mirror of
https://git.jami.net/savoirfairelinux/jami-daemon.git
synced 2025-08-12 22:09:25 +08:00
* #7097: cleanup, removed redundant code in audio backend
This commit is contained in:
Tristan Matthews
@ -37,23 +37,21 @@
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#include <cstring>
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#include <cassert>
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AudioLoop::AudioLoop() :_buffer (0), _size (0), _pos (0), _sampleRate (0)
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AudioLoop::AudioLoop() : buffer_(0), size_(0), pos_(0), sampleRate_(0)
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{
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}
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AudioLoop::~AudioLoop()
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{
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delete [] _buffer;
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delete [] buffer_;
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}
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void
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AudioLoop::getNext (SFLDataFormat* output, size_t total_samples, short volume)
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{
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size_t pos = _pos;
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size_t pos = pos_;
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assert(_size);
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if(_size == 0) {
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if (size_ == 0) {
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_error("AudioLoop: Error: Audio loop size is 0");
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return;
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}
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@ -61,26 +59,26 @@ AudioLoop::getNext (SFLDataFormat* output, size_t total_samples, short volume)
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while (total_samples) {
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size_t samples = total_samples;
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if (samples > (_size-pos)) {
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samples = _size-pos;
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}
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if (samples > (size_ - pos))
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samples = size_ - pos;
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memcpy(output, _buffer+pos, samples*sizeof (SFLDataFormat)); // short>char conversion
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// short->char conversion
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memcpy(output, buffer_ + pos, samples * sizeof(SFLDataFormat));
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if (volume!=100) {
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for (size_t i=0; i<samples; i++) {
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*output = (*output * volume) /100;
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if (volume != 100) {
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double gain = volume * 0.01;
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for (size_t i = 0; i < samples; i++) {
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*output *= gain;
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output++;
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}
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} else {
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} else
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output += samples; // this is the destination...
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}
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pos = (pos + samples) % _size;
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pos = (pos + samples) % size_;
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total_samples -= samples;
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}
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_pos = pos;
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pos_ = pos;
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}
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@ -61,36 +61,36 @@ class AudioLoop
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* @param nb of int16 to send
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* @param volume The volume
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*/
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void getNext (SFLDataFormat* output, size_t samples, short volume=100);
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void getNext(SFLDataFormat* output, size_t samples, short volume=100);
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/**
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* Reset the pointer position
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*/
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void reset() {
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_pos = 0;
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pos_ = 0;
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}
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/**
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* Accessor to the size of the buffer
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* @return unsigned int The size
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*/
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size_t getSize() {
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return _size;
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size_t getSize() const {
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return size_;
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}
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protected:
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/** The data buffer */
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SFLDataFormat* _buffer;
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SFLDataFormat* buffer_;
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/** Number of samples inside the buffer */
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size_t _size;
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size_t size_;
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/** current position, set to 0, when initialize */
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size_t _pos;
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size_t pos_;
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/** Sample rate */
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unsigned int _sampleRate;
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unsigned int sampleRate_;
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private:
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@ -73,17 +73,9 @@ AudioRecord::AudioRecord() : fp (NULL)
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AudioRecord::~AudioRecord()
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{
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if (mixBuffer_) {
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delete [] mixBuffer_;
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}
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if (micBuffer_) {
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delete [] micBuffer_;
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}
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if (spkBuffer_) {
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delete [] spkBuffer_;
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}
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delete [] mixBuffer_;
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delete [] micBuffer_;
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delete [] spkBuffer_;
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}
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void AudioRecord::setSndSamplingRate (int smplRate)
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@ -96,16 +88,12 @@ int AudioRecord::getSndSamplingRate() const
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return sndSmplRate_;
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}
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void AudioRecord::setRecordingOption (FILE_TYPE type, SOUND_FORMAT format, int sndSmplRate, std::string path)
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void AudioRecord::setRecordingOption(FILE_TYPE type, int sndSmplRate, const std::string &path)
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{
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fileType_ = type;
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sndFormat_ = format;
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channels_ = 1;
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sndSmplRate_ = sndSmplRate;
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savePath_ = path + "/";
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}
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@ -309,11 +297,6 @@ bool AudioRecord::setRawFile()
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return false;
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}
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if (sndFormat_ != INT16) { // TODO need to change INT16 to SINT16
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sndFormat_ = INT16;
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_debug ("AudioRecord::setRawFile() : using 16-bit signed integer data format for file.");
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}
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_debug ("AudioRecord:setRawFile() : created RAW file.");
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return true;
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@ -342,9 +325,7 @@ bool AudioRecord::setWavFile()
