Use backticks not vertical bars to denote variables in comments for /modules/audio_coding
Bug: webrtc:12338
Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34621}
diff --git a/modules/audio_coding/acm2/acm_receive_test.h b/modules/audio_coding/acm2/acm_receive_test.h
index 043092c..6349c63 100644
--- a/modules/audio_coding/acm2/acm_receive_test.h
+++ b/modules/audio_coding/acm2/acm_receive_test.h
@@ -71,8 +71,8 @@
RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
};
-// This test toggles the output frequency every |toggle_period_ms|. The test
-// starts with |output_freq_hz_1|. Except for the toggling, it does the same
+// This test toggles the output frequency every `toggle_period_ms`. The test
+// starts with `output_freq_hz_1`. Except for the toggling, it does the same
// thing as AcmReceiveTestOldApi.
class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
public:
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 80cb3c5..6d9211c 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -131,7 +131,7 @@
/*num_channels=*/format->num_channels,
/*sdp_format=*/std::move(format->sdp_format)};
}
- } // |mutex_| is released.
+ } // `mutex_` is released.
if (neteq_->InsertPacket(rtp_header, incoming_payload) < 0) {
RTC_LOG(LERROR) << "AcmReceiver::InsertPacket "
@@ -201,7 +201,7 @@
// We might end up here ONLY if codec is changed.
}
- // Store current audio in |last_audio_buffer_| for next time.
+ // Store current audio in `last_audio_buffer_` for next time.
memcpy(last_audio_buffer_.get(), audio_frame->data(),
sizeof(int16_t) * audio_frame->samples_per_channel_ *
audio_frame->num_channels_);
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 19dc577..9963603 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -177,9 +177,9 @@
// enabled then the maximum NACK list size is modified accordingly.
//
// If the sequence number of last received packet is N, the sequence numbers
- // of NACK list are in the range of [N - |max_nack_list_size|, N).
+ // of NACK list are in the range of [N - `max_nack_list_size`, N).
//
- // |max_nack_list_size| should be positive (none zero) and less than or
+ // `max_nack_list_size` should be positive (none zero) and less than or
// equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1
// is returned. 0 is returned at success.
//
@@ -189,12 +189,12 @@
void DisableNack();
//
- // Get a list of packets to be retransmitted. |round_trip_time_ms| is an
+ // Get a list of packets to be retransmitted. `round_trip_time_ms` is an
// estimate of the round-trip-time (in milliseconds). Missing packets which
// will be playout in a shorter time than the round-trip-time (with respect
// to the time this API is called) will not be included in the list.
//
- // Negative |round_trip_time_ms| results is an error message and empty list
+ // Negative `round_trip_time_ms` results is an error message and empty list
// is returned.
//
std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index b5c0c3b..d629139 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -125,7 +125,7 @@
int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
- // TODO(bugs.webrtc.org/10739): change |absolute_capture_timestamp_ms| to
+ // TODO(bugs.webrtc.org/10739): change `absolute_capture_timestamp_ms` to
// int64_t when it always receives a valid value.
int Encode(const InputData& input_data,
absl::optional<int64_t> absolute_capture_timestamp_ms)
@@ -141,8 +141,8 @@
//
// in_frame: input audio-frame
// ptr_out: pointer to output audio_frame. If no preprocessing is required
- // |ptr_out| will be pointing to |in_frame|, otherwise pointing to
- // |preprocess_frame_|.
+ // `ptr_out` will be pointing to `in_frame`, otherwise pointing to
+ // `preprocess_frame_`.
//
// Return value:
// -1: if encountering an error.
@@ -152,7 +152,7 @@
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
// Change required states after starting to receive the codec corresponding
- // to |index|.
+ // to `index`.
int UpdateUponReceivingCodec(int index);
mutable Mutex acm_mutex_;
@@ -397,7 +397,7 @@
// output data if needed.
ReMixFrame(*ptr_frame, current_num_channels, &input_data->buffer);
- // For pushing data to primary, point the |ptr_audio| to correct buffer.
+ // For pushing data to primary, point the `ptr_audio` to correct buffer.
input_data->audio = input_data->buffer.data();
RTC_DCHECK_GE(input_data->buffer.size(),
input_data->length_per_channel * input_data->audio_channel);
@@ -414,7 +414,7 @@
// encoder is mono and input is stereo. In case of dual-streaming, both
// encoders has to be mono for down-mix to take place.
// |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing
-// is required, |*ptr_out| points to |in_frame|.
+// is required, |*ptr_out| points to `in_frame`.
// TODO(yujo): Make this more efficient for muted frames.
int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
const AudioFrame** ptr_out) {
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 7465456..a0a8854 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -342,7 +342,7 @@
// Introduce this class to set different expectations on the number of encoded
// bytes. This class expects all encoded packets to be 9 bytes (matching one
-// CNG SID frame) or 0 bytes. This test depends on |input_frame_| containing
+// CNG SID frame) or 0 bytes. This test depends on `input_frame_` containing
// (near-)zero values. It also introduces a way to register comfort noise with
// a custom payload type.
class AudioCodingModuleTestWithComfortNoiseOldApi
@@ -593,7 +593,7 @@
InsertAudio();
ASSERT_LT(loop_counter++, 10);
}
- // Set |last_packet_number_| to one less that |num_calls| so that the packet
+ // Set `last_packet_number_` to one less that `num_calls` so that the packet
// will be fetched in the next InsertPacket() call.
last_packet_number_ = packet_cb_.num_calls() - 1;
@@ -617,7 +617,7 @@
if (num_calls > last_packet_number_) {
// Get the new payload out from the callback handler.
// Note that since we swap buffers here instead of directly inserting
- // a pointer to the data in |packet_cb_|, we avoid locking the callback
+ // a pointer to the data in `packet_cb_`, we avoid locking the callback
// for the duration of the IncomingPacket() call.
packet_cb_.SwapBuffers(&last_payload_vec_);
ASSERT_GT(last_payload_vec_.size(), 0u);
@@ -1140,8 +1140,8 @@
// Sets up the test::AcmSendTest object. Returns true on success, otherwise
// false.
bool SetUpSender(std::string input_file_name, int source_rate) {
- // Note that |audio_source_| will loop forever. The test duration is set
- // explicitly by |kTestDurationMs|.
+ // Note that `audio_source_` will loop forever. The test duration is set
+ // explicitly by `kTestDurationMs`.
audio_source_.reset(new test::InputAudioFile(input_file_name));
send_test_.reset(new test::AcmSendTestOldApi(audio_source_.get(),
source_rate, kTestDurationMs));
@@ -1243,7 +1243,7 @@
VerifyPacket(packet.get());
// TODO(henrik.lundin) Save the packet to file as well.
- // Pass it on to the caller. The caller becomes the owner of |packet|.
+ // Pass it on to the caller. The caller becomes the owner of `packet`.
return packet;
}
@@ -1631,8 +1631,8 @@
bool SetUpSender() {
const std::string input_file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
- // Note that |audio_source_| will loop forever. The test duration is set
- // explicitly by |kTestDurationMs|.
+ // Note that `audio_source_` will loop forever. The test duration is set
+ // explicitly by `kTestDurationMs`.
audio_source_.reset(new test::InputAudioFile(input_file_name));
static const int kSourceRateHz = 32000;
send_test_.reset(new test::AcmSendTestOldApi(
@@ -1859,7 +1859,7 @@
// This test fixture is implemented to run ACM and change the desired output
// frequency during the call. The input packets are simply PCM16b-wb encoded
-// payloads with a constant value of |kSampleValue|. The test fixture itself
+// payloads with a constant value of `kSampleValue`. The test fixture itself
// acts as PacketSource in between the receive test class and the constant-
// payload packet source class. The output is both written to file, and analyzed
// in this test fixture.
diff --git a/modules/audio_coding/acm2/call_statistics.cc b/modules/audio_coding/acm2/call_statistics.cc
index e97e529..0aad594 100644
--- a/modules/audio_coding/acm2/call_statistics.cc
+++ b/modules/audio_coding/acm2/call_statistics.cc
@@ -44,7 +44,7 @@
break;
}
case AudioFrame::kUndefined: {
- // If the audio is decoded by NetEq, |kUndefined| is not an option.
+ // If the audio is decoded by NetEq, `kUndefined` is not an option.
RTC_NOTREACHED();
}
}
diff --git a/modules/audio_coding/acm2/call_statistics.h b/modules/audio_coding/acm2/call_statistics.h
index 5d94ac4..a2db2a29 100644
--- a/modules/audio_coding/acm2/call_statistics.h
+++ b/modules/audio_coding/acm2/call_statistics.h
@@ -36,8 +36,8 @@
CallStatistics() {}
~CallStatistics() {}
- // Call this method to indicate that NetEq engaged in decoding. |speech_type|
- // is the audio-type according to NetEq, and |muted| indicates if the decoded
+ // Call this method to indicate that NetEq engaged in decoding. `speech_type`
+ // is the audio-type according to NetEq, and `muted` indicates if the decoded
// frame was produced in muted state.
void DecodedByNetEq(AudioFrame::SpeechType speech_type, bool muted);
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
index 40c8659..88ca38d 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
@@ -50,7 +50,7 @@
}
void BitrateController::MakeDecision(AudioEncoderRuntimeConfig* config) {
- // Decision on |bitrate_bps| should not have been made.
+ // Decision on `bitrate_bps` should not have been made.
RTC_DCHECK(!config->bitrate_bps);
if (target_audio_bitrate_bps_ && overhead_bytes_per_packet_) {
if (config->frame_length_ms)
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller.cc b/modules/audio_coding/audio_network_adaptor/channel_controller.cc
index 2f5af67..2ef2f4c 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/channel_controller.cc
@@ -28,7 +28,7 @@
ChannelController::ChannelController(const Config& config)
: config_(config), channels_to_encode_(config_.intial_channels_to_encode) {
RTC_DCHECK_GT(config_.intial_channels_to_encode, 0lu);
- // Currently, we require |intial_channels_to_encode| to be <= 2.
+ // Currently, we require `intial_channels_to_encode` to be <= 2.
RTC_DCHECK_LE(config_.intial_channels_to_encode, 2lu);
RTC_DCHECK_GE(config_.num_encoder_channels,
config_.intial_channels_to_encode);
@@ -43,7 +43,7 @@
}
void ChannelController::MakeDecision(AudioEncoderRuntimeConfig* config) {
- // Decision on |num_channels| should not have been made.
+ // Decision on `num_channels` should not have been made.
RTC_DCHECK(!config->num_channels);
if (uplink_bandwidth_bps_) {
diff --git a/modules/audio_coding/audio_network_adaptor/config.proto b/modules/audio_coding/audio_network_adaptor/config.proto
index 347372e..4f8b2c7 100644
--- a/modules/audio_coding/audio_network_adaptor/config.proto
+++ b/modules/audio_coding/audio_network_adaptor/config.proto
@@ -23,8 +23,8 @@
optional float high_bandwidth_packet_loss = 4;
}
- // |fec_enabling_threshold| defines a curve, above which FEC should be
- // enabled. |fec_disabling_threshold| defines a curve, under which FEC
+ // `fec_enabling_threshold` defines a curve, above which FEC should be
+ // enabled. `fec_disabling_threshold` defines a curve, under which FEC
// should be disabled. See below
//
// packet-loss ^ | |
@@ -36,7 +36,7 @@
optional Threshold fec_enabling_threshold = 1;
optional Threshold fec_disabling_threshold = 2;
- // |time_constant_ms| is the time constant for an exponential filter, which
+ // `time_constant_ms` is the time constant for an exponential filter, which
// is used for smoothing the packet loss fraction.
optional int32 time_constant_ms = 3;
}
@@ -62,8 +62,8 @@
optional float high_bandwidth_recoverable_packet_loss = 4;
}
- // |fec_enabling_threshold| defines a curve, above which FEC should be
- // enabled. |fec_disabling_threshold| defines a curve, under which FEC
+ // `fec_enabling_threshold` defines a curve, above which FEC should be
+ // enabled. `fec_disabling_threshold` defines a curve, under which FEC
// should be disabled. See below
//
// packet-loss ^ | |
@@ -122,7 +122,7 @@
// FrameLengthControllerV2 chooses the frame length by taking the target
// bitrate and subtracting the overhead bitrate to obtain the remaining
// bitrate for the payload. The chosen frame length is the shortest possible
- // where the payload bitrate is more than |min_payload_bitrate_bps|.
+ // where the payload bitrate is more than `min_payload_bitrate_bps`.
optional int32 min_payload_bitrate_bps = 1;
// If true, uses the stable target bitrate to decide the frame length. This
@@ -158,17 +158,17 @@
message Controller {
message ScoringPoint {
- // |ScoringPoint| is a subspace of network condition. It is used for
+ // `ScoringPoint` is a subspace of network condition. It is used for
// comparing the significance of controllers.
optional int32 uplink_bandwidth_bps = 1;
optional float uplink_packet_loss_fraction = 2;
}
- // The distance from |scoring_point| to a given network condition defines
+ // The distance from `scoring_point` to a given network condition defines
// the significance of this controller with respect that network condition.
// Shorter distance means higher significance. The significances of
// controllers determine their order in the processing pipeline. Controllers
- // without |scoring_point| follow their default order in
+ // without `scoring_point` follow their default order in
// |ControllerManager::controllers|.
optional ScoringPoint scoring_point = 1;
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index 415b9fc..6708bc0 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -373,14 +373,14 @@
config_.min_reordering_squared_distance)
return sorted_controllers_;
- // Sort controllers according to the distances of |scoring_point| to the
+ // Sort controllers according to the distances of `scoring_point` to the
// scoring points of controllers.
//
// A controller that does not associate with any scoring point
// are treated as if
// 1) they are less important than any controller that has a scoring point,
// 2) they are equally important to any controller that has no scoring point,
- // and their relative order will follow |default_sorted_controllers_|.
+ // and their relative order will follow `default_sorted_controllers_`.
std::vector<Controller*> sorted_controllers(default_sorted_controllers_);
std::stable_sort(
sorted_controllers.begin(), sorted_controllers.end(),
@@ -430,7 +430,7 @@
}
float NormalizePacketLossFraction(float uplink_packet_loss_fraction) {
- // |uplink_packet_loss_fraction| is seldom larger than 0.3, so we scale it up
+ // `uplink_packet_loss_fraction` is seldom larger than 0.3, so we scale it up
// by 3.3333f.
return std::min(uplink_packet_loss_fraction * 3.3333f, 1.0f);
}
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.h b/modules/audio_coding/audio_network_adaptor/controller_manager.h
index f46450d..c168ebc 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.h
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.h
@@ -111,7 +111,7 @@
std::vector<Controller*> sorted_controllers_;
- // |scoring_points_| saves the scoring points of various
+ // `scoring_points_` saves the scoring points of various
// controllers.
std::map<const Controller*, ScoringPoint> controller_scoring_points_;
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
index c71bbc9..7b7ced9 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
@@ -43,7 +43,7 @@
constexpr int kFactor = 100;
constexpr float kMinReorderingSquareDistance = 1.0f / kFactor / kFactor;
-// |kMinUplinkBandwidthBps| and |kMaxUplinkBandwidthBps| are copied from
+// `kMinUplinkBandwidthBps` and `kMaxUplinkBandwidthBps` are copied from
// controller_manager.cc
constexpr int kMinUplinkBandwidthBps = 0;
constexpr int kMaxUplinkBandwidthBps = 120000;
@@ -82,7 +82,7 @@
return states;
}
-// |expected_order| contains the expected indices of all controllers in the
+// `expected_order` contains the expected indices of all controllers in the
// vector of controllers returned by GetSortedControllers(). A negative index
// means that we do not care about its exact place, but we do check that it
// exists in the vector.
@@ -112,8 +112,8 @@
TEST(ControllerManagerTest, GetControllersReturnAllControllers) {
auto states = CreateControllerManager();
auto check = states.controller_manager->GetControllers();
- // Verify that controllers in |check| are one-to-one mapped to those in
- // |mock_controllers_|.
+ // Verify that controllers in `check` are one-to-one mapped to those in
+ // `mock_controllers_`.
EXPECT_EQ(states.mock_controllers.size(), check.size());
for (auto& controller : check)
EXPECT_NE(states.mock_controllers.end(),
@@ -123,7 +123,7 @@
TEST(ControllerManagerTest, ControllersInDefaultOrderOnEmptyNetworkMetrics) {
auto states = CreateControllerManager();
- // |network_metrics| are empty, and the controllers are supposed to follow the
+ // `network_metrics` are empty, and the controllers are supposed to follow the
// default order.
CheckControllersOrder(&states, absl::nullopt, absl::nullopt, {0, 1, 2, 3});
}
@@ -304,7 +304,7 @@
for (size_t i = 0; i < controllers.size(); ++i) {
AudioEncoderRuntimeConfig encoder_config;
- // We check the order of |controllers| by judging their decisions.
+ // We check the order of `controllers` by judging their decisions.
controllers[i]->MakeDecision(&encoder_config);
// Since controllers are not provided with network metrics, they give the
diff --git a/modules/audio_coding/audio_network_adaptor/debug_dump.proto b/modules/audio_coding/audio_network_adaptor/debug_dump.proto
index 93b31c3..3aa6a50 100644
--- a/modules/audio_coding/audio_network_adaptor/debug_dump.proto
+++ b/modules/audio_coding/audio_network_adaptor/debug_dump.proto
@@ -21,7 +21,7 @@
optional bool enable_fec = 4;
optional bool enable_dtx = 5;
// Some encoders can encode fewer channels than the actual input to make
- // better use of the bandwidth. |num_channels| sets the number of channels
+ // better use of the bandwidth. `num_channels` sets the number of channels
// to encode.
optional uint32 num_channels = 6;
}
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
index 48384c9..b0a7d5d 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
@@ -33,7 +33,7 @@
}
void DtxController::MakeDecision(AudioEncoderRuntimeConfig* config) {
- // Decision on |enable_dtx| should not have been made.
+ // Decision on `enable_dtx` should not have been made.
RTC_DCHECK(!config->enable_dtx);
if (uplink_bandwidth_bps_) {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
index 87afe2e..85d235e 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
@@ -25,8 +25,8 @@
class FecControllerPlrBased final : public Controller {
public:
struct Config {
- // |fec_enabling_threshold| defines a curve, above which FEC should be
- // enabled. |fec_disabling_threshold| defines a curve, under which FEC
+ // `fec_enabling_threshold` defines a curve, above which FEC should be
+ // enabled. `fec_disabling_threshold` defines a curve, under which FEC
// should be disabled. See below
//
// packet-loss ^ | |
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
index d95cbce..355431a 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
@@ -100,9 +100,9 @@
}
}
-// Checks that the FEC decision and |uplink_packet_loss_fraction| given by
-// |states->controller->MakeDecision| matches |expected_enable_fec| and
-// |expected_uplink_packet_loss_fraction|, respectively.
+// Checks that the FEC decision and `uplink_packet_loss_fraction` given by
+// |states->controller->MakeDecision| matches `expected_enable_fec` and
+// `expected_uplink_packet_loss_fraction`, respectively.
void CheckDecision(FecControllerPlrBasedTestStates* states,
bool expected_enable_fec,
float expected_uplink_packet_loss_fraction) {
@@ -221,7 +221,7 @@
TEST(FecControllerPlrBasedTest, MaintainFecOffForVeryLowBandwidth) {
auto states = CreateFecControllerPlrBased(false);
- // Below |kEnablingBandwidthLow|, no packet loss fraction can cause FEC to
+ // Below `kEnablingBandwidthLow`, no packet loss fraction can cause FEC to
// turn on.
UpdateNetworkMetrics(&states, kEnablingBandwidthLow - 1, 1.0);
CheckDecision(&states, false, 1.0);
@@ -272,7 +272,7 @@
TEST(FecControllerPlrBasedTest, DisableFecForVeryLowBandwidth) {
auto states = CreateFecControllerPlrBased(true);
- // Below |kEnablingBandwidthLow|, any packet loss fraction can cause FEC to
+ // Below `kEnablingBandwidthLow`, any packet loss fraction can cause FEC to
// turn off.
UpdateNetworkMetrics(&states, kDisablingBandwidthLow - 1, 1.0);
CheckDecision(&states, false, 1.0);
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
index 36e9eb9..c47434f 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
@@ -54,7 +54,7 @@
frame_length_ms_ = std::find(config_.encoder_frame_lengths_ms.begin(),
config_.encoder_frame_lengths_ms.end(),
config_.initial_frame_length_ms);
- // |encoder_frame_lengths_ms| must contain |initial_frame_length_ms|.
+ // `encoder_frame_lengths_ms` must contain `initial_frame_length_ms`.
RTC_DCHECK(frame_length_ms_ != config_.encoder_frame_lengths_ms.end());
}
@@ -71,7 +71,7 @@
}
void FrameLengthController::MakeDecision(AudioEncoderRuntimeConfig* config) {
- // Decision on |frame_length_ms| should not have been made.
+ // Decision on `frame_length_ms` should not have been made.
RTC_DCHECK(!config->frame_length_ms);
if (FrameLengthIncreasingDecision(*config)) {
@@ -99,12 +99,12 @@
bool FrameLengthController::FrameLengthIncreasingDecision(
const AudioEncoderRuntimeConfig& config) {
// Increase frame length if
- // 1. |uplink_bandwidth_bps| is known to be smaller or equal than
- // |min_encoder_bitrate_bps| plus |prevent_overuse_margin_bps| plus the
+ // 1. `uplink_bandwidth_bps` is known to be smaller or equal than
+ // `min_encoder_bitrate_bps` plus `prevent_overuse_margin_bps` plus the
// current overhead rate OR all the following:
// 2. longer frame length is available AND
- // 3. |uplink_bandwidth_bps| is known to be smaller than a threshold AND
- // 4. |uplink_packet_loss_fraction| is known to be smaller than a threshold.
+ // 3. `uplink_bandwidth_bps` is known to be smaller than a threshold AND
+ // 4. `uplink_packet_loss_fraction` is known to be smaller than a threshold.
// Find next frame length to which a criterion is defined to shift from
// current frame length.
@@ -156,12 +156,12 @@
const AudioEncoderRuntimeConfig& config) {
// Decrease frame length if
// 1. shorter frame length is available AND
- // 2. |uplink_bandwidth_bps| is known to be bigger than
- // |min_encoder_bitrate_bps| plus |prevent_overuse_margin_bps| plus the
+ // 2. `uplink_bandwidth_bps` is known to be bigger than
+ // `min_encoder_bitrate_bps` plus `prevent_overuse_margin_bps` plus the
// overhead which would be produced with the shorter frame length AND
// one or more of the followings:
- // 3. |uplink_bandwidth_bps| is known to be larger than a threshold,
- // 4. |uplink_packet_loss_fraction| is known to be larger than a threshold,
+ // 3. `uplink_bandwidth_bps` is known to be larger than a threshold,
+ // 4. `uplink_packet_loss_fraction` is known to be larger than a threshold,
// Find next frame length to which a criterion is defined to shift from
// current frame length.
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
index 0ffa54a..2312393 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
@@ -184,8 +184,8 @@
TEST(FrameLengthControllerTest, IncreaseTo40MsOnMultipleConditions) {
// Increase to 40ms frame length if
- // 1. |uplink_bandwidth_bps| is known to be smaller than a threshold AND
- // 2. |uplink_packet_loss_fraction| is known to be smaller than a threshold
+ // 1. `uplink_bandwidth_bps` is known to be smaller than a threshold AND
+ // 2. `uplink_packet_loss_fraction` is known to be smaller than a threshold
// AND
// 3. FEC is not decided or OFF.
auto controller = CreateController(CreateChangeCriteriaFor20msAnd40ms(),
@@ -206,8 +206,8 @@
TEST(FrameLengthControllerTest, Maintain60MsOnMultipleConditions) {
// Maintain 60ms frame length if
- // 1. |uplink_bandwidth_bps| is at medium level,
- // 2. |uplink_packet_loss_fraction| is at medium,
+ // 1. `uplink_bandwidth_bps` is at medium level,
+ // 2. `uplink_packet_loss_fraction` is at medium,
// 3. FEC is not decided ON.
auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
kDefaultEncoderFrameLengthsMs, 60);
@@ -218,8 +218,8 @@
TEST(FrameLengthControllerTest, IncreaseTo60MsOnMultipleConditions) {
// Increase to 60ms frame length if
- // 1. |uplink_bandwidth_bps| is known to be smaller than a threshold AND
- // 2. |uplink_packet_loss_fraction| is known to be smaller than a threshold
+ // 1. `uplink_bandwidth_bps` is known to be smaller than a threshold AND
+ // 2. `uplink_packet_loss_fraction` is known to be smaller than a threshold
// AND
// 3. FEC is not decided or OFF.
auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
@@ -365,8 +365,8 @@
TEST(FrameLengthControllerTest, From20MsTo120MsOnMultipleConditions) {
// Increase to 120ms frame length if
- // 1. |uplink_bandwidth_bps| is known to be smaller than a threshold AND
- // 2. |uplink_packet_loss_fraction| is known to be smaller than a threshold.
+ // 1. `uplink_bandwidth_bps` is known to be smaller than a threshold AND
+ // 2. `uplink_packet_loss_fraction` is known to be smaller than a threshold.
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
kDefaultEncoderFrameLengthsMs, 20);
// It takes two steps for frame length to go from 20ms to 120ms.
diff --git a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
index 94e8ed9..bd16292 100644
--- a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
+++ b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
@@ -32,7 +32,7 @@
absl::optional<bool> enable_dtx;
// Some encoders can encode fewer channels than the actual input to make
- // better use of the bandwidth. |num_channels| sets the number of channels
+ // better use of the bandwidth. `num_channels` sets the number of channels
// to encode.
absl::optional<size_t> num_channels;
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index 547fedd..a34c563 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -92,8 +92,8 @@
timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_);
}
- // Expect |num_calls| calls to the encoder, all successful. The last call
- // claims to have encoded |kMockReturnEncodedBytes| bytes, and all the
+ // Expect `num_calls` calls to the encoder, all successful. The last call
+ // claims to have encoded `kMockReturnEncodedBytes` bytes, and all the
// preceding ones 0 bytes.
void ExpectEncodeCalls(size_t num_calls) {
InSequence s;
@@ -108,7 +108,7 @@
}
// Verifies that the cng_ object waits until it has collected
- // |blocks_per_frame| blocks of audio, and then dispatches all of them to
+ // `blocks_per_frame` blocks of audio, and then dispatches all of them to
// the underlying codec (speech or cng).
void CheckBlockGrouping(size_t blocks_per_frame, bool active_speech) {
EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
@@ -169,7 +169,7 @@
.WillOnce(Return(Vad::kPassive));
}
- // With this call to Encode(), |mock_vad_| should be called according to the
+ // With this call to Encode(), `mock_vad_` should be called according to the
// above expectations.
Encode();
}
@@ -201,7 +201,7 @@
std::unique_ptr<AudioEncoder> cng_;
std::unique_ptr<MockAudioEncoder> mock_encoder_owner_;
MockAudioEncoder* mock_encoder_;
- MockVad* mock_vad_; // Ownership is transferred to |cng_|.
+ MockVad* mock_vad_; // Ownership is transferred to `cng_`.
uint32_t timestamp_;
int16_t audio_[kMaxNumSamples];
size_t num_audio_samples_10ms_;
@@ -294,7 +294,7 @@
for (size_t i = 0; i < 100; ++i) {
Encode();
// Check if it was time to call the cng encoder. This is done once every
- // |kBlocksPerFrame| calls.
+ // `kBlocksPerFrame` calls.
if ((i + 1) % kBlocksPerFrame == 0) {
// Now check if a SID interval has elapsed.
if ((i % (sid_frame_interval_ms / 10)) < kBlocksPerFrame) {
@@ -334,7 +334,7 @@
EXPECT_TRUE(CheckMixedActivePassive(Vad::kPassive, Vad::kActive));
EXPECT_TRUE(encoded_info_.speech);
- // All of the frame is passive speech. Expect no calls to |mock_encoder_|.
+ // All of the frame is passive speech. Expect no calls to `mock_encoder_`.
EXPECT_FALSE(CheckMixedActivePassive(Vad::kPassive, Vad::kPassive));
EXPECT_FALSE(encoded_info_.speech);
}
@@ -442,7 +442,7 @@
}
// Override AudioEncoderCngTest::TearDown, since that one expects a call to
- // the destructor of |mock_vad_|. In this case, that object is already
+ // the destructor of `mock_vad_`. In this case, that object is already
// deleted.
void TearDown() override { cng_.reset(); }
diff --git a/modules/audio_coding/codecs/cng/webrtc_cng.cc b/modules/audio_coding/codecs/cng/webrtc_cng.cc
index 2acaf2b..bfe77c7 100644
--- a/modules/audio_coding/codecs/cng/webrtc_cng.cc
+++ b/modules/audio_coding/codecs/cng/webrtc_cng.cc
@@ -193,10 +193,10 @@
WebRtcSpl_ScaleVector(excitation, excitation, dec_used_scale_factor_,
num_samples, 13);
- /* |lpPoly| - Coefficients in Q12.
- * |excitation| - Speech samples.
+ /* `lpPoly` - Coefficients in Q12.
+ * `excitation` - Speech samples.
* |nst->dec_filtstate| - State preservation.
- * |out_data| - Filtered speech samples. */
+ * `out_data` - Filtered speech samples. */
WebRtcSpl_FilterAR(lpPoly, WEBRTC_CNG_MAX_LPC_ORDER + 1, excitation,
num_samples, dec_filtstate_, WEBRTC_CNG_MAX_LPC_ORDER,
dec_filtstateLow_, WEBRTC_CNG_MAX_LPC_ORDER,
@@ -395,7 +395,7 @@
}
namespace {
-/* Values in |k| are Q15, and |a| Q12. */
+/* Values in `k` are Q15, and `a` Q12. */
void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a) {
int16_t any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
int16_t* aptr;
diff --git a/modules/audio_coding/codecs/cng/webrtc_cng.h b/modules/audio_coding/codecs/cng/webrtc_cng.h
index 563f676..7afd243 100644
--- a/modules/audio_coding/codecs/cng/webrtc_cng.h
+++ b/modules/audio_coding/codecs/cng/webrtc_cng.h
@@ -33,13 +33,13 @@
void Reset();
// Updates the CN state when a new SID packet arrives.
- // |sid| is a view of the SID packet without the headers.
+ // `sid` is a view of the SID packet without the headers.
void UpdateSid(rtc::ArrayView<const uint8_t> sid);
// Generates comfort noise.
