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AudioEncoder subclass for G722

Browse Source 
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7779 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org 2014-12-02 11:45:51 +00:00
parent 84515f841d
commit 0cd5558f2b
10 changed files with 214 additions and 66 deletions
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2
webrtc/modules/audio_coding/BUILD.gn
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@ -172,6 +172,8 @@ config("g722_config") {
source_set("g722") {
sources = [
"codecs/g722/audio_encoder_g722.cc",
"codecs/g722/include/audio_encoder_g722.h",
"codecs/g722/include/g722_interface.h",
"codecs/g722/g722_interface.c",
"codecs/g722/g722_encode.c",

119
webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc Normal file
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@ -0,0 +1,119 @@
/*
* Copyright (c) 2014 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/g722/include/audio_encoder_g722.h"
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
namespace webrtc {
namespace {
const int kSampleRateHz = 16000;
} // namespace
AudioEncoderG722::EncoderState::EncoderState() {
CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
CHECK_EQ(0, WebRtcG722_EncoderInit(encoder));
}
AudioEncoderG722::EncoderState::~EncoderState() {
CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
}
AudioEncoderG722::AudioEncoderG722(const Config& config)
: num_channels_(config.num_channels),
num_10ms_frames_per_packet_(config.frame_size_ms / 10),
num_10ms_frames_buffered_(0),
first_timestamp_in_buffer_(0),
encoders_(new EncoderState[num_channels_]),
interleave_buffer_(new uint8_t[2 * num_channels_]) {
CHECK_EQ(config.frame_size_ms % 10, 0)
<< "Frame size must be an integer multiple of 10 ms.";
const int samples_per_channel =
kSampleRateHz / 100 * num_10ms_frames_per_packet_;
for (int i = 0; i < num_channels_; ++i) {
encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
encoders_[i].encoded_buffer.reset(new uint8_t[samples_per_channel / 2]);
}
}
AudioEncoderG722::~AudioEncoderG722() {}
int AudioEncoderG722::sample_rate_hz() const {
return kSampleRateHz;
}
int AudioEncoderG722::num_channels() const {
return num_channels_;
}
int AudioEncoderG722::Num10MsFramesInNextPacket() const {
return num_10ms_frames_per_packet_;
}
bool AudioEncoderG722::Encode(uint32_t timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
size_t* encoded_bytes,
EncodedInfo* info) {
const int samples_per_channel =
kSampleRateHz / 100 * num_10ms_frames_per_packet_;
CHECK_GE(max_encoded_bytes,
static_cast<size_t>(samples_per_channel) / 2 * num_channels_);
if (num_10ms_frames_buffered_ == 0)
first_timestamp_in_buffer_ = timestamp;
// Deinterleave samples and save them in each channel's buffer.
const int start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
for (int i = 0; i < kSampleRateHz / 100; ++i)
for (int j = 0; j < num_channels_; ++j)
encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
// If we don't yet have enough samples for a packet, we're done for now.
if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
*encoded_bytes = 0;
return true;
}
// Encode each channel separately.
CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
num_10ms_frames_buffered_ = 0;
for (int i = 0; i < num_channels_; ++i) {
const int encoded = WebRtcG722_Encode(
encoders_[i].encoder, encoders_[i].speech_buffer.get(),
samples_per_channel, encoders_[i].encoded_buffer.get());
if (encoded < 0)
return false;
CHECK_EQ(encoded, samples_per_channel / 2);
}
// Interleave the encoded bytes of the different channels. Each separate
// channel and the interleaved stream encodes two samples per byte, most
// significant half first.
for (int i = 0; i < samples_per_channel / 2; ++i) {
for (int j = 0; j < num_channels_; ++j) {
uint8_t two_samples = encoders_[j].encoded_buffer[i];
interleave_buffer_[j] = two_samples >> 4;
interleave_buffer_[num_channels_ + j] = two_samples & 0xf;
}
for (int j = 0; j < num_channels_; ++j)
encoded[i * num_channels_ + j] =
interleave_buffer_[2 * j] << 4 | interleave_buffer_[2 * j + 1];
}
*encoded_bytes = samples_per_channel / 2 * num_channels_;
info->encoded_timestamp = first_timestamp_in_buffer_;
return true;
}
} // namespace webrtc

