NetEq4: Removing templatization for AudioMultiVector
This saves approx 6% runtime for neteq4_speed_test.
$ time out/Release/neteq4_speed_test --runtime_ms=50000000
BUG=1363
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2320006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4885 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/accelerate.cc b/webrtc/modules/audio_coding/neteq4/accelerate.cc
index 37ce2e7..88cfa4d 100644
--- a/webrtc/modules/audio_coding/neteq4/accelerate.cc
+++ b/webrtc/modules/audio_coding/neteq4/accelerate.cc
@@ -17,7 +17,7 @@
Accelerate::ReturnCodes Accelerate::Process(
const int16_t* input,
size_t input_length,
- AudioMultiVector<int16_t>* output,
+ AudioMultiVector* output,
int16_t* length_change_samples) {
// Input length must be (almost) 30 ms.
static const int k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate.
@@ -43,7 +43,7 @@
Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
const int16_t* input, size_t input_length, size_t peak_index,
int16_t best_correlation, bool active_speech,
- AudioMultiVector<int16_t>* output) const {
+ AudioMultiVector* output) const {
// Check for strong correlation or passive speech.
if ((best_correlation > kCorrelationThreshold) || !active_speech) {
// Do accelerate operation by overlap add.
@@ -56,7 +56,7 @@
// Copy first part; 0 to 15 ms.
output->PushBackInterleaved(input, fs_mult_120 * num_channels_);
// Copy the |peak_index| starting at 15 ms to |temp_vector|.
- AudioMultiVector<int16_t> temp_vector(num_channels_);
+ AudioMultiVector temp_vector(num_channels_);
temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_],
peak_index * num_channels_);
// Cross-fade |temp_vector| onto the end of |output|.
diff --git a/webrtc/modules/audio_coding/neteq4/accelerate.h b/webrtc/modules/audio_coding/neteq4/accelerate.h
index 375a6f0..83e3e38 100644
--- a/webrtc/modules/audio_coding/neteq4/accelerate.h
+++ b/webrtc/modules/audio_coding/neteq4/accelerate.h
@@ -43,7 +43,7 @@
// the outcome of the operation as an enumerator value.
ReturnCodes Process(const int16_t* input,
size_t input_length,
- AudioMultiVector<int16_t>* output,
+ AudioMultiVector* output,
int16_t* length_change_samples);
protected:
@@ -58,7 +58,7 @@
virtual ReturnCodes CheckCriteriaAndStretch(
const int16_t* input, size_t input_length, size_t peak_index,
int16_t best_correlation, bool active_speech,
- AudioMultiVector<int16_t>* output) const OVERRIDE;
+ AudioMultiVector* output) const OVERRIDE;
private:
DISALLOW_COPY_AND_ASSIGN(Accelerate);
diff --git a/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc b/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc
index 7eb2142..03baebd 100644
--- a/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc
+++ b/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc
@@ -18,50 +18,44 @@
namespace webrtc {
-template<typename T>
-AudioMultiVector<T>::AudioMultiVector(size_t N) {
+AudioMultiVector::AudioMultiVector(size_t N) {
assert(N > 0);
if (N < 1) N = 1;
for (size_t n = 0; n < N; ++n) {
- channels_.push_back(new AudioVector<T>);
+ channels_.push_back(new AudioVector<int16_t>);
}
}
-template<typename T>
-AudioMultiVector<T>::AudioMultiVector(size_t N, size_t initial_size) {
+AudioMultiVector::AudioMultiVector(size_t N, size_t initial_size) {
assert(N > 0);
if (N < 1) N = 1;
for (size_t n = 0; n < N; ++n) {
- channels_.push_back(new AudioVector<T>(initial_size));
+ channels_.push_back(new AudioVector<int16_t>(initial_size));
}
}
-template<typename T>
-AudioMultiVector<T>::~AudioMultiVector() {
- typename std::vector<AudioVector<T>*>::iterator it = channels_.begin();
+AudioMultiVector::~AudioMultiVector() {
+ std::vector<AudioVector<int16_t>*>::iterator it = channels_.begin();
while (it != channels_.end()) {
delete (*it);
++it;
}
}
-template<typename T>
-void AudioMultiVector<T>::Clear() {
+void AudioMultiVector::Clear() {
for (size_t i = 0; i < Channels(); ++i) {
channels_[i]->Clear();
}
}
-template<typename T>
-void AudioMultiVector<T>::Zeros(size_t length) {
+void AudioMultiVector::Zeros(size_t length) {
for (size_t i = 0; i < Channels(); ++i) {
channels_[i]->Clear();
channels_[i]->Extend(length);
}
}
-template<typename T>
-void AudioMultiVector<T>::CopyFrom(AudioMultiVector<T>* copy_to) const {
+void AudioMultiVector::CopyFrom(AudioMultiVector* copy_to) const {
if (copy_to) {
for (size_t i = 0; i < Channels(); ++i) {
channels_[i]->CopyFrom(&(*copy_to)[i]);
@@ -69,9 +63,8 @@
}
}
-template<typename T>
-void AudioMultiVector<T>::PushBackInterleaved(const T* append_this,
- size_t length) {
+void AudioMultiVector::PushBackInterleaved(const int16_t* append_this,
+ size_t length) {
assert(length % Channels() == 0);
if (Channels() == 1) {
// Special case to avoid extra allocation and data shuffling.
@@ -79,11 +72,12 @@
return;
}
size_t length_per_channel = length / Channels();
- T* temp_array = new T[length_per_channel]; // Intermediate storage.
+ int16_t* temp_array =
+ new int16_t[length_per_channel]; // Intermediate storage.
for (size_t channel = 0; channel < Channels(); ++channel) {
// Copy elements to |temp_array|.
// Set |source_ptr| to first element of this channel.
- const T* source_ptr = &append_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;
source_ptr += Channels(); // Jump to next element of this channel.
