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Merge branch 'master' into multichannel

Browse Source 
Resolved Conflicts:
	daemon/src/sfl_types.h
This commit is contained in:
Tristan Matthews
2013-07-10 14:20:50 -04:00
parent 7841a3a04b 868bf2cc1a
commit 37cc77e3ec
16 changed files with 47 additions and 39 deletions
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2
daemon/src/audio/audiorecord.cpp
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@ -420,7 +420,7 @@ void AudioRecord::closeWavFile()
WARN("Can't close file");
}
//void AudioRecord::recData(SFLDataFormat* buffer, size_t nSamples)
//void AudioRecord::recData(SFLAudioSample* buffer, size_t nSamples)
void AudioRecord::recData(AudioBuffer& buffer)
{
if (recordingEnabled_) {

5
daemon/src/audio/codecs/audiocodec.h
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@ -36,6 +36,7 @@
#include <vector>
#include "cc_config.h"
#include "sfl_types.h"
#include <ccrtp/formats.h> // for ost::DynamicPayloadFormat
#define XSTR(s) STR(s)
@ -70,14 +71,14 @@ class AudioCodec {
* @param buffer_size : the size of the input buffer
* @return the number of samples decoded
*/
virtual int decode(short *dst, unsigned char *buf, size_t buffer_size) = 0;
virtual int decode(SFLAudioSample *dst, unsigned char *buf, size_t buffer_size) = 0;
/**
* Encode an input buffer and fill the output buffer with the encoded data
* @param buffer_size : the maximum size of encoded data buffer (dst)
* @return the number of bytes encoded
*/
virtual int encode(unsigned char *dst, short *src, size_t buffer_size) = 0;
virtual int encode(unsigned char *dst, SFLAudioSample *src, size_t buffer_size) = 0;
/**
* Multichannel version of decode().

9
daemon/src/audio/codecs/g729.cpp
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@ -28,12 +28,13 @@
* as that of the covered work.
*/
#include "g729.h"
#include "sfl_types.h"
#include <iostream>
#include <dlfcn.h>
#include <stdexcept>
#define G729_TYPE_ENCODER (void (*)(bcg729EncoderChannelContextStruct*, int16_t[], uint8_t[]))
#define G729_TYPE_DECODER (void (*)(bcg729DecoderChannelContextStruct*, uint8_t[], uint8_t, int16_t[]))
#define G729_TYPE_ENCODER (void (*)(bcg729EncoderChannelContextStruct*, SFLAudioSample[], uint8_t[]))
#define G729_TYPE_DECODER (void (*)(bcg729DecoderChannelContextStruct*, uint8_t[], uint8_t, SFLAudioSample[]))
#define G729_TYPE_DECODER_INIT (bcg729DecoderChannelContextStruct*(*)())
#define G729_TYPE_ENCODER_INIT (bcg729EncoderChannelContextStruct*(*)())
@ -72,14 +73,14 @@ G729::~G729()
dlclose(handler_);
}
int G729::decode(short *dst, unsigned char *buf, size_t buffer_size)
int G729::decode(SFLAudioSample *dst, unsigned char *buf, size_t buffer_size)
{
decoder_(decoderContext_, buf, false, dst);
decoder_(decoderContext_, buf + (buffer_size / 2), false, dst + 80);
return 160;
}
int G729::encode(unsigned char *dst, short *src, size_t buffer_size)
int G729::encode(unsigned char *dst, SFLAudioSample *src, size_t buffer_size)
{
encoder_(encoderContext_, src, dst);
encoder_(encoderContext_, src + (buffer_size / 2), dst + 10);

9
daemon/src/audio/codecs/g729.h
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@ -31,6 +31,7 @@
#define G729_H_
#include <cstdlib>
#include "sfl_types.h"
#include "noncopyable.h"
#include "audiocodec.h"
@ -43,8 +44,8 @@ public:
G729();
~G729();
private:
virtual int decode(short *dst, unsigned char *buf, size_t buffer_size);
virtual int encode(unsigned char *dst, short *src, size_t buffer_size);
virtual int decode(SFLAudioSample *dst, unsigned char *buf, size_t buffer_size);
virtual int encode(unsigned char *dst, SFLAudioSample *src, size_t buffer_size);
NON_COPYABLE(G729);
//Attributes
@ -53,8 +54,8 @@ private:
void* handler_;
//Extern functions
void (*encoder_) (bcg729EncoderChannelContextStruct *encoderChannelContext, int16_t inputFrame[], uint8_t bitStream[]);
void (*decoder_) (bcg729DecoderChannelContextStruct *decoderChannelContext, uint8_t bitStream[], uint8_t frameErasureFlag, int16_t signal[]);
void (*encoder_) (bcg729EncoderChannelContextStruct *encoderChannelContext, SFLAudioSample inputFrame[], uint8_t bitStream[]);
void (*decoder_) (bcg729DecoderChannelContextStruct *decoderChannelContext, uint8_t bitStream[], uint8_t frameErasureFlag, SFLAudioSample signal[]);
static void loadError(const char *error);
};

