mode_ == kModePreemptiveExpandFail) ||
- (last_mode_ == kModeRfc3389Cng) ||
- (last_mode_ == kModeCodecInternalCng)) {
+ if ((last_mode_ == Mode::kNormal) || (last_mode_ == Mode::kAccelerateFail) ||
+ (last_mode_ == Mode::kPreemptiveExpandFail) ||
+ (last_mode_ == Mode::kRfc3389Cng) ||
+ (last_mode_ == Mode::kCodecInternalCng)) {
background_noise_->Update(*sync_buffer_, *vad_.get());
}
- if (operation == kDtmf) {
+ if (operation == Operation::kDtmf) {
// DTMF data was written the end of |sync_buffer_|.
// Update index to end of DTMF data in |sync_buffer_|.
sync_buffer_->set_dtmf_index(sync_buffer_->Size());
}
- if (last_mode_ != kModeExpand && last_mode_ != kModeCodecPlc) {
+ if (last_mode_ != Mode::kExpand && last_mode_ != Mode::kCodecPlc) {
// If last operation was not expand, calculate the |playout_timestamp_| from
// the |sync_buffer_|. However, do not update the |playout_timestamp_| if it
// would be moved "backwards".
@@ -978,8 +981,9 @@
: timestamp_scaler_->ToExternal(playout_timestamp_) -
static_cast<uint32_t>(audio_frame->samples_per_channel_);
- if (!(last_mode_ == kModeRfc3389Cng || last_mode_ == kModeCodecInternalCng ||
- last_mode_ == kModeExpand || last_mode_ == kModeCodecPlc)) {
+ if (!(last_mode_ == Mode::kRfc3389Cng ||
+ last_mode_ == Mode::kCodecInternalCng || last_mode_ == Mode::kExpand ||
+ last_mode_ == Mode::kCodecPlc)) {
generated_noise_stopwatch_.reset();
}
@@ -988,14 +992,14 @@
return return_value;
}
-int NetEqImpl::GetDecision(Operations* operation,
+int NetEqImpl::GetDecision(Operation* operation,
PacketList* packet_list,
DtmfEvent* dtmf_event,
bool* play_dtmf,
- absl::optional<Operations> action_override) {
+ absl::optional<Operation> action_override) {
// Initialize output variables.
*play_dtmf = false;
- *operation = kUndefined;
+ *operation = Operation::kUndefined;
assert(sync_buffer_.get());
uint32_t end_timestamp = sync_buffer_->end_timestamp();
@@ -1014,7 +1018,7 @@
controller_->noise_fast_forward()
: 0;
- if (controller_->CngRfc3389On() || last_mode_ == kModeRfc3389Cng) {
+ if (controller_->CngRfc3389On() || last_mode_ == Mode::kRfc3389Cng) {
// Because of timestamp peculiarities, we have to "manually" disallow using
// a CNG packet with the same timestamp as the one that was last played.
// This can happen when using redundancy and will cause the timing to shift.
@@ -1038,10 +1042,10 @@
assert(expand_.get());
const int samples_left = static_cast<int>(sync_buffer_->FutureLength() -
expand_->overlap_length());
- if (last_mode_ == kModeAccelerateSuccess ||
- last_mode_ == kModeAccelerateLowEnergy ||
- last_mode_ == kModePreemptiveExpandSuccess ||
- last_mode_ == kModePreemptiveExpandLowEnergy) {
+ if (last_mode_ == Mode::kAccelerateSuccess ||
+ last_mode_ == Mode::kAccelerateLowEnergy ||
+ last_mode_ == Mode::kPreemptiveExpandSuccess ||
+ last_mode_ == Mode::kPreemptiveExpandLowEnergy) {
// Subtract (samples_left + output_size_samples_) from sampleMemory.
controller_->AddSampleMemory(
-(samples_left + rtc::dchecked_cast<int>(output_size_samples_)));
@@ -1091,9 +1095,11 @@
// during DTX. When we have a better way to update buffer level during DTX,
// this can be discarded.
if (packet && packet->frame && packet->frame->IsDtxPacket() &&
- (*operation == kMerge || *operation == kAccelerate ||
- *operation == kFastAccelerate || *operation == kPreemptiveExpand)) {
- *operation = kNormal;
+ (*operation == Operation::kMerge ||
+ *operation == Operation::kAccelerate ||
+ *operation == Operation::kFastAccelerate ||
+ *operation == Operation::kPreemptiveExpand)) {
+ *operation = Operation::kNormal;
}
if (action_override) {
@@ -1104,16 +1110,17 @@
// change decision to normal, unless the decision was merge, accelerate, or
// preemptive expand.
if (samples_left >= rtc::dchecked_cast<int>(output_size_samples_) &&
- *operation != kMerge && *operation != kAccelerate &&
- *operation != kFastAccelerate && *operation != kPreemptiveExpand) {
- *operation = kNormal;
+ *operation != Operation::kMerge && *operation != Operation::kAccelerate &&
+ *operation != Operation::kFastAccelerate &&
+ *operation != Operation::kPreemptiveExpand) {
+ *operation = Operation::kNormal;
return 0;
}
controller_->ExpandDecision(*operation);
// Check conditions for reset.
- if (new_codec_ || *operation == kUndefined) {
+ if (new_codec_ || *operation == Operation::kUndefined) {
// The only valid reason to get kUndefined is that new_codec_ is set.
assert(new_codec_);
if (*play_dtmf && !packet) {
@@ -1124,13 +1131,13 @@
return -1;
}
timestamp_ = packet->timestamp;
- if (*operation == kRfc3389CngNoPacket &&
+ if (*operation == Operation::kRfc3389CngNoPacket &&
decoder_database_->IsComfortNoise(packet->payload_type)) {
// Change decision to CNG packet, since we do have a CNG packet, but it
// was considered too early to use. Now, use it anyway.
- *operation = kRfc3389Cng;
- } else if (*operation != kRfc3389Cng) {
- *operation = kNormal;
+ *operation = Operation::kRfc3389Cng;
+ } else if (*operation != Operation::kRfc3389Cng) {
+ *operation = Operation::kNormal;
}
}
// Adjust |sync_buffer_| timestamp before setting |end_timestamp| to the
@@ -1148,15 +1155,15 @@
const size_t samples_30_ms = 3 * samples_10_ms;
switch (*operation) {
- case kExpand: {
+ case Operation::kExpand: {
timestamp_ = end_timestamp;
return 0;
}
- case kRfc3389CngNoPacket:
- case kCodecInternalCng: {
+ case Operation::kRfc3389CngNoPacket:
+ case Operation::kCodecInternalCng: {
return 0;
}
- case kDtmf: {
+ case Operation::kDtmf: {
// TODO(hlundin): Write test for this.