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hdr.num_chans = channels_;
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if (sndFormat_ == INT16) { // TODO need to write INT16 to SINT16
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hdr.bits_per_samp = 16;
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}
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hdr.bits_per_samp = 16;
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hdr.bytes_per_samp = (SINT16) (channels_ * hdr.bits_per_samp / 8);
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@ -499,15 +480,11 @@ void AudioRecord::recData (SFLDataFormat* buffer, int nSamples)
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return;
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}
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if (sndFormat_ == INT16) { // TODO change INT16 to SINT16
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if (fwrite (buffer, sizeof (SFLDataFormat), nSamples, fp) != (unsigned int) nSamples)
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_warn ("AudioRecord: Could not record data! ");
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else {
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fflush (fp);
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byteCounter_ += (unsigned long) (nSamples*sizeof (SFLDataFormat));
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}
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if (fwrite (buffer, sizeof (SFLDataFormat), nSamples, fp) != (unsigned int) nSamples)
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_warn ("AudioRecord: Could not record data! ");
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else {
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fflush (fp);
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byteCounter_ += (unsigned long) (nSamples*sizeof (SFLDataFormat));
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}
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}
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@ -527,25 +504,16 @@ void AudioRecord::recData (SFLDataFormat* buffer_1, SFLDataFormat* buffer_2, int
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return;
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}
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for (int k = 0; k < nSamples_1; k++) {
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mixBuffer_[k] = (buffer_1[k]+buffer_2[k]);
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if (sndFormat_ == INT16) { // TODO change INT16 to SINT16
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for (int k=0; k<nSamples_1; k++) {
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mixBuffer_[k] = (buffer_1[k]+buffer_2[k]);
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if (fwrite (&mixBuffer_[k], 2, 1, fp) != 1)
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_warn ("AudioRecord: Could not record data!");
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else {
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fflush (fp);
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}
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}
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if (fwrite (&mixBuffer_[k], 2, 1, fp) != 1)
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_warn ("AudioRecord: Could not record data!");
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else
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fflush (fp);
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}
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byteCounter_ += (unsigned long) (nSamples_1*sizeof (SFLDataFormat));
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byteCounter_ += (unsigned long) (nSamples_1 * sizeof(SFLDataFormat));
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}
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return;
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}
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@ -40,25 +40,19 @@ class AudioRecord
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{
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public:
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enum FILE_TYPE { FILE_RAW, FILE_WAV };
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AudioRecord();
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~AudioRecord();
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/**
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* Set the sampling rate for this recorder
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*/
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void setSndSamplingRate (int smplRate);
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/**
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* Get the recrding sampling rate
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*/
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void setSndSamplingRate(int smplRate);
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/**
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* Get the recrding sampling rate
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*/
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int getSndSamplingRate(void) const;
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/**
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* Set the recording option
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*/
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void setRecordingOption (FILE_TYPE type, SOUND_FORMAT format, int sndSmplRate, std::string path);
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void setRecordingOption(FILE_TYPE type, int sndSmplRate, const std::string &path);
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/**
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* Init recording file path
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@ -184,11 +178,6 @@ class AudioRecord
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*/
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FILE_TYPE fileType_;
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/**
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* Sound format (SINT16/SINT32)
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*/
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SOUND_FORMAT sndFormat_;
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/**
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* Number of channels
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*/
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@ -27,6 +27,8 @@
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* shall include the source code for the parts of OpenSSL used as well
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* as that of the covered work.
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*/
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#include "config.h"
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#include "AudioZrtpSession.h"
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#include "ZrtpSessionCallback.h"
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@ -49,14 +51,13 @@ namespace sfl
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{
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AudioZrtpSession::AudioZrtpSession (SIPCall * sipcall, const std::string& zidFilename) :
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// ost::SymmetricZRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()),
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AudioRtpSession(sipcall, Zrtp, static_cast<ost::RTPDataQueue *>(this), static_cast<ost::Thread *>(this))
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,ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::ZrtpQueue> (ost::InetHostAddress (sipcall->getLocalIp().c_str()),
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sipcall->getLocalAudioPort(),
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0,
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ost::MembershipBookkeeping::defaultMembersHashSize,
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ost::defaultApplication())
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, _zidFilename (zidFilename)
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AudioRtpSession(sipcall, Zrtp, this, this),
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ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::ZrtpQueue>(ost::InetHostAddress(sipcall->getLocalIp().c_str()),
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sipcall->getLocalAudioPort(),
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0,
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ost::MembershipBookkeeping::defaultMembersHashSize,
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ost::defaultApplication()),
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_zidFilename (zidFilename)
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{
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_debug ("AudioZrtpSession initialized");
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initializeZid();
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@ -96,7 +97,7 @@ void AudioZrtpSession::initializeZid (void)
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// xdg_config = std::string (HOMEDIR) + DIR_SEPARATOR_STR + ".cache/sflphone";
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std::string xdg_config = std::string (HOMEDIR) + DIR_SEPARATOR_STR + ".cache" + DIR_SEPARATOR_STR + PROGDIR + "/" + _zidFilename;
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std::string xdg_config = std::string(HOMEDIR) + DIR_SEPARATOR_STR + ".cache" + DIR_SEPARATOR_STR + PACKAGE + "/" + _zidFilename;
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_debug (" xdg_config %s", xdg_config.c_str());
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@ -31,6 +31,7 @@
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* as that of the covered work.