- // |out_data| will be filled with samples - its size determines the number of
- // samples generated. When |new_period| is true, CNG history will be reset
- // before any audio is generated. Returns |false| if outData is too large -
+ // `out_data` will be filled with samples - its size determines the number of
+ // samples generated. When `new_period` is true, CNG history will be reset
+ // before any audio is generated. Returns `false` if outData is too large -
// currently 640 bytes (equalling 10ms at 64kHz).
// TODO(ossu): Specify better limits for the size of out_data. Either let it
// be unbounded or limit to 10ms in the current sample rate.
@@ -61,9 +61,9 @@
class ComfortNoiseEncoder {
public:
// Creates a comfort noise encoder.
- // |fs| selects sample rate: 8000 for narrowband or 16000 for wideband.
- // |interval| sets the interval at which to generate SID data (in ms).
- // |quality| selects the number of refl. coeffs. Maximum allowed is 12.
+ // `fs` selects sample rate: 8000 for narrowband or 16000 for wideband.
+ // `interval` sets the interval at which to generate SID data (in ms).
+ // `quality` selects the number of refl. coeffs. Maximum allowed is 12.
ComfortNoiseEncoder(int fs, int interval, int quality);
~ComfortNoiseEncoder() = default;
@@ -74,8 +74,8 @@
// Parameters are set as during construction.
void Reset(int fs, int interval, int quality);
- // Analyzes background noise from |speech| and appends coefficients to
- // |output|. Returns the number of coefficients generated. If |force_sid| is
+ // Analyzes background noise from `speech` and appends coefficients to
+ // `output`. Returns the number of coefficients generated. If `force_sid` is
// true, a SID frame is forced and the internal sid interval counter is reset.
// Will fail if the input size is too large (> 640 samples, see
// ComfortNoiseDecoder::Generate).
diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/modules/audio_coding/codecs/g722/audio_decoder_g722.h
index 6911e0b..eeca139 100644
--- a/modules/audio_coding/codecs/g722/audio_decoder_g722.h
+++ b/modules/audio_coding/codecs/g722/audio_decoder_g722.h
@@ -60,11 +60,11 @@
SpeechType* speech_type) override;
private:
- // Splits the stereo-interleaved payload in |encoded| into separate payloads
+ // Splits the stereo-interleaved payload in `encoded` into separate payloads
// for left and right channels. The separated payloads are written to
- // |encoded_deinterleaved|, which must hold at least |encoded_len| samples.
+ // `encoded_deinterleaved`, which must hold at least `encoded_len` samples.
// The left channel starts at offset 0, while the right channel starts at
- // offset encoded_len / 2 into |encoded_deinterleaved|.
+ // offset encoded_len / 2 into `encoded_deinterleaved`.
void SplitStereoPacket(const uint8_t* encoded,
size_t encoded_len,
uint8_t* encoded_deinterleaved);
diff --git a/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
index 8033c95..7e21fae 100644
--- a/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
+++ b/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
@@ -39,7 +39,7 @@
const int16_t *ppo, *ppi;
int16_t cbVecTmp[4];
/* Interpolation starts 4 elements before cbVec+index, but must not start
- outside |cbVec|; clamping interp_len to stay within |cbVec|.
+ outside `cbVec`; clamping interp_len to stay within `cbVec`.
*/
size_t interp_len = WEBRTC_SPL_MIN(index, 4);
@@ -69,12 +69,12 @@
/* copy the second noninterpolated part */
ppo = buffer - index;
- /* |tempbuff2| is declared in WebRtcIlbcfix_GetCbVec and is SUBL+5 elements
- long. |buffer| points one element past the end of that vector, i.e., at
+ /* `tempbuff2` is declared in WebRtcIlbcfix_GetCbVec and is SUBL+5 elements
+ long. `buffer` points one element past the end of that vector, i.e., at
tempbuff2+SUBL+5. Since ppo=buffer-index, we cannot read any more than
- |index| elements from |ppo|.
+ `index` elements from `ppo`.
- |cbVec| is declared to be SUBL elements long in WebRtcIlbcfix_CbConstruct.
+ `cbVec` is declared to be SUBL elements long in WebRtcIlbcfix_CbConstruct.
Therefore, we can only write SUBL-index elements to cbVec+index.
These two conditions limit the number of elements to copy.
diff --git a/modules/audio_coding/codecs/ilbc/get_cd_vec.c b/modules/audio_coding/codecs/ilbc/get_cd_vec.c
index 145cb96..e9cd200 100644
--- a/modules/audio_coding/codecs/ilbc/get_cd_vec.c
+++ b/modules/audio_coding/codecs/ilbc/get_cd_vec.c
@@ -99,7 +99,7 @@
// We're going to fill in cbveclen + 5 elements of tempbuff2 in
// WebRtcSpl_FilterMAFastQ12, less than the SUBL + 5 elements we'll be
// using in WebRtcIlbcfix_CreateAugmentedVec. This error is caused by
- // bad values in |index| (which come from the encoded stream). Tell the
+ // bad values in `index` (which come from the encoded stream). Tell the
// caller that things went south, and that the decoder state is now
// corrupt (because it's half-way through an update that we can't
// complete).
diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index cb15445..842e77f 100644
--- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@@ -831,15 +831,15 @@
two matrixes, multiply them, and write the results into an output buffer.
Note that two factors (or, multipliers) determine the initialization values of
-the variable |matrix1_index| in the code. The relationship is
-|matrix1_index| = |matrix1_index_factor1| * |matrix1_index_factor2|, where
-|matrix1_index_factor1| is given by the argument while |matrix1_index_factor2|
-is determined by the value of argument |matrix1_index_init_case|;
-|matrix1_index_factor2| is the value of the outmost loop counter j (when
-|matrix1_index_init_case| is 0), or the value of the middle loop counter k (when
-|matrix1_index_init_case| is non-zero).
+the variable `matrix1_index` in the code. The relationship is
+`matrix1_index` = `matrix1_index_factor1` * `matrix1_index_factor2`, where
+`matrix1_index_factor1` is given by the argument while `matrix1_index_factor2`
+is determined by the value of argument `matrix1_index_init_case`;
+`matrix1_index_factor2` is the value of the outmost loop counter j (when
+`matrix1_index_init_case` is 0), or the value of the middle loop counter k (when
+`matrix1_index_init_case` is non-zero).
-|matrix0_index| is determined the same way.
+`matrix0_index` is determined the same way.
Arguments:
matrix0[]: matrix0 data in Q15 domain.
diff --git a/modules/audio_coding/codecs/isac/fix/source/filters.c b/modules/audio_coding/codecs/isac/fix/source/filters.c
index 85860f7..838ba4b 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filters.c
+++ b/modules/audio_coding/codecs/isac/fix/source/filters.c
@@ -75,7 +75,7 @@
a = WEBRTC_SPL_MUL_16_32_RSFT16(InOut16[n], APSectionFactors[j]); //Q0*Q31=Q31 shifted 16 gives Q15
a <<= 1; // Q15 -> Q16
b = WebRtcSpl_AddSatW32(a, FilterState[j]); //Q16+Q16=Q16
- // |a| in Q15 (Q0*Q31=Q31 shifted 16 gives Q15).
+ // `a` in Q15 (Q0*Q31=Q31 shifted 16 gives Q15).
a = WEBRTC_SPL_MUL_16_32_RSFT16(b >> 16, -APSectionFactors[j]);
// FilterState[j]: Q15<<1 + Q0<<16 = Q16 + Q16 = Q16
FilterState[j] = WebRtcSpl_AddSatW32(a << 1, (uint32_t)InOut16[n] << 16);
diff --git a/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index 067d8f3..9a66591 100644
--- a/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -34,7 +34,7 @@
MatrixProduct1 WebRtcIsacfix_MatrixProduct1;
MatrixProduct2 WebRtcIsacfix_MatrixProduct2;
-/* This method assumes that |stream_size_bytes| is in valid range,
+/* This method assumes that `stream_size_bytes` is in valid range,
* i.e. >= 0 && <= STREAM_MAXW16_60MS
*/
static void InitializeDecoderBitstream(size_t stream_size_bytes,
@@ -294,8 +294,8 @@
return statusInit;
}
-/* Read the given number of bytes of big-endian 16-bit integers from |src| and
- write them to |dest| in host endian. If |nbytes| is odd, the number of
+/* Read the given number of bytes of big-endian 16-bit integers from `src` and
+ write them to `dest` in host endian. If `nbytes` is odd, the number of
output elements is rounded up, and the least significant byte of the last
element is set to 0. */
static void read_be16(const uint8_t* src, size_t nbytes, uint16_t* dest) {
@@ -306,8 +306,8 @@
dest[nbytes / 2] = src[nbytes - 1] << 8;
}
-/* Read the given number of bytes of host-endian 16-bit integers from |src| and
- write them to |dest| in big endian. If |nbytes| is odd, the number of source
+/* Read the given number of bytes of host-endian 16-bit integers from `src` and
+ write them to `dest` in big endian. If `nbytes` is odd, the number of source
elements is rounded up (but only the most significant byte of the last
element is used), and the number of output bytes written will be
nbytes + 1. */
diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
index b538085..f151cd1 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
@@ -663,7 +663,7 @@
/* less noise for lower frequencies, by filtering/scaling autocorrelation sequences */
/* Calculate corrlo2[0] = tmpQQlo * corrlo[0] - 2.0*tmpQQlo * corrlo[1];*/
- // |corrlo2QQ| in Q(QdomLO-5).
+ // `corrlo2QQ` in Q(QdomLO-5).
corrlo2QQ[0] = (WEBRTC_SPL_MUL_16_32_RSFT16(tmpQQlo, corrloQQ[0]) >> 1) -
(WEBRTC_SPL_MUL_16_32_RSFT16(aaQ14, corrloQQ[1]) >> 2);
@@ -721,12 +721,12 @@
tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha, tmp);
} else if ((sh-shMem)<7){
tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], shMem); // Shift up CorrBufLoQQ as much as possible
- // Shift |alpha| the number of times required to get |tmp| in QdomLO.
+ // Shift `alpha` the number of times required to get `tmp` in QdomLO.
tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha << (sh - shMem), tmp);
} else {
tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible
- // Shift |alpha| as much as possible without overflow the number of
- // times required to get |tmp| in QdomLO.
+ // Shift `alpha` as much as possible without overflow the number of
+ // times required to get `tmp` in QdomLO.
tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha << 6, tmp);
tmpCorr = corrloQQ[n] >> (sh - shMem - 6);
tmp = tmp + tmpCorr;
@@ -774,7 +774,7 @@
maskdata->CorrBufHiQdom[n] = QdomHI;
} else if ((sh-shMem)<7) {
tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible
- // Shift |alpha| the number of times required to get |tmp| in QdomHI.
+ // Shift `alpha` the number of times required to get `tmp` in QdomHI.
tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha << (sh - shMem), tmp);
tmpCorr = corrhiQQ[n];
tmp = tmp + tmpCorr;
@@ -782,8 +782,8 @@
maskdata->CorrBufHiQdom[n] = QdomHI;
} else {
tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible
- // Shift |alpha| as much as possible without overflow the number of
- // times required to get |tmp| in QdomHI.
+ // Shift `alpha` as much as possible without overflow the number of
+ // times required to get `tmp` in QdomHI.
tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha << 6, tmp);
tmpCorr = corrhiQQ[n] >> (sh - shMem - 6);
tmp = tmp + tmpCorr;
@@ -919,7 +919,7 @@
tmp32a = varscaleQ14 >> 1; // H_T_HQ19=65536 (16-17=-1)
- ssh = sh_hi >> 1; // |sqrt_nrg| is in Qssh.
+ ssh = sh_hi >> 1; // `sqrt_nrg` is in Qssh.
sh = ssh - 14;
tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh
tmp32c = sqrt_nrg + tmp32b; // Qssh (denominator)
diff --git a/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc b/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
index a2e1e08..40381d8 100644
--- a/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
+++ b/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
@@ -203,7 +203,7 @@
e->Encode(/*rtp_timestamp=*/0, AudioFrameToView(in), &encoded);
num_bytes += encoded.size();
}
- // Inverse of the duration of |kNumFrames| 10 ms frames (unit: seconds^-1).
+ // Inverse of the duration of `kNumFrames` 10 ms frames (unit: seconds^-1).
constexpr float kAudioDurationInv = 100.f / kNumFrames;
const int measured_bitrate_bps = 8 * num_bytes * kAudioDurationInv;
EXPECT_LT(measured_bitrate_bps, bitrate_bps + 2000); // Max 2 kbps extra.
diff --git a/modules/audio_coding/codecs/isac/main/include/isac.h b/modules/audio_coding/codecs/isac/main/include/isac.h
index 3d2caef..f45bbb3 100644
--- a/modules/audio_coding/codecs/isac/main/include/isac.h
+++ b/modules/audio_coding/codecs/isac/main/include/isac.h
@@ -606,7 +606,7 @@
int16_t* decoded,
int16_t* speechType);
-/* If |inst| is a decoder but not an encoder: tell it what sample rate the
+/* If `inst` is a decoder but not an encoder: tell it what sample rate the
encoder is using, for bandwidth estimation purposes. */
void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst, int sample_rate_hz);
diff --git a/modules/audio_coding/codecs/isac/main/source/entropy_coding.c b/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
index 6692a51..188c8f6 100644
--- a/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
+++ b/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
@@ -1446,7 +1446,7 @@
index[k] = WebRtcIsac_kQArRcInitIndex[k];
// The safe-guards in following while conditions are to suppress gcc 4.8.3
// warnings, Issue 2888. Otherwise, first and last elements of
- // |WebRtcIsac_kQArBoundaryLevels| are such that the following search
+ // `WebRtcIsac_kQArBoundaryLevels` are such that the following search
// *never* cause an out-of-boundary read.
if (RCQ15[k] > WebRtcIsac_kQArBoundaryLevels[index[k]]) {
while (index[k] + 1 < NUM_AR_RC_QUANT_BAUNDARY &&
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_filter.c b/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
index 61cd533..899d842 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
@@ -25,8 +25,8 @@
* Post-filtering:
* y(z) = x(z) - damper(z) * gain * (x(z) + y(z)) * z ^ (-lag);
*
- * Note that |lag| is a floating number so we perform an interpolation to
- * obtain the correct |lag|.
+ * Note that `lag` is a floating number so we perform an interpolation to
+ * obtain the correct `lag`.
*
*/
@@ -86,7 +86,7 @@
* buffer : a buffer where the sum of previous inputs and outputs
* are stored.
* damper_state : the state of the damping filter. The filter is defined by
- * |kDampFilter|.
+ * `kDampFilter`.
* interpol_coeff : pointer to a set of coefficient which are used to utilize
* fractional pitch by interpolation.
* gain : pitch-gain to be applied to the current segment of input.
@@ -353,7 +353,7 @@
if ((mode == kPitchFilterPreGain) || (mode == kPitchFilterPreLa)) {
/* Filter the lookahead segment, this is treated as the last sub-frame. So
- * set |pf_param| to last sub-frame. */
+ * set `pf_param` to last sub-frame. */
filter_parameters.sub_frame = PITCH_SUBFRAMES - 1;
filter_parameters.num_samples = QLOOKAHEAD;
FilterSegment(in_data, &filter_parameters, out_data, out_dg);
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
index d9efc21..dacf325 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
@@ -59,7 +59,7 @@
new LegacyEncodedAudioFrame(decoder, std::move(payload)));
results.emplace_back(timestamp, 0, std::move(frame));
} else {
- // Reduce the split size by half as long as |split_size_bytes| is at least
+ // Reduce the split size by half as long as `split_size_bytes` is at least
// twice the minimum chunk size (so that the resulting size is at least as
// large as the minimum chunk size).
while (split_size_bytes >= 2 * min_chunk_size) {
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 7c62e98..e4d3b9e 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -564,9 +564,9 @@
void AudioEncoderOpusImpl::SetReceiverFrameLengthRange(
int min_frame_length_ms,
int max_frame_length_ms) {
- // Ensure that |SetReceiverFrameLengthRange| is called before
- // |EnableAudioNetworkAdaptor|, otherwise we need to recreate
- // |audio_network_adaptor_|, which is not a needed use case.
+ // Ensure that `SetReceiverFrameLengthRange` is called before
+ // `EnableAudioNetworkAdaptor`, otherwise we need to recreate
+ // `audio_network_adaptor_`, which is not a needed use case.
RTC_DCHECK(!audio_network_adaptor_);
FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
&config_.supported_frame_lengths_ms);
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index f1953ea..daca6aa 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -228,8 +228,8 @@
TEST_P(AudioEncoderOpusTest, SetReceiverFrameLengthRange) {
auto states = CreateCodec(sample_rate_hz_, 2);
- // Before calling to |SetReceiverFrameLengthRange|,
- // |supported_frame_lengths_ms| should contain only the frame length being
+ // Before calling to `SetReceiverFrameLengthRange`,
+ // `supported_frame_lengths_ms` should contain only the frame length being
// used.
using ::testing::ElementsAre;
EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
@@ -348,7 +348,7 @@
// will fail.
constexpr float kPacketLossFraction_1 = 0.02f;
constexpr float kPacketLossFraction_2 = 0.198f;
- // |kSecondSampleTimeMs| is chosen to ease the calculation since
+ // `kSecondSampleTimeMs` is chosen to ease the calculation since
// 0.9999 ^ 6931 = 0.5.
constexpr int64_t kSecondSampleTimeMs = 6931;
@@ -380,7 +380,7 @@
states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusRate * 2,
absl::nullopt);
- // Since |OnReceivedOverhead| has not been called, the codec bitrate should
+ // Since `OnReceivedOverhead` has not been called, the codec bitrate should
// not change.
EXPECT_EQ(kDefaultOpusRate, states->encoder->GetTargetBitrate());
}
diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc
index 1923647..0636935 100644
--- a/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -218,8 +218,8 @@
time_now_ms += block_duration_ms_;
- // |data_pointer_| is incremented and wrapped across
- // |loop_length_samples_|.
+ // `data_pointer_` is incremented and wrapped across
+ // `loop_length_samples_`.
data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) %
loop_length_samples_;
}
diff --git a/modules/audio_coding/codecs/opus/opus_interface.cc b/modules/audio_coding/codecs/opus/opus_interface.cc
index f684452..0337919 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.cc
+++ b/modules/audio_coding/codecs/opus/opus_interface.cc
@@ -574,8 +574,8 @@
/* For decoder to determine if it is to output speech or comfort noise. */
static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) {
- // Audio type becomes comfort noise if |encoded_byte| is 1 and keeps
- // to be so if the following |encoded_byte| are 0 or 1.
+ // Audio type becomes comfort noise if `encoded_byte` is 1 and keeps
+ // to be so if the following `encoded_byte` are 0 or 1.
if (encoded_bytes == 0 && inst->in_dtx_mode) {
return 2; // Comfort noise.
} else if (encoded_bytes == 1 || encoded_bytes == 2) {
@@ -595,7 +595,7 @@
}
}
-/* |frame_size| is set to maximum Opus frame size in the normal case, and
+/* `frame_size` is set to maximum Opus frame size in the normal case, and
* is set to the number of samples needed for PLC in case of losses.
* It is up to the caller to make sure the value is correct. */
static int DecodeNative(OpusDecInst* inst,
@@ -632,9 +632,9 @@
FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
if (inst->plc_use_prev_decoded_samples) {
- /* The number of samples we ask for is |number_of_lost_frames| times
- * |prev_decoded_samples_|. Limit the number of samples to maximum
- * |MaxFrameSizePerChannel()|. */
+ /* The number of samples we ask for is `number_of_lost_frames` times
+ * `prev_decoded_samples_`. Limit the number of samples to maximum
+ * `MaxFrameSizePerChannel()`. */
plc_samples = inst->prev_decoded_samples;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
@@ -729,9 +729,9 @@
int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
if (inst->plc_use_prev_decoded_samples) {
- /* The number of samples we ask for is |number_of_lost_frames| times
- * |prev_decoded_samples_|. Limit the number of samples to maximum
- * |MaxFrameSizePerChannel()|. */
+ /* The number of samples we ask for is `number_of_lost_frames` times
+ * `prev_decoded_samples_`. Limit the number of samples to maximum
+ * `MaxFrameSizePerChannel()`. */
const int plc_samples = inst->prev_decoded_samples;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
@@ -826,8 +826,8 @@
// as binary values with uniform probability, they can be extracted directly
// from the most significant bits of the first byte of compressed data.
for (int n = 0; n < channels; n++) {
- // The LBRR bit for channel 1 is on the (|silk_frames| + 1)-th bit, and
- // that of channel 2 is on the |(|silk_frames| + 1) * 2 + 1|-th bit.
+ // The LBRR bit for channel 1 is on the (`silk_frames` + 1)-th bit, and
+ // that of channel 2 is on the |(`silk_frames` + 1) * 2 + 1|-th bit.
if (frame_data[0][0] & (0x80 >> ((n + 1) * (silk_frames + 1) - 1)))
return 1;
}
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index 80cab50..b507a32 100644
--- a/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -115,10 +115,10 @@
void TestCbrEffect(bool dtx, int block_length_ms);
- // Prepare |speech_data_| for encoding, read from a hard-coded file.
+ // Prepare `speech_data_` for encoding, read from a hard-coded file.
// After preparation, |speech_data_.GetNextBlock()| returns a pointer to a
- // block of |block_length_ms| milliseconds. The data is looped every
- // |loop_length_ms| milliseconds.
+ // block of `block_length_ms` milliseconds. The data is looped every
+ // `loop_length_ms` milliseconds.
void PrepareSpeechData(int block_length_ms, int loop_length_ms);
int EncodeDecode(WebRtcOpusEncInst* encoder,
@@ -310,24 +310,24 @@
// one with an arbitrary size and the other of 1-byte, then stops sending for
// a certain number of frames.
- // |max_dtx_frames| is the maximum number of frames Opus can stay in DTX.
+ // `max_dtx_frames` is the maximum number of frames Opus can stay in DTX.
// TODO(kwiberg): Why does this number depend on the encoding sample rate?
const int max_dtx_frames =
(encoder_sample_rate_hz_ == 16000 ? 800 : 400) / block_length_ms + 1;
- // We run |kRunTimeMs| milliseconds of pure silence.
+ // We run `kRunTimeMs` milliseconds of pure silence.
const int kRunTimeMs = 4500;
- // We check that, after a |kCheckTimeMs| milliseconds (given that the CNG in
+ // We check that, after a `kCheckTimeMs` milliseconds (given that the CNG in
// Opus needs time to adapt), the absolute values of DTX decoded signal are
- // bounded by |kOutputValueBound|.
+ // bounded by `kOutputValueBound`.
const int kCheckTimeMs = 4000;
#if defined(OPUS_FIXED_POINT)
// Fixed-point Opus generates a random (comfort) noise, which has a less
// predictable value bound than its floating-point Opus. This value depends on
// input signal, and the time window for checking the output values (between
- // |kCheckTimeMs| and |kRunTimeMs|).
+ // `kCheckTimeMs` and `kRunTimeMs`).
const uint16_t kOutputValueBound = 30;
#else
@@ -336,7 +336,7 @@
int time = 0;
while (time < kRunTimeMs) {
- // DTX mode is maintained for maximum |max_dtx_frames| frames.
+ // DTX mode is maintained for maximum `max_dtx_frames` frames.
int i = 0;
for (; i < max_dtx_frames; ++i) {
time += block_length_ms;
diff --git a/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h b/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h
index a89dfd8..a280ca2 100644
--- a/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h
+++ b/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h
@@ -29,11 +29,11 @@
AudioRingBuffer(size_t channels, size_t max_frames);
~AudioRingBuffer();
- // Copies |data| to the buffer and advances the write pointer. |channels| must
+ // Copies `data` to the buffer and advances the write pointer. `channels` must
// be the same as at creation time.
void Write(const float* const* data, size_t channels, size_t frames);
- // Copies from the buffer to |data| and advances the read pointer. |channels|
+ // Copies from the buffer to `data` and advances the read pointer. `channels`
// must be the same as at creation time.
void Read(float* const* data, size_t channels, size_t frames);
diff --git a/modules/audio_coding/codecs/opus/test/blocker.cc b/modules/audio_coding/codecs/opus/test/blocker.cc
index 7f102b5..33406ce 100644
--- a/modules/audio_coding/codecs/opus/test/blocker.cc
+++ b/modules/audio_coding/codecs/opus/test/blocker.cc
@@ -16,7 +16,7 @@
namespace {
-// Adds |a| and |b| frame by frame into |result| (basically matrix addition).
+// Adds `a` and `b` frame by frame into `result` (basically matrix addition).
void AddFrames(const float* const* a,
size_t a_start_index,
const float* const* b,
@@ -33,7 +33,7 @@
}
}
-// Copies |src| into |dst| channel by channel.
+// Copies `src` into `dst` channel by channel.
void CopyFrames(const float* const* src,
size_t src_start_index,
size_t num_frames,
@@ -46,7 +46,7 @@
}
}
-// Moves |src| into |dst| channel by channel.
+// Moves `src` into `dst` channel by channel.
void MoveFrames(const float* const* src,
size_t src_start_index,
size_t num_frames,
@@ -69,8 +69,8 @@
}
}
-// Pointwise multiplies each channel of |frames| with |window|. Results are
-// stored in |frames|.
+// Pointwise multiplies each channel of `frames` with `window`. Results are
+// stored in `frames`.
void ApplyWindow(const float* window,
size_t num_frames,
size_t num_channels,
@@ -134,7 +134,7 @@
// On each call to ProcessChunk():
// 1. New input gets read into sections _b_ and _c_ of the input buffer.
// 2. We block starting from frame_offset.
-// 3. We block until we reach a block |bl| that doesn't contain any frames
+// 3. We block until we reach a block `bl` that doesn't contain any frames
// from sections _a_ or _b_ of the input buffer.
// 4. We window the current block, fire the callback for processing, window
// again, and overlap/add to the output buffer.
@@ -142,7 +142,7 @@
// 6. For both the input and the output buffers, we copy section _c_ into
// section _a_.
// 7. We set the new frame_offset to be the difference between the first frame
-// of |bl| and the border between sections _b_ and _c_.
+// of `bl` and the border between sections _b_ and _c_.
//
// When block_size > chunk_size the input and output buffers look like this:
//
@@ -153,13 +153,13 @@
// On each call to ProcessChunk():
// The procedure is the same as above, except for:
// 1. New input gets read into section _c_ of the input buffer.
-// 3. We block until we reach a block |bl| that doesn't contain any frames
+// 3. We block until we reach a block `bl` that doesn't contain any frames
// from section _a_ of the input buffer.
// 5. We copy section _a_ of the output buffer into output.
// 6. For both the input and the output buffers, we copy sections _b_ and _c_
// into section _a_ and _b_.
// 7. We set the new frame_offset to be the difference between the first frame
-// of |bl| and the border between sections _a_ and _b_.
+// of `bl` and the border between sections _a_ and _b_.
//
// * delay here refers to inintial_delay_
//
diff --git a/modules/audio_coding/codecs/opus/test/blocker.h b/modules/audio_coding/codecs/opus/test/blocker.h
index 26177bc..59b7e29 100644
--- a/modules/audio_coding/codecs/opus/test/blocker.h
+++ b/modules/audio_coding/codecs/opus/test/blocker.h
@@ -39,7 +39,7 @@
// of audio, which is not a power of 2. Blocker allows us to specify the
// transform and all other necessary processing via the Process() callback
// function without any constraints on the transform-size
-// (read: |block_size_|) or received-audio-size (read: |chunk_size_|).
+// (read: `block_size_`) or received-audio-size (read: `chunk_size_`).
// We handle this for the multichannel audio case, allowing for different
// numbers of input and output channels (for example, beamforming takes 2 or
// more input channels and returns 1 output channel). Audio signals are
@@ -53,8 +53,8 @@
// sending back a processed chunk
//
// To use blocker:
-// 1. Impelment a BlockerCallback object |bc|.
-// 2. Instantiate a Blocker object |b|, passing in |bc|.
+// 1. Impelment a BlockerCallback object `bc`.
+// 2. Instantiate a Blocker object `b`, passing in `bc`.
// 3. As you receive audio, call b.ProcessChunk() to get processed audio.
//
// A small amount of delay is added to the first received chunk to deal with
@@ -101,7 +101,7 @@
// input and output buffers are responsible for saving those frames between
// calls to ProcessChunk().
//
- // Both contain |initial delay| + |chunk_size| frames. The input is a fairly
+ // Both contain |initial delay| + `chunk_size` frames. The input is a fairly
// standard FIFO, but due to the overlap-add it's harder to use an
// AudioRingBuffer for the output.
AudioRingBuffer input_buffer_;
@@ -116,7 +116,7 @@
std::unique_ptr<float[]> window_;
// The amount of frames between the start of contiguous blocks. For example,
- // |shift_amount_| = |block_size_| / 2 for a Hann window.
+ // `shift_amount_` = `block_size_` / 2 for a Hann window.
size_t shift_amount_;
BlockerCallback* callback_;
diff --git a/modules/audio_coding/codecs/opus/test/lapped_transform.h b/modules/audio_coding/codecs/opus/test/lapped_transform.h
index 3620df3..bb25c34 100644
--- a/modules/audio_coding/codecs/opus/test/lapped_transform.h
+++ b/modules/audio_coding/codecs/opus/test/lapped_transform.h
@@ -84,11 +84,11 @@
std::complex<float>* const* out_block) = 0;
};
- // Construct a transform instance. |chunk_length| is the number of samples in
- // each channel. |window| defines the window, owned by the caller (a copy is
- // made internally); |window| should have length equal to |block_length|.
- // |block_length| defines the length of a block, in samples.
- // |shift_amount| is in samples. |callback| is the caller-owned audio
+ // Construct a transform instance. `chunk_length` is the number of samples in
+ // each channel. `window` defines the window, owned by the caller (a copy is
+ // made internally); `window` should have length equal to `block_length`.
+ // `block_length` defines the length of a block, in samples.
+ // `shift_amount` is in samples. `callback` is the caller-owned audio
// processing function called for each block of the input chunk.
LappedTransform(size_t num_in_channels,
size_t num_out_channels,
@@ -99,10 +99,10 @@
Callback* callback);
~LappedTransform();
- // Main audio processing helper method. Internally slices |in_chunk| into
+ // Main audio processing helper method. Internally slices `in_chunk` into
// blocks, transforms them to frequency domain, calls the callback for each
// block and returns a de-blocked time domain chunk of audio through
- // |out_chunk|. Both buffers are caller-owned.
+ // `out_chunk`. Both buffers are caller-owned.
void ProcessChunk(const float* const* in_chunk, float* const* out_chunk);
// Get the chunk length.