// Update timestamp.
timestamp_ = end_timestamp;
@@ -1166,7 +1173,7 @@
output_size_samples_ +
controller_->noise_fast_forward()
: 0;
- if (generated_noise_samples > 0 && last_mode_ != kModeDtmf) {
+ if (generated_noise_samples > 0 && last_mode_ != Mode::kDtmf) {
// Make a jump in timestamp due to the recently played comfort noise.
uint32_t timestamp_jump =
static_cast<uint32_t>(generated_noise_samples);
@@ -1175,8 +1182,8 @@
}
return 0;
}
- case kAccelerate:
- case kFastAccelerate: {
+ case Operation::kAccelerate:
+ case Operation::kFastAccelerate: {
// In order to do an accelerate we need at least 30 ms of audio data.
if (samples_left >= static_cast<int>(samples_30_ms)) {
// Already have enough data, so we do not need to extract any more.
@@ -1186,7 +1193,7 @@
} else if (samples_left >= static_cast<int>(samples_10_ms) &&
decoder_frame_length_ >= samples_30_ms) {
// Avoid decoding more data as it might overflow the playout buffer.
- *operation = kNormal;
+ *operation = Operation::kNormal;
return 0;
} else if (samples_left < static_cast<int>(samples_20_ms) &&
decoder_frame_length_ < samples_30_ms) {
@@ -1194,7 +1201,7 @@
// not perform accelerate yet, but wait until we only need to do one
// decoding.
required_samples = 2 * output_size_samples_;
- *operation = kNormal;
+ *operation = Operation::kNormal;
}
// If none of the above is true, we have one of two possible situations:
// (1) 20 ms <= samples_left < 30 ms and decoder_frame_length_ < 30 ms; or
@@ -1203,7 +1210,7 @@
// frame now.
break;
}
- case kPreemptiveExpand: {
+ case Operation::kPreemptiveExpand: {
// In order to do a preemptive expand we need at least 30 ms of decoded
// audio data.
if ((samples_left >= static_cast<int>(samples_30_ms)) ||
@@ -1225,7 +1232,7 @@
// Move on with the preemptive expand decision.
break;
}
- case kMerge: {
+ case Operation::kMerge: {
required_samples =
std::max(merge_->RequiredFutureSamples(), required_samples);
break;
@@ -1247,7 +1254,7 @@
stats_->LostSamples(packet->timestamp - end_timestamp);
}
- if (*operation != kRfc3389Cng) {
+ if (*operation != Operation::kRfc3389Cng) {
// We are about to decode and use a non-CNG packet.
controller_->SetCngOff();
}
@@ -1258,18 +1265,20 @@
}
}
- if (*operation == kAccelerate || *operation == kFastAccelerate ||
- *operation == kPreemptiveExpand) {
+ if (*operation == Operation::kAccelerate ||
+ *operation == Operation::kFastAccelerate ||
+ *operation == Operation::kPreemptiveExpand) {
controller_->set_sample_memory(samples_left + extracted_samples);
controller_->set_prev_time_scale(true);
}
- if (*operation == kAccelerate || *operation == kFastAccelerate) {
+ if (*operation == Operation::kAccelerate ||
+ *operation == Operation::kFastAccelerate) {
// Check that we have enough data (30ms) to do accelerate.
if (extracted_samples + samples_left < static_cast<int>(samples_30_ms)) {
// TODO(hlundin): Write test for this.
// Not enough, do normal operation instead.
- *operation = kNormal;
+ *operation = Operation::kNormal;
}
}
@@ -1278,7 +1287,7 @@
}
int NetEqImpl::Decode(PacketList* packet_list,
- Operations* operation,
+ Operation* operation,
int* decoded_length,
AudioDecoder::SpeechType* speech_type) {
*speech_type = AudioDecoder::kSpeech;
@@ -1341,12 +1350,12 @@
*decoded_length = 0;
// Update codec-internal PLC state.
- if ((*operation == kMerge) && decoder && decoder->HasDecodePlc()) {
+ if ((*operation == Operation::kMerge) && decoder && decoder->HasDecodePlc()) {
decoder->DecodePlc(1, &decoded_buffer_[*decoded_length]);
}
int return_value;
- if (*operation == kCodecInternalCng) {
+ if (*operation == Operation::kCodecInternalCng) {
RTC_DCHECK(packet_list->empty());
return_value = DecodeCng(decoder, decoded_length, speech_type);
} else {
@@ -1371,7 +1380,7 @@
return_value = kOtherDecoderError;
RTC_LOG(LS_WARNING) << "Decoder error (no error code)";
}
- *operation = kExpand; // Do expansion to get data instead.
+ *operation = Operation::kExpand; // Do expansion to get data instead.
}
if (*speech_type != AudioDecoder::kComfortNoise) {
// Don't increment timestamp if codec returned CNG speech type
@@ -1417,7 +1426,7 @@
}
int NetEqImpl::DecodeLoop(PacketList* packet_list,
- const Operations& operation,
+ const Operation& operation,
AudioDecoder* decoder,
int* decoded_length,
AudioDecoder::SpeechType* speech_type) {
@@ -1432,9 +1441,11 @@
// number decoder channels.
assert(sync_buffer_->Channels() == decoder->Channels());
assert(decoded_buffer_length_ >= kMaxFrameSize * decoder->Channels());
- assert(operation == kNormal || operation == kAccelerate ||
- operation == kFastAccelerate || operation == kMerge ||
- operation == kPreemptiveExpand);
+ assert(operation == Operation::kNormal ||
+ operation == Operation::kAccelerate ||
+ operation == Operation::kFastAccelerate ||
+ operation == Operation::kMerge ||
+ operation == Operation::kPreemptiveExpand);
auto opt_result = packet_list->front().frame->Decode(
rtc::ArrayView<int16_t>(&decoded_buffer_[*decoded_length],
@@ -1485,14 +1496,14 @@
normal_->Process(decoded_buffer, decoded_length, last_mode_,
algorithm_buffer_.get());
if (decoded_length != 0) {
- last_mode_ = kModeNormal;
+ last_mode_ = Mode::kNormal;
}
// If last packet was decoded as an inband CNG, set mode to CNG instead.
if ((speech_type == AudioDecoder::kComfortNoise) ||
- ((last_mode_ == kModeCodecInternalCng) && (decoded_length == 0))) {
+ ((last_mode_ == Mode::kCodecInternalCng) && (decoded_length == 0))) {
// TODO(hlundin): Remove second part of || statement above.
- last_mode_ = kModeCodecInternalCng;
+ last_mode_ = Mode::kCodecInternalCng;
}
if (!play_dtmf) {
@@ -1521,10 +1532,10 @@
stats_->ExpandedVoiceSamplesCorrection(expand_length_correction);
}
- last_mode_ = kModeMerge;
+ last_mode_ = Mode::kMerge;
// If last packet was decoded as an inband CNG, set mode to CNG instead.
if (speech_type == AudioDecoder::kComfortNoise) {
- last_mode_ = kModeCodecInternalCng;
+ last_mode_ = Mode::kCodecInternalCng;
}
expand_->Reset();
if (!play_dtmf) {
@@ -1555,7 +1566,7 @@
concealment_audio_.size() / channels;
// Update in-call and post-call statistics.