5
daemon/src/audio/codecs/ilbc.cpp
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@ -29,6 +29,7 @@
*/
#include "audiocodec.h"
#include "sfl_types.h"
#include <tr1/array>
#include <algorithm>
@ -50,13 +51,13 @@ class Ilbc: public sfl::AudioCodec {
private:
// iLBC expects floating point data, so we have to convert
int decode(short *dst, unsigned char *src, size_t /*buf_size*/) {
int decode(SFLAudioSample *dst, unsigned char *src, size_t /*buf_size*/) {
const int NORMAL_MODE = 1;
iLBC_decode(dst, reinterpret_cast<WebRtc_UWord16*>(src), &ilbc_dec_, NORMAL_MODE);
return frameSize_;
}
int encode(unsigned char *dst, short* src, size_t /*buf_size*/) {
int encode(unsigned char *dst, SFLAudioSample * src, size_t /*buf_size*/) {
iLBC_encode(reinterpret_cast<WebRtc_UWord16*>(dst), src, &ilbc_enc_);
return frameSize_;
}

5
daemon/src/audio/codecs/opus.cpp
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@ -29,6 +29,7 @@
* as that of the covered work.
*/
#include "opus.h"
#include "sfl_types.h"
#include <stdexcept>
#include <iostream>
@ -60,12 +61,12 @@ Opus::~Opus()
opus_decoder_destroy(decoder_);
}
int Opus::decode(short *dst, unsigned char *buf, size_t buffer_size)
int Opus::decode(SFLAudioSample *dst, unsigned char *buf, size_t buffer_size)
{
return opus_decode(decoder_, buf, buffer_size, dst, FRAME_SIZE, 0);
}
int Opus::encode(unsigned char *dst, short *src, size_t buffer_size)
int Opus::encode(unsigned char *dst, SFLAudioSample *src, size_t buffer_size)
{
return opus_encode(encoder_, src, FRAME_SIZE, dst, buffer_size * 2);
}

5
daemon/src/audio/codecs/opus.h
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@ -32,6 +32,7 @@
#define OPUS_H_
#include "noncopyable.h"
#include "sfl_types.h"
#include "audiocodec.h"
@ -45,8 +46,8 @@ public:
static const uint8 PAYLOAD_TYPE = 104; // dynamic payload type, out of range of video (96-99)
private:
virtual int decode(short *dst, unsigned char *buf, size_t buffer_size);
virtual int encode(unsigned char *dst, short *src, size_t buffer_size);
virtual int decode(SFLAudioSample *dst, unsigned char *buf, size_t buffer_size);
virtual int encode(unsigned char *dst, SFLAudioSample *src, size_t buffer_size);
//multichannel version
virtual int decode(std::vector<std::vector<short> > *dst, unsigned char *buf, size_t buffer_size, size_t dst_offset=0);

6
daemon/src/audio/codecs/speexcodec.h
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@ -29,7 +29,7 @@
* as that of the covered work.
*/
#include "global.h"
#include "sfl_types.h"
#include "audiocodec.h"
#include "noncopyable.h"
#include "array_size.h"
@ -83,13 +83,13 @@ private:
NON_COPYABLE(Speex);
virtual int decode(short *dst, unsigned char *src, size_t buf_size) {
virtual int decode(SFLAudioSample *dst, unsigned char *src, size_t buf_size) {
speex_bits_read_from(&speex_dec_bits_, (char*) src, buf_size);
speex_decode_int(speex_dec_state_, &speex_dec_bits_, dst);
return frameSize_;
}
virtual int encode(unsigned char *dst, short *src, size_t buf_size) {
virtual int encode(unsigned char *dst, SFLAudioSample *src, size_t buf_size) {
speex_bits_reset(&speex_enc_bits_);
speex_encode_int(speex_enc_state_, src, &speex_enc_bits_);
return speex_bits_write(&speex_enc_bits_, (char*) dst, buf_size);

2
daemon/src/audio/dcblocker.cpp
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@ -39,7 +39,7 @@ void DcBlocker::reset()
states.assign(states.size(), (struct StreamState){0, 0, 0, 0});
}
void DcBlocker::doProcess(SFLAudioSample *out, SFLAudioSample *in, int samples, struct StreamState * state)
void DcBlocker::doProcess(SFLAudioSample *out, SFLAudioSample *in, unsigned samples, struct StreamState * state)
{
for (unsigned i = 0; i < samples; ++i) {
state->x_ = in[i];

2
daemon/src/audio/dcblocker.h
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@ -52,7 +52,7 @@ class DcBlocker {
SFLAudioSample y_, x_, xm1_, ym1_;
};
void doProcess(SFLAudioSample *out, SFLAudioSample *in, int samples, struct StreamState * state);
void doProcess(SFLAudioSample *out, SFLAudioSample *in, unsigned samples, struct StreamState * state);
std::vector<StreamState> states;
};