@@ -93,8 +87,7 @@
delete [] temp_array;
}
-template<typename T>
-void AudioMultiVector<T>::PushBack(const AudioMultiVector<T>& append_this) {
+void AudioMultiVector::PushBack(const AudioMultiVector& append_this) {
assert(Channels() == append_this.Channels());
if (Channels() == append_this.Channels()) {
for (size_t i = 0; i < Channels(); ++i) {
@@ -103,10 +96,8 @@
}
}
-template<typename T>
-void AudioMultiVector<T>::PushBackFromIndex(
- const AudioMultiVector<T>& append_this,
- size_t index) {
+void AudioMultiVector::PushBackFromIndex(const AudioMultiVector& append_this,
+ size_t index) {
assert(index < append_this.Size());
index = std::min(index, append_this.Size() - 1);
size_t length = append_this.Size() - index;
@@ -118,30 +109,26 @@
}
}
-template<typename T>
-void AudioMultiVector<T>::PopFront(size_t length) {
+void AudioMultiVector::PopFront(size_t length) {
for (size_t i = 0; i < Channels(); ++i) {
channels_[i]->PopFront(length);
}
}
-template<typename T>
-void AudioMultiVector<T>::PopBack(size_t length) {
+void AudioMultiVector::PopBack(size_t length) {
for (size_t i = 0; i < Channels(); ++i) {
channels_[i]->PopBack(length);
}
}
-template<typename T>
-size_t AudioMultiVector<T>::ReadInterleaved(size_t length,
- T* destination) const {
+size_t AudioMultiVector::ReadInterleaved(size_t length,
+ int16_t* destination) const {
return ReadInterleavedFromIndex(0, length, destination);
}
-template<typename T>
-size_t AudioMultiVector<T>::ReadInterleavedFromIndex(size_t start_index,
- size_t length,
- T* destination) const {
+size_t AudioMultiVector::ReadInterleavedFromIndex(size_t start_index,
+ size_t length,
+ int16_t* destination) const {
if (!destination) {
return 0;
}
@@ -160,17 +147,15 @@
return index;
}
-template<typename T>
-size_t AudioMultiVector<T>::ReadInterleavedFromEnd(size_t length,
- T* destination) const {
+size_t AudioMultiVector::ReadInterleavedFromEnd(size_t length,
+ int16_t* destination) const {
length = std::min(length, Size()); // Cannot read more than Size() elements.
return ReadInterleavedFromIndex(Size() - length, length, destination);
}
-template<typename T>
-void AudioMultiVector<T>::OverwriteAt(const AudioMultiVector<T>& insert_this,
- size_t length,
- size_t position) {
+void AudioMultiVector::OverwriteAt(const AudioMultiVector& insert_this,
+ size_t length,
+ size_t position) {
assert(Channels() == insert_this.Channels());
// Cap |length| at the length of |insert_this|.
assert(length <= insert_this.Size());
@@ -182,9 +167,8 @@
}
}
-template<typename T>
-void AudioMultiVector<T>::CrossFade(const AudioMultiVector<T>& append_this,
- size_t fade_length) {
+void AudioMultiVector::CrossFade(const AudioMultiVector& append_this,
+ size_t fade_length) {
assert(Channels() == append_this.Channels());
if (Channels() == append_this.Channels()) {
for (size_t i = 0; i < Channels(); ++i) {
@@ -193,14 +177,12 @@
}
}
-template<typename T>
-size_t AudioMultiVector<T>::Size() const {
+size_t AudioMultiVector::Size() const {
assert(channels_[0]);
return channels_[0]->Size();
}
-template<typename T>
-void AudioMultiVector<T>::AssertSize(size_t required_size) {
+void AudioMultiVector::AssertSize(size_t required_size) {
if (Size() < required_size) {
size_t extend_length = required_size - Size();
for (size_t channel = 0; channel < Channels(); ++channel) {
@@ -209,25 +191,17 @@
}
}
-template<typename T>
-bool AudioMultiVector<T>::Empty() const {
+bool AudioMultiVector::Empty() const {
assert(channels_[0]);
return channels_[0]->Empty();
}
-template<typename T>
-const AudioVector<T>& AudioMultiVector<T>::operator[](size_t index) const {
+const AudioVector<int16_t>& AudioMultiVector::operator[](size_t index) const {
return *(channels_[index]);
}
-template<typename T>
-AudioVector<T>& AudioMultiVector<T>::operator[](size_t index) {
+AudioVector<int16_t>& AudioMultiVector::operator[](size_t index) {
return *(channels_[index]);
}
-// Instantiate the template for a few types.
-template class AudioMultiVector<int16_t>;
-template class AudioMultiVector<int32_t>;
-template class AudioMultiVector<double>;
-
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/audio_multi_vector.h b/webrtc/modules/audio_coding/neteq4/audio_multi_vector.h
index 7167c67..ef3a1cf 100644
--- a/webrtc/modules/audio_coding/neteq4/audio_multi_vector.h
+++ b/webrtc/modules/audio_coding/neteq4/audio_multi_vector.h
@@ -17,10 +17,10 @@
#include "webrtc/modules/audio_coding/neteq4/audio_vector.h"
#include "webrtc/system_wrappers/interface/constructor_magic.h"
+#include "webrtc/typedefs.h"
namespace webrtc {
-template <typename T>
class AudioMultiVector {
public:
// Creates an empty AudioMultiVector with |N| audio channels. |N| must be
@@ -43,23 +43,23 @@
// 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 CopyFrom(AudioMultiVector<T>* copy_to) const;
+ virtual void CopyFrom(AudioMultiVector* copy_to) const;
// Appends the contents of array |append_this| to the end of this
// object. The array is assumed to be channel-interleaved. |length| must be
// an even multiple of this object's number of channels.
// The length of this object is increased with the |length| divided by the
// number of channels.
- virtual void PushBackInterleaved(const T* append_this, size_t length);
+ virtual void PushBackInterleaved(const int16_t* append_this, size_t length);
// 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<T>& 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
// object is increased.
- virtual void PushBackFromIndex(const AudioMultiVector<T>& append_this,
+ virtual void PushBackFromIndex(const AudioMultiVector& append_this,
size_t index);
// Removes |length| elements from the beginning of this object, from each
@@ -75,18 +75,18 @@
// 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, T* destination) const;
+ virtual size_t ReadInterleaved(size_t length, int16_t* destination) const;
// Like ReadInterleaved() above, but reads from |start_index| instead of from
// the beginning.
virtual size_t ReadInterleavedFromIndex(size_t start_index,
size_t length,
- T* destination) const;
+ int16_t* destination) const;
// Like ReadInterleaved() above, but reads from the end instead of from
// the beginning.
virtual size_t ReadInterleavedFromEnd(size_t length,
- T* destination) const;
+ 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
@@ -95,14 +95,14 @@
// 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|.
