diff --git a/daemon/src/audio/Makefile.am b/daemon/src/audio/Makefile.am
index 6dc37940aaba4c817066db363f1121fa1a2ba104..51400d2115991e5ee31e8a3a0010c09043d33ea4 100644
--- a/daemon/src/audio/Makefile.am
+++ b/daemon/src/audio/Makefile.am
@@ -29,7 +29,7 @@ libaudio_la_SOURCES = \
audiobuffer.cpp \
audioloop.cpp \
ringbuffer.cpp \
- mainbuffer.cpp \
+ ringbufferpool.cpp \
audiorecord.cpp \
audiorecorder.cpp \
recordable.cpp \
@@ -42,7 +42,7 @@ noinst_HEADERS = \
audiobuffer.h \
audioloop.h \
ringbuffer.h \
- mainbuffer.h \
+ ringbufferpool.h \
audiorecord.h \
audiorecorder.h \
audiolayer.h \
diff --git a/daemon/src/audio/alsa/alsalayer.cpp b/daemon/src/audio/alsa/alsalayer.cpp
index af03a9504d99bc2512ac0316fac019334b334a49..5a62df201d94168fb35fa7f45bb6afec56e7d317 100644
--- a/daemon/src/audio/alsa/alsalayer.cpp
+++ b/daemon/src/audio/alsa/alsalayer.cpp
@@ -35,6 +35,7 @@
#include "manager.h"
#include "noncopyable.h"
#include "client/configurationmanager.h"
+#include "audio/ringbufferpool.h"
#include "audio/ringbuffer.h"
#include <thread>
@@ -158,7 +159,7 @@ AlsaLayer::AlsaLayer(const AudioPreference &pref)
, is_playback_open_(false)
, is_capture_open_(false)
, audioThread_(nullptr)
- , mainRingBuffer_(Manager::instance().getMainBuffer().getRingBuffer(MainBuffer::DEFAULT_ID))
+ , mainRingBuffer_(Manager::instance().getRingBufferPool().getRingBuffer(RingBufferPool::DEFAULT_ID))
{}
AlsaLayer::~AlsaLayer()
@@ -674,7 +675,7 @@ AlsaLayer::getAudioDeviceName(int index, DeviceType type) const
void AlsaLayer::capture()
{
- AudioFormat mainBufferFormat = Manager::instance().getMainBuffer().getInternalAudioFormat();
+ AudioFormat mainBufferFormat = Manager::instance().getRingBufferPool().getInternalAudioFormat();
int toGetFrames = snd_pcm_avail_update(captureHandle_);
@@ -713,7 +714,7 @@ void AlsaLayer::capture()
void AlsaLayer::playback(int maxFrames)
{
- unsigned framesToGet = Manager::instance().getMainBuffer().availableForGet(MainBuffer::DEFAULT_ID);
+ unsigned framesToGet = Manager::instance().getRingBufferPool().availableForGet(RingBufferPool::DEFAULT_ID);
// no audio available, play tone or silence
if (framesToGet <= 0) {
@@ -736,7 +737,7 @@ void AlsaLayer::playback(int maxFrames)
write(playbackIBuff_.data(), playbackBuff_.frames(), playbackHandle_);
} else {
// play the regular sound samples
- const AudioFormat mainBufferFormat = Manager::instance().getMainBuffer().getInternalAudioFormat();
+ const AudioFormat mainBufferFormat = Manager::instance().getRingBufferPool().getInternalAudioFormat();
const bool resample = audioFormat_.sample_rate != mainBufferFormat.sample_rate;
double resampleFactor = 1.0;
@@ -752,7 +753,7 @@ void AlsaLayer::playback(int maxFrames)
playbackBuff_.setFormat(mainBufferFormat);
playbackBuff_.resize(framesToGet);
- Manager::instance().getMainBuffer().getData(playbackBuff_, MainBuffer::DEFAULT_ID);
+ Manager::instance().getRingBufferPool().getData(playbackBuff_, RingBufferPool::DEFAULT_ID);
playbackBuff_.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
if (audioFormat_.nb_channels != mainBufferFormat.nb_channels) {
@@ -785,19 +786,19 @@ void AlsaLayer::audioCallback()
if (not safeUpdate(playbackHandle_, playbackAvailFrames))
return;
- unsigned framesToGet = urgentRingBuffer_.availableForGet(MainBuffer::DEFAULT_ID);
+ unsigned framesToGet = urgentRingBuffer_.availableForGet(RingBufferPool::DEFAULT_ID);
if (framesToGet > 0) {
// Urgent data (dtmf, incoming call signal) come first.
framesToGet = std::min(framesToGet, (unsigned)playbackAvailFrames);
playbackBuff_.setFormat(audioFormat_);
playbackBuff_.resize(framesToGet);
- urgentRingBuffer_.get(playbackBuff_, MainBuffer::DEFAULT_ID);
+ urgentRingBuffer_.get(playbackBuff_, RingBufferPool::DEFAULT_ID);
playbackBuff_.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
playbackBuff_.interleave(playbackIBuff_);
write(playbackIBuff_.data(), framesToGet, playbackHandle_);
// Consume the regular one as well (same amount of frames)
- Manager::instance().getMainBuffer().discard(framesToGet, MainBuffer::DEFAULT_ID);
+ Manager::instance().getRingBufferPool().discard(framesToGet, RingBufferPool::DEFAULT_ID);
} else {
// regular audio data
playback(playbackAvailFrames);
diff --git a/daemon/src/audio/audiolayer.cpp b/daemon/src/audio/audiolayer.cpp
index 972cfbed78812d4d3dd005b6ed0bdb6ee4f0421a..9a3120951babe714a9e303c68a9d81fdbe9ac1f9 100644
--- a/daemon/src/audio/audiolayer.cpp
+++ b/daemon/src/audio/audiolayer.cpp
@@ -33,6 +33,7 @@
#include "audio/dcblocker.h"
#include "logger.h"
#include "manager.h"
+#include "audio/ringbufferpool.h"
#include <ctime>
@@ -42,14 +43,14 @@ AudioLayer::AudioLayer(const AudioPreference &pref)
, captureGain_(pref.getVolumemic())
, playbackGain_(pref.getVolumespkr())
, isStarted_(false)
- , audioFormat_(Manager::instance().getMainBuffer().getInternalAudioFormat())
+ , audioFormat_(Manager::instance().getRingBufferPool().getInternalAudioFormat())
, urgentRingBuffer_("urgentRingBuffer_id", SIZEBUF, audioFormat_)
, mutex_()
, dcblocker_()
, resampler_(audioFormat_.sample_rate)
, lastNotificationTime_(0)
{
- urgentRingBuffer_.createReadOffset(MainBuffer::DEFAULT_ID);
+ urgentRingBuffer_.createReadOffset(RingBufferPool::DEFAULT_ID);
}
AudioLayer::~AudioLayer()
@@ -68,7 +69,7 @@ void AudioLayer::flushMain()
{
std::lock_guard<std::mutex> lock(mutex_);
// should pass call id
- Manager::instance().getMainBuffer().flushAllBuffers();
+ Manager::instance().getRingBufferPool().flushAllBuffers();
}
void AudioLayer::flushUrgent()
diff --git a/daemon/src/audio/audiolayer.h b/daemon/src/audio/audiolayer.h
index c0fb3533741b52500906c56dd3a3f1a1b9b2c12f..91259d546ed33e826f5b39ff93251b3d2dc629b2 100644
--- a/daemon/src/audio/audiolayer.h
+++ b/daemon/src/audio/audiolayer.h
@@ -49,7 +49,6 @@
* @brief Main sound class. Manages the data transfers between the application and the hardware.
*/
-class MainBuffer;
class AudioPreference;
namespace ost {
diff --git a/daemon/src/audio/audiorecorder.cpp b/daemon/src/audio/audiorecorder.cpp
index 0931e34cb30c1a124f126b7bcafdd30f5a4b05a2..cd051d69f7e16b13a8801eaa990ed445fd7c5d45 100644
--- a/daemon/src/audio/audiorecorder.cpp
+++ b/daemon/src/audio/audiorecorder.cpp
@@ -30,7 +30,7 @@
#include "audiorecorder.h"
#include "audiorecord.h"
-#include "mainbuffer.h"
+#include "ringbufferpool.h"
#include "logger.h"
#include <chrono>
@@ -39,8 +39,9 @@
int AudioRecorder::count_ = 0;
-AudioRecorder::AudioRecorder(AudioRecord *arec, MainBuffer &mb) :
- recorderId_(), mbuffer_(mb), arecord_(arec), running_(false), thread_()
+AudioRecorder::AudioRecorder(AudioRecord *arec, RingBufferPool &rbp)
+ : recorderId_(), ringBufferPool_(rbp), arecord_(arec), running_(false)
+ , thread_()
{
++count_;
@@ -65,7 +66,7 @@ AudioRecorder::~AudioRecorder()
void AudioRecorder::init() {
if (!arecord_->isRecording()) {
- arecord_->setSndFormat(mbuffer_.getInternalAudioFormat());
+ arecord_->setSndFormat(ringBufferPool_.getInternalAudioFormat());
}
}
@@ -84,12 +85,12 @@ void AudioRecorder::run()
static const size_t BUFFER_LENGTH = 10000;
static const std::chrono::milliseconds SLEEP_TIME(20); // 20 ms
- AudioBuffer buffer(BUFFER_LENGTH, mbuffer_.getInternalAudioFormat());
+ AudioBuffer buffer(BUFFER_LENGTH, ringBufferPool_.getInternalAudioFormat());
while (running_) {
- const size_t availableSamples = mbuffer_.availableForGet(recorderId_);
+ const size_t availableSamples = ringBufferPool_.availableForGet(recorderId_);
buffer.resize(std::min(availableSamples, BUFFER_LENGTH));
- mbuffer_.getData(buffer, recorderId_);
+ ringBufferPool_.getData(buffer, recorderId_);
if (availableSamples > 0)
arecord_->recData(buffer);
diff --git a/daemon/src/audio/audiorecorder.h b/daemon/src/audio/audiorecorder.h
index dc9c27949e3289e4930ac051b821214a9ddb6cac..50e6b69fe36925a19822d6ad5b980d9ca3088331 100644
--- a/daemon/src/audio/audiorecorder.h
+++ b/daemon/src/audio/audiorecorder.h
@@ -37,13 +37,13 @@
#include <atomic>
#include <string>
-class MainBuffer;
+class RingBufferPool;
class AudioRecord;
class AudioRecorder {
public:
- AudioRecorder(AudioRecord *arec, MainBuffer &mb);
+ AudioRecorder(AudioRecord *arec, RingBufferPool &rbp);
~AudioRecorder();
std::string getRecorderID() const {
return recorderId_;
@@ -66,7 +66,7 @@ class AudioRecorder {
static int count_;
std::string recorderId_;
- MainBuffer &mbuffer_;
+ RingBufferPool &ringBufferPool_;
AudioRecord *arecord_;
std::atomic<bool> running_;
std::thread thread_;
diff --git a/daemon/src/audio/audiortp/audio_rtp_factory.h b/daemon/src/audio/audiortp/audio_rtp_factory.h
index 59bd813542c0703fdc44452287f1e0c50eb918ea..437558ac2708004766550b01abf11bd9754affa3 100644
--- a/daemon/src/audio/audiortp/audio_rtp_factory.h
+++ b/daemon/src/audio/audiortp/audio_rtp_factory.h
@@ -133,9 +133,11 @@ class AudioRtpFactory {
void setDtmfPayloadType(unsigned int);
/**
- * Send DTMF over RTP (RFC2833). The timestamp and sequence number must be
- * incremented as if it was microphone audio. This function change the payload type of the rtp session,
- * send the appropriate DTMF digit using this payload, discard coresponding data from mainbuffer and get
+ * Send DTMF over RTP (RFC2833).