@@ -132,8 +132,8 @@
// Returns the initial delay.
//
- // This is the delay introduced by the |blocker_| to be able to get and return
- // chunks of |chunk_length|, but process blocks of |block_length|.
+ // This is the delay introduced by the `blocker_` to be able to get and return
+ // chunks of `chunk_length`, but process blocks of `block_length`.
size_t initial_delay() const { return blocker_.initial_delay(); }
private:
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index c72768e..1ca7a84 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -145,7 +145,7 @@
info.redundant.push_back(it->first);
}
- // |info| will be implicitly cast to an EncodedInfoLeaf struct, effectively
+ // `info` will be implicitly cast to an EncodedInfoLeaf struct, effectively
// discarding the (empty) vector of redundant information. This is
// intentional.
if (header_length_bytes > 0) {
diff --git a/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
index 59c2f16..c5f1d7c 100644
--- a/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
+++ b/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -31,9 +31,9 @@
virtual void TearDown();
// EncodeABlock(...) does the following:
- // 1. encodes a block of audio, saved in |in_data|,
- // 2. save the bit stream to |bit_stream| of |max_bytes| bytes in size,
- // 3. assign |encoded_bytes| with the length of the bit stream (in bytes),
+ // 1. encodes a block of audio, saved in `in_data`,
+ // 2. save the bit stream to `bit_stream` of `max_bytes` bytes in size,
+ // 3. assign `encoded_bytes` with the length of the bit stream (in bytes),
// 4. return the cost of time (in millisecond) spent on actual encoding.
virtual float EncodeABlock(int16_t* in_data,
uint8_t* bit_stream,
@@ -41,15 +41,15 @@
size_t* encoded_bytes) = 0;
// DecodeABlock(...) does the following:
- // 1. decodes the bit stream in |bit_stream| with a length of |encoded_bytes|
+ // 1. decodes the bit stream in `bit_stream` with a length of `encoded_bytes`
// (in bytes),
- // 2. save the decoded audio in |out_data|,
+ // 2. save the decoded audio in `out_data`,
// 3. return the cost of time (in millisecond) spent on actual decoding.
virtual float DecodeABlock(const uint8_t* bit_stream,
size_t encoded_bytes,
int16_t* out_data) = 0;
- // Encoding and decode an audio of |audio_duration| (in seconds) and
+ // Encoding and decode an audio of `audio_duration` (in seconds) and
// record the runtime for encoding and decoding separately.
void EncodeDecode(size_t audio_duration);
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 102e2de..7551814 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -83,9 +83,9 @@
// Sender
//
- // |modifier| is called exactly once with one argument: a pointer to the
+ // `modifier` is called exactly once with one argument: a pointer to the
// unique_ptr that holds the current encoder (which is null if there is no
- // current encoder). For the duration of the call, |modifier| has exclusive
+ // current encoder). For the duration of the call, `modifier` has exclusive
// access to the unique_ptr; it may call the encoder, steal the encoder and
// replace it with another encoder or with nullptr, etc.
virtual void ModifyEncoder(
diff --git a/modules/audio_coding/neteq/accelerate.cc b/modules/audio_coding/neteq/accelerate.cc
index 954b148..f4ef6cd 100644
--- a/modules/audio_coding/neteq/accelerate.cc
+++ b/modules/audio_coding/neteq/accelerate.cc
@@ -57,12 +57,12 @@
if ((best_correlation > correlation_threshold) || !active_speech) {
// Do accelerate operation by overlap add.
- // Pre-calculate common multiplication with |fs_mult_|.
+ // Pre-calculate common multiplication with `fs_mult_`.
// 120 corresponds to 15 ms.
size_t fs_mult_120 = fs_mult_ * 120;
if (fast_mode) {
- // Fit as many multiples of |peak_index| as possible in fs_mult_120.
+ // Fit as many multiples of `peak_index` as possible in fs_mult_120.
// TODO(henrik.lundin) Consider finding multiple correlation peaks and
// pick the one with the longest correlation lag in this case.
peak_index = (fs_mult_120 / peak_index) * peak_index;
@@ -72,11 +72,11 @@
// Copy first part; 0 to 15 ms.
output->PushBackInterleaved(
rtc::ArrayView<const int16_t>(input, fs_mult_120 * num_channels_));
- // Copy the |peak_index| starting at 15 ms to |temp_vector|.
+ // Copy the `peak_index` starting at 15 ms to `temp_vector`.
AudioMultiVector temp_vector(num_channels_);
temp_vector.PushBackInterleaved(rtc::ArrayView<const int16_t>(
&input[fs_mult_120 * num_channels_], peak_index * num_channels_));
- // Cross-fade |temp_vector| onto the end of |output|.
+ // Cross-fade `temp_vector` onto the end of `output`.
output->CrossFade(temp_vector, peak_index);
// Copy the last unmodified part, 15 ms + pitch period until the end.
output->PushBackInterleaved(rtc::ArrayView<const int16_t>(
diff --git a/modules/audio_coding/neteq/accelerate.h b/modules/audio_coding/neteq/accelerate.h
index 124b633..e03f609 100644
--- a/modules/audio_coding/neteq/accelerate.h
+++ b/modules/audio_coding/neteq/accelerate.h
@@ -34,10 +34,10 @@
: TimeStretch(sample_rate_hz, num_channels, background_noise) {}
// This method performs the actual Accelerate operation. The samples are
- // read from |input|, of length |input_length| elements, and are written to
- // |output|. The number of samples removed through time-stretching is
- // is provided in the output |length_change_samples|. The method returns
- // the outcome of the operation as an enumerator value. If |fast_accelerate|
+ // read from `input`, of length `input_length` elements, and are written to
+ // `output`. The number of samples removed through time-stretching is
+ // is provided in the output `length_change_samples`. The method returns
+ // the outcome of the operation as an enumerator value. If `fast_accelerate`
// is true, the algorithm will relax the requirements on finding strong
// correlations, and may remove multiple pitch periods if possible.
ReturnCodes Process(const int16_t* input,
@@ -47,7 +47,7 @@
size_t* length_change_samples);
protected:
- // Sets the parameters |best_correlation| and |peak_index| to suitable
+ // Sets the parameters `best_correlation` and `peak_index` to suitable
// values when the signal contains no active speech.
void SetParametersForPassiveSpeech(size_t len,
int16_t* best_correlation,
diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc
index b13fe44..66b99b4 100644
--- a/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -40,7 +40,7 @@
constexpr int kOverheadBytesPerPacket = 50;
// The absolute difference between the input and output (the first channel) is
-// compared vs |tolerance|. The parameter |delay| is used to correct for codec
+// compared vs `tolerance`. The parameter `delay` is used to correct for codec
// delays.
void CompareInputOutput(const std::vector<int16_t>& input,
const std::vector<int16_t>& output,
@@ -56,8 +56,8 @@
}
}
-// The absolute difference between the first two channels in |output| is
-// compared vs |tolerance|.
+// The absolute difference between the first two channels in `output` is
+// compared vs `tolerance`.
void CompareTwoChannels(const std::vector<int16_t>& output,
size_t samples_per_channel,
size_t channels,
@@ -70,7 +70,7 @@
}
// Calculates mean-squared error between input and output (the first channel).
-// The parameter |delay| is used to correct for codec delays.
+// The parameter `delay` is used to correct for codec delays.
double MseInputOutput(const std::vector<int16_t>& input,
const std::vector<int16_t>& output,
size_t num_samples,
@@ -152,10 +152,10 @@
}
// Encodes and decodes audio. The absolute difference between the input and
- // output is compared vs |tolerance|, and the mean-squared error is compared
- // with |mse|. The encoded stream should contain |expected_bytes|. For stereo
+ // output is compared vs `tolerance`, and the mean-squared error is compared
+ // with `mse`. The encoded stream should contain `expected_bytes`. For stereo
// audio, the absolute difference between the two channels is compared vs
- // |channel_diff_tolerance|.
+ // `channel_diff_tolerance`.
void EncodeDecodeTest(size_t expected_bytes,
int tolerance,
double mse,
@@ -170,7 +170,7 @@
std::vector<int16_t> input;
std::vector<int16_t> decoded;
while (processed_samples + frame_size_ <= data_length_) {
- // Extend input vector with |frame_size_|.
+ // Extend input vector with `frame_size_`.
input.resize(input.size() + frame_size_, 0);
// Read from input file.
ASSERT_GE(input.size() - processed_samples, frame_size_);
diff --git a/modules/audio_coding/neteq/audio_multi_vector.cc b/modules/audio_coding/neteq/audio_multi_vector.cc
index a3b5ce3..220d5a1 100644
--- a/modules/audio_coding/neteq/audio_multi_vector.cc
+++ b/modules/audio_coding/neteq/audio_multi_vector.cc
@@ -77,8 +77,8 @@
size_t length_per_channel = append_this.size() / num_channels_;
int16_t* temp_array = new int16_t[length_per_channel]; // Temporary storage.
for (size_t channel = 0; channel < num_channels_; ++channel) {
- // Copy elements to |temp_array|.
- // Set |source_ptr| to first element of this channel.
+ // Copy elements to `temp_array`.
+ // Set `source_ptr` to first element of this channel.
const int16_t* source_ptr = &append_this[channel];
for (size_t i = 0; i < length_per_channel; ++i) {
temp_array[i] = *source_ptr;
@@ -132,7 +132,7 @@
size_t length,
int16_t* destination) const {
RTC_DCHECK(destination);
- size_t index = 0; // Number of elements written to |destination| so far.
+ size_t index = 0; // Number of elements written to `destination` so far.
RTC_DCHECK_LE(start_index, Size());
start_index = std::min(start_index, Size());
if (length + start_index > Size()) {
@@ -162,7 +162,7 @@
size_t length,
size_t position) {
RTC_DCHECK_EQ(num_channels_, insert_this.num_channels_);
- // Cap |length| at the length of |insert_this|.
+ // Cap `length` at the length of `insert_this`.
RTC_DCHECK_LE(length, insert_this.Size());
length = std::min(length, insert_this.Size());
if (num_channels_ == insert_this.num_channels_) {
diff --git a/modules/audio_coding/neteq/audio_multi_vector.h b/modules/audio_coding/neteq/audio_multi_vector.h
index 0bb0b28..10179d7 100644
--- a/modules/audio_coding/neteq/audio_multi_vector.h
+++ b/modules/audio_coding/neteq/audio_multi_vector.h
@@ -24,12 +24,12 @@
class AudioMultiVector {
public:
- // Creates an empty AudioMultiVector with |N| audio channels. |N| must be
+ // Creates an empty AudioMultiVector with `N` audio channels. `N` must be
// larger than 0.
explicit AudioMultiVector(size_t N);
- // Creates an AudioMultiVector with |N| audio channels, each channel having
- // an initial size. |N| must be larger than 0.
+ // Creates an AudioMultiVector with `N` audio channels, each channel having
+ // an initial size. `N` must be larger than 0.
AudioMultiVector(size_t N, size_t initial_size);
virtual ~AudioMultiVector();
@@ -37,47 +37,47 @@
// Deletes all values and make the vector empty.
virtual void Clear();
- // Clears the vector and inserts |length| zeros into each channel.
+ // Clears the vector and inserts `length` zeros into each channel.
virtual void Zeros(size_t length);
- // Copies all values from this vector to |copy_to|. Any contents in |copy_to|
- // are deleted. After the operation is done, |copy_to| will be an exact
+ // Copies all values from this vector to `copy_to`. Any contents in `copy_to`
+ // are deleted. After the operation is done, `copy_to` will be an exact
// replica of this object. The source and the destination must have the same
// number of channels.
virtual void CopyTo(AudioMultiVector* copy_to) const;
- // Appends the contents of |append_this| to the end of this object. The array
+ // Appends the contents of `append_this` to the end of this object. The array
// is assumed to be channel-interleaved. The length must be an even multiple
// of this object's number of channels. The length of this object is increased
// with the length of the array divided by the number of channels.
void PushBackInterleaved(rtc::ArrayView<const int16_t> append_this);
- // Appends the contents of AudioMultiVector |append_this| to this object. The
- // length of this object is increased with the length of |append_this|.
+ // Appends the contents of AudioMultiVector `append_this` to this object. The
+ // length of this object is increased with the length of `append_this`.
virtual void PushBack(const AudioMultiVector& append_this);
- // Appends the contents of AudioMultiVector |append_this| to this object,
- // taken from |index| up until the end of |append_this|. The length of this
+ // Appends the contents of AudioMultiVector `append_this` to this object,
+ // taken from `index` up until the end of `append_this`. The length of this
// object is increased.
virtual void PushBackFromIndex(const AudioMultiVector& append_this,
size_t index);
- // Removes |length| elements from the beginning of this object, from each
+ // Removes `length` elements from the beginning of this object, from each
// channel.
virtual void PopFront(size_t length);
- // Removes |length| elements from the end of this object, from each
+ // Removes `length` elements from the end of this object, from each
// channel.
virtual void PopBack(size_t length);
- // Reads |length| samples from each channel and writes them interleaved to
- // |destination|. The total number of elements written to |destination| is
- // returned, i.e., |length| * number of channels. If the AudioMultiVector
- // contains less than |length| samples per channel, this is reflected in the
+ // Reads `length` samples from each channel and writes them interleaved to
+ // `destination`. The total number of elements written to `destination` is
+ // returned, i.e., `length` * number of channels. If the AudioMultiVector
+ // contains less than `length` samples per channel, this is reflected in the
// return value.
virtual size_t ReadInterleaved(size_t length, int16_t* destination) const;
- // Like ReadInterleaved() above, but reads from |start_index| instead of from
+ // Like ReadInterleaved() above, but reads from `start_index` instead of from
// the beginning.
virtual size_t ReadInterleavedFromIndex(size_t start_index,
size_t length,
@@ -89,18 +89,18 @@
int16_t* destination) const;
// Overwrites each channel in this AudioMultiVector with values taken from
- // |insert_this|. The values are taken from the beginning of |insert_this| and
- // are inserted starting at |position|. |length| values are written into each
- // channel. If |length| and |position| are selected such that the new data
+ // `insert_this`. The values are taken from the beginning of `insert_this` and
+ // are inserted starting at `position`. `length` values are written into each
+ // channel. If `length` and `position` are selected such that the new data
// extends beyond the end of the current AudioVector, the vector is extended
- // to accommodate the new data. |length| is limited to the length of
- // |insert_this|.
+ // to accommodate the new data. `length` is limited to the length of
+ // `insert_this`.
virtual void OverwriteAt(const AudioMultiVector& insert_this,
size_t length,
size_t position);
- // Appends |append_this| to the end of the current vector. Lets the two
- // vectors overlap by |fade_length| samples (per channel), and cross-fade
+ // Appends `append_this` to the end of the current vector. Lets the two
+ // vectors overlap by `fade_length` samples (per channel), and cross-fade
// linearly in this region.
virtual void CrossFade(const AudioMultiVector& append_this,
size_t fade_length);
@@ -111,14 +111,14 @@
// Returns the number of elements per channel in this AudioMultiVector.
virtual size_t Size() const;
- // Verify that each channel can hold at least |required_size| elements. If
+ // Verify that each channel can hold at least `required_size` elements. If
// not, extend accordingly.
virtual void AssertSize(size_t required_size);
virtual bool Empty() const;
// Copies the data between two channels in the AudioMultiVector. The method
- // does not add any new channel. Thus, |from_channel| and |to_channel| must
+ // does not add any new channel. Thus, `from_channel` and `to_channel` must
// both be valid channel numbers.
virtual void CopyChannel(size_t from_channel, size_t to_channel);
diff --git a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
index d1351d8..329377a 100644
--- a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
+++ b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
@@ -94,7 +94,7 @@
AudioMultiVector vec(num_channels_);
vec.PushBackInterleaved(array_interleaved_);
AudioMultiVector vec_copy(num_channels_);
- vec.CopyTo(&vec_copy); // Copy from |vec| to |vec_copy|.
+ vec.CopyTo(&vec_copy); // Copy from `vec` to `vec_copy`.
ASSERT_EQ(num_channels_, vec.Channels());
ASSERT_EQ(array_length(), vec.Size());
ASSERT_EQ(num_channels_, vec_copy.Channels());
@@ -106,7 +106,7 @@
}
}
- // Clear |vec| and verify that it is empty.
+ // Clear `vec` and verify that it is empty.
vec.Clear();
EXPECT_TRUE(vec.Empty());
@@ -208,7 +208,7 @@
vec.PushBackInterleaved(array_interleaved_);
vec.PopFront(1); // Remove one element from each channel.
ASSERT_EQ(array_length() - 1u, vec.Size());
- // Let |ptr| point to the second element of the first channel in the
+ // Let `ptr` point to the second element of the first channel in the
// interleaved array.
int16_t* ptr = &array_interleaved_[num_channels_];
for (size_t i = 0; i < array_length() - 1; ++i) {
@@ -227,7 +227,7 @@
vec.PushBackInterleaved(array_interleaved_);
vec.PopBack(1); // Remove one element from each channel.
ASSERT_EQ(array_length() - 1u, vec.Size());
- // Let |ptr| point to the first element of the first channel in the
+ // Let `ptr` point to the first element of the first channel in the
// interleaved array.
int16_t* ptr = array_interleaved_.data();
for (size_t i = 0; i < array_length() - 1; ++i) {
diff --git a/modules/audio_coding/neteq/audio_vector.cc b/modules/audio_coding/neteq/audio_vector.cc
index ce27a88..10e8936 100644
--- a/modules/audio_coding/neteq/audio_vector.cc
+++ b/modules/audio_coding/neteq/audio_vector.cc
@@ -245,14 +245,14 @@
void AudioVector::CrossFade(const AudioVector& append_this,
size_t fade_length) {
- // Fade length cannot be longer than the current vector or |append_this|.
+ // Fade length cannot be longer than the current vector or `append_this`.
RTC_DCHECK_LE(fade_length, Size());
RTC_DCHECK_LE(fade_length, append_this.Size());
fade_length = std::min(fade_length, Size());
fade_length = std::min(fade_length, append_this.Size());
size_t position = Size() - fade_length + begin_index_;
// Cross fade the overlapping regions.
- // |alpha| is the mixing factor in Q14.
+ // `alpha` is the mixing factor in Q14.
// TODO(hlundin): Consider skipping +1 in the denominator to produce a
// smoother cross-fade, in particular at the end of the fade.
int alpha_step = 16384 / (static_cast<int>(fade_length) + 1);
@@ -265,7 +265,7 @@
14;
}
RTC_DCHECK_GE(alpha, 0); // Verify that the slope was correct.
- // Append what is left of |append_this|.
+ // Append what is left of `append_this`.
size_t samples_to_push_back = append_this.Size() - fade_length;
if (samples_to_push_back > 0)
PushBack(append_this, samples_to_push_back, fade_length);
@@ -286,8 +286,8 @@
return;
const size_t length = Size();
// Reserve one more sample to remove the ambiguity between empty vector and
- // full vector. Therefore |begin_index_| == |end_index_| indicates empty
- // vector, and |begin_index_| == (|end_index_| + 1) % capacity indicates
+ // full vector. Therefore `begin_index_` == `end_index_` indicates empty
+ // vector, and `begin_index_` == (`end_index_` + 1) % capacity indicates
// full vector.
std::unique_ptr<int16_t[]> temp_array(new int16_t[n + 1]);
CopyTo(length, 0, temp_array.get());
diff --git a/modules/audio_coding/neteq/audio_vector.h b/modules/audio_coding/neteq/audio_vector.h
index a257586..c722b56 100644
--- a/modules/audio_coding/neteq/audio_vector.h
+++ b/modules/audio_coding/neteq/audio_vector.h
@@ -34,27 +34,27 @@
// Deletes all values and make the vector empty.
virtual void Clear();
- // Copies all values from this vector to |copy_to|. Any contents in |copy_to|
+ // Copies all values from this vector to `copy_to`. Any contents in `copy_to`
// are deleted before the copy operation. After the operation is done,
- // |copy_to| will be an exact replica of this object.
+ // `copy_to` will be an exact replica of this object.
virtual void CopyTo(AudioVector* copy_to) const;
- // Copies |length| values from |position| in this vector to |copy_to|.
+ // Copies `length` values from `position` in this vector to `copy_to`.
virtual void CopyTo(size_t length, size_t position, int16_t* copy_to) const;
- // Prepends the contents of AudioVector |prepend_this| to this object. The
- // length of this object is increased with the length of |prepend_this|.
+ // Prepends the contents of AudioVector `prepend_this` to this object. The
+ // length of this object is increased with the length of `prepend_this`.
virtual void PushFront(const AudioVector& prepend_this);
- // Same as above, but with an array |prepend_this| with |length| elements as
+ // Same as above, but with an array `prepend_this` with `length` elements as
// source.
virtual void PushFront(const int16_t* prepend_this, size_t length);
// Same as PushFront but will append to the end of this object.
virtual void PushBack(const AudioVector& append_this);
- // Appends a segment of |append_this| to the end of this object. The segment
- // starts from |position| and has |length| samples.
+ // Appends a segment of `append_this` to the end of this object. The segment
+ // starts from `position` and has `length` samples.
virtual void PushBack(const AudioVector& append_this,
size_t length,
size_t position);
@@ -62,47 +62,47 @@
// Same as PushFront but will append to the end of this object.
virtual void PushBack(const int16_t* append_this, size_t length);
- // Removes |length| elements from the beginning of this object.
+ // Removes `length` elements from the beginning of this object.
virtual void PopFront(size_t length);
- // Removes |length| elements from the end of this object.
+ // Removes `length` elements from the end of this object.
virtual void PopBack(size_t length);
- // Extends this object with |extra_length| elements at the end. The new
+ // Extends this object with `extra_length` elements at the end. The new
// elements are initialized to zero.
virtual void Extend(size_t extra_length);
- // Inserts |length| elements taken from the array |insert_this| and insert
- // them at |position|. The length of the AudioVector is increased by |length|.
- // |position| = 0 means that the new values are prepended to the vector.
- // |position| = Size() means that the new values are appended to the vector.
+ // Inserts `length` elements taken from the array `insert_this` and insert
+ // them at `position`. The length of the AudioVector is increased by `length`.
+ // `position` = 0 means that the new values are prepended to the vector.
+ // `position` = Size() means that the new values are appended to the vector.
virtual void InsertAt(const int16_t* insert_this,
size_t length,
size_t position);
- // Like InsertAt, but inserts |length| zero elements at |position|.
+ // Like InsertAt, but inserts `length` zero elements at `position`.
virtual void InsertZerosAt(size_t length, size_t position);
- // Overwrites |length| elements of this AudioVector starting from |position|
- // with first values in |AudioVector|. The definition of |position|
- // is the same as for InsertAt(). If |length| and |position| are selected
+ // Overwrites `length` elements of this AudioVector starting from `position`
+ // with first values in `AudioVector`. The definition of `position`
+ // is the same as for InsertAt(). If `length` and `position` are selected
// such that the new data extends beyond the end of the current AudioVector,
// the vector is extended to accommodate the new data.
virtual void OverwriteAt(const AudioVector& insert_this,
size_t length,
size_t position);
- // Overwrites |length| elements of this AudioVector with values taken from the
- // array |insert_this|, starting at |position|. The definition of |position|
- // is the same as for InsertAt(). If |length| and |position| are selected
+ // Overwrites `length` elements of this AudioVector with values taken from the
+ // array `insert_this`, starting at `position`. The definition of `position`
+ // is the same as for InsertAt(). If `length` and `position` are selected
// such that the new data extends beyond the end of the current AudioVector,
// the vector is extended to accommodate the new data.
virtual void OverwriteAt(const int16_t* insert_this,
size_t length,
size_t position);
- // Appends |append_this| to the end of the current vector. Lets the two
- // vectors overlap by |fade_length| samples, and cross-fade linearly in this
+ // Appends `append_this` to the end of the current vector. Lets the two
+ // vectors overlap by `fade_length` samples, and cross-fade linearly in this
// region.
virtual void CrossFade(const AudioVector& append_this, size_t fade_length);
@@ -158,11 +158,11 @@
size_t capacity_; // Allocated number of samples in the array.
- // The index of the first sample in |array_|, except when
+ // The index of the first sample in `array_`, except when
// |begin_index_ == end_index_|, which indicates an empty buffer.
size_t begin_index_;
- // The index of the sample after the last sample in |array_|.
+ // The index of the sample after the last sample in `array_`.
size_t end_index_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioVector);
diff --git a/modules/audio_coding/neteq/audio_vector_unittest.cc b/modules/audio_coding/neteq/audio_vector_unittest.cc
index e39774c..ae9dd88 100644
--- a/modules/audio_coding/neteq/audio_vector_unittest.cc
+++ b/modules/audio_coding/neteq/audio_vector_unittest.cc
@@ -62,7 +62,7 @@
AudioVector vec;
AudioVector vec_copy;
vec.PushBack(array_, array_length());
- vec.CopyTo(&vec_copy); // Copy from |vec| to |vec_copy|.
+ vec.CopyTo(&vec_copy); // Copy from `vec` to `vec_copy`.
ASSERT_EQ(array_length(), vec.Size());
ASSERT_EQ(array_length(), vec_copy.Size());
for (size_t i = 0; i < array_length(); ++i) {
@@ -70,7 +70,7 @@
EXPECT_EQ(array_[i], vec_copy[i]);
}
- // Clear |vec| and verify that it is empty.
+ // Clear `vec` and verify that it is empty.
vec.Clear();
EXPECT_TRUE(vec.Empty());
@@ -178,8 +178,8 @@
int insert_position = 5;
vec.InsertAt(new_array, kNewLength, insert_position);
// Verify that the vector looks as follows:
- // {0, 1, ..., |insert_position| - 1, 100, 101, ..., 100 + kNewLength - 1,
- // |insert_position|, |insert_position| + 1, ..., kLength - 1}.
+ // {0, 1, ..., `insert_position` - 1, 100, 101, ..., 100 + kNewLength - 1,
+ // `insert_position`, `insert_position` + 1, ..., kLength - 1}.
size_t pos = 0;
for (int i = 0; i < insert_position; ++i) {
EXPECT_EQ(array_[i], vec[pos]);
@@ -309,8 +309,8 @@
size_t insert_position = 2;
vec.OverwriteAt(new_array, kNewLength, insert_position);
// Verify that the vector looks as follows:
- // {0, ..., |insert_position| - 1, 100, 101, ..., 100 + kNewLength - 1,
- // |insert_position|, |insert_position| + 1, ..., kLength - 1}.
+ // {0, ..., `insert_position` - 1, 100, 101, ..., 100 + kNewLength - 1,
+ // `insert_position`, `insert_position` + 1, ..., kLength - 1}.
size_t pos = 0;
for (pos = 0; pos < insert_position; ++pos) {
EXPECT_EQ(array_[pos], vec[pos]);
@@ -340,8 +340,8 @@
vec.OverwriteAt(new_array, kNewLength, insert_position);
ASSERT_EQ(array_length() - 2u + kNewLength, vec.Size());
// Verify that the vector looks as follows:
- // {0, ..., |insert_position| - 1, 100, 101, ..., 100 + kNewLength - 1,
- // |insert_position|, |insert_position| + 1, ..., kLength - 1}.
+ // {0, ..., `insert_position` - 1, 100, 101, ..., 100 + kNewLength - 1,
+ // `insert_position`, `insert_position` + 1, ..., kLength - 1}.
int pos = 0;
for (pos = 0; pos < insert_position; ++pos) {
EXPECT_EQ(array_[pos], vec[pos]);
@@ -350,7 +350,7 @@
EXPECT_EQ(new_array[i], vec[pos]);
++pos;
}
- // Verify that we checked to the end of |vec|.
+ // Verify that we checked to the end of `vec`.
EXPECT_EQ(vec.Size(), static_cast<size_t>(pos));
}
@@ -359,7 +359,7 @@
static const size_t kFadeLength = 10;
AudioVector vec1(kLength);
AudioVector vec2(kLength);
- // Set all vector elements to 0 in |vec1| and 100 in |vec2|.
+ // Set all vector elements to 0 in `vec1` and 100 in `vec2`.
for (size_t i = 0; i < kLength; ++i) {
vec1[i] = 0;
vec2[i] = 100;
diff --git a/modules/audio_coding/neteq/background_noise.cc b/modules/audio_coding/neteq/background_noise.cc
index 8f61598..2c95d3b 100644
--- a/modules/audio_coding/neteq/background_noise.cc
+++ b/modules/audio_coding/neteq/background_noise.cc
@@ -108,8 +108,8 @@
if ((sample_energy > 0) &&
(int64_t{5} * residual_energy >= int64_t{16} * sample_energy)) {
// Spectrum is flat enough; save filter parameters.
- // |temp_signal| + |kVecLen| - |kMaxLpcOrder| points at the first of the
- // |kMaxLpcOrder| samples in the residual signal, which will form the
+ // `temp_signal` + `kVecLen` - `kMaxLpcOrder` points at the first of the
+ // `kMaxLpcOrder` samples in the residual signal, which will form the
// filter state for the next noise generation.
SaveParameters(channel_ix, lpc_coefficients,
temp_signal + kVecLen - kMaxLpcOrder, sample_energy,
@@ -117,7 +117,7 @@
filter_params_saved = true;
}
} else {
- // Will only happen if post-decode VAD is disabled and |sample_energy| is
+ // Will only happen if post-decode VAD is disabled and `sample_energy` is
// not low enough. Increase the threshold for update so that it increases
// by a factor 4 in 4 seconds.
IncrementEnergyThreshold(channel_ix, sample_energy);
@@ -264,8 +264,8 @@
parameters.max_energy = sample_energy;
}
- // Set |energy_update_threshold| to no less than 60 dB lower than
- // |max_energy_|. Adding 524288 assures proper rounding.
+ // Set `energy_update_threshold` to no less than 60 dB lower than
+ // `max_energy_`. Adding 524288 assures proper rounding.
int32_t energy_update_threshold = (parameters.max_energy + 524288) >> 20;
if (energy_update_threshold > parameters.energy_update_threshold) {
parameters.energy_update_threshold = energy_update_threshold;
@@ -297,9 +297,9 @@
// Calculate scale and shift factor.
parameters.scale = static_cast<int16_t>(WebRtcSpl_SqrtFloor(residual_energy));
- // Add 13 to the |scale_shift_|, since the random numbers table is in
+ // Add 13 to the `scale_shift_`, since the random numbers table is in
// Q13.
- // TODO(hlundin): Move the "13" to where the |scale_shift_| is used?
