From 420c4274e1673752b3e1f9e6fbd80d42c6c0cf9a Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Adrien=20B=C3=A9raud?= <adrien.beraud@savoirfairelinux.com>
Date: Wed, 21 Jan 2015 00:24:38 -0500
Subject: [PATCH] rebranding: ring_types, ring::AudioBuffer
Refs #62691
Change-Id: I69db47f5af094f151df52f709bb3dee91908fc08
---
daemon/src/Makefile.am | 2 +-
daemon/src/iax/iaxaccount.h | 2 +-
daemon/src/iax/iaxvoiplink.cpp | 2 +-
daemon/src/iax/iaxvoiplink.h | 2 +-
daemon/src/media/audio/alsa/alsalayer.cpp | 4 +-
daemon/src/media/audio/alsa/alsalayer.h | 8 ++--
daemon/src/media/audio/audiobuffer.cpp | 30 +++++++--------
daemon/src/media/audio/audiobuffer.h | 38 +++++++++----------
daemon/src/media/audio/audioloop.cpp | 7 +---
daemon/src/media/audio/audioloop.h | 3 +-
daemon/src/media/audio/audiorecord.h | 1 -
daemon/src/media/audio/codecs/alaw.cpp | 10 ++---
daemon/src/media/audio/codecs/audiocodec.cpp | 12 +++---
daemon/src/media/audio/codecs/audiocodec.h | 12 +++---
daemon/src/media/audio/codecs/g722.cpp | 26 ++++++-------
daemon/src/media/audio/codecs/g729.cpp | 10 ++---
daemon/src/media/audio/codecs/g729.h | 10 ++---
daemon/src/media/audio/codecs/gsmcodec.cpp | 6 +--
daemon/src/media/audio/codecs/ilbc.cpp | 6 +--
daemon/src/media/audio/codecs/opuscodec.cpp | 14 +++----
daemon/src/media/audio/codecs/opuscodec.h | 8 ++--
daemon/src/media/audio/codecs/speexcodec.h | 6 +--
daemon/src/media/audio/codecs/ulaw.cpp | 10 ++---
.../src/media/audio/coreaudio/corelayer.cpp | 2 +-
daemon/src/media/audio/coreaudio/corelayer.h | 4 +-
daemon/src/media/audio/dcblocker.cpp | 8 ++--
daemon/src/media/audio/dcblocker.h | 8 ++--
daemon/src/media/audio/jack/jacklayer.cpp | 4 +-
daemon/src/media/audio/opensl/opensllayer.cpp | 4 +-
.../src/media/audio/pulseaudio/pulselayer.cpp | 6 +--
daemon/src/media/audio/resampler.cpp | 2 +-
daemon/src/media/audio/resampler.h | 4 +-
daemon/src/media/audio/ringbufferpool.cpp | 2 +-
daemon/src/media/audio/sound/audiofile.cpp | 2 +-
daemon/src/media/audio/sound/dtmf.cpp | 2 +-
daemon/src/media/audio/sound/dtmf.h | 4 +-
.../src/media/audio/sound/dtmfgenerator.cpp | 8 ++--
daemon/src/media/audio/sound/dtmfgenerator.h | 16 ++++----
daemon/src/media/audio/sound/tone.cpp | 6 +--
daemon/src/media/audio/sound/tone.h | 2 +-
daemon/src/media/video/video_decoder.cpp | 2 +-
daemon/src/media/video/video_encoder.cpp | 4 +-
daemon/src/{sfl_types.h => ring_types.h} | 10 ++---
daemon/src/ringdht/ringaccount.h | 2 +-
daemon/src/sip/sipaccount.h | 2 +-
daemon/src/sip/sipvoiplink.cpp | 2 +-
daemon/src/sip/sipvoiplink.h | 2 +-
daemon/test/audiobuffertest.cpp | 8 ++--
daemon/test/audiocodectest.cpp | 12 +++---
daemon/test/resamplertest.cpp | 10 ++---
daemon/test/ringbufferpooltest.cpp | 12 +++---
51 files changed, 187 insertions(+), 192 deletions(-)
rename daemon/src/{sfl_types.h => ring_types.h} (93%)
diff --git a/daemon/src/Makefile.am b/daemon/src/Makefile.am
index efa64bd009..77cce13c7a 100644
--- a/daemon/src/Makefile.am
+++ b/daemon/src/Makefile.am
@@ -122,7 +122,7 @@ libring_la_SOURCES = conference.cpp \
fileutils.h \
noncopyable.h \
utf8_utils.h \
- sfl_types.h \
+ ring_types.h \
intrin.h \
array_size.h \
account_schema.h \
diff --git a/daemon/src/iax/iaxaccount.h b/daemon/src/iax/iaxaccount.h
index 1ca1e208a4..2e524e571b 100644
--- a/daemon/src/iax/iaxaccount.h
+++ b/daemon/src/iax/iaxaccount.h
@@ -35,7 +35,7 @@
#include "account.h"
#include "iaxvoiplink.h"
-#include "sfl_types.h" // enable_if_base_of
+#include "ring_types.h" // enable_if_base_of
namespace YAML {
class Emitter;
diff --git a/daemon/src/iax/iaxvoiplink.cpp b/daemon/src/iax/iaxvoiplink.cpp
index 9c5961706c..675d865a99 100644
--- a/daemon/src/iax/iaxvoiplink.cpp
+++ b/daemon/src/iax/iaxvoiplink.cpp
@@ -46,7 +46,7 @@
#include "array_size.h"
#include "map_utils.h"
#include "call_factory.h"
-#include "sfl_types.h"
+#include "ring_types.h"
using namespace ring;
diff --git a/daemon/src/iax/iaxvoiplink.h b/daemon/src/iax/iaxvoiplink.h
index 625a232fbd..152fd38a5c 100644
--- a/daemon/src/iax/iaxvoiplink.h
+++ b/daemon/src/iax/iaxvoiplink.h
@@ -39,7 +39,7 @@
#include "audio/audiobuffer.h"
#include "audio/codecs/audiocodec.h" // for RAW_BUFFER_SIZE
-#include "sfl_types.h"
+#include "ring_types.h"
#include <iax/iax-client.h>
diff --git a/daemon/src/media/audio/alsa/alsalayer.cpp b/daemon/src/media/audio/alsa/alsalayer.cpp
index 4c38bd0bcd..de89cfbf9c 100644
--- a/daemon/src/media/audio/alsa/alsalayer.cpp
+++ b/daemon/src/media/audio/alsa/alsalayer.cpp
@@ -412,7 +412,7 @@ bool AlsaLayer::alsa_set_params(snd_pcm_t *pcm_handle)
// TODO first frame causes broken pipe (underrun) because not enough data is sent
// we should wait until the handle is ready
void
-AlsaLayer::write(SFLAudioSample* buffer, int frames, snd_pcm_t * handle)
+AlsaLayer::write(ring::AudioSample* buffer, int frames, snd_pcm_t * handle)
{
// Skip empty buffers
if (!frames)
@@ -473,7 +473,7 @@ AlsaLayer::write(SFLAudioSample* buffer, int frames, snd_pcm_t * handle)
}
int
-AlsaLayer::read(SFLAudioSample* buffer, int frames)
+AlsaLayer::read(ring::AudioSample* buffer, int frames)
{
if (snd_pcm_state(captureHandle_) == SND_PCM_STATE_XRUN) {
prepareCaptureStream();
diff --git a/daemon/src/media/audio/alsa/alsalayer.h b/daemon/src/media/audio/alsa/alsalayer.h
index 0f060f36ff..b4fc42d11e 100644
--- a/daemon/src/media/audio/alsa/alsalayer.h
+++ b/daemon/src/media/audio/alsa/alsalayer.h
@@ -205,7 +205,7 @@ class AlsaLayer : public AudioLayer {
* @param buffer The non-interleaved data to be copied
* @param frames Frames in the buffer
*/
- void write(SFLAudioSample* buffer, int frames, snd_pcm_t *handle);
+ void write(ring::AudioSample* buffer, int frames, snd_pcm_t *handle);
/**
* Read data from the internal ring buffer
@@ -214,7 +214,7 @@ class AlsaLayer : public AudioLayer {
* @param frames The number of frames to get
* @return int The number of frames actually read
*/
- int read(SFLAudioSample* buffer, int frames);
+ int read(ring::AudioSample* buffer, int frames);
virtual void updatePreference(AudioPreference &pref, int index, DeviceType type);
@@ -249,8 +249,8 @@ class AlsaLayer : public AudioLayer {
AudioBuffer captureBuff_;
/** Interleaved buffer */
- std::vector<SFLAudioSample> playbackIBuff_;
- std::vector<SFLAudioSample> captureIBuff_;
+ std::vector<ring::AudioSample> playbackIBuff_;
+ std::vector<ring::AudioSample> captureIBuff_;
bool is_playback_prepared_;
bool is_capture_prepared_;
diff --git a/daemon/src/media/audio/audiobuffer.cpp b/daemon/src/media/audio/audiobuffer.cpp
index a0cd6799d6..40d11e1087 100644
--- a/daemon/src/media/audio/audiobuffer.cpp
+++ b/daemon/src/media/audio/audiobuffer.cpp
@@ -41,13 +41,13 @@ std::ostream& operator <<(std::ostream& stream, const AudioFormat& f) {
AudioBuffer::AudioBuffer(size_t sample_num, AudioFormat format)
: sampleRate_(format.sample_rate),
samples_(std::max(1U, format.nb_channels),
- std::vector<SFLAudioSample>(sample_num, 0))
+ std::vector<ring::AudioSample>(sample_num, 0))
{
}
-AudioBuffer::AudioBuffer(const SFLAudioSample* in, size_t sample_num, AudioFormat format)
+AudioBuffer::AudioBuffer(const ring::AudioSample* in, size_t sample_num, AudioFormat format)
: sampleRate_(format.sample_rate),
- samples_((std::max(1U, format.nb_channels)), std::vector<SFLAudioSample>(sample_num, 0))
+ samples_((std::max(1U, format.nb_channels)), std::vector<ring::AudioSample>(sample_num, 0))
{
deinterleave(in, sample_num, format.nb_channels);
}
@@ -55,7 +55,7 @@ AudioBuffer::AudioBuffer(const SFLAudioSample* in, size_t sample_num, AudioForma
AudioBuffer::AudioBuffer(const AudioBuffer& other, bool copy_content /* = false */)
: sampleRate_(other.sampleRate_),
samples_(copy_content ? other.samples_ :
- std::vector<std::vector<SFLAudioSample> >(other.samples_.size(), std::vector<SFLAudioSample>(other.frames())))