+ * The timestamp and sequence number must be incremented
+ * as if it was microphone audio. This function change the payload type
+ * of the rtp session, send the appropriate DTMF digit using
+ * this payload, discard corresponding data from RingBufferPool and get
* back the codec payload for further audio processing.
*/
void sendDtmfDigit(int digit);
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.h b/daemon/src/audio/audiortp/audio_rtp_session.h
index d4033a2988f7b92e8552739325068eda4fda1942..ccdf15c5b57dfb92bfd537408fc0b37afedbb8a6 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.h
+++ b/daemon/src/audio/audiortp/audio_rtp_session.h
@@ -141,9 +141,11 @@ class AudioRtpSession {
virtual void startRTPLoop() = 0;
/**
- * Send DTMF over RTP (RFC2833). The timestamp and sequence number must be
- * incremented as if it was microphone audio. This function change the payload type of the rtp session,
- * send the appropriate DTMF digit using this payload, discard coresponding data from mainbuffer and get
+ * Send DTMF over RTP (RFC2833).
+ * The timestamp and sequence number must be incremented
+ * as if it was microphone audio. This function change the payload type
+ * of the rtp session, send the appropriate DTMF digit using
+ * this payload, discard corresponding data from RingBufferPool and get
* back the codec payload for further audio processing.
*/
void sendDtmfEvent();
diff --git a/daemon/src/audio/audiortp/audio_rtp_stream.cpp b/daemon/src/audio/audiortp/audio_rtp_stream.cpp
index bfa89f2b847470863485840b08092f4a422f6592..1e64ee3440807854771cf3c44301721dbb1b99eb 100644
--- a/daemon/src/audio/audiortp/audio_rtp_stream.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_stream.cpp
@@ -36,6 +36,7 @@
#include "audio/audiolayer.h"
#include "audio/resampler.h"
+#include "audio/ringbufferpool.h"
#include "audio/ringbuffer.h"
#if HAVE_SPEEXDSP
@@ -69,7 +70,7 @@ AudioRtpStream::AudioRtpStream(const std::string &id) :
, plcPool_(nullptr)
, plcDec_()
{
- ringbuffer_ = Manager::instance().getMainBuffer().createRingBuffer(id_);
+ ringbuffer_ = Manager::instance().getRingBufferPool().createRingBuffer(id_);
pj_caching_pool_init(&plcCachePool_, &pj_pool_factory_default_policy, 0);
plcPool_ = pj_pool_create(&plcCachePool_.factory, "plc", 64, 1024, nullptr);
}
@@ -131,7 +132,7 @@ bool AudioRtpStream::tryToSwitchDecoder(int newPt)
for (unsigned i=0, n=audioCodecs_.size(); i<n; i++) {
auto codec = audioCodecs_[i];
if (codec == nullptr || codec->getPayloadType() != newPt) continue;
- AudioFormat f = Manager::instance().getMainBuffer().getInternalAudioFormat();
+ AudioFormat f = Manager::instance().getRingBufferPool().getInternalAudioFormat();
codec->setOptimalFormat(f.sample_rate, f.nb_channels);
decoder_.payloadType_ = codec->getPayloadType();
decoder_.frameSize_ = codec->getFrameSize();
@@ -250,7 +251,7 @@ void AudioRtpStream::setRtpMedia(const std::vector<AudioCodec*> &audioCodecs)
currentEncoderIndex_ = currentDecoderIndex_ = 0;
AudioCodec& codec = *audioCodecs[currentEncoderIndex_];
- AudioFormat f = Manager::instance().getMainBuffer().getInternalAudioFormat();
+ AudioFormat f = Manager::instance().getRingBufferPool().getInternalAudioFormat();
codec.setOptimalFormat(f.sample_rate, f.nb_channels);
const int pt = codec.getPayloadType();
@@ -288,7 +289,7 @@ void AudioRtpStream::resetDSP()
bool AudioRtpStream::waitForDataEncode(const std::chrono::milliseconds& max_wait) const
{
- const auto &mainBuffer = Manager::instance().getMainBuffer();
+ const auto &mainBuffer = Manager::instance().getRingBufferPool();
const AudioFormat mainBuffFormat = mainBuffer.getInternalAudioFormat();
const double resampleFactor = (double) mainBuffFormat.sample_rate / encoder_.format_.sample_rate;
const size_t samplesToGet = resampleFactor * encoder_.frameSize_;
@@ -301,22 +302,23 @@ size_t AudioRtpStream::processDataEncode()
if (isDead())
return 0;
- AudioFormat mainBuffFormat = Manager::instance().getMainBuffer().getInternalAudioFormat();
+ AudioFormat mainBuffFormat = Manager::instance().getRingBufferPool().getInternalAudioFormat();
double resampleFactor = (double) mainBuffFormat.sample_rate / encoder_.format_.sample_rate;
// compute nb of byte to get corresponding to 1 audio frame
const size_t samplesToGet = resampleFactor * encoder_.frameSize_;
- if (Manager::instance().getMainBuffer().availableForGet(id_) < samplesToGet)
+ if (Manager::instance().getRingBufferPool().availableForGet(id_) < samplesToGet)
return 0;
micData_.setFormat(mainBuffFormat);
micData_.resize(samplesToGet);
- const size_t samples = Manager::instance().getMainBuffer().getData(micData_, id_);
+ const size_t samples = Manager::instance().getRingBufferPool().getData(micData_, id_);
if (samples != samplesToGet) {
- ERROR("Asked for %d samples from mainbuffer, got %d", samplesToGet, samples);
+ ERROR("Asked for %d samples from bindings on call '%s', got %d",
+ samplesToGet, id_.c_str(), samples);
return 0;
}
@@ -416,7 +418,7 @@ void AudioRtpStream::processDataDecode(unsigned char *spkrData, size_t size, int
// test if resampling or up/down-mixing is required
AudioBuffer *out = &rawBuffer_;
AudioFormat decFormat = out->getFormat();
- AudioFormat mainBuffFormat = Manager::instance().getMainBuffer().getInternalAudioFormat();
+ AudioFormat mainBuffFormat = Manager::instance().getRingBufferPool().getInternalAudioFormat();
if (decFormat.sample_rate != mainBuffFormat.sample_rate) {
if (!decoder_.resampler_) {
ERROR("Resampler already destroyed");
diff --git a/daemon/src/audio/audiortp/audio_rtp_stream.h b/daemon/src/audio/audiortp/audio_rtp_stream.h
index b1db256f2a42089a7dfaad90ad517bb1555adda3..64fea6c5ac82cf1030bed0caa148bbf5bda271da 100644
--- a/daemon/src/audio/audiortp/audio_rtp_stream.h
+++ b/daemon/src/audio/audiortp/audio_rtp_stream.h
@@ -107,12 +107,12 @@ class AudioRtpStream {
void processDataDecode(uint8_t *spkrData, size_t size, int payloadType);
/**
- * Wait for available data to be encoded from mainbuffer.
+ * Wait for available data to be encoded from RingBufferPool.
*/
bool waitForDataEncode(const std::chrono::milliseconds& max_wait) const;
/**
- * Encode audio data from mainbuffer
+ * Encode audio data from RingBufferPool
* @return size of encoded data, in bytes.