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*/
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#include "config.h"
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#include "audiocodecfactory.h"
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#include <cstdlib>
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#include <algorithm> // for std::find
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@ -142,7 +143,7 @@ std::vector<sfl::Codec*> AudioCodecFactory::scanCodecDirectory()
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std::vector<sfl::Codec*> codecs;
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std::vector<std::string> dirToScan;
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dirToScan.push_back(std::string(HOMEDIR) + DIR_SEPARATOR_STR "." PROGDIR "/");
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dirToScan.push_back(std::string(HOMEDIR) + DIR_SEPARATOR_STR "." PACKAGE "/");
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dirToScan.push_back(CODECS_DIR "/");
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const char *envDir = getenv("CODECS_PATH");
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if (envDir)
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@ -162,7 +163,7 @@ std::vector<sfl::Codec*> AudioCodecFactory::scanCodecDirectory()
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dirent *dirStruct;
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while ( (dirStruct = readdir (dir))) {
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std::string file = dirStruct->d_name ;
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if (file == CURRENT_DIR or file == PARENT_DIR)
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if (file == "." or file == "..")
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continue;
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if (seemsValid (file) && !alreadyInCache (file)) {
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@ -243,8 +244,9 @@ bool AudioCodecFactory::seemsValid (const std::string &lib)
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{
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// The name of the shared library seems valid <==> it looks like libcodec_xxx.so
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// We check this
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std::string prefix = SFL_CODEC_VALID_PREFIX;
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std::string suffix = SFL_CODEC_VALID_EXTEN;
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static const std::string prefix("libcodec_");
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static const std::string suffix(".so");
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ssize_t len = lib.length() - prefix.length() - suffix.length();
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if (len < 0)
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@ -257,17 +259,17 @@ bool AudioCodecFactory::seemsValid (const std::string &lib)
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#ifndef HAVE_SPEEX_CODEC
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if (lib.substr (prefix.length() , len) == SPEEX_STRING_DESCRIPTION)
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if (lib.substr (prefix.length() , len) == "speex")
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return false;
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#endif
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#ifndef HAVE_GSM_CODEC
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if (lib.substr (prefix.length() , len) == GSM_STRING_DESCRIPTION)
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if (lib.substr (prefix.length() , len) == "gsm")
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return false;
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#endif
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#ifndef BUILD_ILBC
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if (lib.substr (prefix.length() , len) == ILBC_STRING_DESCRIPTION)
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if (lib.substr (prefix.length() , len) == "ilbc")
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return false;
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#endif
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@ -32,13 +32,8 @@
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Recordable::Recordable() : recorder (&recAudio, Manager::instance().getMainBuffer())
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{
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_debug("-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Recordable Constructor -=-=-=-=-=-=-=-=-=--=-=-=-");
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FILE_TYPE fileType = FILE_WAV;
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SOUND_FORMAT soundFormat = INT16;
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recAudio.setRecordingOption (fileType, soundFormat, 8000, Manager::instance().audioPreference.getRecordpath());
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recAudio.setRecordingOption (AudioRecord::FILE_WAV, 8000, Manager::instance().audioPreference.getRecordpath());
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}
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@ -39,8 +39,8 @@ SamplerateConverter::SamplerateConverter (int freq) : _maxFreq(freq)
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_samples = (freq * 20) / 1000; // start with 20 ms buffers
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_floatBufferIn = new float32[_samples];
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_floatBufferOut = new float32[_samples];
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_floatBufferIn = new float[_samples];
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_floatBufferOut = new float[_samples];
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}
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SamplerateConverter::~SamplerateConverter (void)
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@ -61,7 +61,6 @@ SamplerateConverter::Short2FloatArray (const short *in, float *out, int len)
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out[len] = (float) in[len] * .000030517578125f;
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}
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//TODO Add ifdef for int16 or float32 type
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void SamplerateConverter::resample (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int inputFreq , int outputFreq , int nbSamples)
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{
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double sampleFactor = (double) outputFreq / inputFreq;
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@ -77,8 +76,8 @@ void SamplerateConverter::resample (SFLDataFormat* dataIn , SFLDataFormat* dataO
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_samples = maxSamples;
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delete [] _floatBufferIn;
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delete [] _floatBufferOut;
