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Move AudioDecoderOpus next to AudioEncoderOpus

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All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.

BUG=webrtc:4557
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1342933005 .

Cr-Commit-Position: refs/heads/master@{#9944}
This commit is contained in:
Karl Wiberg
2015-09-15 17:28:18 +02:00
parent ec0feb6ddf
commit 0b05879cd7
9 changed files with 153 additions and 117 deletions
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94
webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc Normal file
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@ -0,0 +1,94 @@
/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
#include "webrtc/base/checks.h"
namespace webrtc {
AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
: channels_(num_channels) {
DCHECK(num_channels == 1 || num_channels == 2);
WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
WebRtcOpus_DecoderInit(dec_state_);
}
AudioDecoderOpus::~AudioDecoderOpus() {
WebRtcOpus_DecoderFree(dec_state_);
}
int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
int ret =
WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
if (ret > 0)
ret *= static_cast<int>(channels_); // Return total number of samples.
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
if (!PacketHasFec(encoded, encoded_len)) {
// This packet is a RED packet.
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
speech_type);
}
DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
&temp_type);
if (ret > 0)
ret *= static_cast<int>(channels_); // Return total number of samples.
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
void AudioDecoderOpus::Reset() {
WebRtcOpus_DecoderInit(dec_state_);
}
int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
}
int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
if (!PacketHasFec(encoded, encoded_len)) {
// This packet is a RED packet.
return PacketDuration(encoded, encoded_len);
}
return WebRtcOpus_FecDurationEst(encoded, encoded_len);
}
bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
size_t encoded_len) const {
int fec;
fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
return (fec == 1);
}
size_t AudioDecoderOpus::Channels() const {
return channels_;
}
} // namespace webrtc

51
webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h Normal file
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@ -0,0 +1,51 @@
/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
namespace webrtc {
class AudioDecoderOpus : public AudioDecoder {
public:
explicit AudioDecoderOpus(size_t num_channels);
~AudioDecoderOpus() override;
void Reset() override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
int PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const override;
bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
size_t Channels() const override;
protected:
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
int DecodeRedundantInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
private:
OpusDecInst* dec_state_;
const size_t channels_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H

2
webrtc/modules/audio_coding/codecs/opus/opus.gypi
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@ -43,7 +43,9 @@
'<(webrtc_root)',
],
'sources': [
'audio_decoder_opus.cc',
'audio_encoder_opus.cc',
'interface/audio_decoder_opus.h',
'interface/audio_encoder_opus.h',
'interface/opus_interface.h',
'opus_inst.h',