*/
size_t processDataEncode();
diff --git a/daemon/src/audio/jack/jacklayer.cpp b/daemon/src/audio/jack/jacklayer.cpp
index 97302bafdff85fc09bf48ed667103026226eb5e9..0ec4b2aafcc7cc6f137582a5fcb6265fe4e7761c 100644
--- a/daemon/src/audio/jack/jacklayer.cpp
+++ b/daemon/src/audio/jack/jacklayer.cpp
@@ -36,7 +36,7 @@
#include <cassert>
#include <climits>
#include "logger.h"
-#include "audio/mainbuffer.h"
+#include "audio/ringbufferpool.h"
#include "audio/ringbuffer.h"
#include "manager.h"
#include "array_size.h"
@@ -83,19 +83,19 @@ void JackLayer::fillWithUrgent(AudioBuffer &buffer, size_t samplesToGet)
// Urgent data (dtmf, incoming call signal) come first.
samplesToGet = std::min(samplesToGet, hardwareBufferSize_);
buffer.resize(samplesToGet);
- urgentRingBuffer_.get(buffer, MainBuffer::DEFAULT_ID);
+ urgentRingBuffer_.get(buffer, RingBufferPool::DEFAULT_ID);
buffer.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
// Consume the regular one as well (same amount of samples)
- Manager::instance().getMainBuffer().discard(samplesToGet, MainBuffer::DEFAULT_ID);
+ Manager::instance().getRingBufferPool().discard(samplesToGet, RingBufferPool::DEFAULT_ID);
}
void JackLayer::fillWithVoice(AudioBuffer &buffer, size_t samplesAvail)
{
- MainBuffer &mainBuffer = Manager::instance().getMainBuffer();
+ RingBufferPool &mainBuffer = Manager::instance().getRingBufferPool();
buffer.resize(samplesAvail);
- mainBuffer.getData(buffer, MainBuffer::DEFAULT_ID);
+ mainBuffer.getData(buffer, RingBufferPool::DEFAULT_ID);
buffer.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
if (audioFormat_.sample_rate != (unsigned) mainBuffer.getInternalSamplingRate()) {
@@ -129,8 +129,8 @@ JackLayer::playback()
{
notifyIncomingCall();
- const size_t samplesToGet = Manager::instance().getMainBuffer().availableForGet(MainBuffer::DEFAULT_ID);
- const size_t urgentSamplesToGet = urgentRingBuffer_.availableForGet(MainBuffer::DEFAULT_ID);
+ const size_t samplesToGet = Manager::instance().getRingBufferPool().availableForGet(RingBufferPool::DEFAULT_ID);
+ const size_t urgentSamplesToGet = urgentRingBuffer_.availableForGet(RingBufferPool::DEFAULT_ID);
if (urgentSamplesToGet > 0) {
fillWithUrgent(playbackBuffer_, urgentSamplesToGet);
@@ -152,7 +152,7 @@ JackLayer::capture()
// get audio from jack ringbuffer
read(captureBuffer_);
- const AudioFormat mainBufferFormat = Manager::instance().getMainBuffer().getInternalAudioFormat();
+ const AudioFormat mainBufferFormat = Manager::instance().getRingBufferPool().getInternalAudioFormat();
const bool resample = mainBufferFormat.sample_rate != audioFormat_.sample_rate;
captureBuffer_.applyGain(isCaptureMuted_ ? 0.0 : captureGain_);
@@ -326,7 +326,7 @@ JackLayer::JackLayer(const AudioPreference &p) :
captureBuffer_(0, audioFormat_),
captureFloatBuffer_(),
hardwareBufferSize_(0),
- mainRingBuffer_(Manager::instance().getMainBuffer().getRingBuffer(MainBuffer::DEFAULT_ID))
+ mainRingBuffer_(Manager::instance().getRingBufferPool().getRingBuffer(RingBufferPool::DEFAULT_ID))
{
playbackClient_ = jack_client_open(PACKAGE_NAME,
(jack_options_t) (JackNullOption | JackNoStartServer), NULL);
diff --git a/daemon/src/audio/opensl/opensllayer.cpp b/daemon/src/audio/opensl/opensllayer.cpp
index 608ebd42231a7a0cd158f3fb9ebd01c05ac61cd7..7d5cd1336f96d4d7b9420f0228eb195330d191ba 100644
--- a/daemon/src/audio/opensl/opensllayer.cpp
+++ b/daemon/src/audio/opensl/opensllayer.cpp
@@ -33,7 +33,7 @@
#include "client/configurationmanager.h"
#include "manager.h"
-#include "audio/mainbuffer.h"
+#include "audio/ringbufferpool.h"
#include "audio/ringbuffer.h"
#include "audio/dcblocker.h"
#include "logger.h"
@@ -77,7 +77,7 @@ OpenSLLayer::OpenSLLayer(const AudioPreference &pref)
, hardwareBuffSize_(BUFFER_SIZE)
, playbackBufferStack_(ANDROID_BUFFER_QUEUE_LENGTH, AudioBuffer(hardwareBuffSize_, AudioFormat::MONO()))
, recordBufferStack_(ANDROID_BUFFER_QUEUE_LENGTH, AudioBuffer(hardwareBuffSize_, AudioFormat::MONO()))
- , mainRingBuffer_(Manager::instance().getMainBuffer().getRingBuffer(MainBuffer::DEFAULT_ID))
+ , mainRingBuffer_(Manager::instance().getRingBufferPool().getRingBuffer(RingBufferPool::DEFAULT_ID))
{
}
@@ -345,7 +345,7 @@ OpenSLLayer::initAudioCapture() const
};
DEBUG("Capture-> Sampling Rate: %d", audioFormat_.sample_rate);
- DEBUG("Capture-> getInternalSamplingRate: %d", Manager::instance().getMainBuffer().getInternalSamplingRate());
+ DEBUG("Capture-> getInternalSamplingRate: %d", Manager::instance().getRingBufferPool().getInternalSamplingRate());
SLDataFormat_PCM audioFormat = {SL_DATAFORMAT_PCM, 1,
audioFormat_.sample_rate * 1000,
SL_PCMSAMPLEFORMAT_FIXED_16,
@@ -638,13 +638,13 @@ OpenSLLayer::playback(SLAndroidSimpleBufferQueueItf queue)
notifyIncomingCall();
AudioBuffer &buffer = getNextPlaybackBuffer();
- size_t urgentSamplesToGet = urgentRingBuffer_.availableForGet(MainBuffer::DEFAULT_ID);
+ size_t urgentSamplesToGet = urgentRingBuffer_.availableForGet(RingBufferPool::DEFAULT_ID);
bufferIsFilled_ = false;
if (urgentSamplesToGet > 0) {
bufferIsFilled_ = audioPlaybackFillWithUrgent(buffer, std::min(urgentSamplesToGet, hardwareBuffSize_));
} else {
- auto& main_buffer = Manager::instance().getMainBuffer();
+ auto& main_buffer = Manager::instance().getRingBufferPool();
buffer.resize(hardwareBuffSize_);
size_t samplesToGet = audioPlaybackFillWithVoice(buffer);
if (samplesToGet == 0) {
@@ -716,7 +716,7 @@ OpenSLLayer::updatePreference(AudioPreference &preference, int index, DeviceType
void OpenSLLayer::audioCaptureFillBuffer(AudioBuffer &buffer)
{
- MainBuffer &mbuffer = Manager::instance().getMainBuffer();
+ RingBufferPool &mbuffer = Manager::instance().getRingBufferPool();
//const unsigned mainBufferSampleRate = mbuffer.getInternalSamplingRate();
const AudioFormat mainBufferFormat = mbuffer.getInternalAudioFormat();
@@ -760,19 +760,19 @@ bool OpenSLLayer::audioPlaybackFillWithUrgent(AudioBuffer &buffer, size_t sample
// Urgent data (dtmf, incoming call signal) come first.
samplesToGet = std::min(samplesToGet, hardwareBuffSize_);
buffer.resize(samplesToGet);
- urgentRingBuffer_.get(buffer, MainBuffer::DEFAULT_ID);
+ urgentRingBuffer_.get(buffer, RingBufferPool::DEFAULT_ID);
buffer.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
// Consume the regular one as well (same amount of samples)
- Manager::instance().getMainBuffer().discard(samplesToGet, MainBuffer::DEFAULT_ID);
+ Manager::instance().getRingBufferPool().discard(samplesToGet, RingBufferPool::DEFAULT_ID);
return true;
}
size_t OpenSLLayer::audioPlaybackFillWithVoice(AudioBuffer &buffer)
{
- MainBuffer &mainBuffer = Manager::instance().getMainBuffer();
- size_t got = mainBuffer.getAvailableData(buffer, MainBuffer::DEFAULT_ID);
+ RingBufferPool &mainBuffer = Manager::instance().getRingBufferPool();
+ size_t got = mainBuffer.getAvailableData(buffer, RingBufferPool::DEFAULT_ID);
buffer.resize(got);
buffer.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
if (audioFormat_.sample_rate != mainBuffer.getInternalSamplingRate()) {
diff --git a/daemon/src/audio/pulseaudio/pulselayer.cpp b/daemon/src/audio/pulseaudio/pulselayer.cpp
index 24ceb6d976998848311d863962b7e7cfe9f81aa1..a1444bccd41c9beae13e1bc039546faf28c54256 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.cpp
+++ b/daemon/src/audio/pulseaudio/pulselayer.cpp
@@ -38,6 +38,7 @@
#include "pulselayer.h"
#include "audio/resampler.h"
#include "audio/dcblocker.h"
+#include "audio/ringbufferpool.h"
#include "audio/ringbuffer.h"
#include "logger.h"
#include "manager.h"
@@ -103,7 +104,7 @@ PulseLayer::PulseLayer(AudioPreference &pref)
, enumeratingSinks_(false)
, enumeratingSources_(false)
, preference_(pref)
- , mainRingBuffer_(Manager::instance().getMainBuffer().getRingBuffer(MainBuffer::DEFAULT_ID))
+ , mainRingBuffer_(Manager::instance().getRingBufferPool().getRingBuffer(RingBufferPool::DEFAULT_ID))
{
setCaptureGain(pref.getVolumemic());
setPlaybackGain(pref.getVolumespkr());
@@ -464,7 +465,7 @@ void PulseLayer::writeToSpeaker()
notifyIncomingCall();
- size_t urgentSamples = urgentRingBuffer_.availableForGet(MainBuffer::DEFAULT_ID);
+ size_t urgentSamples = urgentRingBuffer_.availableForGet(RingBufferPool::DEFAULT_ID);
size_t urgentBytes = urgentSamples * sample_size;
if (urgentSamples > writableSamples) {
@@ -477,12 +478,12 @@ void PulseLayer::writeToSpeaker()
if (urgentBytes) {
AudioBuffer linearbuff(urgentSamples, format);
pa_stream_begin_write(s, (void**)&data, &urgentBytes);
- urgentRingBuffer_.get(linearbuff, MainBuffer::DEFAULT_ID); // retrive only the first sample_spec->channels channels
+ urgentRingBuffer_.get(linearbuff, RingBufferPool::DEFAULT_ID); // retrive only the first sample_spec->channels channels
linearbuff.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
linearbuff.interleave(data);
pa_stream_write(s, data, urgentBytes, nullptr, 0, PA_SEEK_RELATIVE);
// Consume the regular one as well (same amount of samples)
- Manager::instance().getMainBuffer().discard(urgentSamples, MainBuffer::DEFAULT_ID);
+ Manager::instance().getRingBufferPool().discard(urgentSamples, RingBufferPool::DEFAULT_ID);
return;
}
@@ -504,7 +505,7 @@ void PulseLayer::writeToSpeaker()
flushUrgent(); // flush remaining samples in _urgentRingBuffer
- size_t availSamples = Manager::instance().getMainBuffer().availableForGet(MainBuffer::DEFAULT_ID);
+ size_t availSamples = Manager::instance().getRingBufferPool().availableForGet(RingBufferPool::DEFAULT_ID);
if (availSamples == 0) {
pa_stream_begin_write(s, (void**)&data, &writableBytes);
@@ -518,7 +519,7 @@ void PulseLayer::writeToSpeaker()
double resampleFactor = 1.;
- AudioFormat mainBufferAudioFormat = Manager::instance().getMainBuffer().getInternalAudioFormat();
+ AudioFormat mainBufferAudioFormat = Manager::instance().getRingBufferPool().getInternalAudioFormat();
bool resample = audioFormat_.sample_rate != mainBufferAudioFormat.sample_rate;
if (resample) {
@@ -533,7 +534,7 @@ void PulseLayer::writeToSpeaker()
pa_stream_begin_write(s, (void**)&data, &resampledBytes);
