728d9037c0
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}
89 lines
3.4 KiB
C++
89 lines
3.4 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
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#include <assert.h>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
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#include "webrtc/modules/audio_coding/neteq/time_stretch.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Forward declarations.
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class BackgroundNoise;
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// This class implements the PreemptiveExpand operation. Most of the work is
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// done in the base class TimeStretch, which is shared with the Accelerate
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// operation. In the PreemptiveExpand class, the operations that are specific to
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// PreemptiveExpand are implemented.
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class PreemptiveExpand : public TimeStretch {
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public:
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PreemptiveExpand(int sample_rate_hz,
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size_t num_channels,
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const BackgroundNoise& background_noise,
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int overlap_samples)
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: TimeStretch(sample_rate_hz, num_channels, background_noise),
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old_data_length_per_channel_(-1),
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overlap_samples_(overlap_samples) {
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}
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// This method performs the actual PreemptiveExpand operation. The samples are
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// read from |input|, of length |input_length| elements, and are written to
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// |output|. The number of samples added through time-stretching is
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// is provided in the output |length_change_samples|. The method returns
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// the outcome of the operation as an enumerator value.
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ReturnCodes Process(const int16_t *pw16_decoded,
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int len,
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int old_data_len,
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AudioMultiVector* output,
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int16_t* length_change_samples);
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protected:
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// Sets the parameters |best_correlation| and |peak_index| to suitable
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// values when the signal contains no active speech.
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void SetParametersForPassiveSpeech(size_t input_length,
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int16_t* best_correlation,
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int* peak_index) const override;
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// Checks the criteria for performing the time-stretching operation and,
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// if possible, performs the time-stretching.
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ReturnCodes CheckCriteriaAndStretch(const int16_t* input,
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size_t input_length,
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size_t peak_index,
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int16_t best_correlation,
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bool active_speech,
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bool /*fast_mode*/,
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AudioMultiVector* output) const override;
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private:
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int old_data_length_per_channel_;
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int overlap_samples_;
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DISALLOW_COPY_AND_ASSIGN(PreemptiveExpand);
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};
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struct PreemptiveExpandFactory {
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PreemptiveExpandFactory() {}
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virtual ~PreemptiveExpandFactory() {}
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virtual PreemptiveExpand* Create(
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int sample_rate_hz,
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size_t num_channels,
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const BackgroundNoise& background_noise,
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int overlap_samples) const;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
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