2
webrtc/modules/audio_coding/codecs/g722/g722.gypi
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@ -21,6 +21,8 @@
],
},
'sources': [
'audio_encoder_g722.cc',
'include/audio_encoder_g722.h',
'include/g722_interface.h',
'g722_interface.c',
'g722_encode.c',

4
webrtc/modules/audio_coding/codecs/g722/g722_interface.c
View File

@ -46,9 +46,9 @@ int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
}
int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
int16_t *speechIn,
const int16_t* speechIn,
int16_t len,
int16_t *encoded)
uint8_t* encoded)
{
unsigned char *codechar = (unsigned char*) encoded;
// Encode the input speech vector

64
webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h Normal file
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@ -0,0 +1,64 @@
/*
* Copyright (c) 2014 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_G722_INCLUDE_AUDIO_ENCODER_G722_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class AudioEncoderG722 : public AudioEncoder {
public:
struct Config {
Config() : payload_type(9), frame_size_ms(20), num_channels(1) {}
int payload_type;
int frame_size_ms;
int num_channels;
};
explicit AudioEncoderG722(const Config& config);
virtual ~AudioEncoderG722();
virtual int sample_rate_hz() const OVERRIDE;
virtual int num_channels() const OVERRIDE;
virtual int Num10MsFramesInNextPacket() const OVERRIDE;
protected:
virtual bool Encode(uint32_t timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
size_t* encoded_bytes,
EncodedInfo* info) OVERRIDE;
private:
// The encoder state for one channel.
struct EncoderState {
G722EncInst* encoder;
scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
scoped_ptr<uint8_t[]> encoded_buffer; // Already encoded.
EncoderState();
~EncoderState();
};
const int num_channels_;
const int num_10ms_frames_per_packet_;
int num_10ms_frames_buffered_;
uint32_t first_timestamp_in_buffer_;
const scoped_ptr<EncoderState[]> encoders_;
const scoped_ptr<uint8_t[]> interleave_buffer_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_

6
webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
View File

@ -95,10 +95,10 @@ int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
* -1 - Error
*/
int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
int16_t *speechIn,
int16_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
const int16_t* speechIn,
int16_t len,
int16_t *encoded);
uint8_t* encoded);
/****************************************************************************

6
webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
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@ -62,7 +62,7 @@ int main(int argc, char* argv[])
int16_t stream_len = 0;
int16_t shortdata[960];
int16_t decoded[960];
int16_t streamdata[80*3];
uint8_t streamdata[80 * 6];
int16_t speechType[1];
/* handling wrong input arguments in the command line */
@ -124,7 +124,9 @@ int main(int argc, char* argv[])
/* G.722 encoding + decoding */
stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
err = WebRtcG722_Decode((G722DecInst *)G722dec_inst, streamdata, stream_len, decoded, speechType);
err = WebRtcG722_Decode(G722dec_inst,
reinterpret_cast<int16_t*>(streamdata),
stream_len, decoded, speechType);
/* Stop clock after call to encoder and decoder */
runtime += (double)((clock()/(double)CLOCKS_PER_SEC_G722)-starttime);