AudioBuffer linearbuff(readableSamples, format);
- Manager::instance().getMainBuffer().getData(linearbuff, MainBuffer::DEFAULT_ID);
+ Manager::instance().getRingBufferPool().getData(linearbuff, RingBufferPool::DEFAULT_ID);
if (resample) {
AudioBuffer rsmpl_out(nResampled, format);
@@ -569,7 +570,7 @@ void PulseLayer::readFromMic()
AudioBuffer in(samples, format);
in.deinterleave((SFLAudioSample*)data, samples, format.nb_channels);
- unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer().getInternalSamplingRate();
+ unsigned int mainBufferSampleRate = Manager::instance().getRingBufferPool().getInternalSamplingRate();
bool resample = audioFormat_.sample_rate != mainBufferSampleRate;
in.applyGain(isCaptureMuted_ ? 0.0 : captureGain_);
diff --git a/daemon/src/audio/recordable.cpp b/daemon/src/audio/recordable.cpp
index e89c97bcf6e927cd333f7164eb77590e538192b3..bfd7f9105ba1287191325aa4c88c80ab2e373d46 100644
--- a/daemon/src/audio/recordable.cpp
+++ b/daemon/src/audio/recordable.cpp
@@ -31,7 +31,7 @@
#include "manager.h"
#include "logger.h"
-Recordable::Recordable() : recAudio_(), recorder_(&recAudio_, Manager::instance().getMainBuffer())
+Recordable::Recordable() : recAudio_(), recorder_(&recAudio_, Manager::instance().getRingBufferPool())
{
DEBUG("Set recording options: %s", Manager::instance().audioPreference.getRecordPath().c_str());
recAudio_.setRecordingOptions(AudioFormat::MONO(), Manager::instance().audioPreference.getRecordPath());
diff --git a/daemon/src/audio/mainbuffer.cpp b/daemon/src/audio/ringbufferpool.cpp
similarity index 82%
rename from daemon/src/audio/mainbuffer.cpp
rename to daemon/src/audio/ringbufferpool.cpp
index 38a7cabf086dc573b78576ddc8f0e9a189286d9d..0e4b1e4808cd327ba8b83e4dd532cc64d44b2f23 100644
--- a/daemon/src/audio/mainbuffer.cpp
+++ b/daemon/src/audio/ringbufferpool.cpp
@@ -1,6 +1,7 @@
/*
* Copyright (C) 2004-2014 Savoir-Faire Linux Inc.
- * Author : Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ * Author: Guillaume Roguez <guillaume.roguez@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -28,7 +29,7 @@
* as that of the covered work.
*/
-#include "mainbuffer.h"
+#include "ringbufferpool.h"
#include "ringbuffer.h"
#include "sfl_types.h" // for SIZEBUF
#include "logger.h"
@@ -37,12 +38,13 @@
#include <utility> // for std::pair
#include <cstring>
-const char * const MainBuffer::DEFAULT_ID = "audiolayer_id";
+const char * const RingBufferPool::DEFAULT_ID = "audiolayer_id";
-MainBuffer::MainBuffer() : defaultRingBuffer_(createRingBuffer(DEFAULT_ID))
+RingBufferPool::RingBufferPool()
+ : defaultRingBuffer_(createRingBuffer(DEFAULT_ID))
{}
-MainBuffer::~MainBuffer()
+RingBufferPool::~RingBufferPool()
{
readBindingsMap_.clear();
defaultRingBuffer_.reset();
@@ -57,7 +59,7 @@ MainBuffer::~MainBuffer()
}
void
-MainBuffer::setInternalSamplingRate(unsigned sr)
+RingBufferPool::setInternalSamplingRate(unsigned sr)
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -68,7 +70,7 @@ MainBuffer::setInternalSamplingRate(unsigned sr)
}
void
-MainBuffer::setInternalAudioFormat(AudioFormat format)
+RingBufferPool::setInternalAudioFormat(AudioFormat format)
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -79,7 +81,7 @@ MainBuffer::setInternalAudioFormat(AudioFormat format)
}
std::shared_ptr<RingBuffer>
-MainBuffer::getRingBuffer(const std::string& id)
+RingBufferPool::getRingBuffer(const std::string& id)
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -94,7 +96,7 @@ MainBuffer::getRingBuffer(const std::string& id)
}
std::shared_ptr<RingBuffer>
-MainBuffer::getRingBuffer(const std::string& id) const
+RingBufferPool::getRingBuffer(const std::string& id) const
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -106,7 +108,7 @@ MainBuffer::getRingBuffer(const std::string& id) const
}
std::shared_ptr<RingBuffer>
-MainBuffer::createRingBuffer(const std::string& id)
+RingBufferPool::createRingBuffer(const std::string& id)
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -122,22 +124,22 @@ MainBuffer::createRingBuffer(const std::string& id)
return rbuf;
}
-const MainBuffer::ReadBindings*
-MainBuffer::getReadBindings(const std::string& call_id) const
+const RingBufferPool::ReadBindings*
+RingBufferPool::getReadBindings(const std::string& call_id) const
{
const auto& iter = readBindingsMap_.find(call_id);
return iter != readBindingsMap_.cend() ? &iter->second : nullptr;
}
-MainBuffer::ReadBindings*
-MainBuffer::getReadBindings(const std::string& call_id)
+RingBufferPool::ReadBindings*
+RingBufferPool::getReadBindings(const std::string& call_id)
{
const auto& iter = readBindingsMap_.find(call_id);
return iter != readBindingsMap_.cend() ? &iter->second : nullptr;
}
void
-MainBuffer::removeReadBindings(const std::string& call_id)
+RingBufferPool::removeReadBindings(const std::string& call_id)
{
if (not readBindingsMap_.erase(call_id))
ERROR("CallID set %s does not exist!", call_id.c_str());
@@ -147,7 +149,7 @@ MainBuffer::removeReadBindings(const std::string& call_id)
* Make given call ID a reader of given ring buffer
*/
void
-MainBuffer::addReaderToRingBuffer(std::shared_ptr<RingBuffer> rbuf,
+RingBufferPool::addReaderToRingBuffer(std::shared_ptr<RingBuffer> rbuf,
const std::string& call_id)
{
if (call_id != DEFAULT_ID and rbuf->id == call_id)
@@ -159,7 +161,7 @@ MainBuffer::addReaderToRingBuffer(std::shared_ptr<RingBuffer> rbuf,
}
void
-MainBuffer::removeReaderFromRingBuffer(std::shared_ptr<RingBuffer> rbuf,
+RingBufferPool::removeReaderFromRingBuffer(std::shared_ptr<RingBuffer> rbuf,
const std::string& call_id)
{
if (auto bindings = getReadBindings(call_id)) {
@@ -172,7 +174,8 @@ MainBuffer::removeReaderFromRingBuffer(std::shared_ptr<RingBuffer> rbuf,
}
void
-MainBuffer::bindCallID(const std::string& call_id1, const std::string& call_id2)
+RingBufferPool::bindCallID(const std::string& call_id1,
+ const std::string& call_id2)
{
const auto& rb_call1 = getRingBuffer(call_id1);
if (not rb_call1) {
@@ -193,7 +196,7 @@ MainBuffer::bindCallID(const std::string& call_id1, const std::string& call_id2)
}
void
-MainBuffer::bindHalfDuplexOut(const std::string& process_id,
+RingBufferPool::bindHalfDuplexOut(const std::string& process_id,
const std::string& call_id)
{
/* This method is used only for active calls, if this call does not exist,
@@ -206,7 +209,7 @@ MainBuffer::bindHalfDuplexOut(const std::string& process_id,
}
void
-MainBuffer::unBindCallID(const std::string& call_id1,
+RingBufferPool::unBindCallID(const std::string& call_id1,
const std::string& call_id2)
{
const auto& rb_call1 = getRingBuffer(call_id1);
@@ -228,7 +231,7 @@ MainBuffer::unBindCallID(const std::string& call_id1,
}
void
-MainBuffer::unBindHalfDuplexOut(const std::string& process_id,
+RingBufferPool::unBindHalfDuplexOut(const std::string& process_id,
const std::string& call_id)
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -238,7 +241,7 @@ MainBuffer::unBindHalfDuplexOut(const std::string& process_id,
}
void
-MainBuffer::unBindAll(const std::string& call_id)
+RingBufferPool::unBindAll(const std::string& call_id)
{
const auto& rb_call = getRingBuffer(call_id);
if (not rb_call) {
@@ -260,7 +263,7 @@ MainBuffer::unBindAll(const std::string& call_id)
}
size_t
-MainBuffer::getData(AudioBuffer& buffer, const std::string& call_id)
+RingBufferPool::getData(AudioBuffer& buffer, const std::string& call_id)
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -289,9 +292,9 @@ MainBuffer::getData(AudioBuffer& buffer, const std::string& call_id)
}
bool
-MainBuffer::waitForDataAvailable(const std::string& call_id,
- size_t min_frames,
- const std::chrono::microseconds& max_wait) const
+RingBufferPool::waitForDataAvailable(const std::string& call_id,
+ size_t min_frames,
+ const std::chrono::microseconds& max_wait) const
{
std::unique_lock<std::recursive_mutex> lk(stateLock_);
@@ -313,7 +316,7 @@ MainBuffer::waitForDataAvailable(const std::string& call_id,
}
size_t
-MainBuffer::getAvailableData(AudioBuffer& buffer, const std::string& call_id)
+RingBufferPool::getAvailableData(AudioBuffer& buffer, const std::string& call_id)
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -329,7 +332,8 @@ MainBuffer::getAvailableData(AudioBuffer& buffer, const std::string& call_id)
size_t availableSamples = std::numeric_limits<size_t>::max();
for (const auto& rbuf : *bindings)
- availableSamples = std::min(availableSamples, rbuf->availableForGet(call_id));
+ availableSamples = std::min(availableSamples,
+ rbuf->availableForGet(call_id));
if (availableSamples == std::numeric_limits<size_t>::max())
return 0;
@@ -351,7 +355,7 @@ MainBuffer::getAvailableData(AudioBuffer& buffer, const std::string& call_id)
}
size_t
-MainBuffer::availableForGet(const std::string& call_id) const
+RingBufferPool::availableForGet(const std::string& call_id) const
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -376,7 +380,7 @@ MainBuffer::availableForGet(const std::string& call_id) const
}
size_t
-MainBuffer::discard(size_t toDiscard, const std::string& call_id)
+RingBufferPool::discard(size_t toDiscard, const std::string& call_id)
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -391,7 +395,7 @@ MainBuffer::discard(size_t toDiscard, const std::string& call_id)
}
void
-MainBuffer::flush(const std::string& call_id)
+RingBufferPool::flush(const std::string& call_id)
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
@@ -404,7 +408,7 @@ MainBuffer::flush(const std::string& call_id)
}
void
-MainBuffer::flushAllBuffers()
+RingBufferPool::flushAllBuffers()
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
diff --git a/daemon/src/audio/mainbuffer.h b/daemon/src/audio/ringbufferpool.h
similarity index 91%
rename from daemon/src/audio/mainbuffer.h
rename to daemon/src/audio/ringbufferpool.h
index 4f9b90fc506c5170eb5fd8b8b19621ec03119cb1..803606f0cd3797a6565cc3e1cd71dbaa8af86f99 100644
--- a/daemon/src/audio/mainbuffer.h
+++ b/daemon/src/audio/ringbufferpool.h
@@ -1,6 +1,7 @@
/*
* Copyright (C) 2004-2014 Savoir-Faire Linux Inc.
