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Move AudioDecoderPcm* next to AudioEncoderPcm*

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

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1348613003

Cr-Commit-Position: refs/heads/master@{#10015}
This commit is contained in:
kwiberg
2015-09-22 06:16:51 -07:00
committed by Commit bot
parent d4818e7304
commit 6faf5bebba
8 changed files with 162 additions and 123 deletions
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73
webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc Normal file
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@ -0,0 +1,73 @@
/*
* 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/g711/include/audio_decoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
namespace webrtc {
void AudioDecoderPcmU::Reset() {}
size_t AudioDecoderPcmU::Channels() const {
return 1;
}
int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return static_cast<int>(ret);
}
int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// One encoded byte per sample per channel.
return static_cast<int>(encoded_len / Channels());
}
size_t AudioDecoderPcmUMultiCh::Channels() const {
return channels_;
}
void AudioDecoderPcmA::Reset() {}
size_t AudioDecoderPcmA::Channels() const {
return 1;
}
int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return static_cast<int>(ret);
}
int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// One encoded byte per sample per channel.
return static_cast<int>(encoded_len / Channels());
}
size_t AudioDecoderPcmAMultiCh::Channels() const {
return channels_;
}
} // namespace webrtc

2
webrtc/modules/audio_coding/codecs/g711/g711.gypi
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@ -26,10 +26,12 @@
},
'sources': [
'include/g711_interface.h',
'include/audio_decoder_pcm.h',
'include/audio_encoder_pcm.h',
'g711_interface.c',
'g711.c',
'g711.h',
'audio_decoder_pcm.cc',
'audio_encoder_pcm.cc',
],
},

82
webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h Normal file
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/*
* 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_G711_INCLUDE_AUDIO_DECODER_PCM_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_DECODER_PCM_H_
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
namespace webrtc {
class AudioDecoderPcmU : public AudioDecoder {
public:
AudioDecoderPcmU() {}
void Reset() override;
int PacketDuration(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;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmU);
};
class AudioDecoderPcmA : public AudioDecoder {
public:
AudioDecoderPcmA() {}
void Reset() override;
int PacketDuration(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;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmA);
};
class AudioDecoderPcmUMultiCh : public AudioDecoderPcmU {
public:
explicit AudioDecoderPcmUMultiCh(size_t channels)
: AudioDecoderPcmU(), channels_(channels) {
RTC_DCHECK_GT(channels, 0u);
}
size_t Channels() const override;
private:
const size_t channels_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmUMultiCh);
};
class AudioDecoderPcmAMultiCh : public AudioDecoderPcmA {
public:
explicit AudioDecoderPcmAMultiCh(size_t channels)
: AudioDecoderPcmA(), channels_(channels) {
RTC_DCHECK_GT(channels, 0u);
}
size_t Channels() const override;
private:
const size_t channels_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmAMultiCh);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_DECODER_PCM_H_