Reland "Upconvert various types to int.", neteq portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/neteq/ are relanded.
The original commit message is below:
Upconvert various types to int.
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
TBR=kwiberg
Review URL: https://codereview.webrtc.org/1181073002
Cr-Original-Commit-Position: refs/heads/master@{#9427}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 36b7cc32643bae0379d8102ce05dae82ecc466a1
diff --git a/modules/audio_coding/neteq/audio_decoder_impl.cc b/modules/audio_coding/neteq/audio_decoder_impl.cc
index c3f1dbb..99ff95a 100644
--- a/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -163,9 +163,9 @@
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
- int16_t ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
- static_cast<int16_t>(encoded_len), decoded,
- &temp_type);
+ int ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
+ static_cast<int16_t>(encoded_len), decoded,
+ &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
@@ -330,11 +330,11 @@
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
- int16_t ret = WebRtcOpus_Decode(dec_state_, encoded,
- static_cast<int16_t>(encoded_len), decoded,
- &temp_type);
+ int ret = WebRtcOpus_Decode(dec_state_, encoded,
+ static_cast<int16_t>(encoded_len), decoded,
+ &temp_type);
if (ret > 0)
- ret *= static_cast<int16_t>(channels_); // Return total number of samples.
+ ret *= static_cast<int>(channels_); // Return total number of samples.
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
@@ -352,11 +352,11 @@
DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
- int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
- static_cast<int16_t>(encoded_len), decoded,
- &temp_type);
+ int ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
+ static_cast<int16_t>(encoded_len), decoded,
+ &temp_type);
if (ret > 0)
- ret *= static_cast<int16_t>(channels_); // Return total number of samples.
+ ret *= static_cast<int>(channels_); // Return total number of samples.
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
diff --git a/modules/audio_coding/neteq/dsp_helper.cc b/modules/audio_coding/neteq/dsp_helper.cc
index 7451ae2..289e66d 100644
--- a/modules/audio_coding/neteq/dsp_helper.cc
+++ b/modules/audio_coding/neteq/dsp_helper.cc
@@ -272,7 +272,7 @@
}
void DspHelper::UnmuteSignal(const int16_t* input, size_t length,
- int16_t* factor, int16_t increment,
+ int16_t* factor, int increment,
int16_t* output) {
uint16_t factor_16b = *factor;
int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32;
@@ -284,7 +284,7 @@
*factor = factor_16b;
}
-void DspHelper::MuteSignal(int16_t* signal, int16_t mute_slope, size_t length) {
+void DspHelper::MuteSignal(int16_t* signal, int mute_slope, size_t length) {
int32_t factor = (16384 << 6) + 32;
for (size_t i = 0; i < length; i++) {
signal[i] = ((factor >> 6) * signal[i] + 8192) >> 14;
diff --git a/modules/audio_coding/neteq/dsp_helper.h b/modules/audio_coding/neteq/dsp_helper.h
index af4f4d6..f903256 100644
--- a/modules/audio_coding/neteq/dsp_helper.h
+++ b/modules/audio_coding/neteq/dsp_helper.h
@@ -110,11 +110,11 @@
// 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,
- int16_t increment, int16_t* output);
+ 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.
- static void MuteSignal(int16_t* signal, int16_t mute_slope, size_t length);
+ 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|
diff --git a/modules/audio_coding/neteq/expand.cc b/modules/audio_coding/neteq/expand.cc
index bde6559..10f6a9f 100644
--- a/modules/audio_coding/neteq/expand.cc
+++ b/modules/audio_coding/neteq/expand.cc
@@ -239,14 +239,12 @@
if (consecutive_expands_ == 3) {
// Let the mute factor decrease from 1.0 to 0.95 in 6.25 ms.
// mute_slope = 0.0010 / fs_mult in Q20.
- parameters.mute_slope = std::max(parameters.mute_slope,
- static_cast<int16_t>(1049 / fs_mult));
+ parameters.mute_slope = std::max(parameters.mute_slope, 1049 / fs_mult);
}
if (consecutive_expands_ == 7) {
// Let the mute factor decrease from 1.0 to 0.90 in 6.25 ms.
// mute_slope = 0.0020 / fs_mult in Q20.
- parameters.mute_slope = std::max(parameters.mute_slope,
- static_cast<int16_t>(2097 / fs_mult));
+ parameters.mute_slope = std::max(parameters.mute_slope, 2097 / fs_mult);
}
// Mute segment according to slope value.