+ std::vector<std::vector<ring::AudioSample> >(other.samples_.size(), std::vector<ring::AudioSample>(other.frames())))
{}
AudioBuffer& AudioBuffer::operator=(const AudioBuffer& other) {
@@ -92,14 +92,14 @@ void AudioBuffer::setChannelNum(unsigned n, bool mix /* = false */)
if (n < c)
samples_.resize(n);
else
- samples_.resize(n, std::vector<SFLAudioSample>(frames(), 0));
+ samples_.resize(n, std::vector<ring::AudioSample>(frames(), 0));
return;
}
// 2ch->1ch
if (n == 1) {
- std::vector<SFLAudioSample>& chan1 = samples_[0];
- std::vector<SFLAudioSample>& chan2 = samples_[1];
+ std::vector<ring::AudioSample>& chan1 = samples_[0];
+ std::vector<ring::AudioSample>& chan2 = samples_[1];
for (unsigned i = 0, f = frames(); i < f; i++)
chan1[i] = chan1[i] / 2 + chan2[i] / 2;
samples_.resize(1);
@@ -132,7 +132,7 @@ void AudioBuffer::resize(size_t sample_num)
s.resize(sample_num, 0);
}
-std::vector<SFLAudioSample> * AudioBuffer::getChannel(unsigned chan /* = 0 */)
+std::vector<ring::AudioSample> * AudioBuffer::getChannel(unsigned chan /* = 0 */)
{
if (chan < samples_.size())
return &samples_[chan];
@@ -163,7 +163,7 @@ size_t AudioBuffer::channelToFloat(float* out, const int& channel) const
return frames() * samples_.size();
}
-size_t AudioBuffer::interleave(SFLAudioSample* out) const
+size_t AudioBuffer::interleave(ring::AudioSample* out) const
{
for (unsigned i=0, f=frames(), c=channels(); i < f; ++i)
for (unsigned j = 0; j < c; ++j)
@@ -172,15 +172,15 @@ size_t AudioBuffer::interleave(SFLAudioSample* out) const
return frames() * channels();
}
-size_t AudioBuffer::interleave(std::vector<SFLAudioSample>& out) const
+size_t AudioBuffer::interleave(std::vector<ring::AudioSample>& out) const
{
out.resize(capacity());
return interleave(out.data());
}
-std::vector<SFLAudioSample> AudioBuffer::interleave() const
+std::vector<ring::AudioSample> AudioBuffer::interleave() const
{
- std::vector<SFLAudioSample> data(capacity());
+ std::vector<ring::AudioSample> data(capacity());
interleave(data.data());
return data;
}
@@ -194,7 +194,7 @@ size_t AudioBuffer::interleaveFloat(float* out) const
return frames() * samples_.size();
}
-void AudioBuffer::deinterleave(const SFLAudioSample* in, size_t frame_num, unsigned nb_channels)
+void AudioBuffer::deinterleave(const ring::AudioSample* in, size_t frame_num, unsigned nb_channels)
{
if (in == nullptr)
return;
@@ -208,7 +208,7 @@ void AudioBuffer::deinterleave(const SFLAudioSample* in, size_t frame_num, unsig
samples_[j][i] = *in++;
}
-void AudioBuffer::deinterleave(const std::vector<SFLAudioSample>& in, AudioFormat format)
+void AudioBuffer::deinterleave(const std::vector<ring::AudioSample>& in, AudioFormat format)
{
sampleRate_ = format.sample_rate;
deinterleave(in.data(), in.size()/format.nb_channels, format.nb_channels);
@@ -255,7 +255,7 @@ size_t AudioBuffer::copy(AudioBuffer& in, int sample_num /* = -1 */, size_t pos_
return to_copy;
}
-size_t AudioBuffer::copy(SFLAudioSample* in, size_t sample_num, size_t pos_out /* = 0 */)
+size_t AudioBuffer::copy(ring::AudioSample* in, size_t sample_num, size_t pos_out /* = 0 */)
{
if (in == nullptr || sample_num == 0) return 0;
diff --git a/daemon/src/media/audio/audiobuffer.h b/daemon/src/media/audio/audiobuffer.h
index 4ec1a4ce05..3d22eb9e47 100644
--- a/daemon/src/media/audio/audiobuffer.h
+++ b/daemon/src/media/audio/audiobuffer.h
@@ -36,14 +36,14 @@
#include <sstream>
#include <cstddef> // for size_t
-#include "sfl_types.h"
+#include "ring_types.h"
namespace ring {
/**
* Structure to hold sample rate and channel number associated with audio data.
*/
-typedef struct AudioFormat {
+struct AudioFormat {
unsigned sample_rate;
unsigned nb_channels;
@@ -67,7 +67,7 @@ typedef struct AudioFormat {
* Returns bytes necessary to hold one frame of audio data.
*/
inline size_t getBytesPerFrame() const {
- return sizeof(SFLAudioSample)*nb_channels;
+ return sizeof(ring::AudioSample)*nb_channels;
}
/**
@@ -82,7 +82,7 @@ typedef struct AudioFormat {
static const AudioFormat MONO() { return AudioFormat{DEFAULT_SAMPLE_RATE, 1}; }
static const AudioFormat STEREO() { return AudioFormat{DEFAULT_SAMPLE_RATE, 2}; }
-} AudioFormat;
+};
std::ostream& operator <<(std::ostream& stream, const AudioFormat& f);
@@ -96,7 +96,7 @@ class AudioBuffer {
/**
* Construtor from existing interleaved data (copied into the buffer).
*/
- AudioBuffer(const SFLAudioSample* in, size_t sample_num, AudioFormat format);
+ AudioBuffer(const ring::AudioSample* in, size_t sample_num, AudioFormat format);
/**
* Copy constructor that by default only copies the buffer parameters (channel number, sample rate and buffer size).
@@ -129,7 +129,7 @@ class AudioBuffer {
}
inline size_t size() const {
- return frames() * channels() * sizeof(SFLAudioSample);
+ return frames() * channels() * sizeof(ring::AudioSample);
}
/**
@@ -217,12 +217,12 @@ class AudioBuffer {
* Return the data (audio samples) for a given channel number.
* Channel data can be modified but size of individual channel vectors should not be changed by the user.
*/
- std::vector<SFLAudioSample> *getChannel(unsigned chan);
+ std::vector<ring::AudioSample> *getChannel(unsigned chan);
/**
* Return a pointer to the raw data in this buffer.
*/
- inline std::vector<std::vector<SFLAudioSample> > &getData() {
+ inline std::vector<std::vector<ring::AudioSample> > &getData() {
return samples_;
}
@@ -231,9 +231,9 @@ class AudioBuffer {
* Caller should not store result because pointer validity is
* limited in time.
*/
- inline const std::vector<SFLAudioSample*> getDataRaw() {
+ inline const std::vector<ring::AudioSample*> getDataRaw() {
const unsigned chans = samples_.size();
- std::vector<SFLAudioSample*> raw_data(chans, nullptr);
+ std::vector<ring::AudioSample*> raw_data(chans, nullptr);
for(unsigned i=0; i<chans; i++)
raw_data[i] = samples_[i].data();
return raw_data;
@@ -241,7 +241,7 @@ class AudioBuffer {
/**
* Convert fixed-point channel to float and write in the out buffer (Float 32-bits).
- * The out buffer must be at least of size capacity()*sizeof(SFLAudioSample) bytes.
+ * The out buffer must be at least of size capacity()*sizeof(float) bytes.
*
* @returns Number of samples writen.
*/
@@ -249,11 +249,11 @@ class AudioBuffer {
/**
* Write interleaved multichannel data to the out buffer (fixed-point 16-bits).
- * The out buffer must be at least of size capacity()*sizeof(SFLAudioSample) bytes.
+ * The out buffer must be at least of size capacity()*sizeof(ring::AudioSample) bytes.
*
* @returns Number of samples writen.
*/
- size_t interleave(SFLAudioSample* out) const;
+ size_t interleave(ring::AudioSample* out) const;
/**
* Write interleaved multichannel data to the out buffer (fixed-point 16-bits).
@@ -261,12 +261,12 @@ class AudioBuffer {
*
* @returns Number of samples writen.
*/
- size_t interleave(std::vector<SFLAudioSample>& out) const;
+ size_t interleave(std::vector<ring::AudioSample>& out) const;
/**
* Returns vector of interleaved data (fixed-point 16-bits).
*/
- std::vector<SFLAudioSample> interleave() const;
+ std::vector<ring::AudioSample> interleave() const;
/**
* Write interleaved multichannel data to the out buffer, while samples are converted to float.
@@ -280,13 +280,13 @@ class AudioBuffer {
* Import interleaved multichannel data. Internal buffer is resized as needed.
* Function will read sample_num*channel_num elements of the in buffer.
*/
- void deinterleave(const SFLAudioSample* in, size_t frame_num, unsigned nb_channels = 1);
+ void deinterleave(const ring::AudioSample* in, size_t frame_num, unsigned nb_channels = 1);
/**
* Import interleaved multichannel data. Internal buffer is resized as needed.
* Sample rate is set according to format.
*/
- void deinterleave(const std::vector<SFLAudioSample>& in, AudioFormat format);
+ void deinterleave(const std::vector<ring::AudioSample>& in, AudioFormat format);
/**
* In-place gain transformation.