6
webrtc/modules/audio_coding/main/acm2/acm_g722.cc
View File

@ -115,11 +115,11 @@ int16_t ACMG722::InternalEncode(uint8_t* bitstream,
}
len_in_bytes = WebRtcG722_Encode(
encoder_inst_ptr_, left_channel, frame_len_smpl_,
reinterpret_cast<int16_t*>(out_left));
out_left);
len_in_bytes += WebRtcG722_Encode(encoder_inst_ptr_right_,
right_channel,
frame_len_smpl_,
reinterpret_cast<int16_t*>(out_right));
out_right);
*bitstream_len_byte = len_in_bytes;
// Interleave the 4 bits per sample from left and right channel
@ -130,7 +130,7 @@ int16_t ACMG722::InternalEncode(uint8_t* bitstream,
} else {
*bitstream_len_byte = WebRtcG722_Encode(
encoder_inst_ptr_, &in_audio_[in_audio_ix_read_], frame_len_smpl_,
reinterpret_cast<int16_t*>(bitstream));
bitstream);
}
// increment the read index this tell the caller how far

69
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
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@ -22,7 +22,7 @@
#endif
#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
@ -483,67 +483,26 @@ class AudioDecoderG722Test : public AudioDecoderTest {
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderG722;
assert(decoder_);
WebRtcG722_CreateEncoder(&encoder_);
AudioEncoderG722::Config config;
config.frame_size_ms = 10;
config.num_channels = 1;
audio_encoder_.reset(new AudioEncoderG722(config));
}
~AudioDecoderG722Test() {
WebRtcG722_FreeEncoder(encoder_);
}
virtual void InitEncoder() {
ASSERT_EQ(0, WebRtcG722_EncoderInit(encoder_));
}
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
int enc_len_bytes =
WebRtcG722_Encode(encoder_, const_cast<int16_t*>(input),
static_cast<int>(input_len_samples),
reinterpret_cast<int16_t*>(output));
EXPECT_EQ(80, enc_len_bytes);
return enc_len_bytes;
}
G722EncInst* encoder_;
};
class AudioDecoderG722StereoTest : public AudioDecoderG722Test {
class AudioDecoderG722StereoTest : public AudioDecoderTest {
protected:
AudioDecoderG722StereoTest() : AudioDecoderG722Test() {
AudioDecoderG722StereoTest() : AudioDecoderTest() {
channels_ = 2;
// Delete the |decoder_| that was created by AudioDecoderG722Test and
// create an AudioDecoderG722Stereo object instead.
delete decoder_;
codec_input_rate_hz_ = 16000;
frame_size_ = 160;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderG722Stereo;
assert(decoder_);
}
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
uint8_t* temp_output = new uint8_t[data_length_ * 2];
// Encode a mono payload using the base test class.
int mono_enc_len_bytes =
AudioDecoderG722Test::EncodeFrame(input, input_len_samples,
temp_output);
// The bit-stream consists of 4-bit samples:
// +--------+--------+--------+
// | s0 s1 | s2 s3 | s4 s5 |
// +--------+--------+--------+
//
// Duplicate them to the |output| such that the stereo stream becomes:
// +--------+--------+--------+
// | s0 s0 | s1 s1 | s2 s2 |
// +--------+--------+--------+
EXPECT_LE(mono_enc_len_bytes * 2, static_cast<int>(data_length_ * 2));
uint8_t* output_ptr = output;
for (int i = 0; i < mono_enc_len_bytes; ++i) {
*output_ptr = (temp_output[i] & 0xF0) + (temp_output[i] >> 4);
++output_ptr;
*output_ptr = (temp_output[i] << 4) + (temp_output[i] & 0x0F);
++output_ptr;
}
delete [] temp_output;
return mono_enc_len_bytes * 2;
AudioEncoderG722::Config config;
config.frame_size_ms = 10;
config.num_channels = 2;
audio_encoder_.reset(new AudioEncoderG722(config));
}
};

2
webrtc/modules/audio_coding/neteq/test/RTPencode.cc
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@ -1615,7 +1615,7 @@ int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * e
#endif
#ifdef CODEC_G722
else if (coder==webrtc::kDecoderG722) { /*g722 */
cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, (int16_t*)encoded);
cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, encoded);
assert(cdlen == frameLen>>1);
}
#endif