- const bool is_new_concealment_event = (last_mode_ != kModeCodecPlc);
+ const bool is_new_concealment_event = (last_mode_ != Mode::kCodecPlc);
if (std::all_of(concealment_audio_.cbegin(), concealment_audio_.cend(),
[](int16_t i) { return i == 0; })) {
// Expand operation generates only noise.
@@ -1566,7 +1577,7 @@
stats_->ExpandedVoiceSamples(concealed_samples_per_channel,
is_new_concealment_event);
}
- last_mode_ = kModeCodecPlc;
+ last_mode_ = Mode::kCodecPlc;
if (!generated_noise_stopwatch_) {
// Start a new stopwatch since we may be covering for a lost CNG packet.
generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch();
@@ -1580,7 +1591,7 @@
algorithm_buffer_->Clear();
int return_value = expand_->Process(algorithm_buffer_.get());
size_t length = algorithm_buffer_->Size();
- bool is_new_concealment_event = (last_mode_ != kModeExpand);
+ bool is_new_concealment_event = (last_mode_ != Mode::kExpand);
// Update in-call and post-call statistics.
if (expand_->MuteFactor(0) == 0) {
@@ -1591,7 +1602,7 @@
stats_->ExpandedVoiceSamples(length, is_new_concealment_event);
}
- last_mode_ = kModeExpand;
+ last_mode_ = Mode::kExpand;
if (return_value < 0) {
return return_value;
@@ -1640,17 +1651,17 @@
stats_->AcceleratedSamples(samples_removed);
switch (return_code) {
case Accelerate::kSuccess:
- last_mode_ = kModeAccelerateSuccess;
+ last_mode_ = Mode::kAccelerateSuccess;
break;
case Accelerate::kSuccessLowEnergy:
- last_mode_ = kModeAccelerateLowEnergy;
+ last_mode_ = Mode::kAccelerateLowEnergy;
break;
case Accelerate::kNoStretch:
- last_mode_ = kModeAccelerateFail;
+ last_mode_ = Mode::kAccelerateFail;
break;
case Accelerate::kError:
- // TODO(hlundin): Map to kModeError instead?
- last_mode_ = kModeAccelerateFail;
+ // TODO(hlundin): Map to Modes::kError instead?
+ last_mode_ = Mode::kAccelerateFail;
return kAccelerateError;
}
@@ -1676,7 +1687,7 @@
// If last packet was decoded as an inband CNG, set mode to CNG instead.
if (speech_type == AudioDecoder::kComfortNoise) {
- last_mode_ = kModeCodecInternalCng;
+ last_mode_ = Mode::kCodecInternalCng;
}
if (!play_dtmf) {
dtmf_tone_generator_->Reset();
@@ -1718,17 +1729,17 @@
stats_->PreemptiveExpandedSamples(samples_added);
switch (return_code) {
case PreemptiveExpand::kSuccess:
- last_mode_ = kModePreemptiveExpandSuccess;
+ last_mode_ = Mode::kPreemptiveExpandSuccess;
break;
case PreemptiveExpand::kSuccessLowEnergy:
- last_mode_ = kModePreemptiveExpandLowEnergy;
+ last_mode_ = Mode::kPreemptiveExpandLowEnergy;
break;
case PreemptiveExpand::kNoStretch:
- last_mode_ = kModePreemptiveExpandFail;
+ last_mode_ = Mode::kPreemptiveExpandFail;
break;
case PreemptiveExpand::kError:
- // TODO(hlundin): Map to kModeError instead?
- last_mode_ = kModePreemptiveExpandFail;
+ // TODO(hlundin): Map to Modes::kError instead?
+ last_mode_ = Mode::kPreemptiveExpandFail;
return kPreemptiveExpandError;
}
@@ -1742,7 +1753,7 @@
// If last packet was decoded as an inband CNG, set mode to CNG instead.
if (speech_type == AudioDecoder::kComfortNoise) {
- last_mode_ = kModeCodecInternalCng;
+ last_mode_ = Mode::kCodecInternalCng;
}
if (!play_dtmf) {
dtmf_tone_generator_->Reset();
@@ -1769,7 +1780,7 @@
int cn_return =
comfort_noise_->Generate(output_size_samples_, algorithm_buffer_.get());
expand_->Reset();
- last_mode_ = kModeRfc3389Cng;
+ last_mode_ = Mode::kRfc3389Cng;
if (!play_dtmf) {
dtmf_tone_generator_->Reset();
}
@@ -1788,7 +1799,7 @@
RTC_DCHECK(normal_.get());
normal_->Process(decoded_buffer, decoded_length, last_mode_,
algorithm_buffer_.get());
- last_mode_ = kModeCodecInternalCng;
+ last_mode_ = Mode::kCodecInternalCng;
expand_->Reset();
}
@@ -1803,7 +1814,8 @@
// switch from audio to DTMF. Issue 1545 is filed to track this.
//
// bool dtmf_switch = false;
- // if ((last_mode_ != kModeDtmf) && dtmf_tone_generator_->initialized()) {
+ // if ((last_mode_ != Modes::kDtmf) &&
+ // dtmf_tone_generator_->initialized()) {
// // Special case; see below.
// // We must catch this before calling Generate, since |initialized| is
// // modified in that call.
@@ -1865,7 +1877,7 @@
sync_buffer_->IncreaseEndTimestamp(
static_cast<uint32_t>(output_size_samples_));
expand_->Reset();
- last_mode_ = kModeDtmf;
+ last_mode_ = Mode::kDtmf;
// Set to false because the DTMF is already in the algorithm buffer.
*play_dtmf = false;
@@ -2024,7 +2036,7 @@
output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
decoder_frame_length_ = 3 * output_size_samples_; // Initialize to 30ms.
- last_mode_ = kModeNormal;
+ last_mode_ = Mode::kNormal;
ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
if (cng_decoder)
@@ -2076,16 +2088,17 @@
NetEqImpl::OutputType NetEqImpl::LastOutputType() {
assert(vad_.get());
assert(expand_.get());
- if (last_mode_ == kModeCodecInternalCng || last_mode_ == kModeRfc3389Cng) {
+ if (last_mode_ == Mode::kCodecInternalCng ||
+ last_mode_ == Mode::kRfc3389Cng) {
return OutputType::kCNG;
- } else if (last_mode_ == kModeExpand && expand_->MuteFactor(0) == 0) {
+ } else if (last_mode_ == Mode::kExpand && expand_->MuteFactor(0) == 0) {
// Expand mode has faded down to background noise only (very long expand).