@@ -325,13 +325,13 @@ class AudioBuffer {
*
* Buffer sample number is increased if required to hold the new requested samples.
*/
- size_t copy(SFLAudioSample* in, size_t sample_num, size_t pos_out = 0);
+ size_t copy(ring::AudioSample* in, size_t sample_num, size_t pos_out = 0);
private:
int sampleRate_;
// buffers holding data for each channels
- std::vector<std::vector<SFLAudioSample> > samples_;
+ std::vector<std::vector<ring::AudioSample> > samples_;
};
}
diff --git a/daemon/src/media/audio/audioloop.cpp b/daemon/src/media/audio/audioloop.cpp
index 33688ee71c..bbf6ff1f11 100644
--- a/daemon/src/media/audio/audioloop.cpp
+++ b/daemon/src/media/audio/audioloop.cpp
@@ -37,13 +37,10 @@
#endif
#include "audioloop.h"
-
-#include <cmath>
-#include <numeric>
-#include <cstring>
-#include <cassert>
#include "logger.h"
+#include <algorithm> // std::min
+
namespace ring {
AudioLoop::AudioLoop(unsigned int sampleRate) :
diff --git a/daemon/src/media/audio/audioloop.h b/daemon/src/media/audio/audioloop.h
index e257a86658..e97c10ba1b 100644
--- a/daemon/src/media/audio/audioloop.h
+++ b/daemon/src/media/audio/audioloop.h
@@ -33,8 +33,7 @@
#ifndef __AUDIOLOOP_H__
#define __AUDIOLOOP_H__
-#include "sfl_types.h"
-#include <cstring>
+#include "ring_types.h"
#include "noncopyable.h"
#include "audiobuffer.h"
diff --git a/daemon/src/media/audio/audiorecord.h b/daemon/src/media/audio/audiorecord.h
index b357313a73..4c08bebbea 100644
--- a/daemon/src/media/audio/audiorecord.h
+++ b/daemon/src/media/audio/audiorecord.h
@@ -32,7 +32,6 @@
#define _AUDIO_RECORD_H
#include "audiobuffer.h"
-#include "sfl_types.h"
#include "noncopyable.h"
#include <memory>
diff --git a/daemon/src/media/audio/codecs/alaw.cpp b/daemon/src/media/audio/codecs/alaw.cpp
index eed361f66f..5fe627791d 100644
--- a/daemon/src/media/audio/codecs/alaw.cpp
+++ b/daemon/src/media/audio/codecs/alaw.cpp
@@ -29,7 +29,7 @@
* as that of the covered work.
*/
-#include "sfl_types.h"
+#include "ring_types.h"
#include "audiocodec.h"
#include "ring_plugin.h"
#include "g711.h"
@@ -50,7 +50,7 @@ class Alaw : public ring::AudioCodec {
return new Alaw;
}
- int decode(SFLAudioSample *pcm, unsigned char *data, size_t len)
+ int decode(ring::AudioSample *pcm, unsigned char *data, size_t len)
{
for (unsigned char* end = data + len; data < end; ++data, ++pcm)
*pcm = ALawDecode(*data);
@@ -58,7 +58,7 @@ class Alaw : public ring::AudioCodec {
return len;
}
- int encode(unsigned char *data, SFLAudioSample *pcm, size_t max_data_bytes)
+ int encode(unsigned char *data, ring::AudioSample *pcm, size_t max_data_bytes)
{
unsigned char *end = std::min(data + frameSize_, data + max_data_bytes);
unsigned char *tmp = data;
@@ -69,11 +69,11 @@ class Alaw : public ring::AudioCodec {
return end - data;
}
- static SFLAudioSample ALawDecode(unsigned char alaw) {
+ static ring::AudioSample ALawDecode(unsigned char alaw) {
return alaw_to_linear(alaw);
}
- static unsigned char ALawEncode(SFLAudioSample pcm16) {
+ static unsigned char ALawEncode(ring::AudioSample pcm16) {
return linear_to_alaw(pcm16);
}
};
diff --git a/daemon/src/media/audio/codecs/audiocodec.cpp b/daemon/src/media/audio/codecs/audiocodec.cpp
index 5b8093ba3a..f8de110ff3 100644
--- a/daemon/src/media/audio/codecs/audiocodec.cpp
+++ b/daemon/src/media/audio/codecs/audiocodec.cpp
@@ -60,14 +60,14 @@ AudioCodec::AudioCodec(const AudioCodec& c) :
hasDynamicPayload_(c.hasDynamicPayload_)
{}
-int AudioCodec::decode(SFLAudioSample *, unsigned char *, size_t)
+int AudioCodec::decode(ring::AudioSample *, unsigned char *, size_t)
{
// Unimplemented!
assert(false);
return 0;
}
-int AudioCodec::encode(unsigned char *, SFLAudioSample *, size_t)
+int AudioCodec::encode(unsigned char *, ring::AudioSample *, size_t)
{
// Unimplemented!
assert(false);
@@ -76,18 +76,18 @@ int AudioCodec::encode(unsigned char *, SFLAudioSample *, size_t)
// Mono only, subclasses must implement multichannel support
-int AudioCodec::decode(std::vector<std::vector<SFLAudioSample> > &pcm, const uint8_t* data, size_t len)
+int AudioCodec::decode(std::vector<std::vector<ring::AudioSample> > &pcm, const uint8_t* data, size_t len)
{
return decode(pcm[0].data(), const_cast<uint8_t*>(data), len);
}
// Mono only, subclasses must implement multichannel support
-size_t AudioCodec::encode(const std::vector<std::vector<SFLAudioSample> > &pcm, uint8_t *data, size_t len)
+size_t AudioCodec::encode(const std::vector<std::vector<ring::AudioSample> > &pcm, uint8_t *data, size_t len)
{
- return encode(data, const_cast<SFLAudioSample*>(pcm[0].data()), len);
+ return encode(data, const_cast<ring::AudioSample*>(pcm[0].data()), len);
}
-int AudioCodec::decode(std::vector<std::vector<SFLAudioSample> > &pcm)
+int AudioCodec::decode(std::vector<std::vector<ring::AudioSample> > &pcm)
{
pcm.clear();
return frameSize_;
diff --git a/daemon/src/media/audio/codecs/audiocodec.h b/daemon/src/media/audio/codecs/audiocodec.h
index b55d626fd6..468aa206cf 100644
--- a/daemon/src/media/audio/codecs/audiocodec.h
+++ b/daemon/src/media/audio/codecs/audiocodec.h
@@ -32,7 +32,7 @@
#ifndef __AUDIO_CODEC_H__
#define __AUDIO_CODEC_H__
-#include "sfl_types.h"
+#include "ring_types.h"
#include <string>
#include <vector>
@@ -63,20 +63,20 @@ class AudioCodec {
* Multichannel version of decode().
* Default implementation calls mono version
*/
- virtual int decode(std::vector<std::vector<SFLAudioSample> > &pcm, const uint8_t* data, size_t len);
+ virtual int decode(std::vector<std::vector<ring::AudioSample> > &pcm, const uint8_t* data, size_t len);
/**
* Inform the codec of a lost packet and perform packet loss concealment.
* Default implementation fills dst with 0.
*/
- virtual int decode(std::vector<std::vector<SFLAudioSample> > &pcm);
+ virtual int decode(std::vector<std::vector<ring::AudioSample> > &pcm);
/**
* Multichannel version of encode().
* Default implementation calls encode() on the first channel (assume 1 channel).