- virtual void OverwriteAt(const AudioMultiVector<T>& 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
// linearly in this region.
- virtual void CrossFade(const AudioMultiVector<T>& append_this,
+ virtual void CrossFade(const AudioMultiVector& append_this,
size_t fade_length);
// Returns the number of channels.
@@ -119,11 +119,11 @@
// Accesses and modifies a channel (i.e., an AudioVector object) of this
// AudioMultiVector.
- const AudioVector<T>& operator[](size_t index) const;
- AudioVector<T>& operator[](size_t index);
+ const AudioVector<int16_t>& operator[](size_t index) const;
+ AudioVector<int16_t>& operator[](size_t index);
protected:
- std::vector<AudioVector<T>*> channels_;
+ std::vector<AudioVector<int16_t>*> channels_;
private:
DISALLOW_COPY_AND_ASSIGN(AudioMultiVector);
diff --git a/webrtc/modules/audio_coding/neteq4/audio_multi_vector_unittest.cc b/webrtc/modules/audio_coding/neteq4/audio_multi_vector_unittest.cc
index d981ccb..15e64d4 100644
--- a/webrtc/modules/audio_coding/neteq4/audio_multi_vector_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/audio_multi_vector_unittest.cc
@@ -30,12 +30,10 @@
class AudioMultiVectorTest : public ::testing::TestWithParam<size_t> {
protected:
- typedef int16_t T; // Use this value type for all tests.
-
AudioMultiVectorTest()
: num_channels_(GetParam()), // Get the test parameter.
interleaved_length_(num_channels_ * array_length()) {
- array_interleaved_ = new T[num_channels_ * array_length()];
+ array_interleaved_ = new int16_t[num_channels_ * array_length()];
}
~AudioMultiVectorTest() {
@@ -45,9 +43,9 @@
virtual void SetUp() {
// Populate test arrays.
for (size_t i = 0; i < array_length(); ++i) {
- array_[i] = static_cast<T>(i);
+ array_[i] = static_cast<int16_t>(i);
}
- T* ptr = array_interleaved_;
+ int16_t* ptr = array_interleaved_;
// Write 100, 101, 102, ... for first channel.
// Write 200, 201, 202, ... for second channel.
// And so on.
@@ -65,20 +63,20 @@
const size_t num_channels_;
size_t interleaved_length_;
- T array_[10];
- T* array_interleaved_;
+ int16_t array_[10];
+ int16_t* array_interleaved_;
};
// Create and destroy AudioMultiVector objects, both empty and with a predefined
// length.
TEST_P(AudioMultiVectorTest, CreateAndDestroy) {
- AudioMultiVector<T> vec1(num_channels_);
+ AudioMultiVector vec1(num_channels_);
EXPECT_TRUE(vec1.Empty());
EXPECT_EQ(num_channels_, vec1.Channels());
EXPECT_EQ(0u, vec1.Size());
size_t initial_size = 17;
- AudioMultiVector<T> vec2(num_channels_, initial_size);
+ AudioMultiVector vec2(num_channels_, initial_size);
EXPECT_FALSE(vec2.Empty());
EXPECT_EQ(num_channels_, vec2.Channels());
EXPECT_EQ(initial_size, vec2.Size());
@@ -86,13 +84,13 @@
// Test the subscript operator [] for getting and setting.
TEST_P(AudioMultiVectorTest, SubscriptOperator) {
- AudioMultiVector<T> vec(num_channels_, array_length());
+ AudioMultiVector vec(num_channels_, array_length());
for (size_t channel = 0; channel < num_channels_; ++channel) {
for (size_t i = 0; i < array_length(); ++i) {
- vec[channel][i] = static_cast<T>(i);
+ vec[channel][i] = static_cast<int16_t>(i);
// Make sure to use the const version.
- const AudioVector<T>& audio_vec = vec[channel];
- EXPECT_EQ(static_cast<T>(i), audio_vec[i]);
+ const AudioVector<int16_t>& audio_vec = vec[channel];
+ EXPECT_EQ(static_cast<int16_t>(i), audio_vec[i]);
}
}
}
@@ -100,9 +98,9 @@
// Test the PushBackInterleaved method and the CopyFrom method. The Clear
// method is also invoked.
TEST_P(AudioMultiVectorTest, PushBackInterleavedAndCopy) {
- AudioMultiVector<T> vec(num_channels_);
+ AudioMultiVector vec(num_channels_);
vec.PushBackInterleaved(array_interleaved_, interleaved_length_);
- AudioMultiVector<T> vec_copy(num_channels_);
+ AudioMultiVector vec_copy(num_channels_);
vec.CopyFrom(&vec_copy); // Copy from |vec| to |vec_copy|.
ASSERT_EQ(num_channels_, vec.Channels());
ASSERT_EQ(array_length(), vec.Size());
@@ -110,7 +108,7 @@
ASSERT_EQ(array_length(), vec_copy.Size());
for (size_t channel = 0; channel < vec.Channels(); ++channel) {
for (size_t i = 0; i < array_length(); ++i) {
- EXPECT_EQ(static_cast<T>((channel + 1) * 100 + i), vec[channel][i]);
+ EXPECT_EQ(static_cast<int16_t>((channel + 1) * 100 + i), vec[channel][i]);
EXPECT_EQ(vec[channel][i], vec_copy[channel][i]);
}
}
@@ -126,24 +124,25 @@
// Try to copy to a NULL pointer. Nothing should happen.
TEST_P(AudioMultiVectorTest, CopyToNull) {
- AudioMultiVector<T> vec(num_channels_);
- AudioMultiVector<T>* vec_copy = NULL;
+ AudioMultiVector vec(num_channels_);
+ AudioMultiVector* vec_copy = NULL;
vec.PushBackInterleaved(array_interleaved_, interleaved_length_);
vec.CopyFrom(vec_copy);
}
// Test the PushBack method with another AudioMultiVector as input argument.
