Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_normal.h b/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
index 0254839..047663f 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
@@ -61,7 +61,8 @@
virtual Operations FuturePacketAvailable(
const SyncBuffer& sync_buffer,
const Expand& expand,
- int decoder_frame_length, Modes prev_mode,
+ int decoder_frame_length,
+ Modes prev_mode,
uint32_t target_timestamp,
uint32_t available_timestamp,
bool play_dtmf);
diff --git a/webrtc/modules/audio_coding/neteq/expand.cc b/webrtc/modules/audio_coding/neteq/expand.cc
index d5f0f9c..bde6559 100644
--- a/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/webrtc/modules/audio_coding/neteq/expand.cc
@@ -214,24 +214,24 @@
// Create combined signal by shifting in more and more of unvoiced part.
temp_shift = 8 - temp_shift; // = getbits(mix_factor_increment).
- size_t temp_lenght = (parameters.current_voice_mix_factor -
+ size_t temp_length = (parameters.current_voice_mix_factor -
parameters.voice_mix_factor) >> temp_shift;
- temp_lenght = std::min(temp_lenght, current_lag);
- DspHelper::CrossFade(voiced_vector, unvoiced_vector, temp_lenght,
+ temp_length = std::min(temp_length, current_lag);
+ DspHelper::CrossFade(voiced_vector, unvoiced_vector, temp_length,
¶meters.current_voice_mix_factor,
mix_factor_increment, temp_data);
// End of cross-fading period was reached before end of expanded signal
// path. Mix the rest with a fixed mixing factor.
- if (temp_lenght < current_lag) {
+ if (temp_length < current_lag) {
if (mix_factor_increment != 0) {
parameters.current_voice_mix_factor = parameters.voice_mix_factor;
}
int16_t temp_scale = 16384 - parameters.current_voice_mix_factor;
WebRtcSpl_ScaleAndAddVectorsWithRound(
- voiced_vector + temp_lenght, parameters.current_voice_mix_factor,
- unvoiced_vector + temp_lenght, temp_scale, 14,
- temp_data + temp_lenght, static_cast<int>(current_lag - temp_lenght));
+ voiced_vector + temp_length, parameters.current_voice_mix_factor,
+ unvoiced_vector + temp_length, temp_scale, 14,
+ temp_data + temp_length, static_cast<int>(current_lag - temp_length));
}
// Select muting slope depending on how many consecutive expands we have
@@ -428,13 +428,12 @@
// Calculate the exact best correlation in the range between
// |correlation_lag| and |distortion_lag|.
- correlation_length = distortion_lag + 10;
- correlation_length = std::min(correlation_length, fs_mult_120);
- correlation_length = std::max(correlation_length, 60 * fs_mult);
+ correlation_length =
+ std::max(std::min(distortion_lag + 10, fs_mult_120), 60 * fs_mult);
int start_index = std::min(distortion_lag, correlation_lag);
- int correlation_lags = WEBRTC_SPL_ABS_W16((distortion_lag-correlation_lag))
- + 1;
+ int correlation_lags =
+ WEBRTC_SPL_ABS_W16((distortion_lag-correlation_lag)) + 1;
assert(correlation_lags <= 99 * fs_mult + 1); // Cannot be larger.
for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
@@ -753,8 +752,10 @@
memset(ar_filter_state, 0, sizeof(ar_filter_state));
}
-int16_t Expand::Correlation(const int16_t* input, size_t input_length,
- int16_t* output, int16_t* output_scale) const {
+void Expand::Correlation(const int16_t* input,
+ size_t input_length,
+ int16_t* output,
+ int16_t* output_scale) const {
// Set parameters depending on sample rate.
const int16_t* filter_coefficients;
int16_t num_coefficients;
@@ -818,7 +819,6 @@
norm_shift2);
// Total scale factor (right shifts) of correlation value.
*output_scale = 2 * norm_shift + kCorrelationShift + norm_shift2;
- return kNumCorrelationLags;
}
void Expand::UpdateLagIndex() {
@@ -850,7 +850,7 @@
int16_t* buffer) {
static const int kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125];
- assert(static_cast<size_t>(kMaxSampleRate / 8000 * 125) >= num_noise_samples);
+ assert(num_noise_samples <= static_cast<size_t>(kMaxSampleRate / 8000 * 125));
int16_t* noise_samples = &buffer[kNoiseLpcOrder];
if (background_noise_->initialized()) {
// Use background noise parameters.
diff --git a/webrtc/modules/audio_coding/neteq/expand.h b/webrtc/modules/audio_coding/neteq/expand.h
index 0000642..b015959 100644
--- a/webrtc/modules/audio_coding/neteq/expand.h
+++ b/webrtc/modules/audio_coding/neteq/expand.h
@@ -119,9 +119,11 @@
// 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|. The scale factor is written to
- // |output_scale|. Returns the length of the correlation vector.
- int16_t Correlation(const int16_t* input, size_t input_length,
- int16_t* output, int16_t* output_scale) const;
+ // |output_scale|.
+ void Correlation(const int16_t* input,
+ size_t input_length,
+ int16_t* output,
+ int16_t* output_scale) const;
void UpdateLagIndex();
diff --git a/webrtc/modules/audio_coding/neteq/merge.cc b/webrtc/modules/audio_coding/neteq/merge.cc
index 8399a78..8e686ba 100644
--- a/webrtc/modules/audio_coding/neteq/merge.cc
+++ b/webrtc/modules/audio_coding/neteq/merge.cc
@@ -312,8 +312,8 @@
int expand_period) const {
// Calculate correlation without any normalization.
const int max_corr_length = kMaxCorrelationLength;
- int stop_position_downsamp = std::min(
- max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
+ int stop_position_downsamp =
+ std::min(max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
int16_t correlation_shift = 0;
if (expanded_max * input_max > 26843546) {
correlation_shift = 3;
@@ -367,9 +367,9 @@
// Ensure that underrun does not occur for 10ms case => we have to get at
// least 10ms + overlap . (This should never happen thanks to the above
// modification of peak-finding starting point.)