* @return the number of bytes encoded
*/
- virtual size_t encode(const std::vector<std::vector<SFLAudioSample> > &pcm, uint8_t *data, size_t len);
+ virtual size_t encode(const std::vector<std::vector<ring::AudioSample> > &pcm, uint8_t *data, size_t len);
uint8_t getPayloadType() const;
@@ -157,7 +157,7 @@ class AudioCodec {
* @param len: length of input buffer
* @return the number of samples decoded
*/
- virtual int decode(SFLAudioSample *pcm, unsigned char *data, size_t len);
+ virtual int decode(ring::AudioSample *pcm, unsigned char *data, size_t len);
/**
* Encode an input buffer and fill the output buffer with the encoded data
@@ -166,7 +166,7 @@ class AudioCodec {
* @param max_data_bytes: the maximum size of the encoded data buffer (data)
* @return the number of bytes encoded
*/
- virtual int encode(unsigned char *data, SFLAudioSample *pcm, size_t max_data_bytes);
+ virtual int encode(unsigned char *data, ring::AudioSample *pcm, size_t max_data_bytes);
/** Holds SDP-compliant codec name */
std::string codecName_; // what we put inside sdp
diff --git a/daemon/src/media/audio/codecs/g722.cpp b/daemon/src/media/audio/codecs/g722.cpp
index 416127e2ed..e3c5597d95 100644
--- a/daemon/src/media/audio/codecs/g722.cpp
+++ b/daemon/src/media/audio/codecs/g722.cpp
@@ -32,7 +32,7 @@
*/
#include "audiocodec.h"
-#include "sfl_types.h"
+#include "ring_types.h"
#include "ring_plugin.h"
#include "g722.h"
@@ -59,12 +59,12 @@ class G722 : public ring::AudioCodec {
return new G722;
}
- int decode(SFLAudioSample *pcm, unsigned char *data, size_t len)
+ int decode(ring::AudioSample *pcm, unsigned char *data, size_t len)
{
return g722_decode(pcm, data, len);
}
- int encode(unsigned char *data, SFLAudioSample *pcm, size_t max_data_bytes)
+ int encode(unsigned char *data, ring::AudioSample *pcm, size_t max_data_bytes)
{
int out = g722_encode(data, pcm, std::min<size_t>(frameSize_, max_data_bytes));
return out;
@@ -94,12 +94,12 @@ class G722 : public ring::AudioCodec {
state.out_bits = 0;
}
- SFLAudioSample saturate(int32_t amp)
+ ring::AudioSample saturate(int32_t amp)
{
- SFLAudioSample amp16 = 0;
+ ring::AudioSample amp16 = 0;
/* Hopefully this is optimised for the common case - not clipping */
- amp16 = (SFLAudioSample) amp;
+ amp16 = (ring::AudioSample) amp;
if (amp == amp16)
return amp16;
@@ -319,7 +319,7 @@ class G722 : public ring::AudioCodec {
decode_state_.band[band].s = saturate(decode_state_.band[band].sp + decode_state_.band[band].sz);
}
- int g722_decode(SFLAudioSample amp[], const uint8_t g722_data[], int len)
+ int g722_decode(ring::AudioSample amp[], const uint8_t g722_data[], int len)
{
static const int wl[8] = {-60, -30, 58, 172, 334, 538, 1198, 3042 };
static const int rl42[16] = {0, 7, 6, 5, 4, 3, 2, 1, 7, 6, 5, 4, 3, 2, 1, 0 };
@@ -514,11 +514,11 @@ class G722 : public ring::AudioCodec {
}
if (decode_state_.itu_test_mode) {
- amp[outlen++] = (SFLAudioSample)(rlow << 1);
- amp[outlen++] = (SFLAudioSample)(rhigh << 1);
+ amp[outlen++] = (ring::AudioSample)(rlow << 1);
+ amp[outlen++] = (ring::AudioSample)(rhigh << 1);
} else {
if (decode_state_.eight_k) {
- amp[outlen++] = (SFLAudioSample) (rlow << 1);
+ amp[outlen++] = (ring::AudioSample) (rlow << 1);
} else {
/* Apply the receive QMF */
for (i = 0; i < 22; i++)
@@ -537,9 +537,9 @@ class G722 : public ring::AudioCodec {
xout1 += decode_state_.x[2*i + 1]*qmf_coeffs[11 - i];
}
- amp[outlen++] = (SFLAudioSample)(xout1 >> 11);
+ amp[outlen++] = (ring::AudioSample)(xout1 >> 11);
- amp[outlen++] = (SFLAudioSample)(xout2 >> 11);
+ amp[outlen++] = (ring::AudioSample)(xout2 >> 11);
}
}
}
@@ -547,7 +547,7 @@ class G722 : public ring::AudioCodec {
return outlen;
}
- int g722_encode(uint8_t g722_data[], const SFLAudioSample amp[], int len)
+ int g722_encode(uint8_t g722_data[], const ring::AudioSample amp[], int len)
{
static const int q6[32] = {
0, 35, 72, 110, 150, 190, 233, 276,
diff --git a/daemon/src/media/audio/codecs/g729.cpp b/daemon/src/media/audio/codecs/g729.cpp
index b1efbc05d0..0ed33fe3d6 100644
--- a/daemon/src/media/audio/codecs/g729.cpp
+++ b/daemon/src/media/audio/codecs/g729.cpp
@@ -28,15 +28,15 @@
* as that of the covered work.
*/
#include "g729.h"
-#include "sfl_types.h"
+#include "ring_types.h"
#include "ring_plugin.h"
#include <iostream>
#include <dlfcn.h>
#include <stdexcept>
-#define G729_TYPE_ENCODER (void (*)(bcg729EncoderChannelContextStruct*, SFLAudioSample[], uint8_t[]))
-#define G729_TYPE_DECODER (void (*)(bcg729DecoderChannelContextStruct*, uint8_t[], uint8_t, SFLAudioSample[]))
+#define G729_TYPE_ENCODER (void (*)(bcg729EncoderChannelContextStruct*, ring::AudioSample[], uint8_t[]))
+#define G729_TYPE_DECODER (void (*)(bcg729DecoderChannelContextStruct*, uint8_t[], uint8_t, ring::AudioSample[]))
#define G729_TYPE_DECODER_INIT (bcg729DecoderChannelContextStruct*(*)())
#define G729_TYPE_ENCODER_INIT (bcg729EncoderChannelContextStruct*(*)())
@@ -81,14 +81,14 @@ G729::clone()
return new G729;
}
-int G729::decode(SFLAudioSample *pcm, unsigned char *data, size_t len)
+int G729::decode(ring::AudioSample *pcm, unsigned char *data, size_t len)
{
decoder_(decoderContext_, data, false, pcm);
decoder_(decoderContext_, data + (len / 2), false, pcm + 80);
return 160;
}
-int G729::encode(unsigned char *data, SFLAudioSample *pcm, size_t)
+int G729::encode(unsigned char *data, ring::AudioSample *pcm, size_t)
{
encoder_(encoderContext_, pcm, data);
encoder_(encoderContext_, pcm + (frameSize_ / 2), data + 10);
diff --git a/daemon/src/media/audio/codecs/g729.h b/daemon/src/media/audio/codecs/g729.h
index 86b7a4075b..23042b7e6f 100644
--- a/daemon/src/media/audio/codecs/g729.h
+++ b/daemon/src/media/audio/codecs/g729.h
@@ -31,7 +31,7 @@
#define G729_H_
#include <cstdlib>
-#include "sfl_types.h"
+#include "ring_types.h"
#include "noncopyable.h"
#include "audiocodec.h"
@@ -45,8 +45,8 @@ public:
~G729();
private:
AudioCodec * clone();
- virtual int decode(SFLAudioSample *pcm, unsigned char *data, size_t len);
- virtual int encode(unsigned char *data, SFLAudioSample *pcm, size_t max_data_bytes);
+ virtual int decode(ring::AudioSample *pcm, unsigned char *data, size_t len);
+ virtual int encode(unsigned char *data, ring::AudioSample *pcm, size_t max_data_bytes);
NON_COPYABLE(G729);
//Attributes
@@ -55,8 +55,8 @@ private:
void* handler_;
//Extern functions
- void (*encoder_) (bcg729EncoderChannelContextStruct *encoderChannelContext, SFLAudioSample inputFrame[], uint8_t bitStream[]);
- void (*decoder_) (bcg729DecoderChannelContextStruct *decoderChannelContext, uint8_t bitStream[], uint8_t frameErasureFlag, SFLAudioSample signal[]);
+ void (*encoder_) (bcg729EncoderChannelContextStruct *encoderChannelContext, ring::AudioSample inputFrame[], uint8_t bitStream[]);
+ void (*decoder_) (bcg729DecoderChannelContextStruct *decoderChannelContext, uint8_t bitStream[], uint8_t frameErasureFlag, ring::AudioSample signal[]);
static void loadError(const char *error);
};
diff --git a/daemon/src/media/audio/codecs/gsmcodec.cpp b/daemon/src/media/audio/codecs/gsmcodec.cpp
index 4c834a0705..67c99e87ca 100644
--- a/daemon/src/media/audio/codecs/gsmcodec.cpp
+++ b/daemon/src/media/audio/codecs/gsmcodec.cpp
@@ -31,7 +31,7 @@
#include "audiocodec.h"
-#include "sfl_types.h"
+#include "ring_types.h"
#include "noncopyable.h"
#include "ring_plugin.h"
@@ -73,7 +73,7 @@ private:
return new Gsm;
}
- int decode(SFLAudioSample *pcm, unsigned char *data, size_t)
+ int decode(ring::AudioSample *pcm, unsigned char *data, size_t)
{
if (gsm_decode(decode_gsmhandle_, (gsm_byte*) data, (gsm_signal*) pcm) < 0)
throw std::runtime_error("Error in gsm_decode\n");
@@ -81,7 +81,7 @@ private:
return frameSize_;
}
- int encode(unsigned char *data, SFLAudioSample *pcm, size_t)
+ int encode(unsigned char *data, ring::AudioSample *pcm, size_t)
{
gsm_encode(encode_gsmhandle_, (gsm_signal*) pcm, (gsm_byte*) data);
return sizeof(gsm_frame);
diff --git a/daemon/src/media/audio/codecs/ilbc.cpp b/daemon/src/media/audio/codecs/ilbc.cpp
index 8bb64b049d..3f98a38e64 100644
--- a/daemon/src/media/audio/codecs/ilbc.cpp
+++ b/daemon/src/media/audio/codecs/ilbc.cpp
@@ -29,7 +29,7 @@
*/
#include "audiocodec.h"
-#include "sfl_types.h"
+#include "ring_types.h"
#include <algorithm>
extern "C" {
@@ -56,13 +56,13 @@ class Ilbc: public ring::AudioCodec {
}
// iLBC expects floating point data, so we have to convert
- int decode(SFLAudioSample *pcm, unsigned char *data, size_t) {
+ int decode(ring::AudioSample *pcm, unsigned char *data, size_t) {
const int NORMAL_MODE = 1;
iLBC_decode(pcm, reinterpret_cast<WebRtc_UWord16*>(data), &ilbc_dec_, NORMAL_MODE);
return frameSize_;
}
- int encode(unsigned char *data, SFLAudioSample *pcm, size_t) {
+ int encode(unsigned char *data, ring::AudioSample *pcm, size_t) {
iLBC_encode(reinterpret_cast<WebRtc_UWord16*>(data), pcm, &ilbc_enc_);
return frameSize_;
}
diff --git a/daemon/src/media/audio/codecs/opuscodec.cpp b/daemon/src/media/audio/codecs/opuscodec.cpp
index 0dfe3c9d88..4c32f85e8d 100644
--- a/daemon/src/media/audio/codecs/opuscodec.cpp
+++ b/daemon/src/media/audio/codecs/opuscodec.cpp
@@ -29,7 +29,7 @@
* as that of the covered work.
*/
#include "opuscodec.h"
-#include "sfl_types.h"
+#include "ring_types.h"
#include "ring_plugin.h"
#include <stdexcept>
@@ -108,7 +108,7 @@ Opus::getSDPChannels() const
return "2";
}
-int Opus::decode(std::vector<std::vector<SFLAudioSample> > &pcm, const uint8_t *data, size_t len)
+int Opus::decode(std::vector<std::vector<ring::AudioSample> > &pcm, const uint8_t *data, size_t len)
{
if (data == nullptr) return 0;
@@ -116,7 +116,7 @@ int Opus::decode(std::vector<std::vector<SFLAudioSample> > &pcm, const uint8_t *
if (channelsCur_ == 1) {
ret = opus_decode(decoder_, data, len, pcm[0].data(), MAX_PACKET_SIZE, 0);
} else {
- std::array<SFLAudioSample, 2 * MAX_PACKET_SIZE> ibuf; // deinterleave on stack, 11.25KiB used.