TEST_P(AudioMultiVectorTest, PushBackVector) {
- AudioMultiVector<T> vec1(num_channels_, array_length());
- AudioMultiVector<T> vec2(num_channels_, array_length());
+ AudioMultiVector vec1(num_channels_, array_length());
+ AudioMultiVector vec2(num_channels_, array_length());
// Set the first vector to [0, 1, ..., array_length() - 1] +
// 100 * channel_number.
// Set the second vector to [array_length(), array_length() + 1, ...,
// 2 * array_length() - 1] + 100 * channel_number.
for (size_t channel = 0; channel < num_channels_; ++channel) {
for (size_t i = 0; i < array_length(); ++i) {
- vec1[channel][i] = static_cast<T>(i + 100 * channel);
- vec2[channel][i] = static_cast<T>(i + 100 * channel + array_length());
+ vec1[channel][i] = static_cast<int16_t>(i + 100 * channel);
+ vec2[channel][i] =
+ static_cast<int16_t>(i + 100 * channel + array_length());
}
}
// Append vec2 to the back of vec1.
@@ -151,16 +150,16 @@
ASSERT_EQ(2u * array_length(), vec1.Size());
for (size_t channel = 0; channel < num_channels_; ++channel) {
for (size_t i = 0; i < 2 * array_length(); ++i) {
- EXPECT_EQ(static_cast<T>(i + 100 * channel), vec1[channel][i]);
+ EXPECT_EQ(static_cast<int16_t>(i + 100 * channel), vec1[channel][i]);
}
}
}
// Test the PushBackFromIndex method.
TEST_P(AudioMultiVectorTest, PushBackFromIndex) {
- AudioMultiVector<T> vec1(num_channels_);
+ AudioMultiVector vec1(num_channels_);
vec1.PushBackInterleaved(array_interleaved_, interleaved_length_);
- AudioMultiVector<T> vec2(num_channels_);
+ AudioMultiVector vec2(num_channels_);
// Append vec1 to the back of vec2 (which is empty). Read vec1 from the second
// last element.
@@ -176,7 +175,7 @@
// Starts with pushing some values to the vector, then test the Zeros method.
TEST_P(AudioMultiVectorTest, Zeros) {
- AudioMultiVector<T> vec(num_channels_);
+ AudioMultiVector vec(num_channels_);
vec.PushBackInterleaved(array_interleaved_, interleaved_length_);
vec.Zeros(2 * array_length());
ASSERT_EQ(num_channels_, vec.Channels());
@@ -190,28 +189,30 @@
// Test the ReadInterleaved method
TEST_P(AudioMultiVectorTest, ReadInterleaved) {
- AudioMultiVector<T> vec(num_channels_);
+ AudioMultiVector vec(num_channels_);
vec.PushBackInterleaved(array_interleaved_, interleaved_length_);
- T* output = new T[interleaved_length_];
+ int16_t* output = new int16_t[interleaved_length_];
// Read 5 samples.
size_t read_samples = 5;
EXPECT_EQ(num_channels_ * read_samples,
vec.ReadInterleaved(read_samples, output));
- EXPECT_EQ(0, memcmp(array_interleaved_, output, read_samples * sizeof(T)));
+ EXPECT_EQ(0,
+ memcmp(array_interleaved_, output, read_samples * sizeof(int16_t)));
// Read too many samples. Expect to get all samples from the vector.
EXPECT_EQ(interleaved_length_,
vec.ReadInterleaved(array_length() + 1, output));
- EXPECT_EQ(0, memcmp(array_interleaved_, output, read_samples * sizeof(T)));
+ EXPECT_EQ(0,
+ memcmp(array_interleaved_, output, read_samples * sizeof(int16_t)));
delete [] output;
}
// Try to read to a NULL pointer. Expected to return 0.
TEST_P(AudioMultiVectorTest, ReadInterleavedToNull) {
- AudioMultiVector<T> vec(num_channels_);
+ AudioMultiVector vec(num_channels_);
vec.PushBackInterleaved(array_interleaved_, interleaved_length_);
- T* output = NULL;
+ int16_t* output = NULL;
// Read 5 samples.
size_t read_samples = 5;
EXPECT_EQ(0u, vec.ReadInterleaved(read_samples, output));
@@ -219,13 +220,13 @@
// Test the PopFront method.
TEST_P(AudioMultiVectorTest, PopFront) {
- AudioMultiVector<T> vec(num_channels_);
+ AudioMultiVector vec(num_channels_);
vec.PushBackInterleaved(array_interleaved_, interleaved_length_);
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
// interleaved array.
- T* ptr = &array_interleaved_[num_channels_];
+ int16_t* ptr = &array_interleaved_[num_channels_];
for (size_t i = 0; i < array_length() - 1; ++i) {
for (size_t channel = 0; channel < num_channels_; ++channel) {
EXPECT_EQ(*ptr, vec[channel][i]);
@@ -238,13 +239,13 @@
// Test the PopBack method.
TEST_P(AudioMultiVectorTest, PopBack) {
- AudioMultiVector<T> vec(num_channels_);
+ AudioMultiVector vec(num_channels_);
vec.PushBackInterleaved(array_interleaved_, interleaved_length_);
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
// interleaved array.
- T* ptr = array_interleaved_;
+ int16_t* ptr = array_interleaved_;
for (size_t i = 0; i < array_length() - 1; ++i) {
for (size_t channel = 0; channel < num_channels_; ++channel) {
EXPECT_EQ(*ptr, vec[channel][i]);
@@ -257,7 +258,7 @@
// Test the AssertSize method.
TEST_P(AudioMultiVectorTest, AssertSize) {
- AudioMultiVector<T> vec(num_channels_, array_length());
+ AudioMultiVector vec(num_channels_, array_length());
EXPECT_EQ(array_length(), vec.Size());
// Start with asserting with smaller sizes than already allocated.
vec.AssertSize(0);
@@ -276,16 +277,16 @@
// Test the PushBack method with another AudioMultiVector as input argument.