return OutputType::kPLCCNG;
- } else if (last_mode_ == kModeExpand) {
+ } else if (last_mode_ == Mode::kExpand) {
return OutputType::kPLC;
} else if (vad_->running() && !vad_->active_speech()) {
return OutputType::kVadPassive;
- } else if (last_mode_ == kModeCodecPlc) {
+ } else if (last_mode_ == Mode::kCodecPlc) {
return OutputType::kCodecPLC;
} else {
return OutputType::kNormalSpeech;
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index aa7eba1..956cb6e 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -19,16 +19,16 @@
#include "absl/types/optional.h"
#include "api/audio/audio_frame.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "api/neteq/neteq_controller_factory.h"
+#include "api/neteq/tick_timer.h"
#include "api/rtp_packet_info.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
-#include "modules/audio_coding/neteq/defines.h" // Modes, Operations
#include "modules/audio_coding/neteq/expand_uma_logger.h"
-#include "modules/audio_coding/neteq/include/neteq.h"
-#include "modules/audio_coding/neteq/neteq_controller.h"
#include "modules/audio_coding/neteq/packet.h"
#include "modules/audio_coding/neteq/random_vector.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
-#include "modules/audio_coding/neteq/tick_timer.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/thread_annotations.h"
@@ -96,10 +96,10 @@
// before sending the struct to the NetEqImpl constructor. However, there
// are dependencies between some of the classes inside the struct, so
// swapping out one may make it necessary to re-create another one.
- Dependencies(
- const NetEq::Config& config,
- Clock* clock,
- const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
+ Dependencies(const NetEq::Config& config,
+ Clock* clock,
+ const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+ const NetEqControllerFactory& controller_factory);
~Dependencies();
Clock* const clock;
@@ -133,7 +133,7 @@
int GetAudio(
AudioFrame* audio_frame,
bool* muted,
- absl::optional<Operations> action_override = absl::nullopt) override;
+ absl::optional<Operation> action_override = absl::nullopt) override;
void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
@@ -195,7 +195,7 @@
// This accessor method is only intended for testing purposes.
const SyncBuffer* sync_buffer_for_test() const;
- Operations last_operation_for_test() const;
+ Operation last_operation_for_test() const;
protected:
static const int kOutputSizeMs = 10;
@@ -216,7 +216,7 @@
// Returns 0 on success, otherwise an error code.
int GetAudioInternal(AudioFrame* audio_frame,
bool* muted,
- absl::optional<Operations> action_override)
+ absl::optional<Operation> action_override)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Provides a decision to the GetAudioInternal method. The decision what to
@@ -224,11 +224,11 @@
// |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
// DTMF should be played, |play_dtmf| is set to true by the method.
// Returns 0 on success, otherwise an error code.
- int GetDecision(Operations* operation,
+ int GetDecision(Operation* operation,
PacketList* packet_list,
DtmfEvent* dtmf_event,
bool* play_dtmf,
- absl::optional<Operations> action_override)
+ absl::optional<Operation> action_override)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Decodes the speech packets in |packet_list|, and writes the results to
@@ -238,7 +238,7 @@
// to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
// comfort noise, those are not decoded.
int Decode(PacketList* packet_list,
- Operations* operation,
+ Operation* operation,
int* decoded_length,
AudioDecoder::SpeechType* speech_type)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
@@ -251,7 +251,7 @@
// Sub-method to Decode(). Performs the actual decoding.
int DecodeLoop(PacketList* packet_list,
- const Operations& operation,
+ const Operation& operation,
AudioDecoder* decoder,
int* decoded_length,
AudioDecoder::SpeechType* speech_type)
@@ -374,8 +374,8 @@
int last_output_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
size_t output_size_samples_ RTC_GUARDED_BY(crit_sect_);
size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_);
- Modes last_mode_ RTC_GUARDED_BY(crit_sect_);
- Operations last_operation_ RTC_GUARDED_BY(crit_sect_);
+ Mode last_mode_ RTC_GUARDED_BY(crit_sect_);
+ Operation last_operation_ RTC_GUARDED_BY(crit_sect_);
size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_);
uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_);
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 4a47a4d..8599807 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -15,18 +15,20 @@
#include <vector>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/default_neteq_controller_factory.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "api/test/neteq_factory_with_codecs.h"
#include "modules/audio_coding/neteq/accelerate.h"
#include "modules/audio_coding/neteq/decision_logic.h"
#include "modules/audio_coding/neteq/expand.h"
#include "modules/audio_coding/neteq/histogram.h"
-#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "modules/audio_coding/neteq/mock/mock_dtmf_buffer.h"
#include "modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h"
#include "modules/audio_coding/neteq/mock/mock_neteq_controller.h"
#include "modules/audio_coding/neteq/mock/mock_packet_buffer.h"
#include "modules/audio_coding/neteq/mock/mock_red_payload_splitter.h"
-#include "modules/audio_coding/neteq/neteq_controller.h"
#include "modules/audio_coding/neteq/preemptive_expand.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
@@ -73,7 +75,8 @@
void CreateInstance(
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
ASSERT_TRUE(decoder_factory);
- NetEqImpl::Dependencies deps(config_, &clock_, decoder_factory);
+ NetEqImpl::Dependencies deps(config_, &clock_, decoder_factory,
+ DefaultNetEqControllerFactory());
// Get a local pointer to NetEq's TickTimer object.
tick_timer_ = deps.tick_timer.get();
@@ -249,9 +252,8 @@
TEST(NetEq, CreateAndDestroy) {
NetEq::Config config;
SimulatedClock clock(0);
- NetEq* neteq =
- NetEq::Create(config, &clock, CreateBuiltinAudioDecoderFactory());
- delete neteq;
+ std::unique_ptr<NetEqFactory> neteq_factory = CreateNetEqFactoryWithCodecs();
+ std::unique_ptr<NetEq> neteq = neteq_factory->CreateNetEq(config, &clock);
}
TEST_F(NetEqImplTest, RegisterPayloadType) {
@@ -1378,7 +1380,7 @@
CreateInstance();
EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
.Times(1)
- .WillOnce(Return(kNormal));
+ .WillOnce(Return(NetEq::Operation::kNormal));
const int kPayloadLengthSamples = 80;
const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples; // PCM 16-bit.