- * Author : Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ * Author: Guillaume Roguez <guillaume.roguez@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -28,8 +29,8 @@
* as that of the covered work.
*/
-#ifndef MAIN_BUFFER_H_
-#define MAIN_BUFFER_H_
+#ifndef RING_BUFFER_POOL_H_
+#define RING_BUFFER_POOL_H_
#include "audiobuffer.h"
#include "noncopyable.h"
@@ -42,14 +43,14 @@
class RingBuffer;
-class MainBuffer {
+class RingBufferPool {
public:
static const char * const DEFAULT_ID;
- MainBuffer();
+ RingBufferPool();
- ~MainBuffer();
+ ~RingBufferPool();
int getInternalSamplingRate() const {
return internalAudioFormat_.sample_rate;
@@ -129,16 +130,16 @@ class MainBuffer {
std::shared_ptr<RingBuffer> getRingBuffer(const std::string& id) const;
private:
- NON_COPYABLE(MainBuffer);
+ NON_COPYABLE(RingBufferPool);
// A set of RingBuffers readable by a call
typedef std::set<std::shared_ptr<RingBuffer>,
std::owner_less<std::shared_ptr<RingBuffer>> > ReadBindings;
- const MainBuffer::ReadBindings*
- getReadBindings(const std::string& call_id) const;
+ const RingBufferPool::ReadBindings*
+ getReadBindings(const std::string& call_id) const;
- MainBuffer::ReadBindings* getReadBindings(const std::string& call_id);
+ RingBufferPool::ReadBindings* getReadBindings(const std::string& call_id);
void removeReadBindings(const std::string& call_id);
@@ -161,4 +162,4 @@ class MainBuffer {
std::shared_ptr<RingBuffer> defaultRingBuffer_;
};
-#endif // MainBuffer
+#endif // RING_BUFFER_POOL_H_
diff --git a/daemon/src/call.cpp b/daemon/src/call.cpp
index 584d9f6ef65ff6320958ca8e2d09ab650cdb9963..62c751206692f8cb6e39c94832297e6a5d0f4ee8 100644
--- a/daemon/src/call.cpp
+++ b/daemon/src/call.cpp
@@ -33,7 +33,7 @@
#include "call.h"
#include "account.h"
#include "manager.h"
-#include "audio/mainbuffer.h"
+#include "audio/ringbufferpool.h"
#include "history/historyitem.h"
#include "sip/sip_utils.h"
@@ -210,17 +210,17 @@ bool
Call::toggleRecording()
{
const bool startRecording = Recordable::toggleRecording();
- MainBuffer &mbuffer = Manager::instance().getMainBuffer();
+ RingBufferPool &rbPool = Manager::instance().getRingBufferPool();
std::string process_id = Recordable::recorder_.getRecorderID();
if (startRecording) {
- mbuffer.bindHalfDuplexOut(process_id, id_);
- mbuffer.bindHalfDuplexOut(process_id, MainBuffer::DEFAULT_ID);
+ rbPool.bindHalfDuplexOut(process_id, id_);
+ rbPool.bindHalfDuplexOut(process_id, RingBufferPool::DEFAULT_ID);
Recordable::recorder_.start();
} else {
- mbuffer.unBindHalfDuplexOut(process_id, id_);
- mbuffer.unBindHalfDuplexOut(process_id, MainBuffer::DEFAULT_ID);
+ rbPool.unBindHalfDuplexOut(process_id, id_);
+ rbPool.unBindHalfDuplexOut(process_id, RingBufferPool::DEFAULT_ID);
}
return startRecording;
diff --git a/daemon/src/conference.cpp b/daemon/src/conference.cpp
index b681e9b4d4ba769b3ad9a877f121ba11f973fe71..9fe09c6b32cfbbac613a4071162ee62c60c4216d 100644
--- a/daemon/src/conference.cpp
+++ b/daemon/src/conference.cpp
@@ -34,7 +34,7 @@
#include "conference.h"
#include "manager.h"
#include "audio/audiolayer.h"
-#include "audio/mainbuffer.h"
+#include "audio/ringbufferpool.h"
#ifdef SFL_VIDEO
#include "sip/sipcall.h"
@@ -100,16 +100,16 @@ void Conference::remove(const std::string &participant_id)
void Conference::bindParticipant(const std::string &participant_id)
{
- auto &mainBuffer = Manager::instance().getMainBuffer();
+ auto &rbPool = Manager::instance().getRingBufferPool();
for (const auto &item : participants_) {
if (participant_id != item)
- mainBuffer.bindCallID(participant_id, item);
- mainBuffer.flush(item);
+ rbPool.bindCallID(participant_id, item);
+ rbPool.flush(item);
}
- mainBuffer.bindCallID(participant_id, MainBuffer::DEFAULT_ID);
- mainBuffer.flush(MainBuffer::DEFAULT_ID);
+ rbPool.bindCallID(participant_id, RingBufferPool::DEFAULT_ID);
+ rbPool.flush(RingBufferPool::DEFAULT_ID);
}
std::string Conference::getStateStr() const
@@ -153,23 +153,23 @@ Conference::getDisplayNames() const
bool Conference::toggleRecording()
{
const bool startRecording = Recordable::toggleRecording();
- MainBuffer &mbuffer = Manager::instance().getMainBuffer();
+ auto& rbPool = Manager::instance().getRingBufferPool();
std::string process_id(Recordable::recorder_.getRecorderID());
// start recording
if (startRecording) {
for (const auto &item : participants_)
- mbuffer.bindHalfDuplexOut(process_id, item);
+ rbPool.bindHalfDuplexOut(process_id, item);
- mbuffer.bindHalfDuplexOut(process_id, MainBuffer::DEFAULT_ID);
+ rbPool.bindHalfDuplexOut(process_id, RingBufferPool::DEFAULT_ID);
Recordable::recorder_.start();
} else {
for (const auto &item : participants_)
- mbuffer.unBindHalfDuplexOut(process_id, item);
+ rbPool.unBindHalfDuplexOut(process_id, item);
- mbuffer.unBindHalfDuplexOut(process_id, MainBuffer::DEFAULT_ID);
+ rbPool.unBindHalfDuplexOut(process_id, RingBufferPool::DEFAULT_ID);
}
return startRecording;
diff --git a/daemon/src/iax/iaxcall.cpp b/daemon/src/iax/iaxcall.cpp
index c8b059ce496cda358c60a72414983ab6a5ca8ca6..9caa76da5fd7dd1d07fbec10fe7f96b5c23e4fce 100644
--- a/daemon/src/iax/iaxcall.cpp
+++ b/daemon/src/iax/iaxcall.cpp
@@ -40,6 +40,7 @@
#include "manager.h"
#include "iaxaccount.h"
#include "iaxvoiplink.h"
+#include "audio/ringbufferpool.h"
#include "audio/ringbuffer.h"
#if HAVE_INSTANT_MESSAGING
@@ -74,7 +75,7 @@ IAXCall::IAXCall(IAXAccount& account, const std::string& id, Call::CallType type
format(0),
session(NULL)
{
- ringbuffer_ = Manager::instance().getMainBuffer().createRingBuffer(getCallId());
+ ringbuffer_ = Manager::instance().getRingBufferPool().createRingBuffer(getCallId());
}
int IAXCall::getSupportedFormat(const std::string &accountID) const
@@ -148,13 +149,13 @@ IAXCall::answer()
setState(Call::ACTIVE);
setConnectionState(Call::CONNECTED);
- Manager::instance().getMainBuffer().flushAllBuffers();
+ Manager::instance().getRingBufferPool().flushAllBuffers();
}
void
IAXCall::hangup(int reason UNUSED)
{
- Manager::instance().getMainBuffer().unBindAll(getCallId());
+ Manager::instance().getRingBufferPool().unBindAll(getCallId());
std::lock_guard<std::mutex> lock(IAXVoIPLink::mutexIAX);
iax_hangup(session, (char*) "Dumped Call");
@@ -192,7 +193,7 @@ IAXCall::attendedTransfer(const std::string& /*targetID*/)
void
IAXCall::onhold()
{
- Manager::instance().getMainBuffer().unBindAll(getCallId());
+ Manager::instance().getRingBufferPool().unBindAll(getCallId());
{
std::lock_guard<std::mutex> lock(IAXVoIPLink::mutexIAX);
@@ -220,7 +221,7 @@ IAXCall::offhold()
void
IAXCall::peerHungup()
{
- Manager::instance().getMainBuffer().unBindAll(getCallId());
+ Manager::instance().getRingBufferPool().unBindAll(getCallId());
session = nullptr;
}
diff --git a/daemon/src/iax/iaxvoiplink.cpp b/daemon/src/iax/iaxvoiplink.cpp
index 51f3c44332493e18d4b4ebbad421b91aeb5d7bfa..364d8ac8678d8eecab48cadf4fdd92a7e137b35e 100644
--- a/daemon/src/iax/iaxvoiplink.cpp
+++ b/daemon/src/iax/iaxvoiplink.cpp
@@ -42,6 +42,7 @@
#include "hooks/urlhook.h"