TEST_P(AudioMultiVectorTest, OverwriteAt) {
- AudioMultiVector<T> vec1(num_channels_);
+ AudioMultiVector vec1(num_channels_);
vec1.PushBackInterleaved(array_interleaved_, interleaved_length_);
- AudioMultiVector<T> vec2(num_channels_);
+ AudioMultiVector vec2(num_channels_);
vec2.Zeros(3); // 3 zeros in each channel.
// Overwrite vec2 at position 5.
vec1.OverwriteAt(vec2, 3, 5);
// Verify result.
// Length remains the same.
ASSERT_EQ(array_length(), vec1.Size());
- T* ptr = array_interleaved_;
+ int16_t* ptr = array_interleaved_;
for (size_t i = 0; i < array_length() - 1; ++i) {
for (size_t channel = 0; channel < num_channels_; ++channel) {
if (i >= 5 && i <= 7) {
diff --git a/webrtc/modules/audio_coding/neteq4/background_noise.cc b/webrtc/modules/audio_coding/neteq4/background_noise.cc
index f306466..2dfb3c1 100644
--- a/webrtc/modules/audio_coding/neteq4/background_noise.cc
+++ b/webrtc/modules/audio_coding/neteq4/background_noise.cc
@@ -38,7 +38,7 @@
// Keep _bgnMode as it is.
}
-void BackgroundNoise::Update(const AudioMultiVector<int16_t>& input,
+void BackgroundNoise::Update(const AudioMultiVector& input,
const PostDecodeVad& vad) {
if (vad.running() && vad.active_speech()) {
// Do not update the background noise parameters if we know that the signal
diff --git a/webrtc/modules/audio_coding/neteq4/background_noise.h b/webrtc/modules/audio_coding/neteq4/background_noise.h
index 8d4237f..ac5446b 100644
--- a/webrtc/modules/audio_coding/neteq4/background_noise.h
+++ b/webrtc/modules/audio_coding/neteq4/background_noise.h
@@ -38,7 +38,7 @@
// Updates the parameter estimates based on the signal currently in the
// |sync_buffer|, and on the latest decision in |vad| if it is running.
- void Update(const AudioMultiVector<int16_t>& sync_buffer,
+ void Update(const AudioMultiVector& sync_buffer,
const PostDecodeVad& vad);
// Returns |energy_| for |channel|.
diff --git a/webrtc/modules/audio_coding/neteq4/comfort_noise.cc b/webrtc/modules/audio_coding/neteq4/comfort_noise.cc
index a56d967..360767a 100644
--- a/webrtc/modules/audio_coding/neteq4/comfort_noise.cc
+++ b/webrtc/modules/audio_coding/neteq4/comfort_noise.cc
@@ -50,7 +50,7 @@
}
int ComfortNoise::Generate(size_t requested_length,
- AudioMultiVector<int16_t>* output) {
+ AudioMultiVector* output) {
// TODO(hlundin): Change to an enumerator and skip assert.
assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
fs_hz_ == 48000);
diff --git a/webrtc/modules/audio_coding/neteq4/comfort_noise.h b/webrtc/modules/audio_coding/neteq4/comfort_noise.h
index af0501f..7e7c294 100644
--- a/webrtc/modules/audio_coding/neteq4/comfort_noise.h
+++ b/webrtc/modules/audio_coding/neteq4/comfort_noise.h
@@ -53,7 +53,7 @@
// |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<int16_t>* output);
+ int Generate(size_t requested_length, AudioMultiVector* output);
// Returns the last error code that was produced by the comfort noise
// decoder. Returns 0 if no error has been encountered since the last reset.
diff --git a/webrtc/modules/audio_coding/neteq4/dsp_helper.cc b/webrtc/modules/audio_coding/neteq4/dsp_helper.cc
index 7493e10..e1aa0e5 100644
--- a/webrtc/modules/audio_coding/neteq4/dsp_helper.cc
+++ b/webrtc/modules/audio_coding/neteq4/dsp_helper.cc
@@ -80,7 +80,7 @@
return RampSignal(signal, length, factor, increment, signal);
}
-int DspHelper::RampSignal(AudioMultiVector<int16_t>* signal,
+int DspHelper::RampSignal(AudioMultiVector* signal,
size_t start_index,
size_t length,
int factor,
diff --git a/webrtc/modules/audio_coding/neteq4/dsp_helper.h b/webrtc/modules/audio_coding/neteq4/dsp_helper.h
index 9df6fd3..60cd995 100644
--- a/webrtc/modules/audio_coding/neteq4/dsp_helper.h
+++ b/webrtc/modules/audio_coding/neteq4/dsp_helper.h
@@ -67,7 +67,7 @@
// Same as above, but processes |length| samples from |signal|, starting at
// |start_index|.
- static int RampSignal(AudioMultiVector<int16_t>* signal,
+ static int RampSignal(AudioMultiVector* signal,
size_t start_index,
size_t length,
int factor,
diff --git a/webrtc/modules/audio_coding/neteq4/dsp_helper_unittest.cc b/webrtc/modules/audio_coding/neteq4/dsp_helper_unittest.cc
index d3c76df..852c2ec 100644
--- a/webrtc/modules/audio_coding/neteq4/dsp_helper_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/dsp_helper_unittest.cc
@@ -48,7 +48,7 @@
TEST(DspHelper, RampSignalAudioMultiVector) {
static const int kLen = 100;
static const int kChannels = 5;
- AudioMultiVector<int16_t> input(kChannels, kLen * 3);
+ AudioMultiVector input(kChannels, kLen * 3);
// Fill input with 1000.
for (int i = 0; i < kLen * 3; ++i) {
for (int channel = 0; channel < kChannels; ++channel) {
diff --git a/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.cc b/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.cc
index f8b13aa..c85534e 100644
--- a/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.cc
+++ b/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.cc
@@ -150,7 +150,7 @@
// Generate num_samples of DTMF signal and write to |output|.
int DtmfToneGenerator::Generate(int num_samples,
- AudioMultiVector<int16_t>* output) {
+ AudioMultiVector* output) {
if (!initialized_) {
return kNotInitialized;
}
diff --git a/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h b/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h
index 60500ec..e93f0b8 100644
--- a/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h
+++ b/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.h
@@ -31,7 +31,7 @@
virtual ~DtmfToneGenerator() {}
virtual int Init(int fs, int event, int attenuation);
virtual void Reset();
- virtual int Generate(int num_samples, AudioMultiVector<int16_t>* output);
+ virtual int Generate(int num_samples, AudioMultiVector* output);
virtual bool initialized() const { return initialized_; }
private:
diff --git a/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator_unittest.cc b/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator_unittest.cc
index 393648c..37e8bbd 100644
--- a/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator_unittest.cc
@@ -27,7 +27,7 @@
TEST(DtmfToneGenerator, TestErrors) {
DtmfToneGenerator tone_gen;
const int kNumSamples = 10;
- AudioMultiVector<int16_t> signal(1); // One channel.