+ // TODO(hlundin): Move the "13" to where the `scale_shift_` is used?
parameters.scale_shift =
static_cast<int16_t>(13 + ((kLogResidualLength + norm_shift) / 2));
diff --git a/modules/audio_coding/neteq/background_noise.h b/modules/audio_coding/neteq/background_noise.h
index 631db0d..005b376 100644
--- a/modules/audio_coding/neteq/background_noise.h
+++ b/modules/audio_coding/neteq/background_noise.h
@@ -37,12 +37,12 @@
void Reset();
// Updates the parameter estimates based on the signal currently in the
- // |sync_buffer|, and on the latest decision in |vad| if it is running.
+ // `sync_buffer`, and on the latest decision in `vad` if it is running.
// Returns true if the filter parameters are updated.
bool Update(const AudioMultiVector& sync_buffer, const PostDecodeVad& vad);
// Generates background noise given a random vector and writes the output to
- // |buffer|.
+ // `buffer`.
void GenerateBackgroundNoise(rtc::ArrayView<const int16_t> random_vector,
size_t channel,
int mute_slope,
@@ -50,29 +50,29 @@
size_t num_noise_samples,
int16_t* buffer);
- // Returns |energy_| for |channel|.
+ // Returns `energy_` for `channel`.
int32_t Energy(size_t channel) const;
- // Sets the value of |mute_factor_| for |channel| to |value|.
+ // Sets the value of `mute_factor_` for `channel` to `value`.
void SetMuteFactor(size_t channel, int16_t value);
- // Returns |mute_factor_| for |channel|.
+ // Returns `mute_factor_` for `channel`.
int16_t MuteFactor(size_t channel) const;
- // Returns a pointer to |filter_| for |channel|.
+ // Returns a pointer to `filter_` for `channel`.
const int16_t* Filter(size_t channel) const;
- // Returns a pointer to |filter_state_| for |channel|.
+ // Returns a pointer to `filter_state_` for `channel`.
const int16_t* FilterState(size_t channel) const;
- // Copies |input| to the filter state. Will not copy more than |kMaxLpcOrder|
+ // Copies `input` to the filter state. Will not copy more than `kMaxLpcOrder`
// elements.
void SetFilterState(size_t channel, rtc::ArrayView<const int16_t> input);
- // Returns |scale_| for |channel|.
+ // Returns `scale_` for `channel`.
int16_t Scale(size_t channel) const;
- // Returns |scale_shift_| for |channel|.
+ // Returns `scale_shift_` for `channel`.
int16_t ScaleShift(size_t channel) const;
// Accessors.
@@ -117,7 +117,7 @@
size_t length,
int32_t* auto_correlation) const;
- // Increments the energy threshold by a factor 1 + |kThresholdIncrement|.
+ // Increments the energy threshold by a factor 1 + `kThresholdIncrement`.
void IncrementEnergyThreshold(size_t channel, int32_t sample_energy);
// Updates the filter parameters.
diff --git a/modules/audio_coding/neteq/buffer_level_filter.cc b/modules/audio_coding/neteq/buffer_level_filter.cc
index 8901c01..0ccc7bb 100644
--- a/modules/audio_coding/neteq/buffer_level_filter.cc
+++ b/modules/audio_coding/neteq/buffer_level_filter.cc
@@ -30,10 +30,10 @@
void BufferLevelFilter::Update(size_t buffer_size_samples,
int time_stretched_samples) {
// Filter:
- // |filtered_current_level_| = |level_factor_| * |filtered_current_level_| +
- // (1 - |level_factor_|) * |buffer_size_samples|
- // |level_factor_| and |filtered_current_level_| are in Q8.
- // |buffer_size_samples| is in Q0.
+ // `filtered_current_level_` = `level_factor_` * `filtered_current_level_` +
+ // (1 - `level_factor_`) * `buffer_size_samples`
+ // `level_factor_` and `filtered_current_level_` are in Q8.
+ // `buffer_size_samples` is in Q0.
const int64_t filtered_current_level =
(level_factor_ * int64_t{filtered_current_level_} >> 8) +
(256 - level_factor_) * rtc::dchecked_cast<int64_t>(buffer_size_samples);
diff --git a/modules/audio_coding/neteq/buffer_level_filter.h b/modules/audio_coding/neteq/buffer_level_filter.h
index 218a142..94a3715 100644
--- a/modules/audio_coding/neteq/buffer_level_filter.h
+++ b/modules/audio_coding/neteq/buffer_level_filter.h
@@ -24,8 +24,8 @@
virtual ~BufferLevelFilter() {}
virtual void Reset();
- // Updates the filter. Current buffer size is |buffer_size_samples|.
- // |time_stretched_samples| is subtracted from the filtered value (thus
+ // Updates the filter. Current buffer size is `buffer_size_samples`.
+ // `time_stretched_samples` is subtracted from the filtered value (thus
// bypassing the filter operation).
virtual void Update(size_t buffer_size_samples, int time_stretched_samples);
diff --git a/modules/audio_coding/neteq/buffer_level_filter_unittest.cc b/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
index 63fc83b..6773e96 100644
--- a/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
+++ b/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
@@ -38,7 +38,7 @@
filter.Update(value, 0 /* time_stretched_samples */);
}
// Expect the filtered value to be (theoretically)
- // (1 - (251/256) ^ |times|) * |value|.
+ // (1 - (251/256) ^ `times`) * `value`.
double expected_value_double = (1 - pow(251.0 / 256.0, times)) * value;
int expected_value = static_cast<int>(expected_value_double);
@@ -62,7 +62,7 @@
filter.Update(kValue, 0 /* time_stretched_samples */);
}
// Expect the filtered value to be
- // (1 - (252/256) ^ |kTimes|) * |kValue|.
+ // (1 - (252/256) ^ `kTimes`) * `kValue`.
int expected_value = 15;
EXPECT_EQ(expected_value, filter.filtered_current_level());
@@ -72,7 +72,7 @@
filter.Update(kValue, 0 /* time_stretched_samples */);
}
// Expect the filtered value to be
- // (1 - (253/256) ^ |kTimes|) * |kValue|.
+ // (1 - (253/256) ^ `kTimes`) * `kValue`.
expected_value = 11;
EXPECT_EQ(expected_value, filter.filtered_current_level());
@@ -82,7 +82,7 @@
filter.Update(kValue, 0 /* time_stretched_samples */);
}
// Expect the filtered value to be
- // (1 - (254/256) ^ |kTimes|) * |kValue|.
+ // (1 - (254/256) ^ `kTimes`) * `kValue`.
expected_value = 8;
EXPECT_EQ(expected_value, filter.filtered_current_level());
}
@@ -98,13 +98,13 @@
filter.Update(kValue, 0);
}
// Expect the filtered value to be
- // (1 - (251/256) ^ |kTimes|) * |kValue|.
+ // (1 - (251/256) ^ `kTimes`) * `kValue`.
const int kExpectedValue = 18;
EXPECT_EQ(kExpectedValue, filter.filtered_current_level());
// Update filter again, now with non-zero value for packet length.
// Set the current filtered value to be the input, in order to isolate the
- // impact of |kTimeStretchedSamples|.
+ // impact of `kTimeStretchedSamples`.
filter.Update(filter.filtered_current_level(), kTimeStretchedSamples);
EXPECT_EQ(kExpectedValue - kTimeStretchedSamples,
filter.filtered_current_level());
diff --git a/modules/audio_coding/neteq/comfort_noise.cc b/modules/audio_coding/neteq/comfort_noise.cc
index 7169f06..a2ce888 100644
--- a/modules/audio_coding/neteq/comfort_noise.cc
+++ b/modules/audio_coding/neteq/comfort_noise.cc
@@ -119,8 +119,8 @@
muting_window += muting_window_increment;
unmuting_window += unmuting_window_increment;
}
- // Remove |overlap_length_| samples from the front of |output| since they
- // were mixed into |sync_buffer_| above.
+ // Remove `overlap_length_` samples from the front of `output` since they
+ // were mixed into `sync_buffer_` above.
output->PopFront(overlap_length_);
}
first_call_ = false;
diff --git a/modules/audio_coding/neteq/comfort_noise.h b/modules/audio_coding/neteq/comfort_noise.h
index f748772..6419d39 100644
--- a/modules/audio_coding/neteq/comfort_noise.h
+++ b/modules/audio_coding/neteq/comfort_noise.h
@@ -45,11 +45,11 @@
// Resets the state. Should be called before each new comfort noise period.
void Reset();
- // Update the comfort noise generator with the parameters in |packet|.
+ // Update the comfort noise generator with the parameters in `packet`.
int UpdateParameters(const Packet& packet);
- // Generates |requested_length| samples of comfort noise and writes to
- // |output|. If this is the first in call after Reset (or first after creating
+ // Generates `requested_length` samples of comfort noise and writes to
+ // `output`. If this is the first in call after Reset (or first after creating
// the object), it will also mix in comfort noise at the end of the
// SyncBuffer object provided in the constructor.
int Generate(size_t requested_length, AudioMultiVector* output);
diff --git a/modules/audio_coding/neteq/comfort_noise_unittest.cc b/modules/audio_coding/neteq/comfort_noise_unittest.cc
index b3fbb4e..b436800 100644
--- a/modules/audio_coding/neteq/comfort_noise_unittest.cc
+++ b/modules/audio_coding/neteq/comfort_noise_unittest.cc
@@ -23,7 +23,7 @@
MockDecoderDatabase db;
SyncBuffer sync_buffer(1, 1000);
ComfortNoise cn(fs, &db, &sync_buffer);
- EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
+ EXPECT_CALL(db, Die()); // Called when `db` goes out of scope.
}
// TODO(hlundin): Write more tests.
diff --git a/modules/audio_coding/neteq/cross_correlation.h b/modules/audio_coding/neteq/cross_correlation.h
index 9ce8be8..5082ce6 100644
--- a/modules/audio_coding/neteq/cross_correlation.h
+++ b/modules/audio_coding/neteq/cross_correlation.h
@@ -17,19 +17,19 @@
namespace webrtc {
// The function calculates the cross-correlation between two sequences
-// |sequence_1| and |sequence_2|. |sequence_1| is taken as reference, with
-// |sequence_1_length| as its length. |sequence_2| slides for the calculation of
-// cross-correlation. The result will be saved in |cross_correlation|.
-// |cross_correlation_length| correlation points are calculated.
+// `sequence_1` and `sequence_2`. `sequence_1` is taken as reference, with
+// `sequence_1_length` as its length. `sequence_2` slides for the calculation of
+// cross-correlation. The result will be saved in `cross_correlation`.
+// `cross_correlation_length` correlation points are calculated.
// The corresponding lag starts from 0, and increases with a step of
-// |cross_correlation_step|. The result is without normalization. To avoid
+// `cross_correlation_step`. The result is without normalization. To avoid
// overflow, the result will be right shifted. The amount of shifts will be
// returned.
//
// Input:
// - sequence_1 : First sequence (reference).
// - sequence_2 : Second sequence (sliding during calculation).
-// - sequence_1_length : Length of |sequence_1|.
+// - sequence_1_length : Length of `sequence_1`.
// - cross_correlation_length : Number of cross-correlations to calculate.
// - cross_correlation_step : Step in the lag for the cross-correlation.
//
diff --git a/modules/audio_coding/neteq/decision_logic.cc b/modules/audio_coding/neteq/decision_logic.cc
index edefbe6..ceefe50 100644
--- a/modules/audio_coding/neteq/decision_logic.cc
+++ b/modules/audio_coding/neteq/decision_logic.cc
@@ -309,8 +309,8 @@
std::max(target_level_samples * 3 / 4,
target_level_samples -
kDecelerationTargetLevelOffsetMs * samples_per_ms);
- // |higher_limit| is equal to |target_level|, but should at
- // least be 20 ms higher than |lower_limit|.
+ // `higher_limit` is equal to `target_level`, but should at
+ // least be 20 ms higher than `lower_limit`.
const int high_limit =
std::max(target_level_samples, low_limit + 20 * samples_per_ms);
diff --git a/modules/audio_coding/neteq/decision_logic.h b/modules/audio_coding/neteq/decision_logic.h
index 8be4511..693f616 100644
--- a/modules/audio_coding/neteq/decision_logic.h
+++ b/modules/audio_coding/neteq/decision_logic.h
@@ -47,23 +47,23 @@
void SetSampleRate(int fs_hz, size_t output_size_samples) override;
// 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
+ // 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|; otherwise it should be NULL. The mode
+ // should be supplied in `packet`; otherwise it should be NULL. 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
+ // `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).
NetEq::Operation GetDecision(const NetEqController::NetEqStatus& status,
bool* reset_decoder) override;
- // These methods test the |cng_state_| for different conditions.
+ // These methods test the `cng_state_` for different conditions.
bool CngRfc3389On() const override { return cng_state_ == kCngRfc3389On; }
bool CngOff() const override { return cng_state_ == kCngOff; }
- // Resets the |cng_state_| to kCngOff.
+ // Resets the `cng_state_` to kCngOff.
void SetCngOff() override { cng_state_ = kCngOff; }
// Reports back to DecisionLogic whether the decision to do expand remains or
@@ -72,7 +72,7 @@
// sync buffer.
void ExpandDecision(NetEq::Operation operation) override;
- // Adds |value| to |sample_memory_|.
+ // Adds `value` to `sample_memory_`.
void AddSampleMemory(int32_t value) override { sample_memory_ += value; }
int TargetLevelMs() const override { return delay_manager_->TargetDelayMs(); }
@@ -120,8 +120,8 @@
enum CngState { kCngOff, kCngRfc3389On, kCngInternalOn };
- // Updates the |buffer_level_filter_| with the current buffer level
- // |buffer_size_samples|.
+ // Updates the `buffer_level_filter_` with the current buffer level
+ // `buffer_size_samples`.
void FilterBufferLevel(size_t buffer_size_samples);
// Returns the operation given that the next available packet is a comfort
@@ -132,7 +132,7 @@
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).
+ // a DTMF event, flagged by setting `play_dtmf` true).
virtual NetEq::Operation NoPacket(bool play_dtmf);
// Returns the operation to do given that the expected packet is available.
@@ -160,13 +160,13 @@
// Checks if the current (filtered) buffer level is under the target level.
bool UnderTargetLevel() const;
- // Checks if |timestamp_leap| is so long into the future that a reset due
+ // Checks if `timestamp_leap` is so long into the future that a reset due
// to exceeding kReinitAfterExpands will be done.
bool ReinitAfterExpands(uint32_t timestamp_leap) const;
// Checks if we still have not done enough expands to cover the distance from
// the last decoded packet to the next available packet, the distance beeing
- // conveyed in |timestamp_leap|.
+ // conveyed in `timestamp_leap`.
bool PacketTooEarly(uint32_t timestamp_leap) const;
// Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
diff --git a/modules/audio_coding/neteq/decoder_database.cc b/modules/audio_coding/neteq/decoder_database.cc
index e755e7b..e9176f4 100644
--- a/modules/audio_coding/neteq/decoder_database.cc
+++ b/modules/audio_coding/neteq/decoder_database.cc
@@ -161,7 +161,7 @@
rtp_payload_type,
DecoderInfo(audio_format, codec_pair_id_, decoder_factory_.get())));
if (ret.second == false) {
- // Database already contains a decoder with type |rtp_payload_type|.
+ // Database already contains a decoder with type `rtp_payload_type`.
return kDecoderExists;
}
return kOK;
@@ -169,7 +169,7 @@
int DecoderDatabase::Remove(uint8_t rtp_payload_type) {
if (decoders_.erase(rtp_payload_type) == 0) {
- // No decoder with that |rtp_payload_type|.
+ // No decoder with that `rtp_payload_type`.
return kDecoderNotFound;
}
if (active_decoder_type_ == rtp_payload_type) {
@@ -199,7 +199,7 @@
int DecoderDatabase::SetActiveDecoder(uint8_t rtp_payload_type,
bool* new_decoder) {
- // Check that |rtp_payload_type| exists in the database.
+ // Check that `rtp_payload_type` exists in the database.
const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
if (!info) {
// Decoder not found.
@@ -231,7 +231,7 @@
}
int DecoderDatabase::SetActiveCngDecoder(uint8_t rtp_payload_type) {
- // Check that |rtp_payload_type| exists in the database.
+ // Check that `rtp_payload_type` exists in the database.
const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
if (!info) {
// Decoder not found.
diff --git a/modules/audio_coding/neteq/decoder_database.h b/modules/audio_coding/neteq/decoder_database.h
index e0a3fe3..a63a9cf 100644
--- a/modules/audio_coding/neteq/decoder_database.h
+++ b/modules/audio_coding/neteq/decoder_database.h
@@ -80,15 +80,15 @@
// Returns true if the decoder's format is RED.
bool IsRed() const { return subtype_ == Subtype::kRed; }
- // Returns true if the decoder's format is named |name|.
+ // Returns true if the decoder's format is named `name`.
bool IsType(const char* name) const;
- // Returns true if the decoder's format is named |name|.
+ // Returns true if the decoder's format is named `name`.
bool IsType(const std::string& name) const;
const std::string& get_name() const { return name_; }
private:
- // TODO(ossu): |name_| is kept here while we retain the old external
+ // TODO(ossu): `name_` is kept here while we retain the old external
// decoder interface. Remove this once using an
// AudioDecoderFactory has supplanted the old functionality.
const std::string name_;
@@ -143,26 +143,26 @@
virtual int RegisterPayload(int rtp_payload_type,
const SdpAudioFormat& audio_format);
- // Removes the entry for |rtp_payload_type| from the database.
+ // Removes the entry for `rtp_payload_type` from the database.
// Returns kDecoderNotFound or kOK depending on the outcome of the operation.
virtual int Remove(uint8_t rtp_payload_type);
// Remove all entries.
virtual void RemoveAll();
- // Returns a pointer to the DecoderInfo struct for |rtp_payload_type|. If
- // no decoder is registered with that |rtp_payload_type|, NULL is returned.
+ // Returns a pointer to the DecoderInfo struct for `rtp_payload_type`. If
+ // no decoder is registered with that `rtp_payload_type`, NULL is returned.
virtual const DecoderInfo* GetDecoderInfo(uint8_t rtp_payload_type) const;
- // Sets the active decoder to be |rtp_payload_type|. If this call results in a
- // change of active decoder, |new_decoder| is set to true. The previous active
+ // Sets the active decoder to be `rtp_payload_type`. If this call results in a
+ // change of active decoder, `new_decoder` is set to true. The previous active
// decoder's AudioDecoder object is deleted.
virtual int SetActiveDecoder(uint8_t rtp_payload_type, bool* new_decoder);
// Returns the current active decoder, or NULL if no active decoder exists.
virtual AudioDecoder* GetActiveDecoder() const;
- // Sets the active comfort noise decoder to be |rtp_payload_type|. If this
+ // Sets the active comfort noise decoder to be `rtp_payload_type`. If this
// call results in a change of active comfort noise decoder, the previous
// active decoder's AudioDecoder object is deleted.
virtual int SetActiveCngDecoder(uint8_t rtp_payload_type);
@@ -176,26 +176,26 @@
// exists.
// Returns a pointer to the AudioDecoder object associated with
- // |rtp_payload_type|, or NULL if none is registered. If the AudioDecoder
+ // `rtp_payload_type`, or NULL if none is registered. If the AudioDecoder
// object does not exist for that decoder, the object is created.
AudioDecoder* GetDecoder(uint8_t rtp_payload_type) const;
- // Returns if |rtp_payload_type| is registered with a format named |name|.
+ // Returns if `rtp_payload_type` is registered with a format named `name`.
bool IsType(uint8_t rtp_payload_type, const char* name) const;
- // Returns if |rtp_payload_type| is registered with a format named |name|.
+ // Returns if `rtp_payload_type` is registered with a format named `name`.
bool IsType(uint8_t rtp_payload_type, const std::string& name) const;
- // Returns true if |rtp_payload_type| is registered as comfort noise.
+ // Returns true if `rtp_payload_type` is registered as comfort noise.
bool IsComfortNoise(uint8_t rtp_payload_type) const;
- // Returns true if |rtp_payload_type| is registered as DTMF.
+ // Returns true if `rtp_payload_type` is registered as DTMF.
bool IsDtmf(uint8_t rtp_payload_type) const;
- // Returns true if |rtp_payload_type| is registered as RED.
+ // Returns true if `rtp_payload_type` is registered as RED.
bool IsRed(uint8_t rtp_payload_type) const;
- // Returns kOK if all packets in |packet_list| carry payload types that are
+ // Returns kOK if all packets in `packet_list` carry payload types that are
// registered in the database. Otherwise, returns kDecoderNotFound.
int CheckPayloadTypes(const PacketList& packet_list) const;
diff --git a/modules/audio_coding/neteq/decoder_database_unittest.cc b/modules/audio_coding/neteq/decoder_database_unittest.cc
index 33bee8d..f28a0fd 100644
--- a/modules/audio_coding/neteq/decoder_database_unittest.cc
+++ b/modules/audio_coding/neteq/decoder_database_unittest.cc
@@ -148,7 +148,7 @@
}
PacketList packet_list;
for (int i = 0; i < kNumPayloads + 1; ++i) {
- // Create packet with payload type |i|. The last packet will have a payload
+ // Create packet with payload type `i`. The last packet will have a payload
// type that is not registered in the decoder database.
Packet packet;
packet.payload_type = i;
diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc
index 41de274..c244902 100644
--- a/modules/audio_coding/neteq/delay_manager.cc
+++ b/modules/audio_coding/neteq/delay_manager.cc
@@ -191,7 +191,7 @@
}
}
- // Calculate new |target_level_ms_| based on updated statistics.
+ // Calculate new `target_level_ms_` based on updated statistics.
int bucket_index = histogram_->Quantile(histogram_quantile_);
target_level_ms_ = (1 + bucket_index) * kBucketSizeMs;
target_level_ms_ = std::max(target_level_ms_, effective_minimum_delay_ms_);
@@ -293,7 +293,7 @@
}
bool DelayManager::SetMaximumDelay(int delay_ms) {
- // If |delay_ms| is zero then it unsets the maximum delay and target level is
+ // If `delay_ms` is zero then it unsets the maximum delay and target level is
// unconstrained by maximum delay.
if (delay_ms != 0 &&
(delay_ms < minimum_delay_ms_ || delay_ms < packet_len_ms_)) {
@@ -321,7 +321,7 @@
}
void DelayManager::UpdateEffectiveMinimumDelay() {
- // Clamp |base_minimum_delay_ms_| into the range which can be effectively
+ // Clamp `base_minimum_delay_ms_` into the range which can be effectively
// used.
const int base_minimum_delay_ms =
rtc::SafeClamp(base_minimum_delay_ms_, 0, MinimumDelayUpperBound());
diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index 9832ced..cb35274 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -34,9 +34,9 @@
std::unique_ptr<Histogram> histogram);
// Create a DelayManager object. Notify the delay manager that the packet
- // buffer can hold no more than |max_packets_in_buffer| packets (i.e., this
+ // buffer can hold no more than `max_packets_in_buffer` packets (i.e., this
// is the number of packet slots in the buffer) and that the target delay
- // should be greater than or equal to |base_minimum_delay_ms|. Supply a
+ // should be greater than or equal to `base_minimum_delay_ms`. Supply a
// PeakDetector object to the DelayManager.
static std::unique_ptr<DelayManager> Create(int max_packets_in_buffer,
int base_minimum_delay_ms,
@@ -44,10 +44,10 @@
virtual ~DelayManager();
- // Updates the delay manager with a new incoming packet, with |timestamp| from
+ // Updates the delay manager with a new incoming packet, with `timestamp` from
// the RTP header. This updates the statistics and a new target buffer level
// is calculated. Returns the relative delay if it can be calculated. If
- // |reset| is true, restarts the relative arrival delay calculation from this
+ // `reset` is true, restarts the relative arrival delay calculation from this
// packet.
virtual absl::optional<int> Update(uint32_t timestamp,
int sample_rate_hz,
@@ -63,7 +63,7 @@
virtual int SetPacketAudioLength(int length_ms);
// Accessors and mutators.
- // Assuming |delay| is in valid range.
+ // Assuming `delay` is in valid range.
virtual bool SetMinimumDelay(int delay_ms);
virtual bool SetMaximumDelay(int delay_ms);
virtual bool SetBaseMinimumDelay(int delay_ms);
@@ -78,25 +78,25 @@
private:
// Provides value which minimum delay can't exceed based on current buffer
- // size and given |maximum_delay_ms_|. Lower bound is a constant 0.
+ // size and given `maximum_delay_ms_`. Lower bound is a constant 0.
int MinimumDelayUpperBound() const;
- // Updates |delay_history_|.
+ // Updates `delay_history_`.
void UpdateDelayHistory(int iat_delay_ms,
uint32_t timestamp,
int sample_rate_hz);
- // Calculate relative packet arrival delay from |delay_history_|.
+ // Calculate relative packet arrival delay from `delay_history_`.
int CalculateRelativePacketArrivalDelay() const;
- // Updates |effective_minimum_delay_ms_| delay based on current
- // |minimum_delay_ms_|, |base_minimum_delay_ms_| and |maximum_delay_ms_|
+ // Updates `effective_minimum_delay_ms_` delay based on current
+ // `minimum_delay_ms_`, `base_minimum_delay_ms_` and `maximum_delay_ms_`
// and buffer size.
void UpdateEffectiveMinimumDelay();
- // Makes sure that |delay_ms| is less than maximum delay, if any maximum
- // is set. Also, if possible check |delay_ms| to be less than 75% of
- // |max_packets_in_buffer_|.
+ // Makes sure that `delay_ms` is less than maximum delay, if any maximum
+ // is set. Also, if possible check `delay_ms` to be less than 75% of
+ // `max_packets_in_buffer_`.
bool IsValidMinimumDelay(int delay_ms) const;
bool IsValidBaseMinimumDelay(int delay_ms) const;
diff --git a/modules/audio_coding/neteq/dsp_helper.cc b/modules/audio_coding/neteq/dsp_helper.cc
index 2b1518e..54ec556 100644
--- a/modules/audio_coding/neteq/dsp_helper.cc
+++ b/modules/audio_coding/neteq/dsp_helper.cc
@@ -94,7 +94,7 @@
return factor;
}
int end_factor = 0;
- // Loop over the channels, starting at the same |factor| each time.
+ // Loop over the channels, starting at the same `factor` each time.
for (size_t channel = 0; channel < signal->Channels(); ++channel) {
end_factor =
RampSignal(&(*signal)[channel], start_index, length, factor, increment);
@@ -116,7 +116,7 @@
// Single peak. The parabola fit assumes that an extra point is
// available; worst case it gets a zero on the high end of the signal.
// TODO(hlundin): This can potentially get much worse. It breaks the
- // API contract, that the length of |data| is |data_length|.
+ // API contract, that the length of `data` is `data_length`.
data_length++;
}
diff --git a/modules/audio_coding/neteq/dsp_helper.h b/modules/audio_coding/neteq/dsp_helper.h
index 82fe14e..7bdeba6 100644
--- a/modules/audio_coding/neteq/dsp_helper.h
+++ b/modules/audio_coding/neteq/dsp_helper.h
@@ -51,8 +51,8 @@
static const int kUnmuteFactorIncrement48kHz = 1057;
// Multiplies the signal with a gradually changing factor.
- // The first sample is multiplied with |factor| (in Q14). For each sample,
- // |factor| is increased (additive) by the |increment| (in Q20), which can
+ // The first sample is multiplied with `factor` (in Q14). For each sample,
+ // `factor` is increased (additive) by the `increment` (in Q20), which can
// be negative. Returns the scale factor after the last increment.
static int RampSignal(const int16_t* input,
size_t length,
@@ -60,14 +60,14 @@
int increment,
int16_t* output);
- // Same as above, but with the samples of |signal| being modified in-place.
+ // Same as above, but with the samples of `signal` being modified in-place.
static int RampSignal(int16_t* signal,
size_t length,
int factor,
int increment);
- // Same as above, but processes |length| samples from |signal|, starting at
- // |start_index|.
+ // Same as above, but processes `length` samples from `signal`, starting at
+ // `start_index`.
static int RampSignal(AudioVector* signal,
size_t start_index,
size_t length,
@@ -81,10 +81,10 @@
int factor,
int increment);
- // Peak detection with parabolic fit. Looks for |num_peaks| maxima in |data|,
- // having length |data_length| and sample rate multiplier |fs_mult|. The peak
- // locations and values are written to the arrays |peak_index| and
- // |peak_value|, respectively. Both arrays must hold at least |num_peaks|
+ // Peak detection with parabolic fit. Looks for `num_peaks` maxima in `data`,
+ // having length `data_length` and sample rate multiplier `fs_mult`. The peak
+ // locations and values are written to the arrays `peak_index` and
+ // `peak_value`, respectively. Both arrays must hold at least `num_peaks`
// elements.
static void PeakDetection(int16_t* data,
size_t data_length,
@@ -94,30 +94,30 @@
int16_t* peak_value);
// Estimates the height and location of a maximum. The three values in the
- // array |signal_points| are used as basis for a parabolic fit, which is then
- // used to find the maximum in an interpolated signal. The |signal_points| are
+ // array `signal_points` are used as basis for a parabolic fit, which is then
+ // used to find the maximum in an interpolated signal. The `signal_points` are
// assumed to be from a 4 kHz signal, while the maximum, written to
- // |peak_index| and |peak_value| is given in the full sample rate, as
- // indicated by the sample rate multiplier |fs_mult|.
+ // `peak_index` and `peak_value` is given in the full sample rate, as
+ // indicated by the sample rate multiplier `fs_mult`.
static void ParabolicFit(int16_t* signal_points,
int fs_mult,
size_t* peak_index,
int16_t* peak_value);
- // Calculates the sum-abs-diff for |signal| when compared to a displaced
+ // Calculates the sum-abs-diff for `signal` when compared to a displaced
// version of itself. Returns the displacement lag that results in the minimum
- // distortion. The resulting distortion is written to |distortion_value|.
- // The values of |min_lag| and |max_lag| are boundaries for the search.
+ // distortion. The resulting distortion is written to `distortion_value`.
+ // The values of `min_lag` and `max_lag` are boundaries for the search.
static size_t MinDistortion(const int16_t* signal,
size_t min_lag,
size_t max_lag,
size_t length,
int32_t* distortion_value);
- // Mixes |length| samples from |input1| and |input2| together and writes the
- // result to |output|. The gain for |input1| starts at |mix_factor| (Q14) and
- // is decreased by |factor_decrement| (Q14) for each sample. The gain for
- // |input2| is the complement 16384 - mix_factor.