@@ -368,7 +366,7 @@
InitializeForAnExpandPeriod();
// Calculate correlation in downsampled domain (4 kHz sample rate).
- int16_t correlation_scale;
+ int correlation_scale;
int correlation_length = 51; // TODO(hlundin): Legacy bit-exactness.
// If it is decided to break bit-exactness |correlation_length| should be
// initialized to the return value of Correlation().
@@ -445,7 +443,7 @@
correlation_length + start_index + correlation_lags - 1);
correlation_scale = ((31 - WebRtcSpl_NormW32(signal_max * signal_max))
+ (31 - WebRtcSpl_NormW32(correlation_length))) - 31;
- correlation_scale = std::max(static_cast<int16_t>(0), correlation_scale);
+ correlation_scale = std::max(0, correlation_scale);
// Calculate the correlation, store in |correlation_vector2|.
WebRtcSpl_CrossCorrelation(
@@ -472,7 +470,7 @@
// Calculate the correlation coefficient between the two portions of the
// signal.
- int16_t corr_coefficient;
+ int32_t corr_coefficient;
if ((energy1 > 0) && (energy2 > 0)) {
int energy1_scale = std::max(16 - WebRtcSpl_NormW32(energy1), 0);
int energy2_scale = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
@@ -481,17 +479,17 @@
// If sum is odd, add 1 to make it even.
energy1_scale += 1;
}
- int16_t scaled_energy1 = energy1 >> energy1_scale;
- int16_t scaled_energy2 = energy2 >> energy2_scale;
- int16_t sqrt_energy_product = WebRtcSpl_SqrtFloor(
- scaled_energy1 * scaled_energy2);
+ int32_t scaled_energy1 = energy1 >> energy1_scale;
+ int32_t scaled_energy2 = energy2 >> energy2_scale;
+ int16_t sqrt_energy_product = static_cast<int16_t>(
+ WebRtcSpl_SqrtFloor(scaled_energy1 * scaled_energy2));
// Calculate max_correlation / sqrt(energy1 * energy2) in Q14.
int cc_shift = 14 - (energy1_scale + energy2_scale) / 2;
max_correlation = WEBRTC_SPL_SHIFT_W32(max_correlation, cc_shift);
corr_coefficient = WebRtcSpl_DivW32W16(max_correlation,
sqrt_energy_product);
- corr_coefficient = std::min(static_cast<int16_t>(16384),
- corr_coefficient); // Cap at 1.0 in Q14.
+ // Cap at 1.0 in Q14.
+ corr_coefficient = std::min(16384, corr_coefficient);
} else {
corr_coefficient = 0;
}
@@ -512,8 +510,8 @@
if ((energy1 / 4 < energy2) && (energy1 > energy2 / 4)) {
// Energy constraint fulfilled. Use both vectors and scale them
// accordingly.
- int16_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
- int16_t scaled_energy1 = scaled_energy2 - 13;
+ int32_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
+ int32_t scaled_energy1 = scaled_energy2 - 13;
// Calculate scaled_energy1 / scaled_energy2 in Q13.
int32_t energy_ratio = WebRtcSpl_DivW32W16(
WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
@@ -682,7 +680,8 @@
// voice_mix_factor = 0;
if (corr_coefficient > 7875) {
int16_t x1, x2, x3;
- x1 = corr_coefficient; // |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;
static const int kCoefficients[4] = { -5179, 19931, -16422, 5776 };
@@ -709,7 +708,7 @@
// the division.
// Shift the denominator from Q13 to Q5 before the division. The result of
// the division will then be in Q20.
- int16_t temp_ratio = WebRtcSpl_DivW32W16(
+ int temp_ratio = WebRtcSpl_DivW32W16(
(slope - 8192) << 12,
static_cast<int16_t>((distortion_lag * slope) >> 8));
if (slope > 14746) {
@@ -730,8 +729,7 @@
// Make sure the mute factor decreases from 1.0 to 0.9 in no more than
// 6.25 ms.
// mute_slope >= 0.005 / fs_mult in Q20.