+ AudioMultiVector signal(1); // One channel.
// Try to generate tones without initializing.
EXPECT_EQ(DtmfToneGenerator::kNotInitialized,
@@ -62,7 +62,7 @@
DtmfToneGenerator tone_gen;
const int kAttenuation = 0;
const int kNumSamples = 10;
- AudioMultiVector<int16_t> signal(1); // One channel.
+ AudioMultiVector signal(1); // One channel.
// Low and high frequencies for events 0 through 15.
const double low_freq_hz[] = { 941.0, 697.0, 697.0, 697.0, 770.0, 770.0,
@@ -106,8 +106,8 @@
TEST(DtmfToneGenerator, TestAmplitudes) {
DtmfToneGenerator tone_gen;
const int kNumSamples = 10;
- AudioMultiVector<int16_t> signal(1); // One channel.
- AudioMultiVector<int16_t> ref_signal(1); // One channel.
+ AudioMultiVector signal(1); // One channel.
+ AudioMultiVector ref_signal(1); // One channel.
const int fs_vec[] = { 8000, 16000, 32000, 48000 };
const int event_vec[] = { 0, 4, 9, 13 }; // Test a few events.
diff --git a/webrtc/modules/audio_coding/neteq4/expand.cc b/webrtc/modules/audio_coding/neteq4/expand.cc
index 3d9fa38..b9157c3 100644
--- a/webrtc/modules/audio_coding/neteq4/expand.cc
+++ b/webrtc/modules/audio_coding/neteq4/expand.cc
@@ -34,7 +34,7 @@
}
}
-int Expand::Process(AudioMultiVector<int16_t>* output) {
+int Expand::Process(AudioMultiVector* output) {
int16_t random_vector[kMaxSampleRate / 8000 * 120 + 30];
int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125];
static const int kTempDataSize = 3600;
diff --git a/webrtc/modules/audio_coding/neteq4/expand.h b/webrtc/modules/audio_coding/neteq4/expand.h
index b51856f..f7f19e2 100644
--- a/webrtc/modules/audio_coding/neteq4/expand.h
+++ b/webrtc/modules/audio_coding/neteq4/expand.h
@@ -61,7 +61,7 @@
// The main method to produce concealment data. The data is appended to the
// end of |output|.
- int Process(AudioMultiVector<int16_t>* output);
+ int Process(AudioMultiVector* output);
// Prepare the object to do extra expansion during normal operation following
// a period of expands.
diff --git a/webrtc/modules/audio_coding/neteq4/merge.cc b/webrtc/modules/audio_coding/neteq4/merge.cc
index c3c8b48..463b2ca 100644
--- a/webrtc/modules/audio_coding/neteq4/merge.cc
+++ b/webrtc/modules/audio_coding/neteq4/merge.cc
@@ -25,7 +25,7 @@
int Merge::Process(int16_t* input, size_t input_length,
int16_t* external_mute_factor_array,
- AudioMultiVector<int16_t>* output) {
+ AudioMultiVector* output) {
// TODO(hlundin): Change to an enumerator and skip assert.
assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
fs_hz_ == 48000);
@@ -37,7 +37,7 @@
int expanded_length = GetExpandedSignal(&old_length, &expand_period);
// Transfer input signal to an AudioMultiVector.
- AudioMultiVector<int16_t> input_vector(num_channels_);
+ AudioMultiVector input_vector(num_channels_);
input_vector.PushBackInterleaved(input, input_length);
size_t input_length_per_channel = input_vector.Size();
assert(input_length_per_channel == input_length / num_channels_);
@@ -162,7 +162,7 @@
// This assert should always be true thanks to the if statement above.
assert(210 * kMaxSampleRate / 8000 - *old_length >= 0);
- AudioMultiVector<int16_t> expanded_temp(num_channels_);
+ AudioMultiVector expanded_temp(num_channels_);
expand_->Process(&expanded_temp);
*expand_period = static_cast<int>(expanded_temp.Size()); // Samples per
// channel.
diff --git a/webrtc/modules/audio_coding/neteq4/merge.h b/webrtc/modules/audio_coding/neteq4/merge.h
index bb81e20..f1f64e6 100644
--- a/webrtc/modules/audio_coding/neteq4/merge.h
+++ b/webrtc/modules/audio_coding/neteq4/merge.h
@@ -53,7 +53,7 @@
// must have |num_channels_| elements.
int Process(int16_t* input, size_t input_length,
int16_t* external_mute_factor_array,
- AudioMultiVector<int16_t>* output);
+ AudioMultiVector* output);
private:
static const int kMaxSampleRate = 48000;
@@ -95,7 +95,7 @@
SyncBuffer* sync_buffer_;
int16_t expanded_downsampled_[kExpandDownsampLength];
int16_t input_downsampled_[kInputDownsampLength];
- AudioMultiVector<int16_t> expanded_;
+ AudioMultiVector expanded_;
DISALLOW_COPY_AND_ASSIGN(Merge);
};
diff --git a/webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_tone_generator.h b/webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_tone_generator.h
index f8ab56f..d34f747 100644
--- a/webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_tone_generator.h
+++ b/webrtc/modules/audio_coding/neteq4/mock/mock_dtmf_tone_generator.h
@@ -26,7 +26,7 @@
MOCK_METHOD0(Reset,
void());
MOCK_METHOD2(Generate,
- int(int num_samples, AudioMultiVector<int16_t>* output));
+ int(int num_samples, AudioMultiVector* output));
MOCK_CONST_METHOD0(initialized,
bool());
};
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index 289b67f..cf0a1db 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -1685,7 +1685,7 @@
overdub_length = output_size_samples_ - static_cast<int>(out_index);
}
- AudioMultiVector<int16_t> dtmf_output(num_channels);
+ AudioMultiVector dtmf_output(num_channels);
int dtmf_return_value = 0;
if (!dtmf_tone_generator_->initialized()) {
dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no,
@@ -1811,7 +1811,7 @@
vad_->Init();
// Delete algorithm buffer and create a new one.