+ // Mixes `length` samples from `input1` and `input2` together and writes the
+ // result to `output`. The gain for `input1` starts at `mix_factor` (Q14) and
+ // is decreased by `factor_decrement` (Q14) for each sample. The gain for
+ // `input2` is the complement 16384 - mix_factor.
static void CrossFade(const int16_t* input1,
const int16_t* input2,
size_t length,
@@ -125,24 +125,24 @@
int16_t factor_decrement,
int16_t* output);
- // Scales |input| with an increasing gain. Applies |factor| (Q14) to the first
- // sample and increases the gain by |increment| (Q20) for each sample. The
- // result is written to |output|. |length| samples are processed.
+ // Scales `input` with an increasing gain. Applies `factor` (Q14) to the first
+ // sample and increases the gain by `increment` (Q20) for each sample. The
+ // result is written to `output`. `length` samples are processed.
static void UnmuteSignal(const int16_t* input,
size_t length,
int16_t* factor,
int increment,
int16_t* output);
- // Starts at unity gain and gradually fades out |signal|. For each sample,
- // the gain is reduced by |mute_slope| (Q14). |length| samples are processed.
+ // Starts at unity gain and gradually fades out `signal`. For each sample,
+ // the gain is reduced by `mute_slope` (Q14). `length` samples are processed.
static void MuteSignal(int16_t* signal, int mute_slope, size_t length);
- // Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input
- // has |input_length| samples, and the method will write |output_length|
- // samples to |output|. Compensates for the phase delay of the downsampling
- // filters if |compensate_delay| is true. Returns -1 if the input is too short
- // to produce |output_length| samples, otherwise 0.
+ // Downsamples `input` from `sample_rate_hz` to 4 kHz sample rate. The input
+ // has `input_length` samples, and the method will write `output_length`
+ // samples to `output`. Compensates for the phase delay of the downsampling
+ // filters if `compensate_delay` is true. Returns -1 if the input is too short
+ // to produce `output_length` samples, otherwise 0.
static int DownsampleTo4kHz(const int16_t* input,
size_t input_length,
size_t output_length,
diff --git a/modules/audio_coding/neteq/dsp_helper_unittest.cc b/modules/audio_coding/neteq/dsp_helper_unittest.cc
index ec434a4..0924741 100644
--- a/modules/audio_coding/neteq/dsp_helper_unittest.cc
+++ b/modules/audio_coding/neteq/dsp_helper_unittest.cc
@@ -24,7 +24,7 @@
input[i] = 1000;
}
int start_factor = 0;
- // Ramp from 0 to 1 (in Q14) over the array. Note that |increment| is in Q20,
+ // Ramp from 0 to 1 (in Q14) over the array. Note that `increment` is in Q20,
// while the factor is in Q14, hence the shift by 6.
int increment = (16384 << 6) / kLen;
@@ -36,7 +36,7 @@
EXPECT_EQ(1000 * i / kLen, output[i]);
}
- // Test second method. (Note that this modifies |input|.)
+ // Test second method. (Note that this modifies `input`.)
stop_factor = DspHelper::RampSignal(input, kLen, start_factor, increment);
EXPECT_EQ(16383, stop_factor); // Almost reach 1 in Q14.
for (int i = 0; i < kLen; ++i) {
@@ -54,31 +54,31 @@
input[channel][i] = 1000;
}
}
- // We want to start ramping at |start_index| and keep ramping for |kLen|
+ // We want to start ramping at `start_index` and keep ramping for `kLen`
// samples.
int start_index = kLen;
int start_factor = 0;
- // Ramp from 0 to 1 (in Q14) in |kLen| samples. Note that |increment| is in
+ // Ramp from 0 to 1 (in Q14) in `kLen` samples. Note that `increment` is in
// Q20, while the factor is in Q14, hence the shift by 6.
int increment = (16384 << 6) / kLen;
int stop_factor =
DspHelper::RampSignal(&input, start_index, kLen, start_factor, increment);
EXPECT_EQ(16383, stop_factor); // Almost reach 1 in Q14.
- // Verify that the first |kLen| samples are left untouched.
+ // Verify that the first `kLen` samples are left untouched.
int i;
for (i = 0; i < kLen; ++i) {
for (int channel = 0; channel < kChannels; ++channel) {
EXPECT_EQ(1000, input[channel][i]);
}
}
- // Verify that the next block of |kLen| samples are ramped.
+ // Verify that the next block of `kLen` samples are ramped.
for (; i < 2 * kLen; ++i) {
for (int channel = 0; channel < kChannels; ++channel) {
EXPECT_EQ(1000 * (i - kLen) / kLen, input[channel][i]);
}
}
- // Verify the last |kLen| samples are left untouched.
+ // Verify the last `kLen` samples are left untouched.
for (; i < 3 * kLen; ++i) {
for (int channel = 0; channel < kChannels; ++channel) {
EXPECT_EQ(1000, input[channel][i]);
diff --git a/modules/audio_coding/neteq/dtmf_buffer.cc b/modules/audio_coding/neteq/dtmf_buffer.cc
index f81036b..9f78aca 100644
--- a/modules/audio_coding/neteq/dtmf_buffer.cc
+++ b/modules/audio_coding/neteq/dtmf_buffer.cc
@@ -32,7 +32,7 @@
buffer_.clear();
}
-// The ParseEvent method parses 4 bytes from |payload| according to this format
+// The ParseEvent method parses 4 bytes from `payload` according to this format
// from RFC 4733:
//
// 0 1 2 3
@@ -119,8 +119,8 @@
bool DtmfBuffer::GetEvent(uint32_t current_timestamp, DtmfEvent* event) {
DtmfList::iterator it = buffer_.begin();
while (it != buffer_.end()) {
- // |event_end| is an estimate of where the current event ends. If the end
- // bit is set, we know that the event ends at |timestamp| + |duration|.
+ // `event_end` is an estimate of where the current event ends. If the end
+ // bit is set, we know that the event ends at `timestamp` + `duration`.
uint32_t event_end = it->timestamp + it->duration;
#ifdef LEGACY_BITEXACT
bool next_available = false;
@@ -226,7 +226,7 @@
}
}
-// Returns true if |a| goes before |b| in the sorting order ("|a| < |b|").
+// Returns true if `a` goes before `b` in the sorting order ("`a` < `b`").
// The events are ranked using their start timestamp (taking wrap-around into
// account). In the unlikely situation that two events share the same start
// timestamp, the event number is used to rank the two. Note that packets
diff --git a/modules/audio_coding/neteq/dtmf_buffer.h b/modules/audio_coding/neteq/dtmf_buffer.h
index 6bf75e1..9209cae 100644
--- a/modules/audio_coding/neteq/dtmf_buffer.h
+++ b/modules/audio_coding/neteq/dtmf_buffer.h
@@ -45,7 +45,7 @@
kInvalidSampleRate
};
- // Set up the buffer for use at sample rate |fs_hz|.
+ // Set up the buffer for use at sample rate `fs_hz`.
explicit DtmfBuffer(int fs_hz);
virtual ~DtmfBuffer();
@@ -53,21 +53,21 @@
// Flushes the buffer.
virtual void Flush();
- // Static method to parse 4 bytes from |payload| as a DTMF event (RFC 4733)
- // and write the parsed information into the struct |event|. Input variable
- // |rtp_timestamp| is simply copied into the struct.
+ // Static method to parse 4 bytes from `payload` as a DTMF event (RFC 4733)
+ // and write the parsed information into the struct `event`. Input variable
+ // `rtp_timestamp` is simply copied into the struct.
static int ParseEvent(uint32_t rtp_timestamp,
const uint8_t* payload,
size_t payload_length_bytes,
DtmfEvent* event);
- // Inserts |event| into the buffer. The method looks for a matching event and
+ // Inserts `event` into the buffer. The method looks for a matching event and
// merges the two if a match is found.
virtual int InsertEvent(const DtmfEvent& event);
- // Checks if a DTMF event should be played at time |current_timestamp|. If so,
+ // Checks if a DTMF event should be played at time `current_timestamp`. If so,
// the method returns true; otherwise false. The parameters of the event to
- // play will be written to |event|.
+ // play will be written to `event`.
virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event);
// Number of events in the buffer.
@@ -87,7 +87,7 @@
// Compares two events and returns true if they are the same.
static bool SameEvent(const DtmfEvent& a, const DtmfEvent& b);
- // Merges |event| to the event pointed out by |it|. The method checks that
+ // Merges `event` to the event pointed out by `it`. The method checks that
// the two events are the same (using the SameEvent method), and merges them
// if that was the case, returning true. If the events are not the same, false
// is returned.
diff --git a/modules/audio_coding/neteq/dtmf_buffer_unittest.cc b/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
index 607a5ec..83745b6 100644
--- a/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
@@ -208,12 +208,12 @@
DtmfEvent event2(timestamp, event_no, volume, duration, end_bit);
EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event2));
EXPECT_EQ(2u, buffer.Length());
- // Now we expect to get the new event when supplying |timestamp_now|.
+ // Now we expect to get the new event when supplying `timestamp_now`.
EXPECT_TRUE(buffer.GetEvent(timestamp_now, &out_event));
EXPECT_TRUE(EqualEvents(event2, out_event));
// Expect the the first event to be erased now.
EXPECT_EQ(1u, buffer.Length());
- // Move |timestamp_now| to more than 560 samples after the end of the second
+ // Move `timestamp_now` to more than 560 samples after the end of the second
// event. Expect that event to be erased.
timestamp_now = timestamp + duration + 600;
#ifdef LEGACY_BITEXACT
diff --git a/modules/audio_coding/neteq/dtmf_tone_generator.cc b/modules/audio_coding/neteq/dtmf_tone_generator.cc
index 6c412e3..49cbf8f 100644
--- a/modules/audio_coding/neteq/dtmf_tone_generator.cc
+++ b/modules/audio_coding/neteq/dtmf_tone_generator.cc
@@ -167,7 +167,7 @@
initialized_ = false;
}
-// Generate num_samples of DTMF signal and write to |output|.
+// Generate num_samples of DTMF signal and write to `output`.
int DtmfToneGenerator::Generate(size_t num_samples, AudioMultiVector* output) {
if (!initialized_) {
return kNotInitialized;
diff --git a/modules/audio_coding/neteq/expand.cc b/modules/audio_coding/neteq/expand.cc
index 37a08d6..9c32746 100644
--- a/modules/audio_coding/neteq/expand.cc
+++ b/modules/audio_coding/neteq/expand.cc
@@ -167,7 +167,7 @@
}
// Smooth the expanded if it has not been muted to a low amplitude and
- // |current_voice_mix_factor| is larger than 0.5.
+ // `current_voice_mix_factor` is larger than 0.5.
if ((parameters.mute_factor > 819) &&
(parameters.current_voice_mix_factor > 8192)) {
size_t start_ix = sync_buffer_->Size() - overlap_length_;
@@ -197,7 +197,7 @@
}
// Unvoiced part.
- // Filter |scaled_random_vector| through |ar_filter_|.
+ // Filter `scaled_random_vector` through `ar_filter_`.
memcpy(unvoiced_vector - kUnvoicedLpcOrder, parameters.ar_filter_state,
sizeof(int16_t) * kUnvoicedLpcOrder);
int32_t add_constant = 0;
@@ -402,7 +402,7 @@
// Calculate correlation in downsampled domain (4 kHz sample rate).
size_t correlation_length = 51; // TODO(hlundin): Legacy bit-exactness.
- // If it is decided to break bit-exactness |correlation_length| should be
+ // If it is decided to break bit-exactness `correlation_length` should be
// initialized to the return value of Correlation().
Correlation(audio_history.get(), signal_length, correlation_vector);
@@ -417,7 +417,7 @@
best_correlation_index[1] += fs_mult_20;
best_correlation_index[2] += fs_mult_20;
- // Calculate distortion around the |kNumCorrelationCandidates| best lags.
+ // Calculate distortion around the `kNumCorrelationCandidates` best lags.
int distortion_scale = 0;
for (size_t i = 0; i < kNumCorrelationCandidates; i++) {
size_t min_index =
@@ -434,7 +434,7 @@
WebRtcSpl_VectorBitShiftW32ToW16(best_distortion, kNumCorrelationCandidates,
best_distortion_w32, distortion_scale);
- // Find the maximizing index |i| of the cost function
+ // Find the maximizing index `i` of the cost function
// f[i] = best_correlation[i] / best_distortion[i].
int32_t best_ratio = std::numeric_limits<int32_t>::min();
size_t best_index = std::numeric_limits<size_t>::max();
@@ -458,7 +458,7 @@
max_lag_ = std::max(distortion_lag, correlation_lag);
// Calculate the exact best correlation in the range between
- // |correlation_lag| and |distortion_lag|.
+ // `correlation_lag` and `distortion_lag`.
correlation_length = std::max(std::min(distortion_lag + 10, fs_mult_120),
static_cast<size_t>(60 * fs_mult));
@@ -487,7 +487,7 @@
(31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31;
correlation_scale = std::max(0, correlation_scale);
- // Calculate the correlation, store in |correlation_vector2|.
+ // Calculate the correlation, store in `correlation_vector2`.
WebRtcSpl_CrossCorrelation(
correlation_vector2,
&(audio_history[signal_length - correlation_length]),
@@ -537,7 +537,7 @@
}
// Extract the two vectors expand_vector0 and expand_vector1 from
- // |audio_history|.
+ // `audio_history`.
size_t expansion_length = max_lag_ + overlap_length_;
const int16_t* vector1 = &(audio_history[signal_length - expansion_length]);
const int16_t* vector2 = vector1 - distortion_lag;
@@ -594,13 +594,13 @@
expand_lags_[1] = distortion_lag;
expand_lags_[2] = distortion_lag;
} else {
- // |distortion_lag| and |correlation_lag| are not equal; use different
+ // `distortion_lag` and `correlation_lag` are not equal; use different
// combinations of the two.
- // First lag is |distortion_lag| only.
+ // First lag is `distortion_lag` only.
expand_lags_[0] = distortion_lag;
// Second lag is the average of the two.
expand_lags_[1] = (distortion_lag + correlation_lag) / 2;
- // Third lag is the average again, but rounding towards |correlation_lag|.
+ // Third lag is the average again, but rounding towards `correlation_lag`.
if (distortion_lag > correlation_lag) {
expand_lags_[2] = (distortion_lag + correlation_lag - 1) / 2;
} else {
@@ -638,7 +638,7 @@
if (stability != 1) {
// Set first coefficient to 4096 (1.0 in Q12).
parameters.ar_filter[0] = 4096;
- // Set remaining |kUnvoicedLpcOrder| coefficients to zero.
+ // Set remaining `kUnvoicedLpcOrder` coefficients to zero.
WebRtcSpl_MemSetW16(parameters.ar_filter + 1, 0, kUnvoicedLpcOrder);
}
}
@@ -656,7 +656,7 @@
sizeof(int16_t) * noise_length);
} else {
// Only applies to SWB where length could be larger than
- // |kRandomTableSize|.
+ // `kRandomTableSize`.
memcpy(random_vector, RandomVector::kRandomTable,
sizeof(int16_t) * RandomVector::kRandomTableSize);
RTC_DCHECK_LE(noise_length, kMaxSampleRate / 8000 * 120 + 30);
@@ -694,7 +694,7 @@
int32_t unvoiced_energy = WebRtcSpl_DotProductWithScale(
unvoiced_vector, unvoiced_vector, 128, unvoiced_prescale);
- // Normalize |unvoiced_energy| to 28 or 29 bits to preserve sqrt() accuracy.
+ // Normalize `unvoiced_energy` to 28 or 29 bits to preserve sqrt() accuracy.
int16_t unvoiced_scale = WebRtcSpl_NormW32(unvoiced_energy) - 3;
// Make sure we do an odd number of shifts since we already have 7 shifts
// from dividing with 128 earlier. This will make the total scale factor
@@ -715,7 +715,7 @@
// voice_mix_factor = 0;
if (corr_coefficient > 7875) {
int16_t x1, x2, x3;
- // |corr_coefficient| is in Q14.
+ // `corr_coefficient` is in Q14.
x1 = static_cast<int16_t>(corr_coefficient);
x2 = (x1 * x1) >> 14; // Shift 14 to keep result in Q14.
x3 = (x1 * x2) >> 14;
@@ -733,13 +733,13 @@
}
// Calculate muting slope. Reuse value from earlier scaling of
- // |expand_vector0| and |expand_vector1|.
+ // `expand_vector0` and `expand_vector1`.
int16_t slope = amplitude_ratio;
if (slope > 12288) {
// slope > 1.5.
// Calculate (1 - (1 / slope)) / distortion_lag =
// (slope - 1) / (distortion_lag * slope).
- // |slope| is in Q13, so 1 corresponds to 8192. Shift up to Q25 before
+ // `slope` is in Q13, so 1 corresponds to 8192. Shift up to Q25 before
// the division.
// Shift the denominator from Q13 to Q5 before the division. The result of
// the division will then be in Q20.
@@ -757,7 +757,7 @@
parameters.onset = true;
} else {
// Calculate (1 - slope) / distortion_lag.
- // Shift |slope| by 7 to Q20 before the division. The result is in Q20.
+ // Shift `slope` by 7 to Q20 before the division. The result is in Q20.
parameters.mute_slope = WebRtcSpl_DivW32W16(
(8192 - slope) * 128, static_cast<int16_t>(distortion_lag));
if (parameters.voice_mix_factor <= 13107) {
@@ -826,7 +826,7 @@
kDownsampledLength, filter_coefficients, num_coefficients,
downsampling_factor, kFilterDelay);
- // Normalize |downsampled_input| to using all 16 bits.
+ // Normalize `downsampled_input` to using all 16 bits.
int16_t max_value =
WebRtcSpl_MaxAbsValueW16(downsampled_input, kDownsampledLength);
int16_t norm_shift = 16 - WebRtcSpl_NormW32(max_value);
diff --git a/modules/audio_coding/neteq/expand.h b/modules/audio_coding/neteq/expand.h
index 35dee65..2d22b11 100644
--- a/modules/audio_coding/neteq/expand.h
+++ b/modules/audio_coding/neteq/expand.h
@@ -45,7 +45,7 @@
virtual void Reset();
// The main method to produce concealment data. The data is appended to the
- // end of |output|.
+ // end of `output`.
virtual int Process(AudioMultiVector* output);
// Prepare the object to do extra expansion during normal operation following
@@ -56,7 +56,7 @@
// a period of expands.
virtual void SetParametersForMergeAfterExpand();
- // Returns the mute factor for |channel|.
+ // Returns the mute factor for `channel`.
int16_t MuteFactor(size_t channel) const {
RTC_DCHECK_LT(channel, num_channels_);
return channel_parameters_[channel].mute_factor;
@@ -81,7 +81,7 @@
bool TooManyExpands();
- // Analyzes the signal history in |sync_buffer_|, and set up all parameters
+ // Analyzes the signal history in `sync_buffer_`, and set up all parameters
// necessary to produce concealment data.
void AnalyzeSignal(int16_t* random_vector);
@@ -115,9 +115,9 @@
int mute_slope; /* Q20 */
};
- // Calculate the auto-correlation of |input|, with length |input_length|
+ // Calculate the auto-correlation of `input`, with length `input_length`
// samples. The correlation is calculated from a downsampled version of
- // |input|, and is written to |output|.
+ // `input`, and is written to `output`.
void Correlation(const int16_t* input,
size_t input_length,
int16_t* output) const;
diff --git a/modules/audio_coding/neteq/expand_unittest.cc b/modules/audio_coding/neteq/expand_unittest.cc
index 55a8866..9c3264f 100644
--- a/modules/audio_coding/neteq/expand_unittest.cc
+++ b/modules/audio_coding/neteq/expand_unittest.cc
@@ -124,7 +124,7 @@
EXPECT_EQ(0, statistics_.last_outage_duration_samples());
}
expand_.SetParametersForNormalAfterExpand();
- // Convert |sum_output_len_samples| to milliseconds.
+ // Convert `sum_output_len_samples` to milliseconds.
EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples),
statistics_.last_outage_duration_samples());
}
@@ -164,7 +164,7 @@
EXPECT_EQ(0, statistics_.last_outage_duration_samples());
}
expand_.SetParametersForNormalAfterExpand();
- // Convert |sum_output_len_samples| to milliseconds.
+ // Convert `sum_output_len_samples` to milliseconds.
EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples),
statistics_.last_outage_duration_samples());
}
diff --git a/modules/audio_coding/neteq/histogram.cc b/modules/audio_coding/neteq/histogram.cc
index 15a2394..e4b7f10 100644
--- a/modules/audio_coding/neteq/histogram.cc
+++ b/modules/audio_coding/neteq/histogram.cc
@@ -34,42 +34,42 @@
Histogram::~Histogram() {}
// Each element in the vector is first multiplied by the forgetting factor
-// |forget_factor_|. Then the vector element indicated by |iat_packets| is then
-// increased (additive) by 1 - |forget_factor_|. This way, the probability of
-// |value| is slightly increased, while the sum of the histogram remains
+// `forget_factor_`. Then the vector element indicated by `iat_packets` is then
+// increased (additive) by 1 - `forget_factor_`. This way, the probability of
+// `value` is slightly increased, while the sum of the histogram remains
// constant (=1).
// Due to inaccuracies in the fixed-point arithmetic, the histogram may no
// longer sum up to 1 (in Q30) after the update. To correct this, a correction
// term is added or subtracted from the first element (or elements) of the
// vector.
-// The forgetting factor |forget_factor_| is also updated. When the DelayManager
+// The forgetting factor `forget_factor_` is also updated. When the DelayManager
// is reset, the factor is set to 0 to facilitate rapid convergence in the
// beginning. With each update of the histogram, the factor is increased towards
-// the steady-state value |base_forget_factor_|.
+// the steady-state value `base_forget_factor_`.
void Histogram::Add(int value) {
RTC_DCHECK(value >= 0);
RTC_DCHECK(value < static_cast<int>(buckets_.size()));
int vector_sum = 0; // Sum up the vector elements as they are processed.
- // Multiply each element in |buckets_| with |forget_factor_|.
+ // Multiply each element in `buckets_` with `forget_factor_`.
for (int& bucket : buckets_) {
bucket = (static_cast<int64_t>(bucket) * forget_factor_) >> 15;
vector_sum += bucket;
}
// Increase the probability for the currently observed inter-arrival time
- // by 1 - |forget_factor_|. The factor is in Q15, |buckets_| in Q30.
+ // by 1 - `forget_factor_`. The factor is in Q15, `buckets_` in Q30.
// Thus, left-shift 15 steps to obtain result in Q30.
buckets_[value] += (32768 - forget_factor_) << 15;
vector_sum += (32768 - forget_factor_) << 15; // Add to vector sum.
- // |buckets_| should sum up to 1 (in Q30), but it may not due to
+ // `buckets_` should sum up to 1 (in Q30), but it may not due to
// fixed-point rounding errors.
vector_sum -= 1 << 30; // Should be zero. Compensate if not.
if (vector_sum != 0) {
- // Modify a few values early in |buckets_|.
+ // Modify a few values early in `buckets_`.
int flip_sign = vector_sum > 0 ? -1 : 1;
for (int& bucket : buckets_) {
- // Add/subtract 1/16 of the element, but not more than |vector_sum|.
+ // Add/subtract 1/16 of the element, but not more than `vector_sum`.
int correction = flip_sign * std::min(std::abs(vector_sum), bucket >> 4);
bucket += correction;
vector_sum += correction;
@@ -82,8 +82,8 @@
++add_count_;
- // Update |forget_factor_| (changes only during the first seconds after a
- // reset). The factor converges to |base_forget_factor_|.
+ // Update `forget_factor_` (changes only during the first seconds after a
+ // reset). The factor converges to `base_forget_factor_`.
if (start_forget_weight_) {
if (forget_factor_ != base_forget_factor_) {
int old_forget_factor = forget_factor_;
@@ -92,7 +92,7 @@
forget_factor_ =
std::max(0, std::min(base_forget_factor_, forget_factor));
// The histogram is updated recursively by forgetting the old histogram
- // with |forget_factor_| and adding a new sample multiplied by |1 -
+ // with `forget_factor_` and adding a new sample multiplied by |1 -
// forget_factor_|. We need to make sure that the effective weight on the
// new sample is no smaller than those on the old samples, i.e., to
// satisfy the following DCHECK.
@@ -106,21 +106,21 @@
int Histogram::Quantile(int probability) {
// Find the bucket for which the probability of observing an
- // inter-arrival time larger than or equal to |index| is larger than or
- // equal to |probability|. The sought probability is estimated using
+ // inter-arrival time larger than or equal to `index` is larger than or
+ // equal to `probability`. The sought probability is estimated using
// the histogram as the reverse cumulant PDF, i.e., the sum of elements from
- // the end up until |index|. Now, since the sum of all elements is 1
+ // the end up until `index`. Now, since the sum of all elements is 1
// (in Q30) by definition, and since the solution is often a low value for
- // |iat_index|, it is more efficient to start with |sum| = 1 and subtract
+ // `iat_index`, it is more efficient to start with `sum` = 1 and subtract
// elements from the start of the histogram.
int inverse_probability = (1 << 30) - probability;
- size_t index = 0; // Start from the beginning of |buckets_|.
+ size_t index = 0; // Start from the beginning of `buckets_`.
int sum = 1 << 30; // Assign to 1 in Q30.
sum -= buckets_[index];
while ((sum > inverse_probability) && (index < buckets_.size() - 1)) {
// Subtract the probabilities one by one until the sum is no longer greater
- // than |inverse_probability|.
+ // than `inverse_probability`.
++index;
sum -= buckets_[index];
}
diff --git a/modules/audio_coding/neteq/histogram.h b/modules/audio_coding/neteq/histogram.h
index 0567e3f..5b2f2b1 100644
--- a/modules/audio_coding/neteq/histogram.h
+++ b/modules/audio_coding/neteq/histogram.h
@@ -21,7 +21,7 @@
class Histogram {
public:
- // Creates histogram with capacity |num_buckets| and |forget_factor| in Q15.
+ // Creates histogram with capacity `num_buckets` and `forget_factor` in Q15.
Histogram(size_t num_buckets,
int forget_factor,
absl::optional<double> start_forget_weight = absl::nullopt);
@@ -31,10 +31,10 @@
// Resets the histogram to the default start distribution.
virtual void Reset();
- // Add entry in bucket |index|.
+ // Add entry in bucket `index`.
virtual void Add(int index);
- // Calculates the quantile at |probability| (in Q30) of the histogram
+ // Calculates the quantile at `probability` (in Q30) of the histogram
// distribution.
virtual int Quantile(int probability);
diff --git a/modules/audio_coding/neteq/merge.cc b/modules/audio_coding/neteq/merge.cc
index 07d8722..ca5ec22 100644
--- a/modules/audio_coding/neteq/merge.cc
+++ b/modules/audio_coding/neteq/merge.cc
@@ -149,13 +149,13 @@
(*output)[channel].OverwriteAt(temp_data_.data(), output_length, 0);
}
- // Copy back the first part of the data to |sync_buffer_| and remove it from
- // |output|.
+ // Copy back the first part of the data to `sync_buffer_` and remove it from
+ // `output`.
sync_buffer_->ReplaceAtIndex(*output, old_length, sync_buffer_->next_index());
output->PopFront(old_length);
- // Return new added length. |old_length| samples were borrowed from
- // |sync_buffer_|.
+ // Return new added length. `old_length` samples were borrowed from
+ // `sync_buffer_`.
RTC_DCHECK_GE(output_length, old_length);
return output_length - old_length;
}
@@ -200,7 +200,7 @@
// Append one more pitch period each time.
expanded_.PushBack(expanded_temp);
}
- // Trim the length to exactly |required_length|.
+ // Trim the length to exactly `required_length`.
expanded_.PopBack(expanded_.Size() - required_length);
}
RTC_DCHECK_GE(expanded_.Size(), required_length);
@@ -240,17 +240,17 @@
// Calculate muting factor to use for new frame.
int16_t mute_factor;
if (energy_input > energy_expanded) {
- // Normalize |energy_input| to 14 bits.
+ // Normalize `energy_input` to 14 bits.
int16_t temp_shift = WebRtcSpl_NormW32(energy_input) - 17;
energy_input = WEBRTC_SPL_SHIFT_W32(energy_input, temp_shift);
- // Put |energy_expanded| in a domain 14 higher, so that
+ // Put `energy_expanded` in a domain 14 higher, so that
// energy_expanded / energy_input is in Q14.
energy_expanded = WEBRTC_SPL_SHIFT_W32(energy_expanded, temp_shift + 14);
// Calculate sqrt(energy_expanded / energy_input) in Q14.
mute_factor = static_cast<int16_t>(
WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14));
} else {
- // Set to 1 (in Q14) when |expanded| has higher energy than |input|.
+ // Set to 1 (in Q14) when `expanded` has higher energy than `input`.
mute_factor = 16384;
}
@@ -295,7 +295,7 @@
// there is not much we can do.
const size_t temp_len =
input_length > signal_offset ? input_length - signal_offset : 0;
- // TODO(hlundin): Should |downsamp_temp_len| be corrected for round-off
+ // TODO(hlundin): Should `downsamp_temp_len` be corrected for round-off
// errors? I.e., (temp_len + decimation_factor - 1) / decimation_factor?
size_t downsamp_temp_len = temp_len / decimation_factor;
if (downsamp_temp_len > 0) {
@@ -351,8 +351,8 @@
// Downscale starting index to 4kHz domain. (fs_mult_ * 2 = fs_hz_ / 4000.)
size_t start_index_downsamp = start_index / (fs_mult_ * 2);
- // Calculate a modified |stop_position_downsamp| to account for the increased
- // start index |start_index_downsamp| and the effective array length.
+ // Calculate a modified `stop_position_downsamp` to account for the increased
+ // start index `start_index_downsamp` and the effective array length.
size_t modified_stop_pos =
std::min(stop_position_downsamp,
kMaxCorrelationLength + pad_length - start_index_downsamp);
diff --git a/modules/audio_coding/neteq/merge.h b/modules/audio_coding/neteq/merge.h
index a062a95..13aa31d 100644
--- a/modules/audio_coding/neteq/merge.h
+++ b/modules/audio_coding/neteq/merge.h
@@ -37,10 +37,10 @@
virtual ~Merge();
// The main method to produce the audio data. The decoded data is supplied in
- // |input|, having |input_length| samples in total for all channels
- // (interleaved). The result is written to |output|. The number of channels
- // allocated in |output| defines the number of channels that will be used when
- // de-interleaving |input|.