@@ -1518,7 +1520,8 @@
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
- EXPECT_EQ(kCodecInternalCng, neteq_->last_operation_for_test());
+ EXPECT_EQ(NetEq::Operation::kCodecInternalCng,
+ neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, Normal) {
@@ -1528,7 +1531,7 @@
InsertPacket(first_timestamp());
GetFirstPacket();
- EXPECT_EQ(kNormal, neteq_->last_operation_for_test());
+ EXPECT_EQ(NetEq::Operation::kNormal, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, Merge) {
@@ -1541,16 +1544,16 @@
GetFirstPacket();
bool muted;
EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
- .WillOnce(Return(kExpand));
+ .WillOnce(Return(NetEq::Operation::kExpand));
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
InsertPacket(first_timestamp() + 2 * timestamp_diff_between_packets());
EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
- .WillOnce(Return(kMerge));
+ .WillOnce(Return(NetEq::Operation::kMerge));
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
- EXPECT_EQ(kMerge, neteq_->last_operation_for_test());
+ EXPECT_EQ(NetEq::Operation::kMerge, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, Expand) {
@@ -1562,7 +1565,7 @@
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
- EXPECT_EQ(kExpand, neteq_->last_operation_for_test());
+ EXPECT_EQ(NetEq::Operation::kExpand, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, FastAccelerate) {
@@ -1575,11 +1578,12 @@
EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
.Times(1)
- .WillOnce(Return(kFastAccelerate));
+ .WillOnce(Return(NetEq::Operation::kFastAccelerate));
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
- EXPECT_EQ(kFastAccelerate, neteq_->last_operation_for_test());
+ EXPECT_EQ(NetEq::Operation::kFastAccelerate,
+ neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, PreemptiveExpand) {
@@ -1593,11 +1597,12 @@
EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
.Times(1)
- .WillOnce(Return(kPreemptiveExpand));
+ .WillOnce(Return(NetEq::Operation::kPreemptiveExpand));
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
- EXPECT_EQ(kPreemptiveExpand, neteq_->last_operation_for_test());
+ EXPECT_EQ(NetEq::Operation::kPreemptiveExpand,
+ neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, Accelerate) {
@@ -1611,11 +1616,11 @@
EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
.Times(1)
- .WillOnce(Return(kAccelerate));
+ .WillOnce(Return(NetEq::Operation::kAccelerate));
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
- EXPECT_EQ(kAccelerate, neteq_->last_operation_for_test());
+ EXPECT_EQ(NetEq::Operation::kAccelerate, neteq_->last_operation_for_test());
}
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index aa61d65..0e24f68 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -13,7 +13,9 @@
#include "absl/memory/memory.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_decoder.h"
-#include "modules/audio_coding/neteq/include/neteq.h"
+#include "api/neteq/custom_neteq_factory.h"
+#include "api/neteq/default_neteq_controller_factory.h"
+#include "api/neteq/neteq.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "rtc_base/ref_counted_object.h"
#include "system_wrappers/include/clock.h"
@@ -23,6 +25,19 @@
namespace webrtc {
namespace test {
+namespace {
+
+std::unique_ptr<NetEq> CreateNetEq(
+ const NetEq::Config& config,
+ Clock* clock,
+ const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+ CustomNetEqFactory neteq_factory(
+ decoder_factory, std::make_unique<DefaultNetEqControllerFactory>());
+ return neteq_factory.CreateNetEq(config, clock);
+}
+
+} // namespace
+
using ::testing::_;
using ::testing::Return;
using ::testing::SetArgPointee;
@@ -162,8 +177,7 @@
packet_loss_interval_(0xffffffff) {
NetEq::Config config;
config.sample_rate_hz = format.clockrate_hz;
- neteq_ = absl::WrapUnique(
- NetEq::Create(config, Clock::GetRealTimeClock(), decoder_factory_));
+ neteq_ = CreateNetEq(config, Clock::GetRealTimeClock(), decoder_factory_);
neteq_->RegisterPayloadType(kPayloadType, format);
}
diff --git a/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index e59637b..14979ae 100644
--- a/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -17,8 +17,9 @@
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "api/test/neteq_factory_with_codecs.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "rtc_base/strings/string_builder.h"
@@ -67,10 +68,10 @@
last_arrival_time_(0) {
NetEq::Config config;
config.sample_rate_hz = sample_rate_hz_;
- rtc::scoped_refptr<AudioDecoderFactory> factory =
- CreateBuiltinAudioDecoderFactory();
- neteq_mono_ = NetEq::Create(config, &clock_, factory);
- neteq_ = NetEq::Create(config, &clock_, factory);
+ std::unique_ptr<NetEqFactory> neteq_factory =
+ CreateNetEqFactoryWithCodecs();
+ neteq_mono_ = neteq_factory->CreateNetEq(config, &clock_);
+ neteq_ = neteq_factory->CreateNetEq(config, &clock_);
input_ = new int16_t[frame_size_samples_];
encoded_ = new uint8_t[2 * frame_size_samples_];
input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];
@@ -79,8 +80,6 @@
}
~NetEqStereoTest() {
- delete neteq_mono_;
- delete neteq_;
delete[] input_;
delete[] encoded_;
delete[] input_multi_channel_;
@@ -206,8 +205,8 @@
const size_t frame_size_samples_;
const size_t output_size_samples_;
SimulatedClock clock_;
- NetEq* neteq_mono_;
- NetEq* neteq_;
+ std::unique_ptr<NetEq> neteq_mono_;
+ std::unique_ptr<NetEq> neteq_;
test::RtpGenerator rtp_generator_mono_;
test::RtpGenerator rtp_generator_;
int16_t* input_;
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 443c1a0..a192611 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "modules/audio_coding/neteq/include/neteq.h"
+#include "api/neteq/neteq.h"
#include <math.h>
#include <stdlib.h>
@@ -23,6 +23,7 @@
#include "absl/flags/flag.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/test/neteq_factory_with_codecs.h"
#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"
@@ -287,7 +288,7 @@
void DuplicateCng();
SimulatedClock clock_;
- NetEq* neteq_;
+ std::unique_ptr<NetEq> neteq_;
NetEq::Config config_;
std::unique_ptr<test::RtpFileSource> rtp_source_;
std::unique_ptr<test::Packet> packet_;
@@ -305,7 +306,6 @@
NetEqDecodingTest::NetEqDecodingTest()
: clock_(0),
- neteq_(NULL),
config_(),
output_sample_rate_(kInitSampleRateHz),
algorithmic_delay_ms_(0) {
@@ -313,17 +313,16 @@
}
void NetEqDecodingTest::SetUp() {
- neteq_ = NetEq::Create(config_, &clock_, CreateBuiltinAudioDecoderFactory());
+ std::unique_ptr<NetEqFactory> neteq_factory = CreateNetEqFactoryWithCodecs();
+ neteq_ = neteq_factory->CreateNetEq(config_, &clock_);
NetEqNetworkStatistics stat;
ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
algorithmic_delay_ms_ = stat.current_buffer_size_ms;
ASSERT_TRUE(neteq_);
- LoadDecoders(neteq_);
+ LoadDecoders(neteq_.get());
}
-void NetEqDecodingTest::TearDown() {
- delete neteq_;
-}
+void NetEqDecodingTest::TearDown() {}
void NetEqDecodingTest::OpenInputFile(const std::string& rtp_file) {
rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
@@ -1366,8 +1365,9 @@
}
void CreateSecondInstance() {
- neteq2_.reset(
- NetEq::Create(config2_, &clock_, CreateBuiltinAudioDecoderFactory()));
+ std::unique_ptr<NetEqFactory> neteq_factory =
+ CreateNetEqFactoryWithCodecs();
+ neteq2_ = neteq_factory->CreateNetEq(config2_, &clock_);
ASSERT_TRUE(neteq2_);
LoadDecoders(neteq2_.get());
}
@@ -1658,7 +1658,7 @@
// We have two packets in the buffer and kAccelerate operation will
// extract 20 ms of data.