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_floatBufferIn = new float32[_samples];
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_floatBufferOut = new float32[_samples];
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_floatBufferIn = new float[_samples];
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_floatBufferOut = new float[_samples];
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}
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SRC_DATA src_data;
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@ -78,8 +78,8 @@ class SamplerateConverter
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SamplerateConverter& operator= (const SamplerateConverter& rh);
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/* temporary buffers */
|
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float32* _floatBufferIn;
|
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float32* _floatBufferOut;
|
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float * _floatBufferIn;
|
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float * _floatBufferOut;
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size_t _samples; // size in samples of temporary buffers
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int _maxFreq; // maximal output frequency
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@ -73,51 +73,40 @@ RawFile::RawFile(const std::string& name, sfl::AudioCodec* codec, unsigned int s
|
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SFLDataFormat *monoBuffer = new SFLDataFormat[decodedSize];
|
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SFLDataFormat *bufpos = monoBuffer;
|
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unsigned char *filepos = reinterpret_cast<unsigned char *>(fileBuffer);
|
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_size = decodedSize;
|
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while(length >= encFrameSize) {
|
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size_ = decodedSize;
|
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while (length >= encFrameSize) {
|
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bufpos += audioCodec->decode (bufpos, filepos, encFrameSize);
|
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filepos += encFrameSize;
|
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length -= encFrameSize;
|
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}
|
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delete[] fileBuffer;
|
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delete [] fileBuffer;
|
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|
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if (sampleRate == audioRate) {
|
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#ifdef DATAFORMAT_IS_FLOAT
|
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_buffer = new SFLDataFormat[_size];
|
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src_short_to_float_array (monoBuffer, _buffer, _size);
|
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delete[] monoBuffer;
|
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#else
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_buffer = monoBuffer;
|
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#endif
|
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|
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} else {
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if (sampleRate == audioRate)
|
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buffer_ = monoBuffer;
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else {
|
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double factord = (double) sampleRate / audioRate;
|
||||
float* floatBufferIn = new float[_size];
|
||||
int sizeOut = ceil(factord*_size);
|
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src_short_to_float_array (monoBuffer, floatBufferIn, _size);
|
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delete[] monoBuffer;
|
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SFLDataFormat* _buffer = new SFLDataFormat[sizeOut];
|
||||
float* floatBufferIn = new float[size_];
|
||||
int sizeOut = ceil(factord * size_);
|
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src_short_to_float_array (monoBuffer, floatBufferIn, size_);
|
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delete [] monoBuffer;
|
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delete [] buffer_;
|
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buffer_ = new SFLDataFormat[sizeOut];
|
||||
|
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SRC_DATA src_data;
|
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src_data.data_in = floatBufferIn;
|
||||
src_data.input_frames = _size;
|
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src_data.input_frames = size_;
|
||||
src_data.output_frames = sizeOut;
|
||||
src_data.src_ratio = factord;
|
||||
|
||||
#ifdef DATAFORMAT_IS_FLOAT
|
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src_data.data_out = _buffer;
|
||||
src_simple (&src_data, SRC_SINC_BEST_QUALITY, 1);
|
||||
#else
|
||||
float* floatBufferOut = new float[sizeOut];
|
||||
src_data.data_out = floatBufferOut;
|
||||
|
||||
src_simple (&src_data, SRC_SINC_BEST_QUALITY, 1);
|
||||
src_float_to_short_array (floatBufferOut, _buffer, src_data.output_frames_gen);
|
||||
src_float_to_short_array (floatBufferOut, buffer_, src_data.output_frames_gen);
|
||||
|
||||
delete [] floatBufferOut;
|
||||
#endif
|
||||
delete [] floatBufferIn;
|
||||
_size = src_data.output_frames_gen;
|
||||
size_ = src_data.output_frames_gen;
|
||||
}
|
||||
}
|
||||
|
||||
@ -148,8 +137,8 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
|
||||
SINT32 chunk_size; // fmt chunk size
|
||||
unsigned short formatTag; // data compression tag
|
||||
|
||||
fileStream.read ( (char*) &chunk_size, 4); // Read fmt chunk size.
|
||||
fileStream.read ( (char*) &formatTag, 2);
|
||||
fileStream.read(reinterpret_cast<char*>(&chunk_size), 4); // Read fmt chunk size.
|
||||
fileStream.read(reinterpret_cast<char*>(&formatTag), 2);
|
||||
|
||||
if (formatTag != 1) // PCM = 1, FLOAT = 3
|
||||
throw AudioFileException("File contains an unsupported data format type");
|
||||
@ -158,12 +147,12 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
|
||||
SINT16 chan;
|
||||
fileStream.read ( (char*) &chan, 2);
|
||||
|
||||
if(chan > 2)
|
||||
if (chan > 2)
|
||||
throw AudioFileException("WaveFile: unsupported number of channels");
|
||||
|
||||
// Get file sample rate from the header.
|
||||
SINT32 srate;
|
||||
fileStream.read ( (char*) &srate, 4);
|
||||
fileStream.read( (char*) &srate, 4);
|
||||
|
||||
SINT32 avgb;
|
||||
fileStream.read ( (char*) &avgb, 4);
|
||||
@ -185,7 +174,7 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
|
||||
|
||||
// Sample rate converter initialized with 88200 sample long
|
||||
int converterSamples = ((unsigned int)srate > audioSamplingRate) ? srate : audioSamplingRate;
|
||||
SamplerateConverter _converter (converterSamples);
|
||||
SamplerateConverter converter_(converterSamples);
|
||||
|
||||
// Get length of data from the header.