+ // `input`, having `input_length` samples in total for all channels
+ // (interleaved). The result is written to `output`. The number of channels
+ // allocated in `output` defines the number of channels that will be used when
+ // de-interleaving `input`.
virtual size_t Process(int16_t* input,
size_t input_length,
AudioMultiVector* output);
@@ -57,29 +57,29 @@
static const size_t kInputDownsampLength = 40;
static const size_t kMaxCorrelationLength = 60;
- // Calls |expand_| to get more expansion data to merge with. The data is
- // written to |expanded_signal_|. Returns the length of the expanded data,
- // while |expand_period| will be the number of samples in one expansion period
- // (typically one pitch period). The value of |old_length| will be the number
- // of samples that were taken from the |sync_buffer_|.
+ // Calls `expand_` to get more expansion data to merge with. The data is
+ // written to `expanded_signal_`. Returns the length of the expanded data,
+ // while `expand_period` will be the number of samples in one expansion period
+ // (typically one pitch period). The value of `old_length` will be the number
+ // of samples that were taken from the `sync_buffer_`.
size_t GetExpandedSignal(size_t* old_length, size_t* expand_period);
- // Analyzes |input| and |expanded_signal| and returns muting factor (Q14) to
+ // Analyzes `input` and `expanded_signal` and returns muting factor (Q14) to
// be used on the new data.
int16_t SignalScaling(const int16_t* input,
size_t input_length,
const int16_t* expanded_signal) const;
- // Downsamples |input| (|input_length| samples) and |expanded_signal| to
+ // Downsamples `input` (`input_length` samples) and `expanded_signal` to
// 4 kHz sample rate. The downsampled signals are written to
- // |input_downsampled_| and |expanded_downsampled_|, respectively.
+ // `input_downsampled_` and `expanded_downsampled_`, respectively.
void Downsample(const int16_t* input,
size_t input_length,
const int16_t* expanded_signal,
size_t expanded_length);
- // Calculates cross-correlation between |input_downsampled_| and
- // |expanded_downsampled_|, and finds the correlation maximum. The maximizing
+ // Calculates cross-correlation between `input_downsampled_` and
+ // `expanded_downsampled_`, and finds the correlation maximum. The maximizing
// lag is returned.
size_t CorrelateAndPeakSearch(size_t start_position,
size_t input_length,
diff --git a/modules/audio_coding/neteq/nack_tracker.cc b/modules/audio_coding/neteq/nack_tracker.cc
index 8d94306..9f04534 100644
--- a/modules/audio_coding/neteq/nack_tracker.cc
+++ b/modules/audio_coding/neteq/nack_tracker.cc
@@ -123,7 +123,7 @@
IsNewerSequenceNumber(sequence_number_current_received_rtp,
sequence_num_last_decoded_rtp_));
- // Packets with sequence numbers older than |upper_bound_missing| are
+ // Packets with sequence numbers older than `upper_bound_missing` are
// considered missing, and the rest are considered late.
uint16_t upper_bound_missing =
sequence_number_current_received_rtp - nack_threshold_packets_;
diff --git a/modules/audio_coding/neteq/nack_tracker.h b/modules/audio_coding/neteq/nack_tracker.h
index 5a56734..ac0a77f 100644
--- a/modules/audio_coding/neteq/nack_tracker.h
+++ b/modules/audio_coding/neteq/nack_tracker.h
@@ -63,9 +63,9 @@
// Set a maximum for the size of the NACK list. If the last received packet
// has sequence number of N, then NACK list will not contain any element
- // with sequence number earlier than N - |max_nack_list_size|.
+ // with sequence number earlier than N - `max_nack_list_size`.
//
- // The largest maximum size is defined by |kNackListSizeLimit|
+ // The largest maximum size is defined by `kNackListSizeLimit`
void SetMaxNackListSize(size_t max_nack_list_size);
// Set the sampling rate.
@@ -90,7 +90,7 @@
std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
// Reset to default values. The NACK list is cleared.
- // |nack_threshold_packets_| & |max_nack_list_size_| preserve their values.
+ // `nack_threshold_packets_` & `max_nack_list_size_` preserve their values.
void Reset();
private:
@@ -110,7 +110,7 @@
int64_t time_to_play_ms;
// A guess about the timestamp of the missing packet, it is used for
- // estimation of |time_to_play_ms|. The estimate might be slightly wrong if
+ // estimation of `time_to_play_ms`. The estimate might be slightly wrong if
// there has been frame-size change since the last received packet and the
// missing packet. However, the risk of this is low, and in case of such
// errors, there will be a minor misestimation in time-to-play of missing
@@ -139,7 +139,7 @@
// computed correctly.
NackList GetNackList() const;
- // Given the |sequence_number_current_received_rtp| of currently received RTP,
+ // Given the `sequence_number_current_received_rtp` of currently received RTP,
// recognize packets which are not arrive and add to the list.
void AddToList(uint16_t sequence_number_current_received_rtp);
@@ -147,23 +147,23 @@
// This is called when 10 ms elapsed with no new RTP packet decoded.
void UpdateEstimatedPlayoutTimeBy10ms();
- // Given the |sequence_number_current_received_rtp| and
- // |timestamp_current_received_rtp| of currently received RTP update number
+ // Given the `sequence_number_current_received_rtp` and
+ // `timestamp_current_received_rtp` of currently received RTP update number
// of samples per packet.
void UpdateSamplesPerPacket(uint16_t sequence_number_current_received_rtp,
uint32_t timestamp_current_received_rtp);
- // Given the |sequence_number_current_received_rtp| of currently received RTP
+ // Given the `sequence_number_current_received_rtp` of currently received RTP
// update the list. That is; some packets will change from late to missing,
// some packets are inserted as missing and some inserted as late.
void UpdateList(uint16_t sequence_number_current_received_rtp);
// Packets which are considered late for too long (according to
- // |nack_threshold_packets_|) are flagged as missing.
+ // `nack_threshold_packets_`) are flagged as missing.
void ChangeFromLateToMissing(uint16_t sequence_number_current_received_rtp);
// Packets which have sequence number older that
- // |sequence_num_last_received_rtp_| - |max_nack_list_size_| are removed
+ // `sequence_num_last_received_rtp_` - `max_nack_list_size_` are removed
// from the NACK list.
void LimitNackListSize();
@@ -173,9 +173,9 @@
// Compute time-to-play given a timestamp.
int64_t TimeToPlay(uint32_t timestamp) const;
- // If packet N is arrived, any packet prior to N - |nack_threshold_packets_|
+ // If packet N is arrived, any packet prior to N - `nack_threshold_packets_`
// which is not arrived is considered missing, and should be in NACK list.
- // Also any packet in the range of N-1 and N - |nack_threshold_packets_|,
+ // Also any packet in the range of N-1 and N - `nack_threshold_packets_`,
// exclusive, which is not arrived is considered late, and should should be
// in the list of late packets.
const int nack_threshold_packets_;
@@ -202,7 +202,7 @@
NackList nack_list_;
// NACK list will not keep track of missing packets prior to
- // |sequence_num_last_received_rtp_| - |max_nack_list_size_|.
+ // `sequence_num_last_received_rtp_` - `max_nack_list_size_`.
size_t max_nack_list_size_;
};
diff --git a/modules/audio_coding/neteq/nack_tracker_unittest.cc b/modules/audio_coding/neteq/nack_tracker_unittest.cc
index a44f41b..3f5a05b 100644
--- a/modules/audio_coding/neteq/nack_tracker_unittest.cc
+++ b/modules/audio_coding/neteq/nack_tracker_unittest.cc
@@ -215,10 +215,10 @@
std::unique_ptr<NackTracker> nack(NackTracker::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
- // Sequence number wrap around if |k| is 2 or 3;
+ // Sequence number wrap around if `k` is 2 or 3;
int seq_num_offset = (k < 2) ? 0 : 65531;
- // Timestamp wrap around if |k| is 1 or 3.
+ // Timestamp wrap around if `k` is 1 or 3.
uint32_t timestamp_offset =
(k & 0x1) ? static_cast<uint32_t>(0xffffffff) - 6 : 0;
@@ -283,7 +283,7 @@
TEST(NackTrackerTest,
MissingPacketsPriorToLastDecodedRtpShouldNotBeInNackList) {
for (int m = 0; m < 2; ++m) {
- uint16_t seq_num_offset = (m == 0) ? 0 : 65531; // Wrap around if |m| is 1.
+ uint16_t seq_num_offset = (m == 0) ? 0 : 65531; // Wrap around if `m` is 1.
std::unique_ptr<NackTracker> nack(NackTracker::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
@@ -361,7 +361,7 @@
TEST(NackTrackerTest, ListSizeAppliedFromBeginning) {
const size_t kNackListSize = 10;
for (int m = 0; m < 2; ++m) {
- uint16_t seq_num_offset = (m == 0) ? 0 : 65525; // Wrap around if |m| is 1.
+ uint16_t seq_num_offset = (m == 0) ? 0 : 65525; // Wrap around if `m` is 1.
std::unique_ptr<NackTracker> nack(NackTracker::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
nack->SetMaxNackListSize(kNackListSize);
@@ -385,7 +385,7 @@
TEST(NackTrackerTest, ChangeOfListSizeAppliedAndOldElementsRemoved) {
const size_t kNackListSize = 10;
for (int m = 0; m < 2; ++m) {
- uint16_t seq_num_offset = (m == 0) ? 0 : 65525; // Wrap around if |m| is 1.
+ uint16_t seq_num_offset = (m == 0) ? 0 : 65525; // Wrap around if `m` is 1.
std::unique_ptr<NackTracker> nack(NackTracker::Create(kNackThreshold));
nack->UpdateSampleRate(kSampleRateHz);
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 7225227..fce857f 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -608,7 +608,7 @@
// Reinitialize NetEq if it's needed (changed SSRC or first call).
if (update_sample_rate_and_channels) {
- // Note: |first_packet_| will be cleared further down in this method, once
+ // Note: `first_packet_` will be cleared further down in this method, once
// the packet has been successfully inserted into the packet buffer.
// Flush the packet buffer and DTMF buffer.
@@ -784,8 +784,8 @@
}
if (update_sample_rate_and_channels && !packet_buffer_->Empty()) {
- // We do not use |current_rtp_payload_type_| to |set payload_type|, but
- // get the next RTP header from |packet_buffer_| to obtain the payload type.
+ // We do not use `current_rtp_payload_type_` to |set payload_type|, but
+ // get the next RTP header from `packet_buffer_` to obtain the payload type.
// The reason for it is the following corner case. If NetEq receives a
// CNG packet with a sample rate different than the current CNG then it
// flushes its buffer, assuming send codec must have been changed. However,
@@ -978,18 +978,18 @@
comfort_noise_->Reset();
}
- // We treat it as if all packets referenced to by |last_decoded_packet_infos_|
- // were mashed together when creating the samples in |algorithm_buffer_|.
+ // We treat it as if all packets referenced to by `last_decoded_packet_infos_`
+ // were mashed together when creating the samples in `algorithm_buffer_`.
RtpPacketInfos packet_infos(last_decoded_packet_infos_);
- // Copy samples from |algorithm_buffer_| to |sync_buffer_|.
+ // Copy samples from `algorithm_buffer_` to `sync_buffer_`.
//
// TODO(bugs.webrtc.org/10757):
- // We would in the future also like to pass |packet_infos| so that we can do
- // sample-perfect tracking of that information across |sync_buffer_|.
+ // We would in the future also like to pass `packet_infos` so that we can do
+ // sample-perfect tracking of that information across `sync_buffer_`.
sync_buffer_->PushBack(*algorithm_buffer_);
- // Extract data from |sync_buffer_| to |output|.
+ // Extract data from `sync_buffer_` to `output`.
size_t num_output_samples_per_channel = output_size_samples_;
size_t num_output_samples = output_size_samples_ * sync_buffer_->Channels();
if (num_output_samples > AudioFrame::kMaxDataSizeSamples) {
@@ -1006,14 +1006,14 @@
audio_frame->sample_rate_hz_ = fs_hz_;
// TODO(bugs.webrtc.org/10757):
// We don't have the ability to properly track individual packets once their
- // audio samples have entered |sync_buffer_|. So for now, treat it as if
- // |packet_infos| from packets decoded by the current |GetAudioInternal()|
+ // audio samples have entered `sync_buffer_`. So for now, treat it as if
+ // `packet_infos` from packets decoded by the current `GetAudioInternal()`
// call were all consumed assembling the current audio frame and the current
// audio frame only.
audio_frame->packet_infos_ = std::move(packet_infos);
if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
- // The sync buffer should always contain |overlap_length| samples, but now
- // too many samples have been extracted. Reinstall the |overlap_length|
+ // The sync buffer should always contain `overlap_length` samples, but now
+ // too many samples have been extracted. Reinstall the `overlap_length`
// lookahead by moving the index.
const size_t missing_lookahead_samples =
expand_->overlap_length() - sync_buffer_->FutureLength();
@@ -1031,7 +1031,7 @@
return kSampleUnderrun;
}
- // Should always have overlap samples left in the |sync_buffer_|.
+ // Should always have overlap samples left in the `sync_buffer_`.
RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
// TODO(yujo): For muted frames, this can be a copy rather than an addition.
@@ -1041,7 +1041,7 @@
}
// Update the background noise parameters if last operation wrote data
- // straight from the decoder to the |sync_buffer_|. That is, none of the
+ // 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_ == Mode::kNormal) || (last_mode_ == Mode::kAccelerateFail) ||
(last_mode_ == Mode::kPreemptiveExpandFail) ||
@@ -1051,14 +1051,14 @@
}
if (operation == Operation::kDtmf) {
- // DTMF data was written the end of |sync_buffer_|.
- // Update index to end of DTMF data in |sync_buffer_|.
+ // 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_ != 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
+ // 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".
uint32_t temp_timestamp =
sync_buffer_->end_timestamp() -
@@ -1067,7 +1067,7 @@
playout_timestamp_ = temp_timestamp;
}
} else {
- // Use dead reckoning to estimate the |playout_timestamp_|.
+ // Use dead reckoning to estimate the `playout_timestamp_`.
playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
}
// Set the timestamp in the audio frame to zero before the first packet has
@@ -1206,7 +1206,7 @@
// Use the provided action instead of the decision NetEq decided on.
*operation = *action_override;
}
- // Check if we already have enough samples in the |sync_buffer_|. If so,
+ // Check if we already have enough samples in the `sync_buffer_`. If so,
// change decision to normal, unless the decision was merge, accelerate, or
// preemptive expand.
if (samples_left >= rtc::dchecked_cast<int>(output_size_samples_) &&
@@ -1245,7 +1245,7 @@
*operation = Operation::kNormal;
}
}
- // Adjust |sync_buffer_| timestamp before setting |end_timestamp| to the
+ // Adjust `sync_buffer_` timestamp before setting `end_timestamp` to the
// new value.
sync_buffer_->IncreaseEndTimestamp(timestamp_ - end_timestamp);
end_timestamp = timestamp_;
@@ -1535,7 +1535,7 @@
while (!packet_list->empty() && !decoder_database_->IsComfortNoise(
packet_list->front().payload_type)) {
RTC_DCHECK(decoder); // At this point, we must have a decoder object.
- // The number of channels in the |sync_buffer_| should be the same as the
+ // The number of channels in the `sync_buffer_` should be the same as the
// number decoder channels.
RTC_DCHECK_EQ(sync_buffer_->Channels(), decoder->Channels());
RTC_DCHECK_GE(decoded_buffer_length_, kMaxFrameSize * decoder->Channels());
@@ -1557,7 +1557,7 @@
*speech_type = result.speech_type;
if (result.num_decoded_samples > 0) {
*decoded_length += rtc::dchecked_cast<int>(result.num_decoded_samples);
- // Update |decoder_frame_length_| with number of samples per channel.
+ // Update `decoder_frame_length_` with number of samples per channel.
decoder_frame_length_ =
result.num_decoded_samples / decoder->Channels();
}
@@ -1733,7 +1733,7 @@
size_t num_channels = algorithm_buffer_->Channels();
size_t decoded_length_per_channel = decoded_length / num_channels;
if (decoded_length_per_channel < required_samples) {
- // Must move data from the |sync_buffer_| in order to get 30 ms.
+ // Must move data from the `sync_buffer_` in order to get 30 ms.
borrowed_samples_per_channel =
static_cast<int>(required_samples - decoded_length_per_channel);
memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
@@ -1765,7 +1765,7 @@
}
if (borrowed_samples_per_channel > 0) {
- // Copy borrowed samples back to the |sync_buffer_|.
+ // Copy borrowed samples back to the `sync_buffer_`.
size_t length = algorithm_buffer_->Size();
if (length < borrowed_samples_per_channel) {
// This destroys the beginning of the buffer, but will not cause any
@@ -1806,7 +1806,7 @@
size_t old_borrowed_samples_per_channel = 0;
size_t decoded_length_per_channel = decoded_length / num_channels;
if (decoded_length_per_channel < required_samples) {
- // Must move data from the |sync_buffer_| in order to get 30 ms.
+ // Must move data from the `sync_buffer_` in order to get 30 ms.
borrowed_samples_per_channel =
required_samples - decoded_length_per_channel;
// Calculate how many of these were already played out.
@@ -1843,7 +1843,7 @@
}
if (borrowed_samples_per_channel > 0) {
- // Copy borrowed samples back to the |sync_buffer_|.
+ // Copy borrowed samples back to the `sync_buffer_`.
sync_buffer_->ReplaceAtIndex(
*algorithm_buffer_, borrowed_samples_per_channel,
sync_buffer_->Size() - borrowed_samples_per_channel);
@@ -1903,10 +1903,10 @@
}
int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
- // This block of the code and the block further down, handling |dtmf_switch|
+ // This block of the code and the block further down, handling `dtmf_switch`
// are commented out. Otherwise playing out-of-band DTMF would fail in VoE
// test, DtmfTest.ManualSuccessfullySendsOutOfBandTelephoneEvents. This is
- // equivalent to |dtmf_switch| always be false.
+ // equivalent to `dtmf_switch` always be false.
//
// See http://webrtc-codereview.appspot.com/1195004/ for discussion
// On this issue. This change might cause some glitches at the point of
@@ -1916,7 +1916,7 @@
// if ((last_mode_ != Modes::kDtmf) &&
// dtmf_tone_generator_->initialized()) {
// // Special case; see below.
- // // We must catch this before calling Generate, since |initialized| is
+ // // We must catch this before calling Generate, since `initialized` is
// // modified in that call.
// dtmf_switch = true;
// }
@@ -1948,7 +1948,7 @@
// // TODO(hlundin): This code seems incorrect. (Legacy.) Write test and
// // verify correct operation.
// RTC_NOTREACHED();
- // // Must generate enough data to replace all of the |sync_buffer_|
+ // // Must generate enough data to replace all of the `sync_buffer_`
// // "future".
// int required_length = sync_buffer_->FutureLength();
// RTC_DCHECK(dtmf_tone_generator_->initialized());
@@ -2033,7 +2033,7 @@
do {
timestamp_ = next_packet->timestamp;
absl::optional<Packet> packet = packet_buffer_->GetNextPacket();
- // |next_packet| may be invalid after the |packet_buffer_| operation.
+ // `next_packet` may be invalid after the `packet_buffer_` operation.
next_packet = nullptr;
if (!packet) {
RTC_LOG(LS_ERROR) << "Should always be able to extract a packet here";
@@ -2180,7 +2180,7 @@
comfort_noise_.reset(
new ComfortNoise(fs_hz, decoder_database_.get(), sync_buffer_.get()));
- // Verify that |decoded_buffer_| is long enough.
+ // Verify that `decoded_buffer_` is long enough.
if (decoded_buffer_length_ < kMaxFrameSize * channels) {
// Reallocate to larger size.
decoded_buffer_length_ = kMaxFrameSize * channels;
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 88da6dc..e3d84b3 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -141,7 +141,7 @@
bool RegisterPayloadType(int rtp_payload_type,
const SdpAudioFormat& audio_format) override;
- // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
+ // Removes `rtp_payload_type` from the codec database. Returns 0 on success,
// -1 on failure.
int RemovePayloadType(uint8_t rtp_payload_type) override;
@@ -159,7 +159,7 @@
int FilteredCurrentDelayMs() const override;
- // Writes the current network statistics to |stats|. The statistics are reset
+ // Writes the current network statistics to `stats`. The statistics are reset
// after the call.
int NetworkStatistics(NetEqNetworkStatistics* stats) override;
@@ -215,7 +215,7 @@
rtc::ArrayView<const uint8_t> payload)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
- // Delivers 10 ms of audio data. The data is written to |audio_frame|.
+ // Delivers 10 ms of audio data. The data is written to `audio_frame`.
// Returns 0 on success, otherwise an error code.
int GetAudioInternal(AudioFrame* audio_frame,
bool* muted,
@@ -223,9 +223,9 @@
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Provides a decision to the GetAudioInternal method. The decision what to
- // do is written to |operation|. Packets to decode are written to
- // |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.
+ // do is written to `operation`. Packets to decode are written to
+ // `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(Operation* operation,
PacketList* packet_list,
@@ -234,11 +234,11 @@
absl::optional<Operation> action_override)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
- // Decodes the speech packets in |packet_list|, and writes the results to
- // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
- // elements. The length of the decoded data is written to |decoded_length|.
+ // Decodes the speech packets in `packet_list`, and writes the results to
+ // `decoded_buffer`, which is allocated to hold `decoded_buffer_length`
+ // elements. The length of the decoded data is written to `decoded_length`.
// The speech type -- speech or (codec-internal) comfort noise -- is written
- // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
+ // to `speech_type`. If `packet_list` contains any SID frames for RFC 3389
// comfort noise, those are not decoded.
int Decode(PacketList* packet_list,
Operation* operation,
@@ -293,7 +293,7 @@
bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
- // noise. |packet_list| can either contain one SID frame to update the
+ // noise. `packet_list` can either contain one SID frame to update the
// noise parameters, or no payload at all, in which case the previously
// received parameters are used.
int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
@@ -308,20 +308,20 @@
int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
- // Overdub DTMF on top of |output|.
+ // Overdub DTMF on top of `output`.
int DtmfOverdub(const DtmfEvent& dtmf_event,
size_t num_channels,
int16_t* output) const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
- // Extracts packets from |packet_buffer_| to produce at least
- // |required_samples| samples. The packets are inserted into |packet_list|.
+ // Extracts packets from `packet_buffer_` to produce at least
+ // `required_samples` samples. The packets are inserted into `packet_list`.
// Returns the number of samples that the packets in the list will produce, or
// -1 in case of an error.
int ExtractPackets(size_t required_samples, PacketList* packet_list)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Resets various variables and objects to new values based on the sample rate
- // |fs_hz| and |channels| number audio channels.
+ // `fs_hz` and `channels` number audio channels.
void SetSampleRateAndChannels(int fs_hz, size_t channels)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 53b4dae..875e62c 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -207,8 +207,8 @@
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
- // DTMF packets are immediately consumed by |InsertPacket()| and won't be
- // returned by |GetAudio()|.
+ // DTMF packets are immediately consumed by `InsertPacket()` and won't be
+ // returned by `GetAudio()`.
EXPECT_THAT(output.packet_infos_, IsEmpty());
// Verify first 64 samples of actual output.
@@ -461,7 +461,7 @@
public:
CountingSamplesDecoder() : next_value_(1) {}
- // Produce as many samples as input bytes (|encoded_len|).
+ // Produce as many samples as input bytes (`encoded_len`).
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int /* sample_rate_hz */,
@@ -578,7 +578,7 @@
.WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
int16_t dummy_output[kPayloadLengthSamples] = {0};
// The below expectation will make the mock decoder write
- // |kPayloadLengthSamples| zeros to the output array, and mark it as speech.
+ // `kPayloadLengthSamples` zeros to the output array, and mark it as speech.
EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
kSampleRateHz, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
@@ -1284,7 +1284,7 @@
.WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
int16_t dummy_output[kPayloadLengthSamples] = {0};
// The below expectation will make the mock decoder write
- // |kPayloadLengthSamples| - 5 zeros to the output array, and mark it as
+ // `kPayloadLengthSamples` - 5 zeros to the output array, and mark it as
// speech. That is, the decoded length is 5 samples shorter than the expected.
EXPECT_CALL(mock_decoder,
DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index 8f72734..862edaf 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -188,11 +188,11 @@
: 0xffffffff);
}
- // |stats_ref|
+ // `stats_ref`
// expects.x = -1, do not care
- // expects.x = 0, 'x' in current stats should equal 'x' in |stats_ref|
- // expects.x = 1, 'x' in current stats should < 'x' in |stats_ref|
- // expects.x = 2, 'x' in current stats should > 'x' in |stats_ref|
+ // expects.x = 0, 'x' in current stats should equal 'x' in `stats_ref`
+ // expects.x = 1, 'x' in current stats should < 'x' in `stats_ref`
+ // expects.x = 2, 'x' in current stats should > 'x' in `stats_ref`
void CheckNetworkStatistics(NetEqNetworkStatsCheck expects) {
NetEqNetworkStatistics stats;
neteq_->NetworkStatistics(&stats);
@@ -229,7 +229,7 @@
uint32_t time_now;
uint32_t next_send_time;
- // Initiate |last_lost_time_|.
+ // Initiate `last_lost_time_`.
time_now = next_send_time = last_lost_time_ = rtp_generator_->GetRtpHeader(
kPayloadType, frame_size_samples_, &rtp_header_);
for (int k = 0; k < num_loops; ++k) {
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index bdd90e9..5ce6b89 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -305,7 +305,7 @@
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.payloadType = 103; // iSAC, but the payload is invalid.
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
- // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
+ // Set all of `out_data_` to 1, and verify that it was set to 0 by the call
// to GetAudio.
int16_t* out_frame_data = out_frame_.mutable_data();
for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
@@ -327,7 +327,7 @@
}
TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
- // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
+ // Set all of `out_data_` to 1, and verify that it was set to 0 by the call
// to GetAudio.
int16_t* out_frame_data = out_frame_.mutable_data();
for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
@@ -371,7 +371,7 @@
AudioFrame output;
test::AudioLoop input;
// We are using the same 32 kHz input file for all tests, regardless of
- // |sampling_rate_hz|. The output may sound weird, but the test is still
+ // `sampling_rate_hz`. The output may sound weird, but the test is still
// valid.
ASSERT_TRUE(input.Init(
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
@@ -534,7 +534,7 @@
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
payload, payload_len)));
- // Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
+ // Pull audio until we have played `kCngPeriodMs` of CNG. Start at 10 ms since
// we have already pulled out CNG once.
for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) {
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index 3ed0e26..6ffae097 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -45,7 +45,7 @@
const int fs_mult = fs_hz_ / 8000;
RTC_DCHECK_GT(fs_mult, 0);
// fs_shift = log2(fs_mult), rounded down.
- // Note that |fs_shift| is not "exact" for 48 kHz.
+ // Note that `fs_shift` is not "exact" for 48 kHz.
// TODO(hlundin): Investigate this further.
const int fs_shift = 30 - WebRtcSpl_NormW32(fs_mult);
@@ -83,7 +83,7 @@
size_t energy_length =
std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
int scaling = 6 + fs_shift - WebRtcSpl_NormW32(decoded_max * decoded_max);
- scaling = std::max(scaling, 0); // |scaling| should always be >= 0.
+ scaling = std::max(scaling, 0); // `scaling` should always be >= 0.
int32_t energy = WebRtcSpl_DotProductWithScale(signal.get(), signal.get(),
energy_length, scaling);
int32_t scaled_energy_length =
diff --git a/modules/audio_coding/neteq/normal.h b/modules/audio_coding/neteq/normal.h
index d6dc84a..3607208 100644
--- a/modules/audio_coding/neteq/normal.h
+++ b/modules/audio_coding/neteq/normal.h
@@ -49,11 +49,11 @@
virtual ~Normal() {}
- // Performs the "Normal" operation. The decoder data is supplied in |input|,
- // having |length| samples in total for all channels (interleaved). The
- // result is written to |output|. The number of channels allocated in
- // |output| defines the number of channels that will be used when
- // de-interleaving |input|. |last_mode| contains the mode used in the previous
+ // Performs the "Normal" operation. The decoder data is supplied in `input`,
+ // having `length` samples in total for all channels (interleaved). The
+ // result is written to `output`. The number of channels allocated in
+ // `output` defines the number of channels that will be used when
+ // de-interleaving `input`. `last_mode` contains the mode used in the previous
// GetAudio call (i.e., not the current one).
int Process(const int16_t* input,
size_t length,
diff --git a/modules/audio_coding/neteq/normal_unittest.cc b/modules/audio_coding/neteq/normal_unittest.cc
index 7e533bb..4554d79 100644
--- a/modules/audio_coding/neteq/normal_unittest.cc
+++ b/modules/audio_coding/neteq/normal_unittest.cc
@@ -51,7 +51,7 @@
StatisticsCalculator statistics;
Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
Normal normal(fs, &db, bgn, &expand, &statistics);
- EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
+ EXPECT_CALL(db, Die()); // Called when `db` goes out of scope.
}
TEST(Normal, AvoidDivideByZero) {
@@ -85,8 +85,8 @@
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.
+ EXPECT_CALL(db, Die()); // Called when `db` goes out of scope.
+ EXPECT_CALL(expand, Die()); // Called when `expand` goes out of scope.
}
TEST(Normal, InputLengthAndChannelsDoNotMatch) {
@@ -109,8 +109,8 @@
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.
- EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope.
+ EXPECT_CALL(db, Die()); // Called when `db` goes out of scope.
+ EXPECT_CALL(expand, Die()); // Called when `expand` goes out of scope.
}
TEST(Normal, LastModeExpand120msPacket) {
@@ -138,8 +138,8 @@
EXPECT_EQ(kPacketsizeBytes, output.Size());
- EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
- EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope.
+ EXPECT_CALL(db, Die()); // Called when `db` goes out of scope.
+ EXPECT_CALL(expand, Die()); // Called when `expand` goes out of scope.
}
// TODO(hlundin): Write more tests.
diff --git a/modules/audio_coding/neteq/packet.h b/modules/audio_coding/neteq/packet.h
index 4455494..0c6f204 100644
--- a/modules/audio_coding/neteq/packet.h
+++ b/modules/audio_coding/neteq/packet.h
@@ -84,8 +84,8 @@
// Packets should generally be moved around but sometimes it's useful to make
// a copy, for example for testing purposes. NOTE: Will only work for
- // un-parsed packets, i.e. |frame| must be unset. The payload will, however,
- // be copied. |waiting_time| will also not be copied.
+ // un-parsed packets, i.e. `frame` must be unset. The payload will, however,
+ // be copied. `waiting_time` will also not be copied.
