zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Move AudioDecoderOpus next to AudioEncoderOpus

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_coding/BUILD.gn
View File

@ -662,7 +662,9 @@ config("opus_config") {
source_set("webrtc_opus") {
sources = [
"codecs/opus/audio_decoder_opus.cc",
"codecs/opus/audio_encoder_opus.cc",
"codecs/opus/interface/audio_decoder_opus.h",
"codecs/opus/interface/audio_encoder_opus.h",
"codecs/opus/interface/opus_interface.h",
"codecs/opus/opus_inst.h",

94
webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc Normal file
View File

@ -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
View File

@ -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
View File

@ -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',

1
webrtc/modules/audio_coding/main/test/opus_test.h
View File

@ -14,6 +14,7 @@
#include <math.h>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"

82
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
View File

@ -29,7 +29,7 @@
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
#endif
#ifdef WEBRTC_CODEC_OPUS
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
#endif
#ifdef WEBRTC_CODEC_PCM16
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
@ -299,86 +299,6 @@ void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
}
#endif
// Opus
#ifdef WEBRTC_CODEC_OPUS
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_;
}
#endif
AudioDecoderCng::AudioDecoderCng() {
CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
WebRtcCng_InitDec(dec_state_);

35
webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
View File

@ -27,9 +27,6 @@
#ifdef WEBRTC_CODEC_ILBC
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
#endif
#ifdef WEBRTC_CODEC_OPUS
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#endif
#include "webrtc/typedefs.h"
namespace webrtc {
@ -206,38 +203,6 @@ class AudioDecoderG722Stereo : public AudioDecoder {
};
#endif
#ifdef WEBRTC_CODEC_OPUS
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);
};
#endif
// AudioDecoderCng is a special type of AudioDecoder. It inherits from
// AudioDecoder just to fit in the DecoderDatabase. None of the class methods
// should be used, except constructor, destructor, and accessors.

1
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
View File

@ -23,6 +23,7 @@
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"

2
webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
View File

@ -10,7 +10,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"