+ std::array<ring::AudioSample, 2 * MAX_PACKET_SIZE> ibuf; // deinterleave on stack, 11.25KiB used.
ret = opus_decode(decoder_, data, len, ibuf.data(), MAX_PACKET_SIZE, 0);
for (int i = 0; i < ret; i++) {
pcm[0][i] = ibuf[2 * i];
@@ -129,14 +129,14 @@ int Opus::decode(std::vector<std::vector<SFLAudioSample> > &pcm, const uint8_t *
return ret;
}
-int Opus::decode(std::vector<std::vector<SFLAudioSample> > &pcm)
+int Opus::decode(std::vector<std::vector<ring::AudioSample> > &pcm)
{
if (!lastDecodedFrameSize_) return 0;
int ret;
if (channelsCur_ == 1) {
ret = opus_decode(decoder_, nullptr, 0, pcm[0].data(), lastDecodedFrameSize_, 0);
} else {
- std::array<SFLAudioSample, 2 * MAX_PACKET_SIZE> ibuf; // deinterleave on stack, 11.25KiB used.
+ std::array<ring::AudioSample, 2 * MAX_PACKET_SIZE> ibuf; // deinterleave on stack, 11.25KiB used.
ret = opus_decode(decoder_, nullptr, 0, ibuf.data(), lastDecodedFrameSize_, 0);
for (int i = 0; i < ret; i++) {
pcm[0][i] = ibuf[2 * i];
@@ -148,14 +148,14 @@ int Opus::decode(std::vector<std::vector<SFLAudioSample> > &pcm)
return ret;
}
-size_t Opus::encode(const std::vector<std::vector<SFLAudioSample> > &pcm, uint8_t *data, size_t len)
+size_t Opus::encode(const std::vector<std::vector<ring::AudioSample> > &pcm, uint8_t *data, size_t len)
{
if (data == nullptr) return 0;
int ret;
if (channelsCur_ == 1) {
ret = opus_encode(encoder_, pcm[0].data(), FRAME_SIZE, data, len);
} else {
- std::array<SFLAudioSample, 2 * FRAME_SIZE> ibuf; // interleave on stack, 1.875KiB used;
+ std::array<ring::AudioSample, 2 * FRAME_SIZE> ibuf; // interleave on stack, 1.875KiB used;
for (unsigned i = 0; i < FRAME_SIZE; i++) {
ibuf[2 * i] = pcm[0][i];
ibuf[2 * i + 1] = pcm[1][i];
diff --git a/daemon/src/media/audio/codecs/opuscodec.h b/daemon/src/media/audio/codecs/opuscodec.h
index 7543d73da8..d2e667c5d2 100644
--- a/daemon/src/media/audio/codecs/opuscodec.h
+++ b/daemon/src/media/audio/codecs/opuscodec.h
@@ -34,7 +34,7 @@
#include "audiocodec.h"
#include "noncopyable.h"
-#include "sfl_types.h"
+#include "ring_types.h"
#include <opus.h>
@@ -52,10 +52,10 @@ public:
private:
ring::AudioCodec * clone();
- virtual int decode(std::vector<std::vector<SFLAudioSample> > &pcm, const uint8_t *data, size_t len);
- virtual int decode(std::vector<std::vector<SFLAudioSample> > &pcm);
+ virtual int decode(std::vector<std::vector<ring::AudioSample> > &pcm, const uint8_t *data, size_t len);
+ virtual int decode(std::vector<std::vector<ring::AudioSample> > &pcm);
- virtual size_t encode(const std::vector<std::vector<SFLAudioSample> > &pcm, uint8_t *data, size_t len);
+ virtual size_t encode(const std::vector<std::vector<ring::AudioSample> > &pcm, uint8_t *data, size_t len);
virtual uint32_t getSDPClockRate() const;
virtual const char *getSDPChannels() const;
diff --git a/daemon/src/media/audio/codecs/speexcodec.h b/daemon/src/media/audio/codecs/speexcodec.h
index 37235c3f59..3b70170ea4 100644
--- a/daemon/src/media/audio/codecs/speexcodec.h
+++ b/daemon/src/media/audio/codecs/speexcodec.h
@@ -29,7 +29,7 @@
* as that of the covered work.
*/
-#include "sfl_types.h"
+#include "ring_types.h"
#include "audiocodec.h"
#include "noncopyable.h"
#include "array_size.h"
@@ -90,13 +90,13 @@ private:
NON_COPYABLE(Speex);
- virtual int decode(SFLAudioSample *pcm, unsigned char *data, size_t len) {
+ virtual int decode(ring::AudioSample *pcm, unsigned char *data, size_t len) {
speex_bits_read_from(&speex_dec_bits_, (char*) data, len);
speex_decode_int(speex_dec_state_, &speex_dec_bits_, pcm);
return frameSize_;
}
- virtual int encode(unsigned char *data, SFLAudioSample *pcm, size_t max_data_bytes) {
+ virtual int encode(unsigned char *data, ring::AudioSample *pcm, size_t max_data_bytes) {
speex_bits_reset(&speex_enc_bits_);
speex_encode_int(speex_enc_state_, pcm, &speex_enc_bits_);
return speex_bits_write(&speex_enc_bits_, (char*) data,
diff --git a/daemon/src/media/audio/codecs/ulaw.cpp b/daemon/src/media/audio/codecs/ulaw.cpp
index 962cd51a86..d840a05de6 100644
--- a/daemon/src/media/audio/codecs/ulaw.cpp
+++ b/daemon/src/media/audio/codecs/ulaw.cpp
@@ -30,7 +30,7 @@
*/
#include "audiocodec.h"
-#include "sfl_types.h"
+#include "ring_types.h"
#include "ring_plugin.h"
#include "g711.h"
@@ -49,7 +49,7 @@ class Ulaw : public ring::AudioCodec {
return new Ulaw;
}
- int decode(SFLAudioSample *pcm, unsigned char *data, size_t len)
+ int decode(ring::AudioSample *pcm, unsigned char *data, size_t len)
{
for (const unsigned char *end = data + len; data < end;
++data, ++pcm)
@@ -58,7 +58,7 @@ class Ulaw : public ring::AudioCodec {
return len;
}
- int encode(unsigned char *data, SFLAudioSample *pcm, size_t max_data_bytes)
+ int encode(unsigned char *data, ring::AudioSample *pcm, size_t max_data_bytes)
{
const unsigned char *end = data +
std::min<size_t>(frameSize_, max_data_bytes);
@@ -70,11 +70,11 @@ class Ulaw : public ring::AudioCodec {
return end - data;
}
- static SFLAudioSample ULawDecode(uint8_t ulaw) {
+ static ring::AudioSample ULawDecode(uint8_t ulaw) {
return ulaw_to_linear(ulaw);
}
- static uint8_t ULawEncode(SFLAudioSample pcm16) {
+ static uint8_t ULawEncode(ring::AudioSample pcm16) {
return linear_to_ulaw(pcm16);
}
};
diff --git a/daemon/src/media/audio/coreaudio/corelayer.cpp b/daemon/src/media/audio/coreaudio/corelayer.cpp
index 47e83b5280..3deb9c8823 100644
--- a/daemon/src/media/audio/coreaudio/corelayer.cpp
+++ b/daemon/src/media/audio/coreaudio/corelayer.cpp
@@ -499,7 +499,7 @@ void CoreLayer::read(AudioUnitRenderActionFlags* ioActionFlags,
for (int i = 0; i < info.mChannelsPerFrame; ++i) {
Float32* data = (Float32*)captureBuff_->mBuffers[i].mData;
for (int j = 0; j < inNumberFrames; ++j) {
- (*inBuff.getChannel(i))[j] = (SFLAudioSample)((data)[j] / .000030517578125f);
+ (*inBuff.getChannel(i))[j] = (ring::AudioSample)((data)[j] / .000030517578125f);
}
}
diff --git a/daemon/src/media/audio/coreaudio/corelayer.h b/daemon/src/media/audio/coreaudio/corelayer.h
index 0b89b4bc79..51ff8be4db 100644
--- a/daemon/src/media/audio/coreaudio/corelayer.h
+++ b/daemon/src/media/audio/coreaudio/corelayer.h
@@ -171,8 +171,8 @@ class CoreLayer : public AudioLayer {
::AudioBufferList* captureBuff_; // CoreAudio buffer.