Packet Clone() const;
Packet& operator=(Packet&& b);
diff --git a/modules/audio_coding/neteq/packet_buffer.cc b/modules/audio_coding/neteq/packet_buffer.cc
index 86ae847..f6b5a47 100644
--- a/modules/audio_coding/neteq/packet_buffer.cc
+++ b/modules/audio_coding/neteq/packet_buffer.cc
@@ -33,7 +33,7 @@
namespace webrtc {
namespace {
// Predicate used when inserting packets in the buffer list.
-// Operator() returns true when |packet| goes before |new_packet|.
+// Operator() returns true when `packet` goes before `new_packet`.
class NewTimestampIsLarger {
public:
explicit NewTimestampIsLarger(const Packet& new_packet)
@@ -183,16 +183,16 @@
PacketList::reverse_iterator rit = std::find_if(
buffer_.rbegin(), buffer_.rend(), NewTimestampIsLarger(packet));
- // The new packet is to be inserted to the right of |rit|. If it has the same
- // timestamp as |rit|, which has a higher priority, do not insert the new
+ // The new packet is to be inserted to the right of `rit`. If it has the same
+ // timestamp as `rit`, which has a higher priority, do not insert the new
// packet to list.
if (rit != buffer_.rend() && packet.timestamp == rit->timestamp) {
LogPacketDiscarded(packet.priority.codec_level, stats);
return return_val;
}
- // The new packet is to be inserted to the left of |it|. If it has the same
- // timestamp as |it|, which has a lower priority, replace |it| with the new
+ // The new packet is to be inserted to the left of `it`. If it has the same
+ // timestamp as `it`, which has a lower priority, replace `it` with the new
// packet.
PacketList::iterator it = rit.base();
if (it != buffer_.end() && packet.timestamp == it->timestamp) {
diff --git a/modules/audio_coding/neteq/packet_buffer.h b/modules/audio_coding/neteq/packet_buffer.h
index cd2adf7..20a0533 100644
--- a/modules/audio_coding/neteq/packet_buffer.h
+++ b/modules/audio_coding/neteq/packet_buffer.h
@@ -45,7 +45,7 @@
};
// Constructor creates a buffer which can hold a maximum of
- // |max_number_of_packets| packets.
+ // `max_number_of_packets` packets.
PacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer);
// Deletes all packets in the buffer before destroying the buffer.
@@ -63,7 +63,7 @@
// Returns true for an empty buffer.
virtual bool Empty() const;
- // Inserts |packet| into the buffer. The buffer will take over ownership of
+ // Inserts `packet` into the buffer. The buffer will take over ownership of
// the packet object.
// Returns PacketBuffer::kOK on success, PacketBuffer::kFlushed if the buffer
// was flushed due to overfilling.
@@ -93,14 +93,14 @@
int target_level_ms);
// Gets the timestamp for the first packet in the buffer and writes it to the
- // output variable |next_timestamp|.
+ // output variable `next_timestamp`.
// Returns PacketBuffer::kBufferEmpty if the buffer is empty,
// PacketBuffer::kOK otherwise.
virtual int NextTimestamp(uint32_t* next_timestamp) const;
// Gets the timestamp for the first packet in the buffer with a timestamp no
- // lower than the input limit |timestamp|. The result is written to the output
- // variable |next_timestamp|.
+ // lower than the input limit `timestamp`. The result is written to the output
+ // variable `next_timestamp`.
// Returns PacketBuffer::kBufferEmpty if the buffer is empty,
// PacketBuffer::kOK otherwise.
virtual int NextHigherTimestamp(uint32_t timestamp,
@@ -154,11 +154,11 @@
virtual bool ContainsDtxOrCngPacket(
const DecoderDatabase* decoder_database) const;
- // Static method returning true if |timestamp| is older than |timestamp_limit|
- // but less than |horizon_samples| behind |timestamp_limit|. For instance,
+ // Static method returning true if `timestamp` is older than `timestamp_limit`
+ // but less than `horizon_samples` behind `timestamp_limit`. For instance,
// with timestamp_limit = 100 and horizon_samples = 10, a timestamp in the
// range (90, 100) is considered obsolete, and will yield true.
- // Setting |horizon_samples| to 0 is the same as setting it to 2^31, i.e.,
+ // Setting `horizon_samples` to 0 is the same as setting it to 2^31, i.e.,
// half the 32-bit timestamp range.
static bool IsObsoleteTimestamp(uint32_t timestamp,
uint32_t timestamp_limit,
diff --git a/modules/audio_coding/neteq/post_decode_vad.h b/modules/audio_coding/neteq/post_decode_vad.h
index ca7cabf..3134d5f 100644
--- a/modules/audio_coding/neteq/post_decode_vad.h
+++ b/modules/audio_coding/neteq/post_decode_vad.h
@@ -40,8 +40,8 @@
// Initializes post-decode VAD.
void Init();
- // Updates post-decode VAD with the audio data in |signal| having |length|
- // samples. The data is of type |speech_type|, at the sample rate |fs_hz|.
+ // Updates post-decode VAD with the audio data in `signal` having `length`
+ // samples. The data is of type `speech_type`, at the sample rate `fs_hz`.
void Update(int16_t* signal,
size_t length,
AudioDecoder::SpeechType speech_type,
diff --git a/modules/audio_coding/neteq/preemptive_expand.cc b/modules/audio_coding/neteq/preemptive_expand.cc
index cad8d6a..232170b 100644
--- a/modules/audio_coding/neteq/preemptive_expand.cc
+++ b/modules/audio_coding/neteq/preemptive_expand.cc
@@ -26,7 +26,7 @@
size_t* length_change_samples) {
old_data_length_per_channel_ = old_data_length;
// Input length must be (almost) 30 ms.
- // Also, the new part must be at least |overlap_samples_| elements.
+ // Also, the new part must be at least `overlap_samples_` elements.
static const size_t k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate.
if (num_channels_ == 0 ||
input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ ||
@@ -64,7 +64,7 @@
bool active_speech,
bool /*fast_mode*/,
AudioMultiVector* output) const {
- // Pre-calculate common multiplication with |fs_mult_|.
+ // Pre-calculate common multiplication with `fs_mult_`.
// 120 corresponds to 15 ms.
size_t fs_mult_120 = static_cast<size_t>(fs_mult_ * 120);
// Check for strong correlation (>0.9 in Q14) and at least 15 ms new data,
@@ -80,12 +80,12 @@
// Copy first part, including cross-fade region.
output->PushBackInterleaved(rtc::ArrayView<const int16_t>(
input, (unmodified_length + peak_index) * num_channels_));
- // Copy the last |peak_index| samples up to 15 ms to |temp_vector|.
+ // Copy the last `peak_index` samples up to 15 ms to `temp_vector`.
AudioMultiVector temp_vector(num_channels_);
temp_vector.PushBackInterleaved(rtc::ArrayView<const int16_t>(
&input[(unmodified_length - peak_index) * num_channels_],
peak_index * num_channels_));
- // Cross-fade |temp_vector| onto the end of |output|.
+ // Cross-fade `temp_vector` onto the end of `output`.
output->CrossFade(temp_vector, peak_index);
// Copy the last unmodified part, 15 ms + pitch period until the end.
output->PushBackInterleaved(rtc::ArrayView<const int16_t>(
diff --git a/modules/audio_coding/neteq/preemptive_expand.h b/modules/audio_coding/neteq/preemptive_expand.h
index e7d2bad..708ebfd 100644
--- a/modules/audio_coding/neteq/preemptive_expand.h
+++ b/modules/audio_coding/neteq/preemptive_expand.h
@@ -37,9 +37,9 @@
overlap_samples_(overlap_samples) {}
// This method performs the actual PreemptiveExpand operation. The samples are
- // read from |input|, of length |input_length| elements, and are written to
- // |output|. The number of samples added through time-stretching is
- // is provided in the output |length_change_samples|. The method returns
+ // read from `input`, of length `input_length` elements, and are written to
+ // `output`. The number of samples added through time-stretching is
+ // is provided in the output `length_change_samples`. The method returns
// the outcome of the operation as an enumerator value.
ReturnCodes Process(const int16_t* pw16_decoded,
size_t len,
@@ -48,7 +48,7 @@
size_t* length_change_samples);
protected:
- // Sets the parameters |best_correlation| and |peak_index| to suitable
+ // Sets the parameters `best_correlation` and `peak_index` to suitable
// values when the signal contains no active speech.
void SetParametersForPassiveSpeech(size_t input_length,
int16_t* best_correlation,
diff --git a/modules/audio_coding/neteq/red_payload_splitter.cc b/modules/audio_coding/neteq/red_payload_splitter.cc
index b517e38..b7b4520 100644
--- a/modules/audio_coding/neteq/red_payload_splitter.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter.cc
@@ -27,9 +27,9 @@
// The method loops through a list of packets {A, B, C, ...}. Each packet is
// split into its corresponding RED payloads, {A1, A2, ...}, which is
-// temporarily held in the list |new_packets|.
-// When the first packet in |packet_list| has been processed, the original
-// packet is replaced by the new ones in |new_packets|, so that |packet_list|
+// temporarily held in the list `new_packets`.
+// When the first packet in `packet_list` has been processed, the original
+// packet is replaced by the new ones in `new_packets`, so that `packet_list`
// becomes: {A1, A2, ..., B, C, ...}. The method then continues with B, and C,
// until all the original packets have been replaced by their split payloads.
bool RedPayloadSplitter::SplitRed(PacketList* packet_list) {
@@ -110,7 +110,7 @@
if (new_headers.size() <= kMaxRedBlocks) {
// Populate the new packets with payload data.
- // |payload_ptr| now points at the first payload byte.
+ // `payload_ptr` now points at the first payload byte.
PacketList new_packets; // An empty list to store the split packets in.
for (size_t i = 0; i != new_headers.size(); ++i) {
const auto& new_header = new_headers[i];
@@ -143,14 +143,14 @@
payload_ptr += payload_length;
}
// Insert new packets into original list, before the element pointed to by
- // iterator |it|.
+ // iterator `it`.
packet_list->splice(it, std::move(new_packets));
} else {
RTC_LOG(LS_WARNING) << "SplitRed too many blocks: " << new_headers.size();
ret = false;
}
- // Remove |it| from the packet list. This operation effectively moves the
- // iterator |it| to the next packet in the list. Thus, we do not have to
+ // Remove `it` from the packet list. This operation effectively moves the
+ // iterator `it` to the next packet in the list. Thus, we do not have to
// increment it manually.
it = packet_list->erase(it);
}
@@ -175,8 +175,8 @@
} else {
if (this_payload_type != main_payload_type) {
// We do not allow redundant payloads of a different type.
- // Remove |it| from the packet list. This operation effectively
- // moves the iterator |it| to the next packet in the list. Thus, we
+ // Remove `it` from the packet list. This operation effectively
+ // moves the iterator `it` to the next packet in the list. Thus, we
// do not have to increment it manually.
it = packet_list->erase(it);
continue;
diff --git a/modules/audio_coding/neteq/red_payload_splitter.h b/modules/audio_coding/neteq/red_payload_splitter.h
index c54ffc0..5566091 100644
--- a/modules/audio_coding/neteq/red_payload_splitter.h
+++ b/modules/audio_coding/neteq/red_payload_splitter.h
@@ -30,15 +30,15 @@
virtual ~RedPayloadSplitter() {}
- // Splits each packet in |packet_list| into its separate RED payloads. Each
+ // Splits each packet in `packet_list` into its separate RED payloads. Each
// RED payload is packetized into a Packet. The original elements in
- // |packet_list| are properly deleted, and replaced by the new packets.
- // Note that all packets in |packet_list| must be RED payloads, i.e., have
+ // `packet_list` are properly deleted, and replaced by the new packets.
+ // Note that all packets in `packet_list` must be RED payloads, i.e., have
// RED headers according to RFC 2198 at the very beginning of the payload.
// Returns kOK or an error.
virtual bool SplitRed(PacketList* packet_list);
- // Checks all packets in |packet_list|. Packets that are DTMF events or
+ // Checks all packets in `packet_list`. Packets that are DTMF events or
// comfort noise payloads are kept. Except that, only one single payload type
// is accepted. Any packet with another payload type is discarded.
virtual void CheckRedPayloads(PacketList* packet_list,
diff --git a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
index 1f16945..a0ba541 100644
--- a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
@@ -70,9 +70,9 @@
// |0| Block PT |
// +-+-+-+-+-+-+-+-+
-// Creates a RED packet, with |num_payloads| payloads, with payload types given
-// by the values in array |payload_types| (which must be of length
-// |num_payloads|). Each redundant payload is |timestamp_offset| samples
+// Creates a RED packet, with `num_payloads` payloads, with payload types given
+// by the values in array `payload_types` (which must be of length
+// `num_payloads`). Each redundant payload is `timestamp_offset` samples
// "behind" the the previous payload.
Packet CreateRedPayload(size_t num_payloads,
uint8_t* payload_types,
@@ -109,7 +109,7 @@
++payload_ptr;
}
for (size_t i = 0; i < num_payloads; ++i) {
- // Write |i| to all bytes in each payload.
+ // Write `i` to all bytes in each payload.
if (embed_opus_fec) {
CreateOpusFecPayload(payload_ptr, kPayloadLength,
static_cast<uint8_t>(i));
@@ -121,7 +121,7 @@
return packet;
}
-// Create a packet with all payload bytes set to |payload_value|.
+// Create a packet with all payload bytes set to `payload_value`.
Packet CreatePacket(uint8_t payload_type,
size_t payload_length,
uint8_t payload_value,
@@ -140,7 +140,7 @@
return packet;
}
-// Checks that |packet| has the attributes given in the remaining parameters.
+// Checks that `packet` has the attributes given in the remaining parameters.
void VerifyPacket(const Packet& packet,
size_t payload_length,
uint8_t payload_type,
@@ -289,7 +289,7 @@
TEST(RedPayloadSplitter, CheckRedPayloads) {
PacketList packet_list;
for (uint8_t i = 0; i <= 3; ++i) {
- // Create packet with payload type |i|, payload length 10 bytes, all 0.
+ // Create packet with payload type `i`, payload length 10 bytes, all 0.
packet_list.push_back(CreatePacket(i, 10, 0));
}
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index 741cdbd..8e28130 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -339,7 +339,7 @@
} else {
std::sort(waiting_times_.begin(), waiting_times_.end());
// Find mid-point elements. If the size is odd, the two values
- // |middle_left| and |middle_right| will both be the one middle element; if
+ // `middle_left` and `middle_right` will both be the one middle element; if
// the size is even, they will be the the two neighboring elements at the
// middle of the list.
const int middle_left = waiting_times_[(waiting_times_.size() - 1) / 2];
diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h
index f0c2734..5c3fb75 100644
--- a/modules/audio_coding/neteq/statistics_calculator.h
+++ b/modules/audio_coding/neteq/statistics_calculator.h
@@ -34,16 +34,16 @@
// Resets the counters that are not handled by Reset().
void ResetMcu();
- // Reports that |num_samples| samples were produced through expansion, and
+ // Reports that `num_samples` samples were produced through expansion, and
// that the expansion produced other than just noise samples.
void ExpandedVoiceSamples(size_t num_samples, bool is_new_concealment_event);
- // Reports that |num_samples| samples were produced through expansion, and
+ // Reports that `num_samples` samples were produced through expansion, and
// that the expansion produced only noise samples.
void ExpandedNoiseSamples(size_t num_samples, bool is_new_concealment_event);
// Corrects the statistics for number of samples produced through non-noise
- // expansion by adding |num_samples| (negative or positive) to the current
+ // expansion by adding `num_samples` (negative or positive) to the current
// value. The result is capped to zero to avoid negative values.
void ExpandedVoiceSamplesCorrection(int num_samples);
@@ -55,24 +55,24 @@
// Mark end of expand event; triggers some stats to be reported.
void EndExpandEvent(int fs_hz);
- // Reports that |num_samples| samples were produced through preemptive
+ // Reports that `num_samples` samples were produced through preemptive
// expansion.
void PreemptiveExpandedSamples(size_t num_samples);
- // Reports that |num_samples| samples were removed through accelerate.
+ // Reports that `num_samples` samples were removed through accelerate.
void AcceleratedSamples(size_t num_samples);
- // Reports that |num_packets| packets were discarded.
+ // Reports that `num_packets` packets were discarded.
virtual void PacketsDiscarded(size_t num_packets);
- // Reports that |num_packets| secondary (FEC) packets were discarded.
+ // Reports that `num_packets` secondary (FEC) packets were discarded.
virtual void SecondaryPacketsDiscarded(size_t num_packets);
- // Reports that |num_packets| secondary (FEC) packets were received.
+ // Reports that `num_packets` secondary (FEC) packets were received.
virtual void SecondaryPacketsReceived(size_t num_packets);
- // Increases the report interval counter with |num_samples| at a sample rate
- // of |fs_hz|. This is how the StatisticsCalculator gets notified that current
+ // Increases the report interval counter with `num_samples` at a sample rate
+ // of `fs_hz`. This is how the StatisticsCalculator gets notified that current
// time is increasing.
void IncreaseCounter(size_t num_samples, int fs_hz);
@@ -84,7 +84,7 @@
// Stores new packet waiting time in waiting time statistics.
void StoreWaitingTime(int waiting_time_ms);
- // Reports that |num_samples| samples were decoded from secondary packets.
+ // Reports that `num_samples` samples were decoded from secondary packets.
void SecondaryDecodedSamples(int num_samples);
// Reports that the packet buffer was flushed.
@@ -93,17 +93,17 @@
// Reports that the jitter buffer received a packet.
void ReceivedPacket();
- // Reports that a received packet was delayed by |delay_ms| milliseconds.
+ // Reports that a received packet was delayed by `delay_ms` milliseconds.
virtual void RelativePacketArrivalDelay(size_t delay_ms);
- // Logs a delayed packet outage event of |num_samples| expanded at a sample
- // rate of |fs_hz|. A delayed packet outage event is defined as an expand
+ // Logs a delayed packet outage event of `num_samples` expanded at a sample
+ // rate of `fs_hz`. A delayed packet outage event is defined as an expand
// period caused not by an actual packet loss, but by a delayed packet.
virtual void LogDelayedPacketOutageEvent(int num_samples, int fs_hz);
- // Returns the current network statistics in |stats|. The number of samples
- // per packet is |samples_per_packet|. The method does not populate
- // |preferred_buffer_size_ms|, |jitter_peaks_found| or |clockdrift_ppm|; use
+ // Returns the current network statistics in `stats`. The number of samples
+ // per packet is `samples_per_packet`. The method does not populate
+ // `preferred_buffer_size_ms`, `jitter_peaks_found` or `clockdrift_ppm`; use
// the PopulateDelayManagerStats method for those.
void GetNetworkStatistics(size_t samples_per_packet,
NetEqNetworkStatistics* stats);
diff --git a/modules/audio_coding/neteq/sync_buffer.cc b/modules/audio_coding/neteq/sync_buffer.cc
index 73e0628..80e1691 100644
--- a/modules/audio_coding/neteq/sync_buffer.cc
+++ b/modules/audio_coding/neteq/sync_buffer.cc
@@ -59,11 +59,11 @@
channels_[channel]->InsertZerosAt(length, position);
}
if (next_index_ >= position) {
- // We are moving the |next_index_| sample.
+ // We are moving the `next_index_` sample.
set_next_index(next_index_ + length); // Overflow handled by subfunction.
}
if (dtmf_index_ > 0 && dtmf_index_ >= position) {
- // We are moving the |dtmf_index_| sample.
+ // We are moving the `dtmf_index_` sample.
set_dtmf_index(dtmf_index_ + length); // Overflow handled by subfunction.
}
}
@@ -71,7 +71,7 @@
void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
size_t length,
size_t position) {
- position = std::min(position, Size()); // Cap |position| in the valid range.
+ position = std::min(position, Size()); // Cap `position` in the valid range.
length = std::min(length, Size() - position);
AudioMultiVector::OverwriteAt(insert_this, length, position);
}
@@ -106,12 +106,12 @@
}
void SyncBuffer::set_next_index(size_t value) {
- // Cannot set |next_index_| larger than the size of the buffer.
+ // Cannot set `next_index_` larger than the size of the buffer.
next_index_ = std::min(value, Size());
}
void SyncBuffer::set_dtmf_index(size_t value) {
- // Cannot set |dtmf_index_| larger than the size of the buffer.
+ // Cannot set `dtmf_index_` larger than the size of the buffer.
dtmf_index_ = std::min(value, Size());
}
diff --git a/modules/audio_coding/neteq/sync_buffer.h b/modules/audio_coding/neteq/sync_buffer.h
index 754716b..7d24730 100644
--- a/modules/audio_coding/neteq/sync_buffer.h
+++ b/modules/audio_coding/neteq/sync_buffer.h
@@ -35,55 +35,55 @@
// Returns the number of samples yet to play out from the buffer.
size_t FutureLength() const;
- // Adds the contents of |append_this| to the back of the SyncBuffer. Removes
+ // Adds the contents of `append_this` to the back of the SyncBuffer. Removes
// the same number of samples from the beginning of the SyncBuffer, to
- // maintain a constant buffer size. The |next_index_| is updated to reflect
+ // maintain a constant buffer size. The `next_index_` is updated to reflect
// the move of the beginning of "future" data.
void PushBack(const AudioMultiVector& append_this) override;
// Like PushBack, but reads the samples channel-interleaved from the input.
void PushBackInterleaved(const rtc::BufferT<int16_t>& append_this);
- // Adds |length| zeros to the beginning of each channel. Removes
+ // Adds `length` zeros to the beginning of each channel. Removes
// the same number of samples from the end of the SyncBuffer, to
- // maintain a constant buffer size. The |next_index_| is updated to reflect
+ // maintain a constant buffer size. The `next_index_` is updated to reflect
// the move of the beginning of "future" data.
// Note that this operation may delete future samples that are waiting to
// be played.
void PushFrontZeros(size_t length);
- // Inserts |length| zeros into each channel at index |position|. The size of
- // the SyncBuffer is kept constant, which means that the last |length|
+ // Inserts `length` zeros into each channel at index `position`. The size of
+ // the SyncBuffer is kept constant, which means that the last `length`
// elements in each channel will be purged.
virtual void InsertZerosAtIndex(size_t length, size_t position);
// Overwrites each channel in this SyncBuffer with values taken from
- // |insert_this|. The values are taken from the beginning of |insert_this| and
- // are inserted starting at |position|. |length| values are written into each
- // channel. The size of the SyncBuffer is kept constant. That is, if |length|
- // and |position| are selected such that the new data would extend beyond the
+ // `insert_this`. The values are taken from the beginning of `insert_this` and
+ // are inserted starting at `position`. `length` values are written into each
+ // channel. The size of the SyncBuffer is kept constant. That is, if `length`
+ // and `position` are selected such that the new data would extend beyond the
// end of the current SyncBuffer, the buffer is not extended.
- // The |next_index_| is not updated.
+ // The `next_index_` is not updated.
virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
size_t length,
size_t position);
- // Same as the above method, but where all of |insert_this| is written (with
+ // Same as the above method, but where all of `insert_this` is written (with
// the same constraints as above, that the SyncBuffer is not extended).
virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
size_t position);
- // Reads |requested_len| samples from each channel and writes them interleaved
- // into |output|. The |next_index_| is updated to point to the sample to read
- // next time. The AudioFrame |output| is first reset, and the |data_|,
- // |num_channels_|, and |samples_per_channel_| fields are updated.
+ // Reads `requested_len` samples from each channel and writes them interleaved
+ // into `output`. The `next_index_` is updated to point to the sample to read
+ // next time. The AudioFrame `output` is first reset, and the `data_`,
+ // `num_channels_`, and `samples_per_channel_` fields are updated.
void GetNextAudioInterleaved(size_t requested_len, AudioFrame* output);
- // Adds |increment| to |end_timestamp_|.
+ // Adds `increment` to `end_timestamp_`.
void IncreaseEndTimestamp(uint32_t increment);
// Flushes the buffer. The buffer will contain only zeros after the flush, and
- // |next_index_| will point to the end, like when the buffer was first
+ // `next_index_` will point to the end, like when the buffer was first
// created.
void Flush();
diff --git a/modules/audio_coding/neteq/sync_buffer_unittest.cc b/modules/audio_coding/neteq/sync_buffer_unittest.cc
index 860dbae..bdcd924 100644
--- a/modules/audio_coding/neteq/sync_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/sync_buffer_unittest.cc
@@ -55,18 +55,18 @@
SyncBuffer sync_buffer(kChannels, kLen);
static const size_t kNewLen = 10;
AudioMultiVector new_data(kChannels, kNewLen);
- // Populate |new_data|.
+ // Populate `new_data`.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen; ++i) {
new_data[channel][i] = rtc::checked_cast<int16_t>(i);
}
}
- // Push back |new_data| into |sync_buffer|. This operation should pop out
- // data from the front of |sync_buffer|, so that the size of the buffer
- // remains the same. The |next_index_| should also move with the same length.
+ // Push back `new_data` into `sync_buffer`. This operation should pop out
+ // data from the front of `sync_buffer`, so that the size of the buffer
+ // remains the same. The `next_index_` should also move with the same length.
sync_buffer.PushBack(new_data);
ASSERT_EQ(kLen, sync_buffer.Size());
- // Verify that |next_index_| moved accordingly.
+ // Verify that `next_index_` moved accordingly.
EXPECT_EQ(kLen - kNewLen, sync_buffer.next_index());
// Verify the new contents.
for (size_t channel = 0; channel < kChannels; ++channel) {
@@ -95,7 +95,7 @@
SyncBuffer sync_buffer(kChannels, kLen);
static const size_t kNewLen = 10;
AudioMultiVector new_data(kChannels, kNewLen);
- // Populate |new_data|.
+ // Populate `new_data`.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen; ++i) {
new_data[channel][i] = rtc::checked_cast<int16_t>(1000 + i);
@@ -104,10 +104,10 @@
sync_buffer.PushBack(new_data);
EXPECT_EQ(kLen, sync_buffer.Size());
- // Push |kNewLen| - 1 zeros into each channel in the front of the SyncBuffer.
+ // Push `kNewLen` - 1 zeros into each channel in the front of the SyncBuffer.
sync_buffer.PushFrontZeros(kNewLen - 1);
EXPECT_EQ(kLen, sync_buffer.Size()); // Size should remain the same.
- // Verify that |next_index_| moved accordingly. Should be at the end - 1.
+ // Verify that `next_index_` moved accordingly. Should be at the end - 1.
EXPECT_EQ(kLen - 1, sync_buffer.next_index());
// Verify the zeros.
for (size_t channel = 0; channel < kChannels; ++channel) {
@@ -128,22 +128,22 @@
SyncBuffer sync_buffer(kChannels, kLen);
static const size_t kNewLen = 10;
AudioMultiVector new_data(kChannels, kNewLen);
- // Populate |new_data|.
+ // Populate `new_data`.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen; ++i) {
new_data[channel][i] = rtc::checked_cast<int16_t>(i);
}
}
- // Push back |new_data| into |sync_buffer|. This operation should pop out
- // data from the front of |sync_buffer|, so that the size of the buffer
- // remains the same. The |next_index_| should also move with the same length.
+ // Push back `new_data` into `sync_buffer`. This operation should pop out
+ // data from the front of `sync_buffer`, so that the size of the buffer
+ // remains the same. The `next_index_` should also move with the same length.
sync_buffer.PushBack(new_data);
// Read to interleaved output. Read in two batches, where each read operation
- // should automatically update the |net_index_| in the SyncBuffer.
- // Note that |samples_read| is the number of samples read from each channel.
- // That is, the number of samples written to |output| is
- // |samples_read| * |kChannels|.
+ // should automatically update the `net_index_` in the SyncBuffer.
+ // Note that `samples_read` is the number of samples read from each channel.
+ // That is, the number of samples written to `output` is
+ // `samples_read` * `kChannels`.
AudioFrame output1;
sync_buffer.GetNextAudioInterleaved(kNewLen / 2, &output1);
EXPECT_EQ(kChannels, output1.num_channels_);
diff --git a/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
index d7f414a..1c70f14 100644
--- a/modules/audio_coding/neteq/test/neteq_decoding_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
@@ -346,8 +346,8 @@
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
if (network_freeze_ms > 0) {
- // First keep pulling audio for |network_freeze_ms| without inserting
- // any data, then insert CNG data corresponding to |network_freeze_ms|
+ // First keep pulling audio for `network_freeze_ms` without inserting
+ // any data, then insert CNG data corresponding to `network_freeze_ms`
// without pulling any output audio.
const double loop_end_time = t_ms + network_freeze_ms;
for (; t_ms < loop_end_time; t_ms += 10) {
@@ -357,7 +357,7 @@
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
}
bool pull_once = pull_audio_during_freeze;
- // If |pull_once| is true, GetAudio will be called once half-way through
+ // If `pull_once` is true, GetAudio will be called once half-way through
// the network recovery period.
double pull_time_ms = (t_ms + next_input_time_ms) / 2;
while (next_input_time_ms <= t_ms) {
diff --git a/modules/audio_coding/neteq/time_stretch.cc b/modules/audio_coding/neteq/time_stretch.cc
index b768029..b89be06 100644
--- a/modules/audio_coding/neteq/time_stretch.cc
+++ b/modules/audio_coding/neteq/time_stretch.cc
@@ -26,7 +26,7 @@
bool fast_mode,
AudioMultiVector* output,
size_t* length_change_samples) {
- // Pre-calculate common multiplication with |fs_mult_|.
+ // Pre-calculate common multiplication with `fs_mult_`.
size_t fs_mult_120 =
static_cast<size_t>(fs_mult_ * 120); // Corresponds to 15 ms.
@@ -37,8 +37,8 @@
signal = input;
signal_len = input_len;
} else {
- // We want |signal| to be only the first channel of |input|, which is
- // interleaved. Thus, we take the first sample, skip forward |num_channels|
+ // We want `signal` to be only the first channel of `input`, which is
+ // interleaved. Thus, we take the first sample, skip forward `num_channels`
// samples, and continue like that.
signal_len = input_len / num_channels_;
signal_array.reset(new int16_t[signal_len]);
@@ -65,37 +65,37 @@
int16_t peak_value;
DspHelper::PeakDetection(auto_correlation_, kCorrelationLen, kNumPeaks,
fs_mult_, &peak_index, &peak_value);
- // Assert that |peak_index| stays within boundaries.
+ // Assert that `peak_index` stays within boundaries.
RTC_DCHECK_LE(peak_index, (2 * kCorrelationLen - 1) * fs_mult_);
// Compensate peak_index for displaced starting position. The displacement
- // happens in AutoCorrelation(). Here, |kMinLag| is in the down-sampled 4 kHz
- // domain, while the |peak_index| is in the original sample rate; hence, the
+ // happens in AutoCorrelation(). Here, `kMinLag` is in the down-sampled 4 kHz
+ // domain, while the `peak_index` is in the original sample rate; hence, the
// multiplication by fs_mult_ * 2.
peak_index += kMinLag * fs_mult_ * 2;
- // Assert that |peak_index| stays within boundaries.