#include "audio/audiolayer.h"
#include "audio/resampler.h"
+#include "audio/ringbufferpool.h"
#include "array_size.h"
#include "map_utils.h"
#include "call_factory.h"
@@ -157,20 +158,20 @@ IAXVoIPLink::sendAudioFromMic()
if (!audioCodec)
continue;
- Manager::instance().getMainBuffer().setInternalSamplingRate(audioCodec->getClockRate());
+ Manager::instance().getRingBufferPool().setInternalSamplingRate(audioCodec->getClockRate());
- unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer().getInternalSamplingRate();
+ unsigned int mainBufferSampleRate = Manager::instance().getRingBufferPool().getInternalSamplingRate();
// we have to get 20ms of data from the mic *20/1000 = /50
// rate/50 shall be lower than IAX__20S_48KHZ_MAX
size_t samples = mainBufferSampleRate * 20 / 1000;
- if (Manager::instance().getMainBuffer().availableForGet(currentCall->getCallId()) < samples)
+ if (Manager::instance().getRingBufferPool().availableForGet(currentCall->getCallId()) < samples)
continue;
// Get bytes from micRingBuffer to data_from_mic
rawBuffer_.resize(samples);
- samples = Manager::instance().getMainBuffer().getData(rawBuffer_, currentCall->getCallId());
+ samples = Manager::instance().getRingBufferPool().getData(rawBuffer_, currentCall->getCallId());
int compSize;
unsigned int audioRate = audioCodec->getClockRate();
@@ -231,7 +232,7 @@ IAXVoIPLink::handleAnswerTransfer(iax_event* event, IAXCall& call)
Manager::instance().addStream(id);
Manager::instance().peerAnsweredCall(id);
Manager::instance().startAudioDriverStream();
- Manager::instance().getMainBuffer().flushAllBuffers();
+ Manager::instance().getRingBufferPool().flushAllBuffers();
}
void
@@ -333,8 +334,8 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall& call)
if (!audioCodec)
return;
- Manager::instance().getMainBuffer().setInternalSamplingRate(audioCodec->getClockRate());
- unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer().getInternalSamplingRate();
+ Manager::instance().getRingBufferPool().setInternalSamplingRate(audioCodec->getClockRate());
+ unsigned int mainBufferSampleRate = Manager::instance().getRingBufferPool().getInternalSamplingRate();
if (event->subclass)
call.format = event->subclass;
diff --git a/daemon/src/managerimpl.cpp b/daemon/src/managerimpl.cpp
index 672a6e43ab3e893360d977aa1d359673472130bf..5972f2f054f3cdffcfdf91657b7e402a0aa9f731 100644
--- a/daemon/src/managerimpl.cpp
+++ b/daemon/src/managerimpl.cpp
@@ -63,6 +63,7 @@
#include "audio/sound/tonelist.h"
#include "audio/sound/audiofile.h"
#include "audio/sound/dtmf.h"
+#include "audio/ringbufferpool.h"
#include "history/historynamecache.h"
#include "history/history.h"
#include "manager.h"
@@ -127,8 +128,9 @@ ManagerImpl::ManagerImpl() :
hasTriedToRegister_(false), audioCodecFactory(), client_(),
currentCallMutex_(), dtmfKey_(), dtmfBuf_(0, AudioFormat::MONO()),
toneMutex_(), telephoneTone_(), audiofile_(), audioLayerMutex_(),
- waitingCalls_(), waitingCallsMutex_(), path_(),
- mainBuffer_(), callFactory(), conferenceMap_(), history_(),
+ waitingCalls_(), waitingCallsMutex_(), path_()
+ , ringbufferpool_(new RingBufferPool)
+ , callFactory(), conferenceMap_(), history_(),
finished_(false), accountFactory_()
{
// initialize random generator for call id
@@ -209,7 +211,7 @@ ManagerImpl::init(const std::string &config_file)
std::lock_guard<std::mutex> toneLock(toneMutex_);
telephoneTone_.reset(new TelephoneTone(preferences.getZoneToneChoice(), audiodriver_->getSampleRate()));
}
- dtmfKey_.reset(new DTMF(getMainBuffer().getInternalSamplingRate()));
+ dtmfKey_.reset(new DTMF(getRingBufferPool().getInternalSamplingRate()));
}
}
@@ -345,7 +347,7 @@ ManagerImpl::outgoingCall(const std::string& preferred_account_id,
if (not isConference(current_call_id) and not isConferenceParticipant(current_call_id))
onHoldCall(current_call_id);
else if (isConference(current_call_id) and not isConferenceParticipant(call_id))
- detachParticipant(MainBuffer::DEFAULT_ID);
+ detachParticipant(RingBufferPool::DEFAULT_ID);
}
try {
@@ -405,7 +407,7 @@ ManagerImpl::answerCall(const std::string& call_id)
} else if (isConference(current_call_id) and not isConferenceParticipant(call_id)) {
// if we are talking to a conference and we are answering an incoming call
DEBUG("Detach main participant from conference");
- detachParticipant(MainBuffer::DEFAULT_ID);
+ detachParticipant(RingBufferPool::DEFAULT_ID);
}
}
@@ -573,7 +575,7 @@ ManagerImpl::offHoldCall(const std::string& callId)
} else if (isConference(currentCallId) && callId != currentCallId) {
holdConference(currentCallId);
} else if (isConference(currentCallId) and not isConferenceParticipant(callId))
- detachParticipant(MainBuffer::DEFAULT_ID);
+ detachParticipant(RingBufferPool::DEFAULT_ID);
}
std::shared_ptr<Call> call;
@@ -711,7 +713,7 @@ ManagerImpl::removeConference(const std::string& conference_id)
// We now need to bind the audio to the remain participant
// Unbind main participant audio from conference
- getMainBuffer().unBindAll(MainBuffer::DEFAULT_ID);
+ getRingBufferPool().unBindAll(RingBufferPool::DEFAULT_ID);
ParticipantSet participants(conf->getParticipantList());
@@ -719,7 +721,7 @@ ManagerImpl::removeConference(const std::string& conference_id)
ParticipantSet::iterator iter_p = participants.begin();
if (iter_p != participants.end())
- getMainBuffer().bindCallID(*iter_p, MainBuffer::DEFAULT_ID);
+ getRingBufferPool().bindCallID(*iter_p, RingBufferPool::DEFAULT_ID);
// Then remove the conference from the conference map
if (conferenceMap_.erase(conference_id))
@@ -845,7 +847,7 @@ ManagerImpl::addParticipant(const std::string& callId,
// detach from prior communication and switch to this conference
if (current_call_id != callId) {
if (isConference(current_call_id))
- detachParticipant(MainBuffer::DEFAULT_ID);
+ detachParticipant(RingBufferPool::DEFAULT_ID);
else
onHoldCall(current_call_id);
}
@@ -866,7 +868,7 @@ ManagerImpl::addParticipant(const std::string& callId,
conf->add(callId);
// Connect new audio streams together
- getMainBuffer().unBindAll(callId);
+ getRingBufferPool().unBindAll(callId);
std::map<std::string, std::string> callDetails(getCallDetails(callId));
std::string callState(callDetails.find("CALL_STATE")->second);
@@ -897,7 +899,7 @@ ManagerImpl::addMainParticipant(const std::string& conference_id)
std::string current_call_id(getCurrentCallId());
if (isConference(current_call_id))
- detachParticipant(MainBuffer::DEFAULT_ID);
+ detachParticipant(RingBufferPool::DEFAULT_ID);
else
onHoldCall(current_call_id);
}
@@ -915,12 +917,12 @@ ManagerImpl::addMainParticipant(const std::string& conference_id)
ParticipantSet participants(conf->getParticipantList());
for (const auto &item_p : participants) {
- getMainBuffer().bindCallID(item_p, MainBuffer::DEFAULT_ID);
+ getRingBufferPool().bindCallID(item_p, RingBufferPool::DEFAULT_ID);
// Reset ringbuffer's readpointers
- getMainBuffer().flush(item_p);
+ getRingBufferPool().flush(item_p);
}
- getMainBuffer().flush(MainBuffer::DEFAULT_ID);
