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6dc82e8f8b
NetEq tapers down the audio produced through loss concealment when the expansion has been going on for some time. When the audio packets starts coming in again, there is a ramp-up that happens. This ramp-up could before this change extend over more than one 10 ms block, which made keeping track of the scaling factor necessary. With this change, we make this ramp-up quicker in the rare cases when it lasted more than 10 ms, so that it always ramps up to 100% within one block. This way, we can remove the mute_factor_array. This change breaks bit-exactness, but careful listening could not reveal an audible difference. This change is a part of a larger refactoring of NetEq's PLC code. Bug: webrtc:9180 Change-Id: I4c513ce3ed8d66f9beec2abfb1f0c7ffaac7a21e Reviewed-on: https://webrtc-review.googlesource.com/77180 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23342}
73 lines
2.4 KiB
C++
73 lines
2.4 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 MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
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#define 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 "modules/audio_coding/neteq/audio_multi_vector.h"
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#include "modules/audio_coding/neteq/defines.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "typedefs.h" // NOLINT(build/include)
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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).
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int Process(const int16_t* input, size_t length,
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Modes last_mode,
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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 // MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
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