+ // Assert that `peak_index` stays within boundaries.
RTC_DCHECK_GE(peak_index, static_cast<size_t>(20 * fs_mult_));
RTC_DCHECK_LE(peak_index,
20 * fs_mult_ + (2 * kCorrelationLen - 1) * fs_mult_);
- // Calculate scaling to ensure that |peak_index| samples can be square-summed
+ // Calculate scaling to ensure that `peak_index` samples can be square-summed
// without overflowing.
int scaling = 31 - WebRtcSpl_NormW32(max_input_value_ * max_input_value_) -
WebRtcSpl_NormW32(static_cast<int32_t>(peak_index));
scaling = std::max(0, scaling);
- // |vec1| starts at 15 ms minus one pitch period.
+ // `vec1` starts at 15 ms minus one pitch period.
const int16_t* vec1 = &signal[fs_mult_120 - peak_index];
- // |vec2| start at 15 ms.
+ // `vec2` start at 15 ms.
const int16_t* vec2 = &signal[fs_mult_120];
- // Calculate energies for |vec1| and |vec2|, assuming they both contain
- // |peak_index| samples.
+ // Calculate energies for `vec1` and `vec2`, assuming they both contain
+ // `peak_index` samples.
int32_t vec1_energy =
WebRtcSpl_DotProductWithScale(vec1, vec1, peak_index, scaling);
int32_t vec2_energy =
WebRtcSpl_DotProductWithScale(vec2, vec2, peak_index, scaling);
- // Calculate cross-correlation between |vec1| and |vec2|.
+ // Calculate cross-correlation between `vec1` and `vec2`.
int32_t cross_corr =
WebRtcSpl_DotProductWithScale(vec1, vec2, peak_index, scaling);
@@ -135,7 +135,7 @@
cross_corr = WEBRTC_SPL_SHIFT_W32(cross_corr, temp_scale);
cross_corr = std::max(0, cross_corr); // Don't use if negative.
best_correlation = WebRtcSpl_DivW32W16(cross_corr, sqrt_energy_prod);
- // Make sure |best_correlation| is no larger than 1 in Q14.
+ // Make sure `best_correlation` is no larger than 1 in Q14.
best_correlation = std::min(static_cast<int16_t>(16384), best_correlation);
}
@@ -165,7 +165,7 @@
&downsampled_input_[kMaxLag], &downsampled_input_[kMaxLag - kMinLag],
kCorrelationLen, kMaxLag - kMinLag, -1, auto_corr);
- // Normalize correlation to 14 bits and write to |auto_correlation_|.
+ // Normalize correlation to 14 bits and write to `auto_correlation_`.
int32_t max_corr = WebRtcSpl_MaxAbsValueW32(auto_corr, kCorrelationLen);
int scaling = std::max(0, 17 - WebRtcSpl_NormW32(max_corr));
WebRtcSpl_VectorBitShiftW32ToW16(auto_correlation_, kCorrelationLen,
@@ -182,8 +182,8 @@
// active speech.
// Rewrite the inequality as:
// (vec1_energy + vec2_energy) / 16 <= peak_index * background_noise_energy.
- // The two sides of the inequality will be denoted |left_side| and
- // |right_side|.
+ // The two sides of the inequality will be denoted `left_side` and
+ // `right_side`.
int32_t left_side = rtc::saturated_cast<int32_t>(
(static_cast<int64_t>(vec1_energy) + vec2_energy) / 16);
int32_t right_side;
@@ -199,11 +199,11 @@
right_side =
rtc::dchecked_cast<int32_t>(peak_index) * (right_side >> right_scale);
- // Scale |left_side| properly before comparing with |right_side|.
- // (|scaling| is the scale factor before energy calculation, thus the scale
+ // Scale `left_side` properly before comparing with `right_side`.
+ // (`scaling` is the scale factor before energy calculation, thus the scale
// factor for the energy is 2 * scaling.)
if (WebRtcSpl_NormW32(left_side) < 2 * scaling) {
- // Cannot scale only |left_side|, must scale |right_side| too.
+ // Cannot scale only `left_side`, must scale `right_side` too.
int temp_scale = WebRtcSpl_NormW32(left_side);
left_side = left_side << temp_scale;
right_side = right_side >> (2 * scaling - temp_scale);
diff --git a/modules/audio_coding/neteq/time_stretch.h b/modules/audio_coding/neteq/time_stretch.h
index 17ea4ec..998d080 100644
--- a/modules/audio_coding/neteq/time_stretch.h
+++ b/modules/audio_coding/neteq/time_stretch.h
@@ -58,7 +58,7 @@
size_t* length_change_samples);
protected:
- // Sets the parameters |best_correlation| and |peak_index| to suitable
+ // Sets the parameters `best_correlation` and `peak_index` to suitable
// values when the signal contains no active speech. This method must be
// implemented by the sub-classes.
virtual void SetParametersForPassiveSpeech(size_t input_length,
@@ -91,13 +91,13 @@
const BackgroundNoise& background_noise_;
int16_t max_input_value_;
int16_t downsampled_input_[kDownsampledLen];
- // Adding 1 to the size of |auto_correlation_| because of how it is used
+ // Adding 1 to the size of `auto_correlation_` because of how it is used
// by the peak-detection algorithm.
int16_t auto_correlation_[kCorrelationLen + 1];
private:
- // Calculates the auto-correlation of |downsampled_input_| and writes the
- // result to |auto_correlation_|.
+ // Calculates the auto-correlation of `downsampled_input_` and writes the
+ // result to `auto_correlation_`.
void AutoCorrelation();
// Performs a simple voice-activity detection based on the input parameters.
diff --git a/modules/audio_coding/neteq/timestamp_scaler.cc b/modules/audio_coding/neteq/timestamp_scaler.cc
index b0461bb..59177d0 100644
--- a/modules/audio_coding/neteq/timestamp_scaler.cc
+++ b/modules/audio_coding/neteq/timestamp_scaler.cc
@@ -79,7 +79,7 @@
const int64_t internal_diff = int64_t{internal_timestamp} - internal_ref_;
RTC_DCHECK_GT(numerator_, 0);
// Do not update references in this method.
- // Switch |denominator_| and |numerator_| to convert the other way.
+ // Switch `denominator_` and `numerator_` to convert the other way.
return external_ref_ + (internal_diff * denominator_) / numerator_;
}
}
diff --git a/modules/audio_coding/neteq/timestamp_scaler.h b/modules/audio_coding/neteq/timestamp_scaler.h
index 93cb953..4d578fc 100644
--- a/modules/audio_coding/neteq/timestamp_scaler.h
+++ b/modules/audio_coding/neteq/timestamp_scaler.h
@@ -37,15 +37,15 @@
// Start over.
virtual void Reset();
- // Scale the timestamp in |packet| from external to internal.
+ // Scale the timestamp in `packet` from external to internal.
virtual void ToInternal(Packet* packet);
- // Scale the timestamp for all packets in |packet_list| from external to
+ // Scale the timestamp for all packets in `packet_list` from external to
// internal.
virtual void ToInternal(PacketList* packet_list);
- // Returns the internal equivalent of |external_timestamp|, given the
- // RTP payload type |rtp_payload_type|.
+ // Returns the internal equivalent of `external_timestamp`, given the
+ // RTP payload type `rtp_payload_type`.
virtual uint32_t ToInternal(uint32_t external_timestamp,
uint8_t rtp_payload_type);
diff --git a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
index 9ba63e3..26dc06d 100644
--- a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
+++ b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
@@ -58,7 +58,7 @@
// Test both sides of the timestamp wrap-around.
static const uint32_t kStep = 160;
uint32_t start_timestamp = 0;
- // |external_timestamp| will be a large positive value.
+ // `external_timestamp` will be a large positive value.
start_timestamp = start_timestamp - 5 * kStep;
for (uint32_t timestamp = start_timestamp; timestamp != 5 * kStep;
timestamp += kStep) {
@@ -111,7 +111,7 @@
// Test both sides of the timestamp wrap-around.
static const uint32_t kStep = 320;
uint32_t external_timestamp = 0;
- // |external_timestamp| will be a large positive value.
+ // `external_timestamp` will be a large positive value.
external_timestamp = external_timestamp - 5 * kStep;
uint32_t internal_timestamp = external_timestamp;
for (; external_timestamp != 5 * kStep; external_timestamp += kStep) {
@@ -290,7 +290,7 @@
// Test both sides of the timestamp wrap-around.
static const uint32_t kStep = 960;
uint32_t external_timestamp = 0;
- // |external_timestamp| will be a large positive value.
+ // `external_timestamp` will be a large positive value.
external_timestamp = external_timestamp - 5 * kStep;
uint32_t internal_timestamp = external_timestamp;
for (; external_timestamp != 5 * kStep; external_timestamp += kStep) {
diff --git a/modules/audio_coding/neteq/tools/audio_loop.h b/modules/audio_coding/neteq/tools/audio_loop.h
index cd764cc..25da463 100644
--- a/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/modules/audio_coding/neteq/tools/audio_loop.h
@@ -29,17 +29,17 @@
virtual ~AudioLoop() {}
- // Initializes the AudioLoop by reading from |file_name|. The loop will be no
- // longer than |max_loop_length_samples|, if the length of the file is
+ // Initializes the AudioLoop by reading from `file_name`. The loop will be no
+ // longer than `max_loop_length_samples`, if the length of the file is
// greater. Otherwise, the loop length is the same as the file length.
- // The audio will be delivered in blocks of |block_length_samples|.
+ // The audio will be delivered in blocks of `block_length_samples`.
// Returns false if the initialization failed, otherwise true.
bool Init(const std::string file_name,
size_t max_loop_length_samples,
size_t block_length_samples);
// Returns a (pointer,size) pair for the next block of audio. The size is
- // equal to the |block_length_samples| Init() argument.
+ // equal to the `block_length_samples` Init() argument.
rtc::ArrayView<const int16_t> GetNextBlock();
private:
diff --git a/modules/audio_coding/neteq/tools/audio_sink.h b/modules/audio_coding/neteq/tools/audio_sink.h
index 68825eb..cd6733b 100644
--- a/modules/audio_coding/neteq/tools/audio_sink.h
+++ b/modules/audio_coding/neteq/tools/audio_sink.h
@@ -24,11 +24,11 @@
AudioSink() {}
virtual ~AudioSink() {}
- // Writes |num_samples| from |audio| to the AudioSink. Returns true if
+ // Writes `num_samples` from `audio` to the AudioSink. Returns true if
// successful, otherwise false.
virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
- // Writes |audio_frame| to the AudioSink. Returns true if successful,
+ // Writes `audio_frame` to the AudioSink. Returns true if successful,
// otherwise false.
bool WriteAudioFrame(const AudioFrame& audio_frame) {
return WriteArray(audio_frame.data(), audio_frame.samples_per_channel_ *
diff --git a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
index 6cbba20..18a9103 100644
--- a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
+++ b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
@@ -43,7 +43,7 @@
for (unsigned i = 0; i < 2 * payload_len_samples_; ++i)
packet_memory[kHeaderLenBytes + i] = encoded_sample_[i % 2];
WriteHeader(packet_memory);
- // |packet| assumes ownership of |packet_memory|.
+ // `packet` assumes ownership of `packet_memory`.
auto packet =
std::make_unique<Packet>(std::move(packet_buffer), next_arrival_time_ms_);
next_arrival_time_ms_ += payload_len_samples_ / samples_per_ms_;
diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
index 0260981..7b53653 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.h
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
@@ -54,9 +54,9 @@
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
- // Helper method. Writes |timestamp|, |samples| and
- // |original_payload_size_bytes| to |encoded| in a format that the
- // FakeDecodeFromFile decoder will understand. |encoded| must be at least 12
+ // Helper method. Writes `timestamp`, `samples` and
+ // `original_payload_size_bytes` to `encoded` in a format that the
+ // FakeDecodeFromFile decoder will understand. `encoded` must be at least 12
// bytes long.
static void PrepareEncoded(uint32_t timestamp,
size_t samples,
diff --git a/modules/audio_coding/neteq/tools/input_audio_file.cc b/modules/audio_coding/neteq/tools/input_audio_file.cc
index d5e2862..0d9f0ed 100644
--- a/modules/audio_coding/neteq/tools/input_audio_file.cc
+++ b/modules/audio_coding/neteq/tools/input_audio_file.cc
@@ -81,9 +81,9 @@
size_t samples,
size_t channels,
int16_t* destination) {
- // Start from the end of |source| and |destination|, and work towards the
+ // Start from the end of `source` and `destination`, and work towards the
// beginning. This is to allow in-place interleaving of the same array (i.e.,
- // |source| and |destination| are the same array).
+ // `source` and `destination` are the same array).
for (int i = static_cast<int>(samples - 1); i >= 0; --i) {
for (int j = static_cast<int>(channels - 1); j >= 0; --j) {
destination[i * channels + j] = source[i];
diff --git a/modules/audio_coding/neteq/tools/input_audio_file.h b/modules/audio_coding/neteq/tools/input_audio_file.h
index 4335a99..010d8cc 100644
--- a/modules/audio_coding/neteq/tools/input_audio_file.h
+++ b/modules/audio_coding/neteq/tools/input_audio_file.h
@@ -27,22 +27,22 @@
virtual ~InputAudioFile();
- // Reads |samples| elements from source file to |destination|. Returns true
+ // Reads `samples` elements from source file to `destination`. Returns true
// if the read was successful, otherwise false. If the file end is reached,
// the file is rewound and reading continues from the beginning.
- // The output |destination| must have the capacity to hold |samples| elements.
+ // The output `destination` must have the capacity to hold `samples` elements.
virtual bool Read(size_t samples, int16_t* destination);
- // Fast-forwards (|samples| > 0) or -backwards (|samples| < 0) the file by the
+ // Fast-forwards (`samples` > 0) or -backwards (`samples` < 0) the file by the
// indicated number of samples. Just like Read(), Seek() starts over at the
// beginning of the file if the end is reached. However, seeking backwards
// past the beginning of the file is not possible.
virtual bool Seek(int samples);
// Creates a multi-channel signal from a mono signal. Each sample is repeated
- // |channels| times to create an interleaved multi-channel signal where all
- // channels are identical. The output |destination| must have the capacity to
- // hold samples * channels elements. Note that |source| and |destination| can
+ // `channels` times to create an interleaved multi-channel signal where all
+ // channels are identical. The output `destination` must have the capacity to
+ // hold samples * channels elements. Note that `source` and `destination` can
// be the same array (i.e., point to the same address).
static void DuplicateInterleaved(const int16_t* source,
size_t samples,
diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
index f6b895a..f56ddb7 100644
--- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
+++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
@@ -47,7 +47,7 @@
// as provided by CreateGraphs.
void CreateMatlabScript(const std::string& script_name) const;
- // Creates a python script with file name |script_name|. When executed in
+ // Creates a python script with file name `script_name`. When executed in
// Python, the script will generate graphs with the same timing information
// as provided by CreateGraphs.
void CreatePythonScript(const std::string& script_name) const;
diff --git a/modules/audio_coding/neteq/tools/neteq_input.h b/modules/audio_coding/neteq/tools/neteq_input.h
index 732b807..3a66264 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_input.h
@@ -51,7 +51,7 @@
absl::optional<int64_t> NextEventTime() const {
const auto a = NextPacketTime();
const auto b = NextOutputEventTime();
- // Return the minimum of non-empty |a| and |b|, or empty if both are empty.
+ // Return the minimum of non-empty `a` and `b`, or empty if both are empty.
if (a) {
return b ? std::min(*a, *b) : a;
}
diff --git a/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 1fb853c..ccaa87b 100644
--- a/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -44,7 +44,7 @@
auto audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
auto neteq =
DefaultNetEqFactory().CreateNetEq(config, audio_decoder_factory, clock);
- // Register decoder in |neteq|.
+ // Register decoder in `neteq`.
if (!neteq->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", kSampRateHz, 1)))
return -1;
diff --git a/modules/audio_coding/neteq/tools/neteq_performance_test.h b/modules/audio_coding/neteq/tools/neteq_performance_test.h
index d2212f0..b5b4d91 100644
--- a/modules/audio_coding/neteq/tools/neteq_performance_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_performance_test.h
@@ -19,9 +19,9 @@
class NetEqPerformanceTest {
public:
// Runs a performance test with parameters as follows:
- // |runtime_ms|: the simulation time, i.e., the duration of the audio data.
- // |lossrate|: drop one out of |lossrate| packets, e.g., one out of 10.
- // |drift_factor|: clock drift in [0, 1].
+ // `runtime_ms`: the simulation time, i.e., the duration of the audio data.
+ // `lossrate`: drop one out of `lossrate` packets, e.g., one out of 10.
+ // `drift_factor`: clock drift in [0, 1].
// Returns the runtime in ms.
static int64_t Run(int runtime_ms, int lossrate, double drift_factor);
};
diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index 3f3077f..8322ac2 100644
--- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -120,8 +120,8 @@
// ProbTrans00Solver() is to calculate the transition probability from no-loss
// state to itself in a modified Gilbert Elliot packet loss model. The result is
-// to achieve the target packet loss rate |loss_rate|, when a packet is not
-// lost only if all |units| drawings within the duration of the packet result in
+// to achieve the target packet loss rate `loss_rate`, when a packet is not
+// lost only if all `units` drawings within the duration of the packet result in
// no-loss.
static double ProbTrans00Solver(int units,
double loss_rate,
@@ -310,10 +310,10 @@
int units = block_duration_ms_ / kPacketLossTimeUnitMs;
switch (absl::GetFlag(FLAGS_random_loss_mode)) {
case kUniformLoss: {
- // |unit_loss_rate| is the packet loss rate for each unit time interval
+ // `unit_loss_rate` is the packet loss rate for each unit time interval
// (kPacketLossTimeUnitMs). Since a packet loss event is generated if any
// of |block_duration_ms_ / kPacketLossTimeUnitMs| unit time intervals of
- // a full packet duration is drawn with a loss, |unit_loss_rate| fulfills
+ // a full packet duration is drawn with a loss, `unit_loss_rate` fulfills
// (1 - unit_loss_rate) ^ (block_duration_ms_ / kPacketLossTimeUnitMs) ==
// 1 - packet_loss_rate.
double unit_loss_rate =
@@ -322,7 +322,7 @@
break;
}
case kGilbertElliotLoss: {
- // |FLAGS_burst_length| should be integer times of kPacketLossTimeUnitMs.
+ // `FLAGS_burst_length` should be integer times of kPacketLossTimeUnitMs.
ASSERT_EQ(0, absl::GetFlag(FLAGS_burst_length) % kPacketLossTimeUnitMs);
// We do not allow 100 percent packet loss in Gilbert Elliot model, which
diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.h b/modules/audio_coding/neteq/tools/neteq_quality_test.h
index 11d347a..edcb117 100644
--- a/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -108,9 +108,9 @@
void SetUp() override;
// EncodeBlock(...) does the following:
- // 1. encodes a block of audio, saved in |in_data| and has a length of
- // |block_size_samples| (samples per channel),
- // 2. save the bit stream to |payload| of |max_bytes| bytes in size,
+ // 1. encodes a block of audio, saved in `in_data` and has a length of
+ // `block_size_samples` (samples per channel),
+ // 2. save the bit stream to `payload` of `max_bytes` bytes in size,
// 3. returns the length of the payload (in bytes),
virtual int EncodeBlock(int16_t* in_data,
size_t block_size_samples,
@@ -122,12 +122,12 @@
bool PacketLost();
// DecodeBlock() decodes a block of audio using the payload stored in
- // |payload_| with the length of |payload_size_bytes_| (bytes). The decoded
- // audio is to be stored in |out_data_|.
+ // `payload_` with the length of `payload_size_bytes_` (bytes). The decoded
+ // audio is to be stored in `out_data_`.
int DecodeBlock();
- // Transmit() uses |rtp_generator_| to generate a packet and passes it to
- // |neteq_|.
+ // Transmit() uses `rtp_generator_` to generate a packet and passes it to
+ // `neteq_`.
int Transmit();
// Runs encoding / transmitting / decoding.
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 16a789f..39f05e5 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -157,7 +157,7 @@
namespace {
// Parses the input string for a valid SSRC (at the start of the string). If a
-// valid SSRC is found, it is written to the output variable |ssrc|, and true is
+// valid SSRC is found, it is written to the output variable `ssrc`, and true is
// returned. Otherwise, false is returned.
bool ParseSsrc(const std::string& str, uint32_t* ssrc) {
if (str.empty())
@@ -247,7 +247,7 @@
<< std::endl;
return false;
}
- // Without |output_audio_filename|, |output_files_base_name| is required when
+ // Without `output_audio_filename`, `output_files_base_name` is required when
// plotting output files must be generated (in order to form a valid output
// file name).
if (output_audio_filename.empty() && plotting &&
diff --git a/modules/audio_coding/neteq/tools/neteq_test_factory.h b/modules/audio_coding/neteq/tools/neteq_test_factory.h
index fdfe650..cb9bb1c 100644
--- a/modules/audio_coding/neteq/tools/neteq_test_factory.h
+++ b/modules/audio_coding/neteq/tools/neteq_test_factory.h
@@ -134,7 +134,7 @@
bool enable_fast_accelerate = false;
// Dumps events that describes the simulation on a step-by-step basis.
bool textlog = false;
- // If specified and |textlog| is true, the output of |textlog| is written to
+ // If specified and `textlog` is true, the output of `textlog` is written to
// the specified file name.
absl::optional<std::string> textlog_filename;
// Base name for the output script files for plotting the delay profile.
diff --git a/modules/audio_coding/neteq/tools/output_audio_file.h b/modules/audio_coding/neteq/tools/output_audio_file.h
index f5b0988..ad97722 100644
--- a/modules/audio_coding/neteq/tools/output_audio_file.h
+++ b/modules/audio_coding/neteq/tools/output_audio_file.h
@@ -23,7 +23,7 @@
class OutputAudioFile : public AudioSink {
public:
- // Creates an OutputAudioFile, opening a file named |file_name| for writing.
+ // Creates an OutputAudioFile, opening a file named `file_name` for writing.
// The file format is 16-bit signed host-endian PCM.
explicit OutputAudioFile(const std::string& file_name) {
out_file_ = fopen(file_name.c_str(), "wb");
diff --git a/modules/audio_coding/neteq/tools/output_wav_file.h b/modules/audio_coding/neteq/tools/output_wav_file.h
index 6982a76..ae2e970 100644
--- a/modules/audio_coding/neteq/tools/output_wav_file.h
+++ b/modules/audio_coding/neteq/tools/output_wav_file.h
@@ -22,7 +22,7 @@
class OutputWavFile : public AudioSink {
public:
- // Creates an OutputWavFile, opening a file named |file_name| for writing.
+ // Creates an OutputWavFile, opening a file named `file_name` for writing.
// The output file is a PCM encoded wav file.
OutputWavFile(const std::string& file_name,
int sample_rate_hz,
diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h
index ef118d9..92e5ee9 100644
--- a/modules/audio_coding/neteq/tools/packet.h
+++ b/modules/audio_coding/neteq/tools/packet.h
@@ -55,12 +55,12 @@
virtual ~Packet();
// Parses the first bytes of the RTP payload, interpreting them as RED headers
- // according to RFC 2198. The headers will be inserted into |headers|. The
+ // according to RFC 2198. The headers will be inserted into `headers`. The
// caller of the method assumes ownership of the objects in the list, and
// must delete them properly.
bool ExtractRedHeaders(std::list<RTPHeader*>* headers) const;
- // Deletes all RTPHeader objects in |headers|, but does not delete |headers|
+ // Deletes all RTPHeader objects in `headers`, but does not delete `headers`
// itself.
static void DeleteRedHeaders(std::list<RTPHeader*>* headers);
diff --git a/modules/audio_coding/neteq/tools/packet_unittest.cc b/modules/audio_coding/neteq/tools/packet_unittest.cc
index 7cc9a48..69cf56b 100644
--- a/modules/audio_coding/neteq/tools/packet_unittest.cc
+++ b/modules/audio_coding/neteq/tools/packet_unittest.cc
@@ -124,17 +124,17 @@
}
namespace {
-// Writes one RED block header starting at |rtp_data|, according to RFC 2198.
+// Writes one RED block header starting at `rtp_data`, according to RFC 2198.
// returns the number of bytes written (1 or 4).
//
-// Format if |last_payoad| is false:
+// Format if `last_payoad` is false:
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |1| block PT | timestamp offset | block length |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
-// Format if |last_payoad| is true:
+// Format if `last_payoad` is true:
// 0 1 2 3 4 5 6 7
// +-+-+-+-+-+-+-+-+
// |0| Block PT |
@@ -183,7 +183,7 @@
last_block, payload_ptr);
}
const double kPacketTime = 1.0;
- // Hand over ownership of |packet_memory| to |packet|.
+ // Hand over ownership of `packet_memory` to `packet`.
Packet packet(packet_memory, kPacketLengthBytes, kPacketTime);
ASSERT_TRUE(packet.valid_header());
EXPECT_EQ(kRedPayloadType, packet.header().payloadType);
diff --git a/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/modules/audio_coding/neteq/tools/rtc_event_log_source.h
index 3c91f73..d4be2a7 100644
--- a/modules/audio_coding/neteq/tools/rtc_event_log_source.h
+++ b/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -31,7 +31,7 @@
class RtcEventLogSource : public PacketSource {
public:
- // Creates an RtcEventLogSource reading from |file_name|. If the file cannot
+ // Creates an RtcEventLogSource reading from `file_name`. If the file cannot
// be opened, or has the wrong format, NULL will be returned.
static std::unique_ptr<RtcEventLogSource> CreateFromFile(
const std::string& file_name,
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.h b/modules/audio_coding/neteq/tools/rtp_file_source.h
index 953e2fa..3777e52 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -30,7 +30,7 @@
class RtpFileSource : public PacketSource {
public:
- // Creates an RtpFileSource reading from |file_name|. If the file cannot be
+ // Creates an RtpFileSource reading from `file_name`. If the file cannot be
// opened, or has the wrong format, NULL will be returned.
static RtpFileSource* Create(
const std::string& file_name,
@@ -42,7 +42,7 @@
~RtpFileSource() override;
- // Registers an RTP header extension and binds it to |id|.
+ // Registers an RTP header extension and binds it to `id`.
virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
std::unique_ptr<Packet> NextPacket() override;
diff --git a/modules/audio_coding/neteq/tools/rtp_generator.cc b/modules/audio_coding/neteq/tools/rtp_generator.cc
index 38c30c4f..e883fc1 100644
--- a/modules/audio_coding/neteq/tools/rtp_generator.cc
+++ b/modules/audio_coding/neteq/tools/rtp_generator.cc
@@ -50,7 +50,7 @@
if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <=
jump_from_timestamp_ &&
timestamp_ > jump_from_timestamp_) {
- // We just moved across the |jump_from_timestamp_| timestamp. Do the jump.
+ // We just moved across the `jump_from_timestamp_` timestamp. Do the jump.
timestamp_ = jump_to_timestamp_;
}
return ret;
diff --git a/modules/audio_coding/neteq/tools/rtp_generator.h b/modules/audio_coding/neteq/tools/rtp_generator.h
index 1454c57..6ca6e1b 100644
--- a/modules/audio_coding/neteq/tools/rtp_generator.h
+++ b/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -34,9 +34,9 @@
virtual ~RtpGenerator() {}
- // Writes the next RTP header to |rtp_header|, which will be of type
- // |payload_type|. Returns the send time for this packet (in ms). The value of
- // |payload_length_samples| determines the send time for the next packet.
+ // Writes the next RTP header to `rtp_header`, which will be of type
+ // `payload_type`. Returns the send time for this packet (in ms). The value of
+ // `payload_length_samples` determines the send time for the next packet.
virtual uint32_t GetRtpHeader(uint8_t payload_type,
size_t payload_length_samples,
RTPHeader* rtp_header);
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index 9cb3752..e93df34 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -320,7 +320,7 @@
// If G.722, store half the size to compensate for the timestamp bug in the
// RFC for G.722.
// If iSAC runs in adaptive mode, packet size in samples can change on the
- // fly, so we exclude this test by setting |packet_size_samples_| to -1.
+ // fly, so we exclude this test by setting `packet_size_samples_` to -1.
int clockrate_hz = sampling_freq_hz;
size_t num_channels = 1;
if (absl::EqualsIgnoreCase(codec_name, "G722")) {
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 61d27aa..1b1222c 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -509,8 +509,8 @@
in_file_mono_->FastForward(100);
while (1) {
- // Simulate packet loss by setting |packet_loss_| to "true" in
- // |percent_loss| percent of the loops.
+ // Simulate packet loss by setting `packet_loss_` to "true" in
+ // `percent_loss` percent of the loops.
if (percent_loss > 0) {
if (counter_ == floor((100 / percent_loss) + 0.5)) {
counter_ = 0;
diff --git a/modules/audio_coding/test/TestVADDTX.h b/modules/audio_coding/test/TestVADDTX.h
index cce802d..9c6791a 100644
--- a/modules/audio_coding/test/TestVADDTX.h
+++ b/modules/audio_coding/test/TestVADDTX.h
@@ -23,7 +23,7 @@
namespace webrtc {
// This class records the frame type, and delegates actual sending to the
-// |next_| AudioPacketizationCallback.
+// `next_` AudioPacketizationCallback.
class MonitoringAudioPacketizationCallback : public AudioPacketizationCallback {
public:
explicit MonitoringAudioPacketizationCallback(
@@ -67,9 +67,9 @@
// the expectation. Saves result to a file.
// expects[x] means
// -1 : do not care,
- // 0 : there have been no packets of type |x|,
- // 1 : there have been packets of type |x|,
- // with |x| indicates the following packet types
+ // 0 : there have been no packets of type `x`,
+ // 1 : there have been packets of type `x`,
+ // with `x` indicates the following packet types
// 0 - kEmptyFrame
// 1 - kAudioFrameSpeech
// 2 - kAudioFrameCN
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 5f70c03..6822bc3 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -277,8 +277,8 @@
ASSERT_GE(bitstream_len_byte_int, 0);
bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);
- // Simulate packet loss by setting |packet_loss_| to "true" in
- // |percent_loss| percent of the loops.
+ // Simulate packet loss by setting `packet_loss_` to "true" in
+ // `percent_loss` percent of the loops.
// TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
if (percent_loss > 0) {
if (counter_ == floor((100 / percent_loss) + 0.5)) {