|
||||
SINT32 bytes;
|
||||
@ -197,31 +186,28 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
|
||||
nbSamples, bytes, blockal, srate, avgb, chunk_size, dt);
|
||||
|
||||
// Should not be longer than a minute
|
||||
if (nbSamples > (unsigned int) (60*srate))
|
||||
nbSamples = 60*srate;
|
||||
if (nbSamples > static_cast<unsigned int>(60 * srate))
|
||||
nbSamples = 60 * srate;
|
||||
|
||||
SFLDataFormat *tempBuffer = new SFLDataFormat[nbSamples];
|
||||
if (!tempBuffer)
|
||||
throw AudioFileException("Could not allocate temporary buffer");
|
||||
|
||||
fileStream.read ( (char *) tempBuffer, nbSamples*sizeof (SFLDataFormat));
|
||||
fileStream.read (reinterpret_cast<char *>(tempBuffer), nbSamples * sizeof(SFLDataFormat));
|
||||
|
||||
// mix two channels together if stereo
|
||||
if(chan == 2) {
|
||||
for(unsigned int i = 0; i < nbSamples-1; i+=2)
|
||||
tempBuffer[i/2] = (tempBuffer[i] + tempBuffer[i+1]) / 2;
|
||||
nbSamples /= 2;
|
||||
if (chan == 2) {
|
||||
for (unsigned int i = 0; i < nbSamples - 1; i += 2)
|
||||
tempBuffer[static_cast<size_t>(i * 0.5)] = (tempBuffer[i] + tempBuffer[i + 1]) * 0.5;
|
||||
nbSamples *= 0.5;
|
||||
}
|
||||
|
||||
if ((unsigned int)srate != audioSamplingRate) {
|
||||
if ((unsigned int) srate != audioSamplingRate) {
|
||||
int outSamples = ((float) nbSamples * ( (float) audioSamplingRate / (float) srate));
|
||||
_buffer = new SFLDataFormat[outSamples];
|
||||
_converter.resample (tempBuffer, _buffer, srate, audioSamplingRate, nbSamples);
|
||||
delete[] tempBuffer;
|
||||
} else {
|
||||
_buffer = tempBuffer;
|
||||
}
|
||||
buffer_ = new SFLDataFormat[outSamples];
|
||||
converter_.resample (tempBuffer, buffer_, srate, audioSamplingRate, nbSamples);
|
||||
delete [] tempBuffer;
|
||||
} else
|
||||
buffer_ = tempBuffer;
|
||||
|
||||
_size = nbSamples;
|
||||
_sampleRate = audioSamplingRate;
|
||||
size_ = nbSamples;
|
||||
sampleRate_ = audioSamplingRate;
|
||||
}
|
||||
|
||||
@ -35,31 +35,26 @@
|
||||
* YM: 2006-11-15: changes unsigned int to std::string::size_type, thanks to Pierre Pomes (AMD64 compilation)
|
||||
*/
|
||||
#include "tone.h"
|
||||
#include <math.h>
|
||||
#include <cmath>
|
||||
#include <cstdlib>
|
||||
#include <cstring>
|
||||
|
||||
#define TABLE_LENGTH 4096
|
||||
double TWOPI = 2 * M_PI;
|
||||
static const double TWOPI = 2.0 * M_PI;
|
||||
|
||||
Tone::Tone (const std::string& definition, unsigned int sampleRate) : AudioLoop(), _sampleRate (sampleRate), _xhigher (0.0), _xlower (0.0)
|
||||
Tone::Tone(const std::string& definition, unsigned int sampleRate) :
|
||||
sampleRate_(sampleRate), xhigher_(0.0), xlower_(0.0)
|
||||
{
|
||||
fillWavetable();
|
||||
genBuffer (definition); // allocate memory with definition parameter
|
||||
}
|
||||
|
||||
Tone::~Tone()
|
||||
{
|
||||
genBuffer(definition); // allocate memory with definition parameter
|
||||
}
|
||||
|
||||
void
|
||||
Tone::genBuffer (const std::string& definition)
|
||||
Tone::genBuffer(const std::string& definition)
|
||||
{
|
||||
if (definition.empty()) {
|
||||
if (definition.empty())
|
||||
return;
|
||||
}
|
||||
|
||||
_size = 0;
|
||||
size_ = 0;
|
||||
|
||||
SFLDataFormat* buffer = new SFLDataFormat[SIZEBUF]; //1kb
|
||||
SFLDataFormat* bufferPos = buffer;
|
||||
@ -76,67 +71,61 @@ Tone::genBuffer (const std::string& definition)
|
||||
do {
|
||||
posEnd = definition.find (',', posStart);
|
||||
|
||||
if (posEnd == std::string::npos) {
|
||||
if (posEnd == std::string::npos)
|
||||
posEnd = deflen;
|
||||
}
|
||||
|
||||