- algorithm_buffer_.reset(new AudioMultiVector<int16_t>(channels));
+ algorithm_buffer_.reset(new AudioMultiVector(channels));
// Delete sync buffer and create a new one.
sync_buffer_.reset(new SyncBuffer(channels, kSyncBufferSize * fs_mult_));
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.h b/webrtc/modules/audio_coding/neteq4/neteq_impl.h
index 5e387f2..83dd58b 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.h
@@ -312,7 +312,7 @@
scoped_ptr<TimestampScaler> timestamp_scaler_;
scoped_ptr<DecisionLogic> decision_logic_;
scoped_ptr<PostDecodeVad> vad_;
- scoped_ptr<AudioMultiVector<int16_t> > algorithm_buffer_;
+ scoped_ptr<AudioMultiVector> algorithm_buffer_;
scoped_ptr<SyncBuffer> sync_buffer_;
scoped_ptr<Expand> expand_;
scoped_ptr<Normal> normal_;
diff --git a/webrtc/modules/audio_coding/neteq4/normal.cc b/webrtc/modules/audio_coding/neteq4/normal.cc
index 426df68..8d9c020 100644
--- a/webrtc/modules/audio_coding/neteq4/normal.cc
+++ b/webrtc/modules/audio_coding/neteq4/normal.cc
@@ -28,7 +28,7 @@
size_t length,
Modes last_mode,
int16_t* external_mute_factor_array,
- AudioMultiVector<int16_t>* output) {
+ AudioMultiVector* output) {
if (length == 0) {
// Nothing to process.
output->Clear();
@@ -55,7 +55,7 @@
expand_->SetParametersForNormalAfterExpand();
// Call Expand.
- AudioMultiVector<int16_t> expanded(output->Channels());
+ AudioMultiVector expanded(output->Channels());
expand_->Process(&expanded);
expand_->Reset();
diff --git a/webrtc/modules/audio_coding/neteq4/normal.h b/webrtc/modules/audio_coding/neteq4/normal.h
index 23fcca5..fa14685 100644
--- a/webrtc/modules/audio_coding/neteq4/normal.h
+++ b/webrtc/modules/audio_coding/neteq4/normal.h
@@ -53,7 +53,7 @@
int Process(const int16_t* input, size_t length,
Modes last_mode,
int16_t* external_mute_factor_array,
- AudioMultiVector<int16_t>* output);
+ AudioMultiVector* output);
private:
int fs_hz_;
diff --git a/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc b/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc
index 195315f..ac787eb 100644
--- a/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc
+++ b/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc
@@ -20,7 +20,7 @@
const int16_t* input,
int input_length,
int old_data_length,
- AudioMultiVector<int16_t>* output,
+ AudioMultiVector* output,
int16_t* length_change_samples) {
old_data_length_per_channel_ = old_data_length;
// Input length must be (almost) 30 ms.
@@ -56,7 +56,7 @@
PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
const int16_t *input, size_t input_length, size_t peak_index,
int16_t best_correlation, bool active_speech,
- AudioMultiVector<int16_t>* output) const {
+ AudioMultiVector* output) const {
// Pre-calculate common multiplication with |fs_mult_|.
// 120 corresponds to 15 ms.
int fs_mult_120 = fs_mult_ * 120;
@@ -75,7 +75,7 @@
output->PushBackInterleaved(
input, (unmodified_length + peak_index) * num_channels_);
// Copy the last |peak_index| samples up to 15 ms to |temp_vector|.
- AudioMultiVector<int16_t> temp_vector(num_channels_);
+ AudioMultiVector temp_vector(num_channels_);
temp_vector.PushBackInterleaved(
&input[(unmodified_length - peak_index) * num_channels_],
peak_index * num_channels_);
diff --git a/webrtc/modules/audio_coding/neteq4/preemptive_expand.h b/webrtc/modules/audio_coding/neteq4/preemptive_expand.h
index 1bb610f..4cd92cc 100644
--- a/webrtc/modules/audio_coding/neteq4/preemptive_expand.h
+++ b/webrtc/modules/audio_coding/neteq4/preemptive_expand.h
@@ -46,7 +46,7 @@
ReturnCodes Process(const int16_t *pw16_decoded,
int len,
int old_data_len,
- AudioMultiVector<int16_t>* output,
+ AudioMultiVector* output,
int16_t* length_change_samples);
protected:
@@ -61,7 +61,7 @@
virtual ReturnCodes CheckCriteriaAndStretch(
const int16_t *pw16_decoded, size_t len, size_t w16_bestIndex,
int16_t w16_bestCorr, bool w16_VAD,
- AudioMultiVector<int16_t>* output) const;
+ AudioMultiVector* output) const;
private:
int old_data_length_per_channel_;
diff --git a/webrtc/modules/audio_coding/neteq4/sync_buffer.cc b/webrtc/modules/audio_coding/neteq4/sync_buffer.cc
index 72fa16a..75ee6ec 100644
--- a/webrtc/modules/audio_coding/neteq4/sync_buffer.cc
+++ b/webrtc/modules/audio_coding/neteq4/sync_buffer.cc
@@ -20,10 +20,10 @@
return Size() - next_index_;
}
-void SyncBuffer::PushBack(const AudioMultiVector<int16_t>& append_this) {
+void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
size_t samples_added = append_this.Size();
- AudioMultiVector<int16_t>::PushBack(append_this);
- AudioMultiVector<int16_t>::PopFront(samples_added);
+ AudioMultiVector::PushBack(append_this);
+ AudioMultiVector::PopFront(samples_added);
if (samples_added <= next_index_) {
next_index_ -= samples_added;
} else {
@@ -44,7 +44,7 @@
void SyncBuffer::InsertZerosAtIndex(size_t length, size_t position) {
position = std::min(position, Size());
length = std::min(length, Size() - position);
- AudioMultiVector<int16_t>::PopBack(length);
+ AudioMultiVector::PopBack(length);
for (size_t channel = 0; channel < Channels(); ++channel) {
channels_[channel]->InsertZerosAt(length, position);
}
@@ -58,15 +58,15 @@
}
}
-void SyncBuffer::ReplaceAtIndex(const AudioMultiVector<int16_t>& insert_this,
+void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
size_t length,
size_t position) {
position = std::min(position, Size()); // Cap |position| in the valid range.