+ getRingBufferPool().flush(RingBufferPool::DEFAULT_ID);
if (conf->getState() == Conference::ACTIVE_DETACHED)
conf->setState(Conference::ACTIVE_ATTACHED);
@@ -981,7 +983,7 @@ ManagerImpl::joinParticipant(const std::string& callId1,
if ((current_call_id != callId1) and (current_call_id != callId2)) {
// If currently in a conference
if (isConference(current_call_id))
- detachParticipant(MainBuffer::DEFAULT_ID);
+ detachParticipant(RingBufferPool::DEFAULT_ID);
else
onHoldCall(current_call_id); // currently in a call
}
@@ -990,10 +992,10 @@ ManagerImpl::joinParticipant(const std::string& callId1,
auto conf = createConference(callId1, callId2);
call1->setConfId(conf->getConfID());
- getMainBuffer().unBindAll(callId1);
+ getRingBufferPool().unBindAll(callId1);
call2->setConfId(conf->getConfID());
- getMainBuffer().unBindAll(callId2);
+ getRingBufferPool().unBindAll(callId2);
// Process call1 according to its state
std::string call1_state_str(call1Details.find("CALL_STATE")->second);
@@ -1036,7 +1038,7 @@ ManagerImpl::joinParticipant(const std::string& callId1,
conf->setState(Conference::ACTIVE_ATTACHED);
// set recording sampling rate
- conf->setRecordingFormat(mainBuffer_.getInternalAudioFormat());
+ conf->setRecordingFormat(ringbufferpool_->getInternalAudioFormat());
return true;
}
@@ -1082,7 +1084,7 @@ ManagerImpl::createConfFromParticipantList(const std::vector< std::string > &par
if (successCounter >= 2) {
conferenceMap_[conf->getConfID()] = conf;
client_.getCallManager()->conferenceCreated(conf->getConfID());
- conf->setRecordingFormat(mainBuffer_.getInternalAudioFormat());
+ conf->setRecordingFormat(ringbufferpool_->getInternalAudioFormat());
}
}
@@ -1091,7 +1093,7 @@ ManagerImpl::detachParticipant(const std::string& call_id)
{
const std::string current_call_id(getCurrentCallId());
- if (call_id != MainBuffer::DEFAULT_ID) {
+ if (call_id != RingBufferPool::DEFAULT_ID) {
auto call = getCallFromCallID(call_id);
if (!call) {
ERROR("Could not find call %s", call_id.c_str());
@@ -1121,7 +1123,7 @@ ManagerImpl::detachParticipant(const std::string& call_id)
} else {
DEBUG("Unbind main participant from conference %d");
- getMainBuffer().unBindAll(MainBuffer::DEFAULT_ID);
+ getRingBufferPool().unBindAll(RingBufferPool::DEFAULT_ID);
if (not isConference(current_call_id)) {
ERROR("Current call id (%s) is not a conference", current_call_id.c_str());
@@ -1194,9 +1196,9 @@ ManagerImpl::processRemainingParticipants(Conference &conf)
if (n > 1) {
// Reset ringbuffer's readpointers
for (const auto &p : participants)
- getMainBuffer().flush(p);
+ getRingBufferPool().flush(p);
- getMainBuffer().flush(MainBuffer::DEFAULT_ID);
+ getRingBufferPool().flush(RingBufferPool::DEFAULT_ID);
} else if (n == 1) {
// this call is the last participant, hence
// the conference is over
@@ -1264,7 +1266,7 @@ ManagerImpl::addStream(const std::string& call_id)
DEBUG("Add stream to call");
// bind to main
- getMainBuffer().bindCallID(call_id, MainBuffer::DEFAULT_ID);
+ getRingBufferPool().bindCallID(call_id, RingBufferPool::DEFAULT_ID);
std::lock_guard<std::mutex> lock(audioLayerMutex_);
audiodriver_->flushUrgent();
@@ -1276,7 +1278,7 @@ void
ManagerImpl::removeStream(const std::string& call_id)
{
DEBUG("Remove audio stream %s", call_id.c_str());
- getMainBuffer().unBindAll(call_id);
+ getRingBufferPool().unBindAll(call_id);
}
void
@@ -2252,7 +2254,7 @@ ManagerImpl::hardwareAudioFormatChanged(AudioFormat format)
void
ManagerImpl::audioFormatUsed(AudioFormat format)
{
- AudioFormat currentFormat = mainBuffer_.getInternalAudioFormat();
+ AudioFormat currentFormat = ringbufferpool_->getInternalAudioFormat();
format.nb_channels = std::max(currentFormat.nb_channels, std::min(format.nb_channels, 2u)); // max 2 channels.
format.sample_rate = std::max(currentFormat.sample_rate, format.sample_rate);
@@ -2261,7 +2263,7 @@ ManagerImpl::audioFormatUsed(AudioFormat format)
DEBUG("Audio format changed: %s -> %s", currentFormat.toString().c_str(), format.toString().c_str());
- mainBuffer_.setInternalAudioFormat(format);
+ ringbufferpool_->setInternalAudioFormat(format);
{
std::lock_guard<std::mutex> toneLock(toneMutex_);
@@ -2707,12 +2709,6 @@ ManagerImpl::getAudioDriver()
return audiodriver_;
}
-MainBuffer &
-ManagerImpl::getMainBuffer()
-{
- return mainBuffer_;
-}
-
Client*
ManagerImpl::getClient()
{
diff --git a/daemon/src/managerimpl.h b/daemon/src/managerimpl.h
index b3ec3598d5ebe722ebdf8f1252535665fd97e511..2bdd17c7c75addff49806dc8a9d447132b117749 100644
--- a/daemon/src/managerimpl.h
+++ b/daemon/src/managerimpl.h
@@ -57,7 +57,6 @@
#include "audio/audiolayer.h"
#include "audio/sound/tone.h" // for Tone::TONEID declaration
#include "audio/codecs/audiocodecfactory.h"
-#include "audio/mainbuffer.h"
#include "preferences.h"
#include "history/history.h"
@@ -73,6 +72,7 @@ class AudioFile;
class AudioLayer;
class History;
class TelephoneTone;
+class RingBufferPool;
/** To send multiple string */
typedef std::list<std::string> TokenList;
@@ -84,6 +84,8 @@ typedef std::set<std::string> CallIDSet;
static const char * const default_conf = "conf";
+typedef std::set<std::string> CallIDSet;
+
/** Manager (controller) of sflphone daemon */
class ManagerImpl {
public:
@@ -838,20 +840,20 @@ class ManagerImpl {
int loadAccountMap(const YAML::Node &node);
/**
- * Instance of the MainBuffer for the whole application
+ * Instance of the RingBufferPool for the whole application
*
- * In order to send signal to other parts of the application, one must pass through the mainbuffer.
- * Audio instances must be registered into the MainBuffer and bound together via the ManagerImpl.
+ * In order to send signal to other parts of the application, one must pass through the RingBufferMananger.
+ * Audio instances must be registered into the RingBufferMananger and bound together via the ManagerImpl.
*
*/
- MainBuffer mainBuffer_;
+ std::unique_ptr<RingBufferPool> ringbufferpool_;
public:
/**
- * Return a pointer to the instance of the mainbuffer
+ * Return a pointer to the instance of the RingBufferPool
*/
- MainBuffer &getMainBuffer();
+ RingBufferPool& getRingBufferPool() { return *ringbufferpool_; }
/**
* Tell if there is a current call processed
diff --git a/daemon/test/Makefile.am b/daemon/test/Makefile.am
index 4186f0984c5e3b7a8a67fb0bb55e539ca8ca4470..094777e82fbb0ee8fff076df5ecc5ed407091c9a 100644
--- a/daemon/test/Makefile.am
+++ b/daemon/test/Makefile.am
@@ -34,8 +34,8 @@ test_SOURCES = constants.h \
siptest.cpp \
sdptest.h \
sdptest.cpp \
- mainbuffertest.h \
- mainbuffertest.cpp \
+ ringbufferpooltest.h \
+ ringbufferpooltest.cpp \
resamplertest.h \
resamplertest.cpp \
hooktest.h \
diff --git a/daemon/test/mainbuffertest.cpp b/daemon/test/mainbuffertest.cpp
deleted file mode 100644
index 448b04e71b21b9014daf55ffa8af907e6a69f92e..0000000000000000000000000000000000000000
--- a/daemon/test/mainbuffertest.cpp
+++ /dev/null
@@ -1,217 +0,0 @@
-/*
- * Copyright (C) 2004-2013 Savoir-Faire Linux Inc.