/* begin scope */
|
||||
{
|
||||
// Sample string: "350+440" or "350+440/2000,244+655/2000"
|
||||
int freq1, freq2, time;
|
||||
s = definition.substr (posStart, posEnd-posStart);
|
||||
s = definition.substr(posStart, posEnd-posStart);
|
||||
|
||||
// The 1st frequency is before the first + or the /
|
||||
std::string::size_type pos_plus = s.find ('+');
|
||||
std::string::size_type pos_slash = s.find ('/');
|
||||
std::string::size_type len = s.length();
|
||||
std::string::size_type endfrequency = 0;
|
||||
size_t pos_plus = s.find('+');
|
||||
size_t pos_slash = s.find('/');
|
||||
size_t len = s.length();
|
||||
size_t endfrequency = 0;
|
||||
|
||||
if (pos_slash == std::string::npos) {
|
||||
time = 0;
|
||||
endfrequency = len;
|
||||
} else {
|
||||
time = atoi ( (s.substr (pos_slash+1,len-pos_slash-1)).data());
|
||||
time = atoi(s.substr(pos_slash + 1, len - pos_slash - 1).c_str());
|
||||
endfrequency = pos_slash;
|
||||
}
|
||||
|
||||
// without a plus = 1 frequency
|
||||
if (pos_plus == std::string::npos) {
|
||||
freq1 = atoi ( (s.substr (0,endfrequency)).data());
|
||||
freq1 = atoi(s.substr(0, endfrequency).c_str());
|
||||
freq2 = 0;
|
||||
} else {
|
||||
freq1 = atoi ( (s.substr (0,pos_plus)).data());
|
||||
freq2 = atoi ( (s.substr (pos_plus+1, endfrequency-pos_plus-1)).data());
|
||||
freq1 = atoi(s.substr(0, pos_plus).c_str());
|
||||
freq2 = atoi(s.substr(pos_plus + 1, endfrequency - pos_plus - 1).c_str());
|
||||
}
|
||||
|
||||
// If there is time or if it's unlimited
|
||||
if (time == 0) {
|
||||
count = _sampleRate;
|
||||
} else {
|
||||
count = (_sampleRate * time) / 1000;
|
||||
}
|
||||
if (time == 0)
|
||||
count = sampleRate_;
|
||||
else
|
||||
count = (sampleRate_ * time) / 1000;
|
||||
|
||||
// Generate SAMPLING_RATE samples of sinus, buffer is the result
|
||||
_debug ("genSin(%d, %d)", freq1, freq2);
|
||||
genSin (bufferPos, freq1, freq2, count);
|
||||
_debug("genSin(%d, %d)", freq1, freq2);
|
||||
genSin(bufferPos, freq1, freq2, count);
|
||||
|
||||
// To concatenate the different buffers for each section.
|
||||
_size += (count);
|
||||
size_ += count;
|
||||
bufferPos += count;
|
||||
} /* end scope */
|
||||
|
||||
bufferPos += (count);
|
||||
}
|
||||
|
||||
posStart = posEnd+1;
|
||||
posStart = posEnd + 1;
|
||||
} while (posStart < deflen);
|
||||
|
||||
_buffer = new SFLDataFormat[_size];
|
||||
buffer_ = new SFLDataFormat[size_];
|
||||
|
||||
memcpy (_buffer, buffer, _size*sizeof (SFLDataFormat)); // copy char, not SFLDataFormat.
|
||||
memcpy(buffer_, buffer, size_ * sizeof(SFLDataFormat)); // copy char, not SFLDataFormat.
|
||||
|
||||
delete[] buffer;
|
||||
|
||||
buffer=0;
|
||||
|
||||
bufferPos=0;
|
||||
delete [] buffer;
|
||||
}
|
||||
|
||||
void
|
||||
@ -144,9 +133,8 @@ Tone::fillWavetable()
|
||||
{
|
||||
double tableSize = (double) TABLE_LENGTH;
|
||||
|
||||
for (int i = 0; i < TABLE_LENGTH; i++) {
|
||||
_wavetable[i] = sin ( ( (double) i / (tableSize - 1.0)) * TWOPI);
|
||||
}
|
||||
for (int i = 0; i < TABLE_LENGTH; ++i)
|
||||
wavetable_[i] = sin((static_cast<double>(i) / (tableSize - 1.0)) * TWOPI);
|
||||
}
|
||||
|
||||
double
|
||||
@ -156,24 +144,24 @@ Tone::interpolate (double x)
|
||||
double yi_0, yi_1, A, B;
|
||||
|
||||
xi_0 = (int) x;
|
||||
xi_1 = xi_0+1;
|
||||
xi_1 = xi_0 + 1;
|
||||
|
||||
yi_0 =_wavetable[xi_0];
|
||||
yi_1 = _wavetable[xi_1];
|
||||
yi_0 = wavetable_[xi_0];
|
||||
yi_1 = wavetable_[xi_1];
|
||||
|
||||
A = (x - xi_0);
|
||||
B = 1.0 - A;
|
||||
|
||||
return A*yi_0 + B*yi_1;