4
daemon/src/audio/gaincontrol.cpp
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@ -69,7 +69,7 @@ void GainControl::process(SFLAudioSample *buf, int samples)
for (int i = 0; i < samples; i++) {
// linear conversion
in = (double)buf[i] / (double)SHRT_MAX;
in = (double) buf[i] / (double) SFL_DATA_FORMAT_MAX;
out = currentGain_ * in;
@ -81,7 +81,7 @@ void GainControl::process(SFLAudioSample *buf, int samples)
out = limiter_.limit(out);
buf[i] = (short)(out * (double)SHRT_MAX);
buf[i] = (SFLAudioSample) (out * (double) SFL_DATA_FORMAT_MAX);
}
diffRms = maxRms - targetLevelLinear_;

6
daemon/src/audio/mainbuffer.cpp
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@ -275,14 +275,14 @@ size_t MainBuffer::getData(AudioBuffer& buffer, const std::string &call_id)
for (CallIDSet::iterator iter_id = callid_set->begin();
iter_id != callid_set->end(); ++iter_id) {
/*size_t nbSmplToCopy = toCopy / sizeof(SFLDataFormat);
SFLDataFormat mixBuffer[nbSmplToCopy];
/*size_t nbSmplToCopy = toCopy / sizeof(SFLAudioSample);
SFLAudioSample mixBuffer[nbSmplToCopy];
memset(mixBuffer, 0, toCopy);
size = getDataByID(mixBuffer, toCopy, *iter_id, call_id);*/
size = getDataByID(mixBuffer, *iter_id, call_id);
if (size > 0) {
/*SFLDataFormat *dest = static_cast<SFLDataFormat*>(buffer);
/*SFLAudioSample *dest = static_cast<SFLAudioSample*>(buffer);
for (size_t k = 0; k < nbSmplToCopy; ++k)
dest[k] += mixBuffer[k];*/
buffer.mix(mixBuffer);

7
daemon/src/audio/samplerateconverter.cpp
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@ -29,6 +29,7 @@
*/
#include "samplerateconverter.h"
#include "sfl_types.h"
#include "manager.h"
#include <cassert>
#include "logger.h"
@ -54,7 +55,7 @@ SamplerateConverter::~SamplerateConverter()
}
void
SamplerateConverter::Short2FloatArray(const short *in, float *out, int len)
SamplerateConverter::Short2FloatArray(const SFLAudioSample *in, float *out, int len)
{
// factor is 1/(2^15), used to rescale the short int range to the
// [-1.0 - 1.0] float range.
@ -64,8 +65,8 @@ SamplerateConverter::Short2FloatArray(const short *in, float *out, int len)
}
void SamplerateConverter::resample(const AudioBuffer &dataIn, AudioBuffer &dataOut)
/*void SamplerateConverter::resample(SFLDataFormat *dataIn,
SFLDataFormat *dataOut,
/*void SamplerateConverter::resample(SFLAudioSample *dataIn,
SFLAudioSample *dataOut,
size_t dataOutSize,
int inputFreq,
int outputFreq,

4
daemon/src/audio/samplerateconverter.h
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@ -60,7 +60,7 @@ class SamplerateConverter {
* @param SamplerateConverter2 The desired sample rate
* @param nbSamples The number of samples to process
*/
//void resample(SFLDataFormat* dataIn, SFLDataFormat* dataOut, size_t dataOutSize, int oldrate, int newrate, size_t nbSamples);
//void resample(SFLAudioSample* dataIn, SFLAudioSample* dataOut, size_t dataOutSize, int oldrate, int newrate, size_t nbSamples);
void resample(const AudioBuffer& dataIn, AudioBuffer& dataOut);
/**
@ -69,7 +69,7 @@ class SamplerateConverter {
* @param out The resulting (float) array
* @param len The number of elements in both tables
*/
void Short2FloatArray(const short *in, float *out, int len);
void Short2FloatArray(const SFLAudioSample *in, float *out, int len);
private:

6
daemon/src/audio/sound/dtmfgenerator.h
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@ -101,7 +101,7 @@ class DTMFGenerator {
/*
* Get n samples of the signal of code code
* @param buffer a SFLDataFormat vector
* @param buffer a SFLAudioSample vector
* @param code dtmf code to get sound
*/
void getSamples(std::vector<SFLAudioSample> &buffer, unsigned char code);
@ -109,7 +109,7 @@ class DTMFGenerator {
/*
* Get next n samples (continues where previous call to
* genSample or genNextSamples stopped
* @param buffer a SFLDataFormat vector
* @param buffer a SFLAudioSample vector
*/
void getNextSamples(std::vector<SFLAudioSample> &buffer);
@ -118,7 +118,7 @@ class DTMFGenerator {
/**
* Fill tone buffer for a given index of the array of tones.
* @param index of the tone in the array tones_
* @return SFLDataFormat* The generated data
* @return SFLAudioSample* The generated data
*/
SFLAudioSample* fillToneBuffer(int index);
};

9
daemon/src/sfl_types.h
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@ -32,12 +32,13 @@
#define SFL_TYPES_H_
#include <cstddef> // for size_t
#include <stdint.h>
typedef short SFLAudioSample;
//typedef size_t AudioChannel;
typedef int16_t SFLAudioSample;
typedef signed short SINT16;
typedef signed int SINT32;
typedef int16_t SINT16;
typedef int32_t SINT32;
#define SFL_DATA_FORMAT_MAX SHRT_MAX
/*
typedef struct {