/** Interleaved buffer */
- std::vector<SFLAudioSample> playbackIBuff_;
- std::vector<SFLAudioSample> captureIBuff_;
+ std::vector<ring::AudioSample> playbackIBuff_;
+ std::vector<ring::AudioSample> captureIBuff_;
AudioUnit outputUnit_;
AudioUnit inputUnit_;
diff --git a/daemon/src/media/audio/dcblocker.cpp b/daemon/src/media/audio/dcblocker.cpp
index 4a86273a8b..24229e08c7 100644
--- a/daemon/src/media/audio/dcblocker.cpp
+++ b/daemon/src/media/audio/dcblocker.cpp
@@ -41,13 +41,13 @@ void DcBlocker::reset()
states.assign(states.size(), (struct StreamState){0, 0, 0, 0});
}
-void DcBlocker::doProcess(SFLAudioSample *out, SFLAudioSample *in, unsigned samples, struct StreamState * state)
+void DcBlocker::doProcess(ring::AudioSample *out, ring::AudioSample *in, unsigned samples, struct StreamState * state)
{
for (unsigned i = 0; i < samples; ++i) {
state->x_ = in[i];
- state->y_ = (SFLAudioSample) ((float) state->x_ - (float) state->xm1_ + 0.9999 * (float) state->y_);
+ state->y_ = (ring::AudioSample) ((float) state->x_ - (float) state->xm1_ + 0.9999 * (float) state->y_);
state->xm1_ = state->x_;
state->ym1_ = state->y_;
@@ -55,7 +55,7 @@ void DcBlocker::doProcess(SFLAudioSample *out, SFLAudioSample *in, unsigned samp
}
}
-void DcBlocker::process(SFLAudioSample *out, SFLAudioSample *in, int samples)
+void DcBlocker::process(ring::AudioSample *out, ring::AudioSample *in, int samples)
{
if (out == NULL or in == NULL or samples == 0) return;
doProcess(out, in, samples, &states[0]);
@@ -70,7 +70,7 @@ void DcBlocker::process(AudioBuffer& buf)
unsigned i;
for(i=0; i<chans; i++) {
- SFLAudioSample *chan = buf.getChannel(i)->data();
+ ring::AudioSample *chan = buf.getChannel(i)->data();
doProcess(chan, chan, samples, &states[i]);
}
}
diff --git a/daemon/src/media/audio/dcblocker.h b/daemon/src/media/audio/dcblocker.h
index 54c39732f4..6cc4255798 100644
--- a/daemon/src/media/audio/dcblocker.h
+++ b/daemon/src/media/audio/dcblocker.h
@@ -32,7 +32,7 @@
#ifndef DCBLOCKER_H
#define DCBLOCKER_H
-#include "sfl_types.h"
+#include "ring_types.h"
#include "audiobuffer.h"
namespace ring {
@@ -42,7 +42,7 @@ class DcBlocker {
DcBlocker(unsigned channels = 1);
void reset();
- void process(SFLAudioSample *out, SFLAudioSample *in, int samples);
+ void process(ring::AudioSample *out, ring::AudioSample *in, int samples);
/**
* In-place processing of all samples in buf (each channel treated independently)
@@ -51,10 +51,10 @@ class DcBlocker {
private:
struct StreamState {
- SFLAudioSample y_, x_, xm1_, ym1_;
+ ring::AudioSample y_, x_, xm1_, ym1_;
};
- void doProcess(SFLAudioSample *out, SFLAudioSample *in, unsigned samples, struct StreamState * state);
+ void doProcess(ring::AudioSample *out, ring::AudioSample *in, unsigned samples, struct StreamState * state);
std::vector<StreamState> states;
};
diff --git a/daemon/src/media/audio/jack/jacklayer.cpp b/daemon/src/media/audio/jack/jacklayer.cpp
index 3e23abbfb3..4b796b7dc9 100644
--- a/daemon/src/media/audio/jack/jacklayer.cpp
+++ b/daemon/src/media/audio/jack/jacklayer.cpp
@@ -173,7 +173,7 @@ JackLayer::capture()
}
static void
-convertToFloat(const std::vector<SFLAudioSample> &src, std::vector<float> &dest)
+convertToFloat(const std::vector<ring::AudioSample> &src, std::vector<float> &dest)
{
static const float INV_SHORT_MAX = 1 / (float) SHRT_MAX;
if (dest.size() != src.size()) {
@@ -185,7 +185,7 @@ convertToFloat(const std::vector<SFLAudioSample> &src, std::vector<float> &dest)
}
static void
-convertFromFloat(std::vector<float> &src, std::vector<SFLAudioSample> &dest)
+convertFromFloat(std::vector<float> &src, std::vector<ring::AudioSample> &dest)
{
if (dest.size() != src.size()) {
RING_ERR("MISMATCH");
diff --git a/daemon/src/media/audio/opensl/opensllayer.cpp b/daemon/src/media/audio/opensl/opensllayer.cpp
index 39c2ab7fb6..284df53d8d 100644
--- a/daemon/src/media/audio/opensl/opensllayer.cpp
+++ b/daemon/src/media/audio/opensl/opensllayer.cpp
@@ -658,7 +658,7 @@ OpenSLLayer::playback(SLAndroidSimpleBufferQueueItf queue)
}
if (bufferIsFilled_) {
- SLresult result = (*queue)->Enqueue(queue, buffer.getChannel(0)->data(), buffer.frames()*sizeof(SFLAudioSample));
+ SLresult result = (*queue)->Enqueue(queue, buffer.getChannel(0)->data(), buffer.frames()*sizeof(ring::AudioSample));
if (SL_RESULT_SUCCESS != result) {
RING_DBG("Error could not enqueue buffers in playback callback\n");
}
@@ -687,7 +687,7 @@ OpenSLLayer::capture(SLAndroidSimpleBufferQueueItf queue)
SLresult result;
// enqueue an empty buffer to be filled by the recorder
// (for streaming recording, we enqueue at least 2 empty buffers to start things off)
- result = (*recorderBufferQueue_)->Enqueue(recorderBufferQueue_, buffer.getChannel(0)->data(), buffer.frames()*sizeof(SFLAudioSample));
+ result = (*recorderBufferQueue_)->Enqueue(recorderBufferQueue_, buffer.getChannel(0)->data(), buffer.frames()*sizeof(ring::AudioSample));
audioCaptureFillBuffer(old_buffer);
diff --git a/daemon/src/media/audio/pulseaudio/pulselayer.cpp b/daemon/src/media/audio/pulseaudio/pulselayer.cpp
index 25709ad15a..4ea5858fbd 100644
--- a/daemon/src/media/audio/pulseaudio/pulselayer.cpp
+++ b/daemon/src/media/audio/pulseaudio/pulselayer.cpp
@@ -490,7 +490,7 @@ void PulseLayer::writeToSpeaker()
urgentBytes = urgentSamples * sample_size;
}
- SFLAudioSample *data = 0;
+ ring::AudioSample *data = 0;
if (urgentBytes) {
AudioBuffer linearbuff(urgentSamples, format);
@@ -585,7 +585,7 @@ void PulseLayer::readFromMic()
const size_t samples = bytes / sample_size / format.nb_channels;
AudioBuffer in(samples, format);
- in.deinterleave((SFLAudioSample*)data, samples, format.nb_channels);
+ in.deinterleave((ring::AudioSample*)data, samples, format.nb_channels);
unsigned int mainBufferSampleRate = Manager::instance().getRingBufferPool().getInternalSamplingRate();
bool resample = audioFormat_.sample_rate != mainBufferSampleRate;
@@ -635,7 +635,7 @@ void PulseLayer::ringtoneToSpeaker()
const unsigned samples = (bytes / sample_size) / ringtone_->channels();
AudioBuffer tmp(samples, ringtone_->getFormat());
fileToPlay->getNext(tmp, playbackGain_);
- tmp.interleave((SFLAudioSample*) data);
+ tmp.interleave((ring::AudioSample*) data);
} else {
memset(data, 0, bytes);
}
diff --git a/daemon/src/media/audio/resampler.cpp b/daemon/src/media/audio/resampler.cpp
index e7a2d745cb..b422c749b5 100644
--- a/daemon/src/media/audio/resampler.cpp
+++ b/daemon/src/media/audio/resampler.cpp
@@ -30,7 +30,7 @@
#include "resampler.h"
#include "logger.h"
-#include "sfl_types.h"
+#include "ring_types.h"
#include <samplerate.h>
diff --git a/daemon/src/media/audio/resampler.h b/daemon/src/media/audio/resampler.h
index 06a74620b5..6d90b90c56 100644
--- a/daemon/src/media/audio/resampler.h
+++ b/daemon/src/media/audio/resampler.h
@@ -36,7 +36,7 @@
#include <memory>
#include "audiobuffer.h"
-#include "sfl_types.h"
+#include "ring_types.h"
#include "noncopyable.h"
namespace ring {
@@ -77,7 +77,7 @@ class Resampler {
/* temporary buffers */
std::vector<float> floatBufferIn_;
std::vector<float> floatBufferOut_;
- std::vector<SFLAudioSample> scratchBuffer_;
+ std::vector<ring::AudioSample> scratchBuffer_;
size_t samples_; // size in samples of temporary buffers
AudioFormat format_; // number of channels and max output frequency
diff --git a/daemon/src/media/audio/ringbufferpool.cpp b/daemon/src/media/audio/ringbufferpool.cpp
index 9460eac29f..f94ab7efd5 100644
--- a/daemon/src/media/audio/ringbufferpool.cpp
+++ b/daemon/src/media/audio/ringbufferpool.cpp
@@ -31,7 +31,7 @@
#include "ringbufferpool.h"
#include "ringbuffer.h"
-#include "sfl_types.h" // for SIZEBUF
+#include "ring_types.h" // for SIZEBUF
#include "logger.h"
#include <limits>
diff --git a/daemon/src/media/audio/sound/audiofile.cpp b/daemon/src/media/audio/sound/audiofile.cpp
index 07a00c8d0b..cc04c9bf59 100644
--- a/daemon/src/media/audio/sound/audiofile.cpp
+++ b/daemon/src/media/audio/sound/audiofile.cpp
@@ -115,7 +115,7 @@ AudioFile::AudioFile(const std::string &fileName, unsigned int sampleRate) :