- neteq_->GetAudio(&out_frame_, &muted, Operations::kAccelerate);
+ neteq_->GetAudio(&out_frame_, &muted, NetEq::Operation::kAccelerate);
// Check jitter buffer delay.
NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index 713bfb6..967deea 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -26,7 +26,7 @@
int Normal::Process(const int16_t* input,
size_t length,
- Modes last_mode,
+ NetEq::Mode last_mode,
AudioMultiVector* output) {
if (length == 0) {
// Nothing to process.
@@ -52,7 +52,7 @@
// Check if last RecOut call resulted in an Expand. If so, we have to take
// care of some cross-fading and unmuting.
- if (last_mode == kModeExpand) {
+ if (last_mode == NetEq::Mode::kExpand) {
// Generate interpolation data using Expand.
// First, set Expand parameters to appropriate values.
expand_->SetParametersForNormalAfterExpand();
@@ -144,7 +144,7 @@
RTC_DCHECK_GT(win_up_Q14,
(1 << 14) - 32); // Worst case rouding is a length of 34
}
- } else if (last_mode == kModeRfc3389Cng) {
+ } else if (last_mode == NetEq::Mode::kRfc3389Cng) {
RTC_DCHECK_EQ(output->Channels(), 1); // Not adapted for multi-channel yet.
static const size_t kCngLength = 48;
RTC_DCHECK_LE(8 * fs_mult, kCngLength);
diff --git a/modules/audio_coding/neteq/normal.h b/modules/audio_coding/neteq/normal.h
index 2059c5a..d8c13e6 100644
--- a/modules/audio_coding/neteq/normal.h
+++ b/modules/audio_coding/neteq/normal.h
@@ -14,7 +14,7 @@
#include <stdint.h>
#include <string.h> // Access to size_t.
-#include "modules/audio_coding/neteq/defines.h"
+#include "api/neteq/neteq.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -54,7 +54,7 @@
// GetAudio call (i.e., not the current one).
int Process(const int16_t* input,
size_t length,
- Modes last_mode,
+ NetEq::Mode last_mode,
AudioMultiVector* output);
private:
diff --git a/modules/audio_coding/neteq/normal_unittest.cc b/modules/audio_coding/neteq/normal_unittest.cc
index 222fcaa..36751f8 100644
--- a/modules/audio_coding/neteq/normal_unittest.cc
+++ b/modules/audio_coding/neteq/normal_unittest.cc
@@ -70,7 +70,7 @@
AudioMultiVector output(channels);
// Zero input length.
- EXPECT_EQ(0, normal.Process(input, 0, kModeExpand, &output));
+ EXPECT_EQ(0, normal.Process(input, 0, NetEq::Mode::kExpand, &output));
EXPECT_EQ(0u, output.Size());
// Try to make energy_length >> scaling = 0;
@@ -82,8 +82,8 @@
// will be zero, and scaling will be >= 6. Thus, energy_length >> scaling = 0,
// and using this as a denominator would lead to problems.
int input_size_samples = 63;
- EXPECT_EQ(input_size_samples,
- normal.Process(input, input_size_samples, kModeExpand, &output));
+ EXPECT_EQ(input_size_samples, normal.Process(input, input_size_samples,
+ NetEq::Mode::kExpand, &output));
EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope.
@@ -106,7 +106,7 @@
// Let the number of samples be one sample less than 80 samples per channel.
size_t input_len = 80 * channels - 1;
- EXPECT_EQ(0, normal.Process(input, input_len, kModeExpand, &output));
+ EXPECT_EQ(0, normal.Process(input, input_len, NetEq::Mode::kExpand, &output));
EXPECT_EQ(0u, output.Size());
EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
@@ -132,8 +132,9 @@
EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
EXPECT_CALL(expand, Process(_)).WillOnce(Invoke(ExpandProcess120ms));
EXPECT_CALL(expand, Reset());
- EXPECT_EQ(static_cast<int>(kPacketsizeBytes),
- normal.Process(input, kPacketsizeBytes, kModeExpand, &output));
+ EXPECT_EQ(
+ static_cast<int>(kPacketsizeBytes),
+ normal.Process(input, kPacketsizeBytes, NetEq::Mode::kExpand, &output));
EXPECT_EQ(kPacketsizeBytes, output.Size());
diff --git a/modules/audio_coding/neteq/packet.h b/modules/audio_coding/neteq/packet.h
index 238e769..4455494 100644
--- a/modules/audio_coding/neteq/packet.h
+++ b/modules/audio_coding/neteq/packet.h
@@ -17,8 +17,8 @@
#include <memory>
#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/tick_timer.h"
#include "api/rtp_packet_info.h"
-#include "modules/audio_coding/neteq/tick_timer.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
diff --git a/modules/audio_coding/neteq/packet_buffer.cc b/modules/audio_coding/neteq/packet_buffer.cc
index 540d279..059308f 100644
--- a/modules/audio_coding/neteq/packet_buffer.cc
+++ b/modules/audio_coding/neteq/packet_buffer.cc
@@ -21,9 +21,9 @@
#include <utility>
#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/decoder_database.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
-#include "modules/audio_coding/neteq/tick_timer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc
index 0b638bf..0aff955 100644
--- a/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -15,10 +15,10 @@
#include <memory>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
#include "modules/audio_coding/neteq/packet.h"
-#include "modules/audio_coding/neteq/tick_timer.h"
#include "test/gmock.h"
#include "test/gtest.h"
diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h
index a438811..38e463c 100644
--- a/modules/audio_coding/neteq/statistics_calculator.h
+++ b/modules/audio_coding/neteq/statistics_calculator.h
@@ -14,7 +14,7 @@
#include <deque>
#include <string>
-#include "modules/audio_coding/neteq/include/neteq.h"
+#include "api/neteq/neteq.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/tick_timer.cc b/modules/audio_coding/neteq/tick_timer.cc
deleted file mode 100644
index 17f83b1..0000000
--- a/modules/audio_coding/neteq/tick_timer.cc
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- * Copyright (c) 2016 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/tick_timer.h"
-
-namespace webrtc {
-
-TickTimer::Stopwatch::Stopwatch(const TickTimer& ticktimer)
- : ticktimer_(ticktimer), starttick_(ticktimer.ticks()) {}
-
-TickTimer::Countdown::Countdown(const TickTimer& ticktimer,
- uint64_t ticks_to_count)
- : stopwatch_(ticktimer.GetNewStopwatch()),
- ticks_to_count_(ticks_to_count) {}
-
-TickTimer::Countdown::~Countdown() = default;
-
-} // namespace webrtc
diff --git a/modules/audio_coding/neteq/tick_timer.h b/modules/audio_coding/neteq/tick_timer.h
deleted file mode 100644
index 2504ce3..0000000
--- a/modules/audio_coding/neteq/tick_timer.h
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright (c) 2016 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_TICK_TIMER_H_
-#define MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
-
-#include <stdint.h>
-
-#include <memory>
-
-#include "rtc_base/checks.h"
-#include "rtc_base/constructor_magic.h"
-
-namespace webrtc {
-
-// Implements a time counter. The counter is advanced with the Increment()
-// methods, and is queried with the ticks() accessor. It is assumed that one
-// "tick" och the counter corresponds to 10 ms.
