c8fa692ec4
I used a command like this to update the paths: perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"` The only manual edit is to add an include of webrtc/rtc_base/checks.h in webrtc/modules/audio_device/android/opensles_common.h, which likely was needed due to changed include paths due to 'git cl format'. BUG=webrtc:7634 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2969653002 Cr-Commit-Position: refs/heads/master@{#18871}
76 lines
2.6 KiB
C++
76 lines
2.6 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_NORMAL_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
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#include <string.h> // Access to size_t.
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#include <vector>
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#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
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#include "webrtc/modules/audio_coding/neteq/defines.h"
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/safe_conversions.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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class DecoderDatabase;
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class Expand;
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// This class provides the "Normal" DSP operation, that is performed when
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// there is no data loss, no need to stretch the timing of the signal, and
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// no other "special circumstances" are at hand.
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class Normal {
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public:
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Normal(int fs_hz,
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DecoderDatabase* decoder_database,
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const BackgroundNoise& background_noise,
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Expand* expand)
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: fs_hz_(fs_hz),
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decoder_database_(decoder_database),
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background_noise_(background_noise),
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expand_(expand),
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samples_per_ms_(rtc::CheckedDivExact(fs_hz_, 1000)),
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default_win_slope_Q14_(
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rtc::dchecked_cast<uint16_t>((1 << 14) / samples_per_ms_)) {}
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virtual ~Normal() {}
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// Performs the "Normal" operation. The decoder data is supplied in |input|,
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// having |length| samples in total for all channels (interleaved). The
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// result is written to |output|. The number of channels allocated in
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// |output| defines the number of channels that will be used when
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// de-interleaving |input|. |last_mode| contains the mode used in the previous
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// GetAudio call (i.e., not the current one), and |external_mute_factor| is
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// a pointer to the mute factor in the NetEqImpl class.
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int Process(const int16_t* input, size_t length,
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Modes last_mode,
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int16_t* external_mute_factor_array,
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AudioMultiVector* output);
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private:
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int fs_hz_;
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DecoderDatabase* decoder_database_;
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const BackgroundNoise& background_noise_;
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Expand* expand_;
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const size_t samples_per_ms_;
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const int16_t default_win_slope_Q14_;
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RTC_DISALLOW_COPY_AND_ASSIGN(Normal);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
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