|
||||
return (A * yi_0) + (B * yi_1);
|
||||
}
|
||||
|
||||
void
|
||||
Tone::genSin (SFLDataFormat* buffer, int frequency1, int frequency2, int nb)
|
||||
{
|
||||
_xhigher = 0.0;
|
||||
_xlower = 0.0;
|
||||
xhigher_ = 0.0;
|
||||
xlower_ = 0.0;
|
||||
|
||||
double sr = (double) _sampleRate;
|
||||
double sr = (double) sampleRate_;
|
||||
double tableSize = (double) TABLE_LENGTH;
|
||||
|
||||
double N_h = sr / (double) (frequency1);
|
||||
@ -182,27 +170,25 @@ Tone::genSin (SFLDataFormat* buffer, int frequency1, int frequency2, int nb)
|
||||
double dx_h = tableSize / N_h;
|
||||
double dx_l = tableSize / N_l;
|
||||
|
||||
double x_h = _xhigher;
|
||||
double x_l = _xlower;
|
||||
double x_h = xhigher_;
|
||||
double x_l = xlower_;
|
||||
|
||||
double amp = (double) SFLDataAmplitude;
|
||||
static const double DATA_AMPLITUDE = 2047;
|
||||
double amp = DATA_AMPLITUDE;
|
||||
|
||||
for (int t = 0; t < nb; t ++) {
|
||||
buffer[t] = (int16) (amp* (interpolate (x_h) + interpolate (x_l)));
|
||||
buffer[t] = static_cast<SFLDataFormat>(amp * (interpolate(x_h) + interpolate(x_l)));
|
||||
x_h += dx_h;
|
||||
x_l += dx_l;
|
||||
|
||||
if (x_h > tableSize) {
|
||||
while (x_h > tableSize)
|
||||
x_h -= tableSize;
|
||||
}
|
||||
|
||||
if (x_l > tableSize) {
|
||||
while (x_l > tableSize)
|
||||
x_l -= tableSize;
|
||||
}
|
||||
}
|
||||
|
||||
_xhigher = x_h;
|
||||
_xlower = x_l;
|
||||
|
||||
xhigher_ = x_h;
|
||||
xlower_ = x_l;
|
||||
}
|
||||
|
||||
|
||||
@ -36,8 +36,6 @@
|
||||
#include <string>
|
||||
#include "audio/audioloop.h"
|
||||
|
||||
#define TABLE_LENGTH 4096
|
||||
|
||||
/**
|
||||
* @file tone.h
|
||||
* @brief Tone sample (dial, busy, ring, congestion)
|
||||
@ -52,11 +50,6 @@ class Tone : public AudioLoop
|
||||
*/
|
||||
Tone (const std::string& definition, unsigned int sampleRate);
|
||||
|
||||
/**
|
||||
* Destructor
|
||||
*/
|
||||
~Tone();
|
||||
|
||||
/** The different kind of tones */
|
||||
enum TONEID {
|
||||
TONE_DIALTONE = 0,
|
||||
@ -74,17 +67,17 @@ class Tone : public AudioLoop
|
||||
* @param nb are the number of int16 (mono) to generate
|
||||
* by example nb=5 generate 10 int16, 5 for the left, 5 for the right
|
||||
*/
|
||||
void genSin (SFLDataFormat* buffer, int frequency1, int frequency2, int nb);
|
||||
void genSin(SFLDataFormat* buffer, int frequency1, int frequency2, int nb);
|
||||
|
||||
/**
|
||||
*
|
||||
*/
|
||||
void fillWavetable (void);
|
||||
void fillWavetable(void);
|
||||
|
||||
/**
|
||||
*
|
||||
*/
|
||||
double interpolate (double x);
|
||||
double interpolate(double x);
|
||||
|
||||
|
||||
private:
|
||||
@ -93,15 +86,16 @@ class Tone : public AudioLoop
|
||||
* allocate the memory with the definition
|
||||
* @param definition String that contain frequency/time of the tone.
|
||||
*/
|
||||
void genBuffer (const std::string& definition);
|
||||
void genBuffer(const std::string& definition);
|
||||
|
||||
/** Sample rate */
|
||||
unsigned int _sampleRate;
|
||||
unsigned int sampleRate_;
|
||||
|
||||
double _wavetable[TABLE_LENGTH];
|
||||
static const int TABLE_LENGTH = 4096;
|
||||
double wavetable_[TABLE_LENGTH];
|
||||
|
||||
double _xhigher;
|
||||
double _xlower;
|
||||
double xhigher_;
|
||||
double xlower_;
|
||||
};
|
||||
|
||||
#endif // __TONE_H__
|
||||
|
||||
@ -49,38 +49,13 @@
|
||||
#define XDG_CACHE_HOME (getenv ("XDG_CACHE_HOME"))
|
||||
const char * const ZRTP_ZID_FILENAME = "sfl.zid";
|
||||
|
||||
typedef float float32;
|
||||
typedef short int16;
|
||||
|
||||
//useful typedefs.