const sf_count_t nbFrames = hasHeader ? fileHandle.frames() : fileSize / fileHandle.channels();
- SFLAudioSample * interleaved = new SFLAudioSample[nbFrames * fileHandle.channels()];
+ ring::AudioSample * interleaved = new ring::AudioSample[nbFrames * fileHandle.channels()];
// get n "items", aka samples (not frames)
fileHandle.read(interleaved, nbFrames * fileHandle.channels());
diff --git a/daemon/src/media/audio/sound/dtmf.cpp b/daemon/src/media/audio/sound/dtmf.cpp
index efba06deb5..4f35d3298b 100644
--- a/daemon/src/media/audio/sound/dtmf.cpp
+++ b/daemon/src/media/audio/sound/dtmf.cpp
@@ -47,7 +47,7 @@ void DTMF::startTone(char code)
using std::vector;
-bool DTMF::generateDTMF(vector<SFLAudioSample> &buffer)
+bool DTMF::generateDTMF(vector<ring::AudioSample> &buffer)
{
try {
if (currentTone_ != 0) {
diff --git a/daemon/src/media/audio/sound/dtmf.h b/daemon/src/media/audio/sound/dtmf.h
index 48c8b3adf4..a0d64c7115 100644
--- a/daemon/src/media/audio/sound/dtmf.h
+++ b/daemon/src/media/audio/sound/dtmf.h
@@ -59,9 +59,9 @@ class DTMF {
/**
* Copy the sound inside the sampling* buffer
- * @param buffer : a vector of SFLAudioSample
+ * @param buffer : a vector of ring::AudioSample
*/
- bool generateDTMF(std::vector<SFLAudioSample> &buffer);
+ bool generateDTMF(std::vector<ring::AudioSample> &buffer);
private:
char currentTone_;
diff --git a/daemon/src/media/audio/sound/dtmfgenerator.cpp b/daemon/src/media/audio/sound/dtmfgenerator.cpp
index 837978d2d1..94bc0ba361 100644
--- a/daemon/src/media/audio/sound/dtmfgenerator.cpp
+++ b/daemon/src/media/audio/sound/dtmfgenerator.cpp
@@ -87,7 +87,7 @@ using std::vector;
/*
* Get n samples of the signal of code code
*/
-void DTMFGenerator::getSamples(vector<SFLAudioSample> &buffer, unsigned char code)
+void DTMFGenerator::getSamples(vector<ring::AudioSample> &buffer, unsigned char code)
{
code = toupper(code);
@@ -124,7 +124,7 @@ void DTMFGenerator::getSamples(vector<SFLAudioSample> &buffer, unsigned char cod
* Get next n samples (continues where previous call to
* genSample or genNextSamples stopped
*/
-void DTMFGenerator::getNextSamples(vector<SFLAudioSample> &buffer)
+void DTMFGenerator::getNextSamples(vector<ring::AudioSample> &buffer)
{
if (state.sample == 0)
throw DTMFException("DTMF generator not initialized");
@@ -138,10 +138,10 @@ void DTMFGenerator::getNextSamples(vector<SFLAudioSample> &buffer)
state.offset = (state.offset + i) % sampleRate_;
}
-SFLAudioSample* DTMFGenerator::fillToneBuffer(int index)
+ring::AudioSample* DTMFGenerator::fillToneBuffer(int index)
{
assert(index >= 0 and index < NUM_TONES);
- SFLAudioSample* ptr = new SFLAudioSample[sampleRate_];
+ ring::AudioSample* ptr = new ring::AudioSample[sampleRate_];
tone_.genSin(ptr, tones_[index].higher, tones_[index].lower, sampleRate_);
return ptr;
}
diff --git a/daemon/src/media/audio/sound/dtmfgenerator.h b/daemon/src/media/audio/sound/dtmfgenerator.h
index 273b926be0..cea0bc481c 100644
--- a/daemon/src/media/audio/sound/dtmfgenerator.h
+++ b/daemon/src/media/audio/sound/dtmfgenerator.h
@@ -72,7 +72,7 @@ class DTMFGenerator {
/** State of the DTMF generator */
struct DTMFState {
unsigned int offset; /** Offset in the sample currently being played */
- SFLAudioSample* sample; /** Currently generated code */
+ ring::AudioSample* sample; /** Currently generated code */
};
/** State of the DTMF generator */
@@ -82,7 +82,7 @@ class DTMFGenerator {
static const DTMFTone tones_[NUM_TONES];
/** Generated samples for each tone */
- SFLAudioSample* toneBuffers_[NUM_TONES];
+ ring::AudioSample* toneBuffers_[NUM_TONES];
/** Sampling rate of generated dtmf */
int sampleRate_;
@@ -104,26 +104,26 @@ class DTMFGenerator {
/*
* Get n samples of the signal of code code
- * @param buffer a SFLAudioSample vector
+ * @param buffer a ring::AudioSample vector
* @param code dtmf code to get sound
*/
- void getSamples(std::vector<SFLAudioSample> &buffer, unsigned char code);
+ void getSamples(std::vector<ring::AudioSample> &buffer, unsigned char code);
/*
* Get next n samples (continues where previous call to
* genSample or genNextSamples stopped
- * @param buffer a SFLAudioSample vector
+ * @param buffer a ring::AudioSample vector
*/
- void getNextSamples(std::vector<SFLAudioSample> &buffer);
+ void getNextSamples(std::vector<ring::AudioSample> &buffer);
private:
/**
* Fill tone buffer for a given index of the array of tones.
* @param index of the tone in the array tones_
- * @return SFLAudioSample* The generated data
+ * @return ring::AudioSample* The generated data
*/
- SFLAudioSample* fillToneBuffer(int index);
+ ring::AudioSample* fillToneBuffer(int index);
};
}
diff --git a/daemon/src/media/audio/sound/tone.cpp b/daemon/src/media/audio/sound/tone.cpp
index c6b9815428..f35ca63cfa 100644
--- a/daemon/src/media/audio/sound/tone.cpp
+++ b/daemon/src/media/audio/sound/tone.cpp
@@ -36,7 +36,7 @@
*/
#include "tone.h"
#include "logger.h"
-#include "sfl_types.h"
+#include "ring_types.h"
#include <cmath>
#include <cassert>
#include <cstdlib>
@@ -61,7 +61,7 @@ Tone::genBuffer(const std::string& definition)
size_t size = 0;
const int sampleRate = buffer_->getSampleRate();
- std::vector<SFLAudioSample> buffer(SIZEBUF);
+ std::vector<ring::AudioSample> buffer(SIZEBUF);
size_t bufferPos(0);
// Number of format sections
@@ -155,7 +155,7 @@ Tone::interpolate(double x) const
}
void
-Tone::genSin(SFLAudioSample* buffer, int lowFrequency, int highFrequency, int nb)
+Tone::genSin(ring::AudioSample* buffer, int lowFrequency, int highFrequency, int nb)
{
xhigher_ = 0.0;
xlower_ = 0.0;
diff --git a/daemon/src/media/audio/sound/tone.h b/daemon/src/media/audio/sound/tone.h
index cc75cdb4eb..decf8f0562 100644
--- a/daemon/src/media/audio/sound/tone.h
+++ b/daemon/src/media/audio/sound/tone.h
@@ -69,7 +69,7 @@ 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(SFLAudioSample* buffer, int frequency1, int frequency2, int nb);
+ void genSin(ring::AudioSample* buffer, int frequency1, int frequency2, int nb);
/**
*
diff --git a/daemon/src/media/video/video_decoder.cpp b/daemon/src/media/video/video_decoder.cpp
index aea85c6cff..821ced2e2c 100644
--- a/daemon/src/media/video/video_decoder.cpp
+++ b/daemon/src/media/video/video_decoder.cpp
@@ -427,7 +427,7 @@ void VideoDecoder::writeToRingBuffer(AVFrame* decoded_frame,
ring::AudioBuffer out(decoded_frame->nb_samples, decoderFormat);
- out.deinterleave(reinterpret_cast<const SFLAudioSample*>(decoded_frame->data[0]),
+ out.deinterleave(reinterpret_cast<const ring::AudioSample*>(decoded_frame->data[0]),
decoded_frame->nb_samples, decoderCtx_->channels);
if ((unsigned)decoded_frame->sample_rate != outFormat.sample_rate) {
if (!resampler_) {
diff --git a/daemon/src/media/video/video_encoder.cpp b/daemon/src/media/video/video_encoder.cpp
index 63aa173adc..4ff26d4eb4 100644
--- a/daemon/src/media/video/video_encoder.cpp
+++ b/daemon/src/media/video/video_encoder.cpp
@@ -329,11 +329,11 @@ int VideoEncoder::encode_audio(const ring::AudioBuffer &buffer)
return -1;
}
- SFLAudioSample *sample_data = reinterpret_cast<SFLAudioSample*>(av_malloc(needed_bytes));
+ ring::AudioSample *sample_data = reinterpret_cast<ring::AudioSample*>(av_malloc(needed_bytes));
if (!sample_data)
return -1;
- SFLAudioSample *offset_ptr = sample_data;
+ ring::AudioSample *offset_ptr = sample_data;
int nb_frames = buffer.frames();
buffer.interleave(sample_data);
diff --git a/daemon/src/sfl_types.h b/daemon/src/ring_types.h
similarity index 93%
rename from daemon/src/sfl_types.h
rename to daemon/src/ring_types.h
index e7fbe829ec..04c26d015b 100644
--- a/daemon/src/sfl_types.h
+++ b/daemon/src/ring_types.h
@@ -31,15 +31,15 @@
#ifndef RING_TYPES_H_
#define RING_TYPES_H_
-#include <cstddef> // for size_t
-#include <stdint.h>
#include <type_traits>
#include <memory>
+#include <cstddef> // for size_t
-typedef int16_t SFLAudioSample;
-#define RING_DATA_FORMAT_MAX SHRT_MAX
+namespace ring {
+typedef int16_t AudioSample;
+}