- while ((best_correlation_index + input_length) <
- static_cast<int>(timestamps_per_call_ + expand_->overlap_length()) ||
- best_correlation_index + input_length < start_position) {
+ while (((best_correlation_index + input_length) <
+ static_cast<int>(timestamps_per_call_ + expand_->overlap_length())) ||
+ ((best_correlation_index + input_length) < start_position)) {
assert(false); // Should never happen.
best_correlation_index += expand_period; // Jump one lag ahead.
}
diff --git a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index 729dbf6..bf68b3b 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -74,7 +74,7 @@
return -1;
}
payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_,
- encoded_);;
+ encoded_);
int next_send_time = rtp_generator_->GetRtpHeader(
kPayloadType, frame_size_samples_, &rtp_header_);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 3a3ad98..29b8d1a 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -315,9 +315,10 @@
int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
CriticalSectionScoped lock(crit_sect_.get());
assert(decoder_database_.get());
- const int total_samples_in_buffers = packet_buffer_->NumSamplesInBuffer(
- decoder_database_.get(), decoder_frame_length_) +
- static_cast<int>(sync_buffer_->FutureLength());
+ const int total_samples_in_buffers =
+ packet_buffer_->NumSamplesInBuffer(decoder_database_.get(),
+ decoder_frame_length_) +
+ static_cast<int>(sync_buffer_->FutureLength());
assert(delay_manager_.get());
assert(decision_logic_.get());
stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
@@ -704,8 +705,10 @@
return 0;
}
-int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
- int* samples_per_channel, int* num_channels) {
+int NetEqImpl::GetAudioInternal(size_t max_length,
+ int16_t* output,
+ int* samples_per_channel,
+ int* num_channels) {
PacketList packet_list;
DtmfEvent dtmf_event;
Operations operation;
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.cc b/webrtc/modules/audio_coding/neteq/packet_buffer.cc
index 08b237f..5792b22 100644
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.cc
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.cc
@@ -250,16 +250,12 @@
Packet* packet = (*it);
AudioDecoder* decoder =
decoder_database->GetDecoder(packet->header.payloadType);
- if (decoder) {
- int duration;
- if (packet->sync_packet) {
- duration = last_duration;
- } else if (packet->primary) {
- duration =
- decoder->PacketDuration(packet->payload, packet->payload_length);
- } else {
+ if (decoder && !packet->sync_packet) {
+ if (!packet->primary) {
continue;
}
+ int duration =
+ decoder->PacketDuration(packet->payload, packet->payload_length);
if (duration >= 0) {
last_duration = duration; // Save the most up-to-date (valid) duration.
}
diff --git a/webrtc/modules/audio_coding/neteq/preemptive_expand.h b/webrtc/modules/audio_coding/neteq/preemptive_expand.h
index c583a48..65da703 100644
--- a/webrtc/modules/audio_coding/neteq/preemptive_expand.h
+++ b/webrtc/modules/audio_coding/neteq/preemptive_expand.h
@@ -52,9 +52,9 @@
protected:
// 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* w16_bestCorr,
- int* w16_bestIndex) const override;
+ void SetParametersForPassiveSpeech(size_t input_length,
+ int16_t* best_correlation,
+ int* peak_index) const override;
// Checks the criteria for performing the time-stretching operation and,
// if possible, performs the time-stretching.
diff --git a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
index f637eb8..ce800dd 100644
--- a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
@@ -149,7 +149,7 @@
stats->speech_expand_rate =
CalculateQ14Ratio(expanded_speech_samples_,
- timestamps_since_last_report_);
+ timestamps_since_last_report_);
stats->secondary_decoded_rate =
CalculateQ14Ratio(secondary_decoded_samples_,
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index 192d374..f25a279 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -152,7 +152,7 @@
#ifdef NETEQ_ISACFIX_CODEC
#include "isacfix.h"
#ifdef CODEC_ISAC
-#error Cannot have both ISAC and ISACfix defined. Please de-select one in the beginning of RTPencode.cpp
+#error Cannot have both ISAC and ISACfix defined. Please de-select one.
#endif
#endif
#ifdef CODEC_G722
diff --git a/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc b/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
index 65c4e9d..af4b8e1 100644
--- a/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
@@ -32,7 +32,7 @@
timestamp_(0),
payload_ssrc_(0xABCD1234) {
int encoded_len = WebRtcPcm16b_Encode(&sample_value, 1, encoded_sample_);
- CHECK_EQ(encoded_len, 2);
+ CHECK_EQ(2, encoded_len);
}
Packet* ConstantPcmPacketSource::NextPacket() {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 080b99b..1c76d76 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -103,8 +103,8 @@
static const int kMaxChannels = 1;
static const int kMaxSamplesPerMs = 48000 / 1000;
static const int kOutputBlockSizeMs = 10;
- static const int kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs *
- kMaxChannels;
+ static const int kOutDataLen =
+ kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
int16_t out_data[kOutDataLen];
int num_channels;
int samples_per_channel;
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 11dd20a..6bcd717 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -552,8 +552,8 @@
// Check if it is time to get output audio.
if (time_now_ms >= next_output_time_ms) {
- static const int kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs *
- kMaxChannels;
+ static const int kOutDataLen =
+ kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
int16_t out_data[kOutDataLen];
int num_channels;
int samples_per_channel;