length = std::min(length, Size() - position);
- AudioMultiVector<int16_t>::OverwriteAt(insert_this, length, position);
+ AudioMultiVector::OverwriteAt(insert_this, length, position);
}
-void SyncBuffer::ReplaceAtIndex(const AudioMultiVector<int16_t>& insert_this,
+void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
size_t position) {
ReplaceAtIndex(insert_this, insert_this.Size(), position);
}
diff --git a/webrtc/modules/audio_coding/neteq4/sync_buffer.h b/webrtc/modules/audio_coding/neteq4/sync_buffer.h
index 7add358..cfd35fc 100644
--- a/webrtc/modules/audio_coding/neteq4/sync_buffer.h
+++ b/webrtc/modules/audio_coding/neteq4/sync_buffer.h
@@ -17,10 +17,10 @@
namespace webrtc {
-class SyncBuffer : public AudioMultiVector<int16_t> {
+class SyncBuffer : public AudioMultiVector {
public:
SyncBuffer(size_t channels, size_t length)
- : AudioMultiVector<int16_t>(channels, length),
+ : AudioMultiVector(channels, length),
next_index_(length),
end_timestamp_(0),
dtmf_index_(0) {}
@@ -34,7 +34,7 @@
// the same number of samples from the beginning of the SyncBuffer, to
// maintain a constant buffer size. The |next_index_| is updated to reflect
// the move of the beginning of "future" data.
- void PushBack(const AudioMultiVector<int16_t>& append_this);
+ void PushBack(const AudioMultiVector& append_this);
// Adds |length| zeros to the beginning of each channel. Removes
// the same number of samples from the end of the SyncBuffer, to
@@ -56,13 +56,13 @@
// 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.
- virtual void ReplaceAtIndex(const AudioMultiVector<int16_t>& insert_this,
+ 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
// the same constraints as above, that the SyncBuffer is not extended).
- virtual void ReplaceAtIndex(const AudioMultiVector<int16_t>& insert_this,
+ virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
size_t position);
// Reads |requested_len| samples from each channel and writes them interleaved
diff --git a/webrtc/modules/audio_coding/neteq4/sync_buffer_unittest.cc b/webrtc/modules/audio_coding/neteq4/sync_buffer_unittest.cc
index 054e69a..1aafa22 100644
--- a/webrtc/modules/audio_coding/neteq4/sync_buffer_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/sync_buffer_unittest.cc
@@ -53,7 +53,7 @@
static const size_t kChannels = 2;
SyncBuffer sync_buffer(kChannels, kLen);
static const size_t kNewLen = 10;
- AudioMultiVector<int16_t> new_data(kChannels, kNewLen);
+ AudioMultiVector new_data(kChannels, kNewLen);
// Populate |new_data|.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen; ++i) {
@@ -93,7 +93,7 @@
static const size_t kChannels = 2;
SyncBuffer sync_buffer(kChannels, kLen);
static const size_t kNewLen = 10;
- AudioMultiVector<int16_t> new_data(kChannels, kNewLen);
+ AudioMultiVector new_data(kChannels, kNewLen);
// Populate |new_data|.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen; ++i) {
@@ -126,7 +126,7 @@
static const size_t kChannels = 2;
SyncBuffer sync_buffer(kChannels, kLen);
static const size_t kNewLen = 10;
- AudioMultiVector<int16_t> new_data(kChannels, kNewLen);
+ AudioMultiVector new_data(kChannels, kNewLen);
// Populate |new_data|.
for (size_t channel = 0; channel < kChannels; ++channel) {
for (size_t i = 0; i < kNewLen; ++i) {
diff --git a/webrtc/modules/audio_coding/neteq4/time_stretch.cc b/webrtc/modules/audio_coding/neteq4/time_stretch.cc
index a248698..5b6b3ba 100644
--- a/webrtc/modules/audio_coding/neteq4/time_stretch.cc
+++ b/webrtc/modules/audio_coding/neteq4/time_stretch.cc
@@ -22,7 +22,7 @@
TimeStretch::ReturnCodes TimeStretch::Process(
const int16_t* input,
size_t input_len,
- AudioMultiVector<int16_t>* output,
+ AudioMultiVector* output,
int16_t* length_change_samples) {
// Pre-calculate common multiplication with |fs_mult_|.
diff --git a/webrtc/modules/audio_coding/neteq4/time_stretch.h b/webrtc/modules/audio_coding/neteq4/time_stretch.h
index 4741da2..f0f58b8 100644
--- a/webrtc/modules/audio_coding/neteq4/time_stretch.h
+++ b/webrtc/modules/audio_coding/neteq4/time_stretch.h
@@ -58,7 +58,7 @@
// PreemptiveExpand.
ReturnCodes Process(const int16_t* input,
size_t input_len,
- AudioMultiVector<int16_t>* output,
+ AudioMultiVector* output,
int16_t* length_change_samples);
protected:
@@ -75,7 +75,7 @@
virtual ReturnCodes CheckCriteriaAndStretch(
const int16_t* input, size_t input_length, size_t peak_index,
int16_t best_correlation, bool active_speech,
- AudioMultiVector<int16_t>* output) const = 0;
+ AudioMultiVector* output) const = 0;
static const int kCorrelationLen = 50;
static const int kLogCorrelationLen = 6; // >= log2(kCorrelationLen).