- parameters.mute_slope = std::max(static_cast<int16_t>(5243 / fs_mult),
- parameters.mute_slope);
+ parameters.mute_slope = std::max(5243 / fs_mult, parameters.mute_slope);
} else if (slope > 8028) {
parameters.mute_slope = 0;
}
@@ -755,7 +753,7 @@
void Expand::Correlation(const int16_t* input,
size_t input_length,
int16_t* output,
- int16_t* output_scale) const {
+ int* output_scale) const {
// Set parameters depending on sample rate.
const int16_t* filter_coefficients;
int16_t num_coefficients;
@@ -844,7 +842,7 @@
// TODO(turajs): This can be moved to BackgroundNoise class.
void Expand::GenerateBackgroundNoise(int16_t* random_vector,
size_t channel,
- int16_t mute_slope,
+ int mute_slope,
bool too_many_expands,
size_t num_noise_samples,
int16_t* buffer) {
@@ -887,7 +885,7 @@
bgn_mute_factor > 0) {
// Fade BGN to zero.
// Calculate muting slope, approximately -2^18 / fs_hz.
- int16_t mute_slope;
+ int mute_slope;
if (fs_hz_ == 8000) {
mute_slope = -32;
} else if (fs_hz_ == 16000) {
diff --git a/modules/audio_coding/neteq/expand.h b/modules/audio_coding/neteq/expand.h
index b015959..5fb117d 100644
--- a/modules/audio_coding/neteq/expand.h
+++ b/modules/audio_coding/neteq/expand.h
@@ -72,7 +72,7 @@
void GenerateBackgroundNoise(int16_t* random_vector,
size_t channel,
- int16_t mute_slope,
+ int mute_slope,
bool too_many_expands,
size_t num_noise_samples,
int16_t* buffer);
@@ -113,7 +113,7 @@
AudioVector expand_vector0;
AudioVector expand_vector1;
bool onset;
- int16_t mute_slope; /* Q20 */
+ int mute_slope; /* Q20 */
};
// Calculate the auto-correlation of |input|, with length |input_length|
@@ -123,7 +123,7 @@
void Correlation(const int16_t* input,
size_t input_length,
int16_t* output,
- int16_t* output_scale) const;
+ int* output_scale) const;
void UpdateLagIndex();
diff --git a/modules/audio_coding/neteq/merge.cc b/modules/audio_coding/neteq/merge.cc
index 8e686ba..2c515c1 100644
--- a/modules/audio_coding/neteq/merge.cc
+++ b/modules/audio_coding/neteq/merge.cc
@@ -314,7 +314,7 @@
const int max_corr_length = kMaxCorrelationLength;
int stop_position_downsamp =
std::min(max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
- int16_t correlation_shift = 0;
+ int correlation_shift = 0;
if (expanded_max * input_max > 26843546) {
correlation_shift = 3;
}
@@ -333,7 +333,7 @@
int16_t* correlation_ptr = &correlation16[pad_length];
int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
stop_position_downsamp);
- int16_t norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
+ int norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
WebRtcSpl_VectorBitShiftW32ToW16(correlation_ptr, stop_position_downsamp,
correlation, norm_shift);
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 29b8d1a..6598a79 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -1278,7 +1278,7 @@
*operation == kPreemptiveExpand);
packet_list->pop_front();
size_t payload_length = packet->payload_length;
- int16_t decode_length;
+ int decode_length;
if (packet->sync_packet) {
// Decode to silence with the same frame size as the last decode.
LOG(LS_VERBOSE) << "Decoding sync-packet: " <<
diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index a0e5d2d..bf455c9 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -111,7 +111,7 @@
}
// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
- int16_t increment = 64 / fs_mult;
+ int increment = static_cast<int>(64 / fs_mult);
for (size_t i = 0; i < length_per_channel; i++) {
// Scale with mute factor.
assert(channel_ix < output->Channels());
@@ -178,7 +178,7 @@
// Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
// still ramping up from previous muting.
// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
- int16_t increment = 64 / fs_mult;
+ int increment = static_cast<int>(64 / fs_mult);
size_t length_per_channel = length / output->Channels();
for (size_t i = 0; i < length_per_channel; i++) {
for (size_t channel_ix = 0; channel_ix < output->Channels();
diff --git a/modules/audio_coding/neteq/test/RTPencode.cc b/modules/audio_coding/neteq/test/RTPencode.cc
index f25a279..1aacb40 100644
--- a/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/modules/audio_coding/neteq/test/RTPencode.cc
@@ -1561,7 +1561,7 @@
int useVAD,
int bitrate,
int numChannels) {
- short cdlen = 0;
+ int cdlen = 0;
int16_t* tempdata;
static int first_cng = 1;
int16_t tempLen;