|
||||
typedef signed short SINT16;
|
||||
typedef signed int SINT32;
|
||||
|
||||
typedef unsigned long FILE_TYPE;
|
||||
typedef unsigned long SOUND_FORMAT;
|
||||
|
||||
const FILE_TYPE FILE_RAW = 1;
|
||||
const FILE_TYPE FILE_WAV = 2;
|
||||
|
||||
static const SOUND_FORMAT INT16 = 0x2; // TODO shold change these symbols
|
||||
static const SOUND_FORMAT INT32 = 0x8;
|
||||
|
||||
#define PIDFILE "sfl.pid"
|
||||
|
||||
#ifdef DATAFORMAT_IS_FLOAT
|
||||
#define SFLDataFormat float32
|
||||
#define SFLDataFormatString "Float32"
|
||||
#define SFLDataAmplitude 0.05
|
||||
#else
|
||||
#define SFLDataFormat int16
|
||||
#define SFLDataFormatString "Int16"
|
||||
#define SFLDataAmplitude (32767 >> 4)
|
||||
#endif
|
||||
|
||||
#define PROGNAME "sflphoned" /** Binary name */
|
||||
#define PROGDIR "sflphone" /** Program directory */
|
||||
#define RINGDIR "ringtones" /** Ringtones directory */
|
||||
#define CODECDIR "codecs" /** Codecs directory */
|
||||
typedef short SFLDataFormat;
|
||||
|
||||
#define SIZEBUF 400000 /** About 12 sec of buffering at 8000 Hz*/
|
||||
|
||||
@ -92,19 +67,11 @@ static const SOUND_FORMAT INT32 = 0x8;
|
||||
#define PCM_DSNOOP "plug:dsnoop" /** Alsa plugin for microphone sharing */
|
||||
#define PCM_DMIX_DSNOOP "dmix/dsnoop" /** Audio profile using Alsa dmix/dsnoop */
|
||||
|
||||
#define SFL_CODEC_VALID_PREFIX "libcodec_" /** Valid prefix for codecs shared library */
|
||||
#define SFL_CODEC_VALID_EXTEN ".so" /** Valid extension for codecs shared library */
|
||||
#define CURRENT_DIR "." /** Current directory */
|
||||
#define PARENT_DIR ".." /** Parent directory */
|
||||
|
||||
#define SFL_PCM_BOTH 0x0021 /** To open both playback and capture devices */
|
||||
#define SFL_PCM_PLAYBACK 0x0022 /** To open playback device only */
|
||||
#define SFL_PCM_CAPTURE 0x0023 /** To open capture device only */
|
||||
#define SFL_PCM_RINGTONE 0x0024
|
||||
#define SFL_PCM_RINGTONE 0x0024
|
||||
|
||||
#define GSM_STRING_DESCRIPTION "gsm" /** GSM codec string description */
|
||||
#define SPEEX_STRING_DESCRIPTION "speex" /** SPEEX codec string description */
|
||||
#define ILBC_STRING_DESCRIPTION "ilbc" /** Ilbc codec string description */
|
||||
#define RINGTONE_ENABLED TRUE_STR /** Custom ringtone enable or not */
|
||||
#define DISPLAY_DIALPAD TRUE_STR /** Display dialpad or not */
|
||||
#define DISPLAY_VOLUME_CONTROLS TRUE_STR /** Display the volume controls or not */
|
||||
@ -126,9 +93,6 @@ static const SOUND_FORMAT INT32 = 0x8;
|
||||
#define DEFAULT_SIP_PORT 5060
|
||||
#define DEFAULT_SIP_TLS_PORT 5061
|
||||
|
||||
#define HOOK_DEFAULT_SIP_FIELD "X-sflphone-url"
|
||||
#define HOOK_DEFAULT_URL_COMMAND "x-www-browser"
|
||||
|
||||
/** Enumeration that contains known audio payloads */
|
||||
enum {
|
||||
// http://www.iana.org/assignments/rtp-parameters
|
||||
|
||||
@ -1743,6 +1743,7 @@ void ManagerImpl::ringtone (const std::string& accountID)
|
||||
std::string ringchoice = account->getRingtonePath();
|
||||
if (ringchoice.find (DIR_SEPARATOR_STR) == std::string::npos) {
|
||||
// check inside global share directory
|
||||
static const char * RINGDIR = "ringtones";
|
||||
ringchoice = std::string (PROGSHAREDIR) + DIR_SEPARATOR_STR
|
||||
+ RINGDIR + DIR_SEPARATOR_STR + ringchoice;
|
||||
}
|
||||
@ -1819,7 +1820,7 @@ ManagerImpl::getTelephoneFile ()
|
||||
std::string ManagerImpl::getConfigFile (void) const
|
||||
{
|
||||
std::string configdir = std::string (HOMEDIR) + DIR_SEPARATOR_STR + ".config"
|
||||
+ DIR_SEPARATOR_STR + PROGDIR;
|
||||
+ DIR_SEPARATOR_STR + PACKAGE;
|
||||
|
||||
if (XDG_CONFIG_HOME != NULL) {
|
||||
std::string xdg_env = std::string (XDG_CONFIG_HOME);
|
||||
@ -1833,6 +1834,7 @@ std::string ManagerImpl::getConfigFile (void) const
|
||||
_debug ("Cannot create directory: %m");
|
||||
}
|
||||
|
||||
static const char * PROGNAME = "sflphoned";
|
||||
return configdir + DIR_SEPARATOR_STR + PROGNAME + ".yml";
|
||||
}
|
||||
|
||||
|
||||
Reference in New Issue
Block a user