-// A TickTimer object can provide two types of associated time-measuring
-// objects: Stopwatch and Countdown.
-class TickTimer {
- public:
- // Stopwatch measures time elapsed since it was started, by querying the
- // associated TickTimer for the current time. The intended use is to request a
- // new Stopwatch object from a TickTimer object with the GetNewStopwatch()
- // method. Note: since the Stopwatch object contains a reference to the
- // TickTimer it is associated with, it cannot outlive the TickTimer.
- class Stopwatch {
- public:
- explicit Stopwatch(const TickTimer& ticktimer);
-
- uint64_t ElapsedTicks() const { return ticktimer_.ticks() - starttick_; }
-
- uint64_t ElapsedMs() const {
- const uint64_t elapsed_ticks = ticktimer_.ticks() - starttick_;
- const int ms_per_tick = ticktimer_.ms_per_tick();
- return elapsed_ticks < UINT64_MAX / ms_per_tick
- ? elapsed_ticks * ms_per_tick
- : UINT64_MAX;
- }
-
- private:
- const TickTimer& ticktimer_;
- const uint64_t starttick_;
- };
-
- // Countdown counts down from a given start value with each tick of the
- // associated TickTimer, until zero is reached. The Finished() method will
- // return true if zero has been reached, false otherwise. The intended use is
- // to request a new Countdown object from a TickTimer object with the
- // GetNewCountdown() method. Note: since the Countdown object contains a
- // reference to the TickTimer it is associated with, it cannot outlive the
- // TickTimer.
- class Countdown {
- public:
- Countdown(const TickTimer& ticktimer, uint64_t ticks_to_count);
-
- ~Countdown();
-
- bool Finished() const {
- return stopwatch_->ElapsedTicks() >= ticks_to_count_;
- }
-
- private:
- const std::unique_ptr<Stopwatch> stopwatch_;
- const uint64_t ticks_to_count_;
- };
-
- TickTimer() : TickTimer(10) {}
- explicit TickTimer(int ms_per_tick) : ms_per_tick_(ms_per_tick) {
- RTC_DCHECK_GT(ms_per_tick_, 0);
- }
-
- void Increment() { ++ticks_; }
-
- // Mainly intended for testing.
- void Increment(uint64_t x) { ticks_ += x; }
-
- uint64_t ticks() const { return ticks_; }
-
- int ms_per_tick() const { return ms_per_tick_; }
-
- // Returns a new Stopwatch object, based on the current TickTimer. Note that
- // the new Stopwatch object contains a reference to the current TickTimer,
- // and must therefore not outlive the TickTimer.
- std::unique_ptr<Stopwatch> GetNewStopwatch() const {
- return std::unique_ptr<Stopwatch>(new Stopwatch(*this));
- }
-
- // Returns a new Countdown object, based on the current TickTimer. Note that
- // the new Countdown object contains a reference to the current TickTimer,
- // and must therefore not outlive the TickTimer.
- std::unique_ptr<Countdown> GetNewCountdown(uint64_t ticks_to_count) const {
- return std::unique_ptr<Countdown>(new Countdown(*this, ticks_to_count));
- }
-
- private:
- uint64_t ticks_ = 0;
- const int ms_per_tick_;
- RTC_DISALLOW_COPY_AND_ASSIGN(TickTimer);
-};
-
-} // namespace webrtc
-#endif // MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
diff --git a/modules/audio_coding/neteq/tick_timer_unittest.cc b/modules/audio_coding/neteq/tick_timer_unittest.cc
deleted file mode 100644
index c501329..0000000
--- a/modules/audio_coding/neteq/tick_timer_unittest.cc
+++ /dev/null
@@ -1,135 +0,0 @@
-/*
- * Copyright (c) 2016 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/tick_timer.h"
-
-#include <memory>
-
-#include "test/gmock.h"
-#include "test/gtest.h"
-
-namespace webrtc {
-
-// Verify that the default value for ms_per_tick is 10.
-TEST(TickTimer, DefaultMsPerTick) {
- TickTimer tt;
- EXPECT_EQ(10, tt.ms_per_tick());
-}
-
-TEST(TickTimer, CustomMsPerTick) {
- TickTimer tt(17);
- EXPECT_EQ(17, tt.ms_per_tick());
-}
-
-TEST(TickTimer, Increment) {
- TickTimer tt;
- EXPECT_EQ(0u, tt.ticks());
- tt.Increment();
- EXPECT_EQ(1u, tt.ticks());
-
- for (int i = 0; i < 17; ++i) {
- tt.Increment();
- }
- EXPECT_EQ(18u, tt.ticks());
-
- tt.Increment(17);
- EXPECT_EQ(35u, tt.ticks());
-}
-
-TEST(TickTimer, WrapAround) {
- TickTimer tt;
- tt.Increment(UINT64_MAX);
- EXPECT_EQ(UINT64_MAX, tt.ticks());
- tt.Increment();
- EXPECT_EQ(0u, tt.ticks());
-}
-
-TEST(TickTimer, Stopwatch) {
- TickTimer tt;
- // Increment it a "random" number of steps.
- tt.Increment(17);
-
- std::unique_ptr<TickTimer::Stopwatch> sw = tt.GetNewStopwatch();
- ASSERT_TRUE(sw);
-
- EXPECT_EQ(0u, sw->ElapsedTicks()); // Starts at zero.
- EXPECT_EQ(0u, sw->ElapsedMs());
- tt.Increment();
- EXPECT_EQ(1u, sw->ElapsedTicks()); // Increases with the TickTimer.
- EXPECT_EQ(10u, sw->ElapsedMs());
-}
-
-TEST(TickTimer, StopwatchWrapAround) {
- TickTimer tt;
- tt.Increment(UINT64_MAX);
-
- std::unique_ptr<TickTimer::Stopwatch> sw = tt.GetNewStopwatch();
- ASSERT_TRUE(sw);
-
- tt.Increment();
- EXPECT_EQ(0u, tt.ticks());
- EXPECT_EQ(1u, sw->ElapsedTicks());
- EXPECT_EQ(10u, sw->ElapsedMs());
-
- tt.Increment();
- EXPECT_EQ(1u, tt.ticks());
- EXPECT_EQ(2u, sw->ElapsedTicks());
- EXPECT_EQ(20u, sw->ElapsedMs());
-}
-
-TEST(TickTimer, StopwatchMsOverflow) {
- TickTimer tt;
- std::unique_ptr<TickTimer::Stopwatch> sw = tt.GetNewStopwatch();
- ASSERT_TRUE(sw);
-
- tt.Increment(UINT64_MAX / 10);
- EXPECT_EQ(UINT64_MAX, sw->ElapsedMs());
-
- tt.Increment();
- EXPECT_EQ(UINT64_MAX, sw->ElapsedMs());
-
- tt.Increment(UINT64_MAX - tt.ticks());
- EXPECT_EQ(UINT64_MAX, tt.ticks());
- EXPECT_EQ(UINT64_MAX, sw->ElapsedMs());
-}
-
-TEST(TickTimer, StopwatchWithCustomTicktime) {
- const int kMsPerTick = 17;
- TickTimer tt(kMsPerTick);
- std::unique_ptr<TickTimer::Stopwatch> sw = tt.GetNewStopwatch();
- ASSERT_TRUE(sw);
-
- EXPECT_EQ(0u, sw->ElapsedMs());
- tt.Increment();
- EXPECT_EQ(static_cast<uint64_t>(kMsPerTick), sw->ElapsedMs());
-}
-
-TEST(TickTimer, Countdown) {
- TickTimer tt;
- // Increment it a "random" number of steps.