-static const size_t SIZEBUF = 32000; /** About 1s of buffering at 48kHz */
+static constexpr size_t SIZEBUF = 32000; /** About 1s of buffering at 48kHz */
/**
* This meta-function is used to enable a template overload
diff --git a/daemon/src/ringdht/ringaccount.h b/daemon/src/ringdht/ringaccount.h
index 7d8125b377..d60baaf9db 100644
--- a/daemon/src/ringdht/ringaccount.h
+++ b/daemon/src/ringdht/ringaccount.h
@@ -39,7 +39,7 @@
#include "sip/sipaccountbase.h"
#include "noncopyable.h"
#include "ip_utils.h"
-#include "sfl_types.h" // enable_if_base_of
+#include "ring_types.h" // enable_if_base_of
#include <opendht/dhtrunner.h>
diff --git a/daemon/src/sip/sipaccount.h b/daemon/src/sip/sipaccount.h
index 4d22ed6985..f1cbda0950 100644
--- a/daemon/src/sip/sipaccount.h
+++ b/daemon/src/sip/sipaccount.h
@@ -42,7 +42,7 @@
#include "sipaccountbase.h"
#include "siptransport.h"
#include "noncopyable.h"
-#include "sfl_types.h" // enable_if_base_of
+#include "ring_types.h" // enable_if_base_of
#include <pjsip/sip_transport_tls.h>
#include <pjsip/sip_types.h>
diff --git a/daemon/src/sip/sipvoiplink.cpp b/daemon/src/sip/sipvoiplink.cpp
index 6646d1c809..5d240ccaf3 100644
--- a/daemon/src/sip/sipvoiplink.cpp
+++ b/daemon/src/sip/sipvoiplink.cpp
@@ -565,7 +565,7 @@ SIPVoIPLink::SIPVoIPLink()
TRY(pjsip_replaces_init_module(endpt_)); // See the Replaces specification in RFC 3891
TRY(pjsip_100rel_init_module(endpt_));
- // Initialize and register sflphone module
+ // Initialize and register ring module
mod_ua_.name = pj_str((char*) PACKAGE);
mod_ua_.id = -1;
mod_ua_.priority = PJSIP_MOD_PRIORITY_APPLICATION;
diff --git a/daemon/src/sip/sipvoiplink.h b/daemon/src/sip/sipvoiplink.h
index 547f978653..99416d296c 100644
--- a/daemon/src/sip/sipvoiplink.h
+++ b/daemon/src/sip/sipvoiplink.h
@@ -41,7 +41,7 @@
#include "config.h"
#endif
-#include "sfl_types.h"
+#include "ring_types.h"
#include "siptransport.h"
#include "ip_utils.h"
diff --git a/daemon/test/audiobuffertest.cpp b/daemon/test/audiobuffertest.cpp
index bbd99e6420..ca1564cfea 100644
--- a/daemon/test/audiobuffertest.cpp
+++ b/daemon/test/audiobuffertest.cpp
@@ -40,7 +40,7 @@ void AudioBufferTest::testAudioBufferConstructors()
{
TITLE();
- SFLAudioSample test_samples2[] = {10, 11, 12, 13, 14, 15, 16, 17};
+ ring::AudioSample test_samples2[] = {10, 11, 12, 13, 14, 15, 16, 17};
AudioBuffer empty_buf(0, AudioFormat::MONO());
CPPUNIT_ASSERT(empty_buf.frames() == 0);
@@ -64,8 +64,8 @@ void AudioBufferTest::testAudioBufferMix()
{
TITLE();
- SFLAudioSample test_samples1[] = {18, 19, 20, 21, 22, 23, 24, 25};
- SFLAudioSample test_samples2[] = {10, 11, 12, 13, 14, 15, 16, 17, 18};
+ ring::AudioSample test_samples1[] = {18, 19, 20, 21, 22, 23, 24, 25};
+ ring::AudioSample test_samples2[] = {10, 11, 12, 13, 14, 15, 16, 17, 18};
AudioBuffer test_buf1(test_samples1, 4, AudioFormat::STEREO());
CPPUNIT_ASSERT(test_buf1.channels() == 2);
@@ -86,7 +86,7 @@ void AudioBufferTest::testAudioBufferMix()
CPPUNIT_ASSERT((*test_buf2.getChannel(2))[0] == test_samples2[2]);
CPPUNIT_ASSERT(test_buf2.capacity() == 9);
- SFLAudioSample *output = new SFLAudioSample[test_buf2.capacity()];
+ ring::AudioSample *output = new ring::AudioSample[test_buf2.capacity()];
test_buf2.interleave(output);
CPPUNIT_ASSERT(std::equal(test_samples2, test_samples2 + sizeof test_samples2 / sizeof *test_samples2, output));
//CPPUNIT_ASSERT(std::equal(std::begin(test_samples2), std::end(test_samples2), std::begin(output))); C++11
diff --git a/daemon/test/audiocodectest.cpp b/daemon/test/audiocodectest.cpp
index 7bc88900ef..c9859ce4db 100644
--- a/daemon/test/audiocodectest.cpp
+++ b/daemon/test/audiocodectest.cpp
@@ -37,7 +37,7 @@
#include "plugin_manager.h"
#include "test_utils.h"
-#include "sfl_types.h" // for SFLAudioSample
+#include "ring_types.h" // for ring::AudioSample
#include <cmath>
#include <climits>
@@ -48,7 +48,7 @@
* http://netwerkt.wordpress.com/2011/08/25/goertzel-filter/
*/
static double
-goertzelFilter(SFLAudioSample *samples, double freq, unsigned N, double sample_rate)
+goertzelFilter(ring::AudioSample *samples, double freq, unsigned N, double sample_rate)
{
double s_prev = 0.0;
double s_prev2 = 0.0;
@@ -76,8 +76,8 @@ void AudioCodecTest::testCodecs()
for (auto p : payloadTypes)
codecs.push_back(factory.getCodec(p));
- std::vector<std::vector<SFLAudioSample>> sine = {};
- std::vector<std::vector<SFLAudioSample>> pcm;
+ std::vector<std::vector<ring::AudioSample>> sine = {};
+ std::vector<std::vector<ring::AudioSample>> pcm;
unsigned sampleRate = 0;
double referencePower = 0.0;
@@ -88,8 +88,8 @@ void AudioCodecTest::testCodecs()
if (sampleRate != c->getCurrentClockRate()) {
sampleRate = c->getCurrentClockRate();
const unsigned nbSamples = sampleRate * 0.02; // 20 ms worth of samples
- sine = {std::vector<SFLAudioSample>(nbSamples)};
- pcm = {std::vector<SFLAudioSample>(nbSamples)};
+ sine = {std::vector<ring::AudioSample>(nbSamples)};
+ pcm = {std::vector<ring::AudioSample>(nbSamples)};
const float theta = M_2_PI * frequency_ / sampleRate;
diff --git a/daemon/test/resamplertest.cpp b/daemon/test/resamplertest.cpp
index b541ba4bca..41410f8ea5 100644
--- a/daemon/test/resamplertest.cpp
+++ b/daemon/test/resamplertest.cpp
@@ -57,7 +57,7 @@ namespace {
{
#ifdef VERBOSE
std::copy(buffer.begin(), buffer.end(),
- std::ostream_iterator<SFLAudioSample>(std::cout, ", "));
+ std::ostream_iterator<ring::AudioSample>(std::cout, ", "));
std::cout << std::endl;
#endif
}
@@ -193,23 +193,23 @@ void ResamplerTest::testDownsamplingSine()
void ResamplerTest::generateRamp()
{
- std::vector<SFLAudioSample>* buf = inputBuffer.getChannel(0);
+ std::vector<ring::AudioSample>* buf = inputBuffer.getChannel(0);
for (size_t i = 0; i < buf->size(); ++i)
(*buf)[i] = i;
}
void ResamplerTest::generateTriangularSignal()
{
- std::vector<SFLAudioSample>* buf = inputBuffer.getChannel(0);
+ std::vector<ring::AudioSample>* buf = inputBuffer.getChannel(0);
for (size_t i = 0; i < buf->size(); ++i)
(*buf)[i] = i * 10;
}
void ResamplerTest::generateSineSignal()
{
- std::vector<SFLAudioSample>* buf = inputBuffer.getChannel(0);
+ std::vector<ring::AudioSample>* buf = inputBuffer.getChannel(0);
for (size_t i = 0; i < buf->size(); ++i)
- (*buf)[i] = (SFLAudioSample) (1000.0 * sin(i));
+ (*buf)[i] = (ring::AudioSample) (1000.0 * sin(i));
}
void ResamplerTest::performUpsampling(Resampler &resampler)
diff --git a/daemon/test/ringbufferpooltest.cpp b/daemon/test/ringbufferpooltest.cpp
index 63f2a6884a..f50ced92db 100644
--- a/daemon/test/ringbufferpooltest.cpp
+++ b/daemon/test/ringbufferpooltest.cpp
@@ -90,8 +90,8 @@ void RingBufferPoolTest::testGetPutData()
rbPool_->bindCallID(test_id, RingBufferPool::DEFAULT_ID);
- SFLAudioSample test_sample1 = 12;
- SFLAudioSample test_sample2 = 13;
+ ring::AudioSample test_sample1 = 12;
+ ring::AudioSample test_sample2 = 13;
AudioBuffer test_input1(&test_sample1, 1, AudioFormat::MONO());
AudioBuffer test_input2(&test_sample2, 1, AudioFormat::MONO());
@@ -127,8 +127,8 @@ void RingBufferPoolTest::testGetAvailableData()
rbPool_->bindCallID(test_id, RingBufferPool::DEFAULT_ID);
- SFLAudioSample test_sample1 = 12;
- SFLAudioSample test_sample2 = 13;
+ ring::AudioSample test_sample1 = 12;
+ ring::AudioSample test_sample2 = 13;
AudioBuffer test_input1(&test_sample1, 1, AudioFormat::MONO());
AudioBuffer test_input2(&test_sample2, 1, AudioFormat::MONO());
@@ -170,7 +170,7 @@ void RingBufferPoolTest::testDiscardFlush()
rbPool_->bindCallID(test_id, RingBufferPool::DEFAULT_ID);
- SFLAudioSample test_sample1 = 12;
+ ring::AudioSample test_sample1 = 12;
AudioBuffer test_input1(&test_sample1, 1, AudioFormat::MONO());
testRingBuffer->put(test_input1);
@@ -205,7 +205,7 @@ void RingBufferPoolTest::testConference()
// test bind Participant A with Participant B
rbPool_->bindCallID(test_id1, test_id2);
- SFLAudioSample testint = 12;
+ ring::AudioSample testint = 12;
AudioBuffer testbuf(&testint, 1, AudioFormat::MONO());
// put data test ring buffers
--
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