- tt.Increment(4711);
-
- std::unique_ptr<TickTimer::Countdown> cd = tt.GetNewCountdown(17);
- ASSERT_TRUE(cd);
-
- EXPECT_FALSE(cd->Finished());
- tt.Increment();
- EXPECT_FALSE(cd->Finished());
-
- tt.Increment(16); // Total increment is now 17.
- EXPECT_TRUE(cd->Finished());
-
- // Further increments do not change the state.
- tt.Increment();
- EXPECT_TRUE(cd->Finished());
- tt.Increment(1234);
- EXPECT_TRUE(cd->Finished());
-}
-} // namespace webrtc
diff --git a/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index dfd61d8..d963903 100644
--- a/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -11,9 +11,9 @@
#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
#include "api/audio/audio_frame.h"
-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "api/test/neteq_factory_with_codecs.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "rtc_base/checks.h"
@@ -40,8 +40,8 @@
NetEq::Config config;
config.sample_rate_hz = kSampRateHz;
webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
- NetEq* neteq =
- NetEq::Create(config, clock, CreateBuiltinAudioDecoderFactory());
+ std::unique_ptr<NetEqFactory> neteq_factory = CreateNetEqFactoryWithCodecs();
+ auto neteq = neteq_factory->CreateNetEq(config, clock);
// Register decoder in |neteq|.
if (!neteq->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", kSampRateHz, 1)))
@@ -119,7 +119,6 @@
}
}
int64_t end_time_ms = clock->TimeInMilliseconds();
- delete neteq;
return end_time_ms - start_time_ms;
}
diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index 3b3d337..ba53954 100644
--- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -15,6 +15,8 @@
#include <cmath>
#include "absl/flags/flag.h"
+#include "api/neteq/custom_neteq_factory.h"
+#include "api/neteq/default_neteq_controller_factory.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "modules/audio_coding/neteq/tools/output_audio_file.h"
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
@@ -87,6 +89,19 @@
namespace webrtc {
namespace test {
+namespace {
+
+std::unique_ptr<NetEq> CreateNetEq(
+ const NetEq::Config& config,
+ Clock* clock,
+ const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+ CustomNetEqFactory neteq_factory(
+ decoder_factory, std::make_unique<DefaultNetEqControllerFactory>());
+ return neteq_factory.CreateNetEq(config, clock);
+}
+
+} // namespace
+
const uint8_t kPayloadType = 95;
const int kOutputSizeMs = 10;
const int kInitSeed = 0x12345678;
@@ -228,8 +243,7 @@
NetEq::Config config;
config.sample_rate_hz = out_sampling_khz_ * 1000;
- neteq_.reset(
- NetEq::Create(config, Clock::GetRealTimeClock(), decoder_factory));
+ neteq_ = CreateNetEq(config, Clock::GetRealTimeClock(), decoder_factory);
max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
in_data_.reset(new int16_t[in_size_samples_ * channels_]);
}
diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.h b/modules/audio_coding/neteq/tools/neteq_quality_test.h
index 8bf5e5a..11d347a 100644
--- a/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -15,7 +15,7 @@
#include <memory>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "modules/audio_coding/neteq/include/neteq.h"
+#include "api/neteq/neteq.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc
index c4fdef0..50d8ba1 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -13,6 +13,8 @@
#include <iomanip>
#include <iostream>
+#include "api/neteq/custom_neteq_factory.h"
+#include "api/neteq/default_neteq_controller_factory.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "system_wrappers/include/clock.h"
@@ -20,23 +22,32 @@
namespace test {
namespace {
-absl::optional<Operations> ActionToOperations(
+absl::optional<NetEq::Operation> ActionToOperations(
absl::optional<NetEqSimulator::Action> a) {
if (!a) {
return absl::nullopt;
}
switch (*a) {
case NetEqSimulator::Action::kAccelerate:
- return absl::make_optional(kAccelerate);
+ return absl::make_optional(NetEq::Operation::kAccelerate);
case NetEqSimulator::Action::kExpand:
- return absl::make_optional(kExpand);
+ return absl::make_optional(NetEq::Operation::kExpand);
case NetEqSimulator::Action::kNormal:
- return absl::make_optional(kNormal);
+ return absl::make_optional(NetEq::Operation::kNormal);
case NetEqSimulator::Action::kPreemptiveExpand:
- return absl::make_optional(kPreemptiveExpand);
+ return absl::make_optional(NetEq::Operation::kPreemptiveExpand);
}
}
+std::unique_ptr<NetEq> CreateNetEq(
+ const NetEq::Config& config,
+ Clock* clock,
+ const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+ CustomNetEqFactory neteq_factory(
+ decoder_factory, std::make_unique<DefaultNetEqControllerFactory>());
+ return neteq_factory.CreateNetEq(config, clock);
+}
+
} // namespace
void DefaultNetEqTestErrorCallback::OnInsertPacketError(
@@ -59,7 +70,7 @@
std::unique_ptr<AudioSink> output,
Callbacks callbacks)
: clock_(0),
- neteq_(NetEq::Create(config, &clock_, decoder_factory)),
+ neteq_(CreateNetEq(config, &clock_, decoder_factory)),
input_(std::move(input)),
output_(std::move(output)),
callbacks_(callbacks),
diff --git a/modules/audio_coding/neteq/tools/neteq_test.h b/modules/audio_coding/neteq/tools/neteq_test.h
index 3cf105c..c6dbca7 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_test.h
@@ -19,8 +19,8 @@
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
#include "api/test/neteq_simulator.h"
-#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/neteq_input.h"
#include "system_wrappers/include/clock.h"
diff --git a/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/modules/audio_coding/neteq/tools/neteq_test_factory.cc
index c50f7d9..55bc46d 100644
--- a/modules/audio_coding/neteq/tools/neteq_test_factory.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test_factory.cc
@@ -23,7 +23,7 @@
#include <utility>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "modules/audio_coding/neteq/include/neteq.h"
+#include "api/neteq/neteq.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
#include "modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h"