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#include <string>
-#include <memory>
-
-#include "mainbuffertest.h"
-#include "audio/mainbuffer.h"
-#include "audio/ringbuffer.h"
-#include "logger.h"
-#include "test_utils.h"
-
-void MainBufferTest::testBindUnbindBuffer()
-{
- TITLE();
-
- std::string test_id1 = "bind unbind 1";
- std::string test_id2 = "bind unbind 2";
-
- // bind test_id1 with MainBuffer::DEFAULT_ID (test_id1 not already created)
- mainbuffer_->bindCallID(test_id1, MainBuffer::DEFAULT_ID);
-
- // unbind test_id1 with MainBuffer::DEFAULT_ID
- mainbuffer_->unBindCallID(test_id1, MainBuffer::DEFAULT_ID);
-
- mainbuffer_->bindCallID(test_id1, MainBuffer::DEFAULT_ID);
- mainbuffer_->bindCallID(test_id1, MainBuffer::DEFAULT_ID);
-
- mainbuffer_->bindCallID(test_id2, MainBuffer::DEFAULT_ID);
- mainbuffer_->bindCallID(test_id2, MainBuffer::DEFAULT_ID);
-
- // bind test_id1 with test_id2 (both testid1 and test_id2 already created)
- // calling it twice not supposed to break anything
- mainbuffer_->bindCallID(test_id1, test_id2);
- mainbuffer_->bindCallID(test_id1, test_id2);
-
- mainbuffer_->unBindCallID(test_id1, test_id2);
- mainbuffer_->unBindCallID(test_id1, test_id2);
-
- mainbuffer_->unBindCallID(MainBuffer::DEFAULT_ID, test_id2);
- mainbuffer_->unBindCallID(MainBuffer::DEFAULT_ID, test_id2);
-
- mainbuffer_->unBindCallID(MainBuffer::DEFAULT_ID, test_id1);
-
- // test unbind all function
- mainbuffer_->bindCallID(MainBuffer::DEFAULT_ID, test_id1);
- mainbuffer_->bindCallID(MainBuffer::DEFAULT_ID, test_id2);
- mainbuffer_->bindCallID(test_id1, test_id2);
-
- mainbuffer_->unBindAll(test_id2);
-}
-
-void MainBufferTest::testGetPutData()
-{
- TITLE();
-
- std::string test_id = "incoming rtp session";
-
- auto mainRingBuffer = mainbuffer_->getRingBuffer(MainBuffer::DEFAULT_ID);
- auto testRingBuffer = mainbuffer_->createRingBuffer(test_id);
-
- mainbuffer_->bindCallID(test_id, MainBuffer::DEFAULT_ID);
-
- SFLAudioSample test_sample1 = 12;
- SFLAudioSample test_sample2 = 13;
-
- AudioBuffer test_input1(&test_sample1, 1, AudioFormat::MONO());
- AudioBuffer test_input2(&test_sample2, 1, AudioFormat::MONO());
- AudioBuffer test_output(100, AudioFormat::MONO());
-
- // get by test_id without preleminary put
- CPPUNIT_ASSERT(mainbuffer_->getData(test_output, test_id) == 0);
-
- // put by MainBuffer::DEFAULT_ID, get by test_id
- mainRingBuffer->put(test_input1);
- CPPUNIT_ASSERT(mainbuffer_->getData(test_output, test_id) == 1);
- CPPUNIT_ASSERT(test_sample1 == (*test_output.getChannel(0))[0]);
-
- // get by MainBuffer::DEFAULT_ID without preleminary put
- CPPUNIT_ASSERT(mainbuffer_->getData(test_output, MainBuffer::DEFAULT_ID) == 0);
-
- // put by test_id, get by MainBuffer::DEFAULT_ID
- testRingBuffer->put(test_input2);
- CPPUNIT_ASSERT(mainbuffer_->getData(test_output, MainBuffer::DEFAULT_ID) == 1);
- CPPUNIT_ASSERT(test_sample2 == (*test_output.getChannel(0))[0]);
-
- mainbuffer_->unBindCallID(test_id, MainBuffer::DEFAULT_ID);
-}
-
-void MainBufferTest::testGetAvailableData()
-{
- TITLE();
- std::string test_id = "getData putData";
- std::string false_id = "false id";
-
- auto mainRingBuffer = mainbuffer_->getRingBuffer(MainBuffer::DEFAULT_ID);
- auto testRingBuffer = mainbuffer_->createRingBuffer(test_id);
-
- mainbuffer_->bindCallID(test_id, MainBuffer::DEFAULT_ID);
-
- SFLAudioSample test_sample1 = 12;
- SFLAudioSample test_sample2 = 13;
-
- AudioBuffer test_input1(&test_sample1, 1, AudioFormat::MONO());
- AudioBuffer test_input2(&test_sample2, 1, AudioFormat::MONO());
- AudioBuffer test_output(1, AudioFormat::MONO());
- AudioBuffer test_output_large(100, AudioFormat::MONO());
-
- // put by MainBuffer::DEFAULT_ID get by test_id without preleminary put
- CPPUNIT_ASSERT(mainbuffer_->availableForGet(test_id) == 0);
- CPPUNIT_ASSERT(mainbuffer_->getAvailableData(test_output, test_id) == 0);
-
- // put by MainBuffer::DEFAULT_ID, get by test_id
- mainRingBuffer->put(test_input1);
- CPPUNIT_ASSERT(mainbuffer_->availableForGet(test_id) == 1);
-
- // get by MainBuffer::DEFAULT_ID without preliminary input
- CPPUNIT_ASSERT(mainbuffer_->availableForGet(test_id) == 0);
- CPPUNIT_ASSERT(mainbuffer_->getData(test_output_large, test_id) == 0);
-
- // put by test_id get by test_id
- testRingBuffer->put(test_input2);
- CPPUNIT_ASSERT(mainbuffer_->availableForGet(test_id) == 1);
- CPPUNIT_ASSERT(mainbuffer_->getData(test_output_large, test_id) == 1);
- CPPUNIT_ASSERT(mainbuffer_->availableForGet(test_id) == 0);
- CPPUNIT_ASSERT((*test_output_large.getChannel(0))[0] == test_sample2);
-
- // get by false id
- CPPUNIT_ASSERT(mainbuffer_->getData(test_output_large, false_id) == 0);
-
- mainbuffer_->unBindCallID(test_id, MainBuffer::DEFAULT_ID);
-}
-
-void MainBufferTest::testDiscardFlush()
-{
- TITLE();
- std::string test_id = "flush discard";
-
- auto mainRingBuffer = mainbuffer_->getRingBuffer(MainBuffer::DEFAULT_ID);
- auto testRingBuffer = mainbuffer_->createRingBuffer(test_id);
-
- mainbuffer_->bindCallID(test_id, MainBuffer::DEFAULT_ID);
-
- SFLAudioSample test_sample1 = 12;
- AudioBuffer test_input1(&test_sample1, 1, AudioFormat::MONO());
-
- testRingBuffer->put(test_input1);
- mainbuffer_->discard(1, MainBuffer::DEFAULT_ID);
-
- mainbuffer_->discard(1, test_id);
-
- mainRingBuffer->put(test_input1);
-
- mainbuffer_->discard(1, test_id);
-
- mainbuffer_->unBindCallID(test_id, MainBuffer::DEFAULT_ID);
-}
-
-void MainBufferTest::testConference()
-{
- TITLE();
-
- std::string test_id1 = "participant A";
- std::string test_id2 = "participant B";
-
- auto mainRingBuffer = mainbuffer_->getRingBuffer(MainBuffer::DEFAULT_ID);
- auto testRingBuffer1 = mainbuffer_->createRingBuffer(test_id1);
- auto testRingBuffer2 = mainbuffer_->createRingBuffer(test_id2);
-
- // test bind Participant A with default
- mainbuffer_->bindCallID(test_id1, MainBuffer::DEFAULT_ID);
-
- // test bind Participant B with default
- mainbuffer_->bindCallID(test_id2, MainBuffer::DEFAULT_ID);
-
- // test bind Participant A with Participant B
- mainbuffer_->bindCallID(test_id1, test_id2);
-
- SFLAudioSample testint = 12;
- AudioBuffer testbuf(&testint, 1, AudioFormat::MONO());
-
- // put data test ring buffers
- mainRingBuffer->put(testbuf);
-
- // put data test ring buffers
- testRingBuffer1->put(testbuf);
- testRingBuffer2->put(testbuf);
-}
-
-MainBufferTest::MainBufferTest() : CppUnit::TestCase("Audio Layer Tests"), mainbuffer_(new MainBuffer) {}
diff --git a/daemon/test/ringbufferpooltest.cpp b/daemon/test/ringbufferpooltest.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..94bc01057a2545c9db4109a5ecfbb79362ef1fe3
--- /dev/null
+++ b/daemon/test/ringbufferpooltest.cpp
@@ -0,0 +1,219 @@
+/*
+ * Copyright (C) 2004-2013 Savoir-Faire Linux Inc.
+ * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ *
+ * Additional permission under GNU GPL version 3 section 7:
+ *
+ * If you modify this program, or any covered work, by linking or
+ * combining it with the OpenSSL project's OpenSSL library (or a
+ * modified version of that library), containing parts covered by the
+ * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
+ * grants you additional permission to convey the resulting work.
+ * Corresponding Source for a non-source form of such a combination
+ * shall include the source code for the parts of OpenSSL used as well
+ * as that of the covered work.
+ */
+
+#include <string>
+#include <memory>
+
+#include "ringbufferpooltest.h"
+#include "audio/ringbufferpool.h"
+#include "audio/ringbuffer.h"
+#include "logger.h"
+#include "test_utils.h"
+
+void RingBufferPoolTest::testBindUnbindBuffer()
+{
+ TITLE();
+
+ std::string test_id1 = "bind unbind 1";
+ std::string test_id2 = "bind unbind 2";
+
+ // bind test_id1 with RingBufferPool::DEFAULT_ID (test_id1 not already created)
+ rbPool_->bindCallID(test_id1, RingBufferPool::DEFAULT_ID);
+
+ // unbind test_id1 with RingBufferPool::DEFAULT_ID
+ rbPool_->unBindCallID(test_id1, RingBufferPool::DEFAULT_ID);
+
+ rbPool_->bindCallID(test_id1, RingBufferPool::DEFAULT_ID);
+ rbPool_->bindCallID(test_id1, RingBufferPool::DEFAULT_ID);
+
+ rbPool_->bindCallID(test_id2, RingBufferPool::DEFAULT_ID);
+ rbPool_->bindCallID(test_id2, RingBufferPool::DEFAULT_ID);
+
+ // bind test_id1 with test_id2 (both testid1 and test_id2 already created)
+ // calling it twice not supposed to break anything
+ rbPool_->bindCallID(test_id1, test_id2);
+ rbPool_->bindCallID(test_id1, test_id2);
+
+ rbPool_->unBindCallID(test_id1, test_id2);
+ rbPool_->unBindCallID(test_id1, test_id2);
+
+ rbPool_->unBindCallID(RingBufferPool::DEFAULT_ID, test_id2);
+ rbPool_->unBindCallID(RingBufferPool::DEFAULT_ID, test_id2);
+
+ rbPool_->unBindCallID(RingBufferPool::DEFAULT_ID, test_id1);
+
+ // test unbind all function
+ rbPool_->bindCallID(RingBufferPool::DEFAULT_ID, test_id1);
+ rbPool_->bindCallID(RingBufferPool::DEFAULT_ID, test_id2);
+ rbPool_->bindCallID(test_id1, test_id2);
+
+ rbPool_->unBindAll(test_id2);
+}
+
+void RingBufferPoolTest::testGetPutData()
+{
+ TITLE();
+
+ std::string test_id = "incoming rtp session";
+
+ auto mainRingBuffer = rbPool_->getRingBuffer(RingBufferPool::DEFAULT_ID);
+ auto testRingBuffer = rbPool_->createRingBuffer(test_id);
+
+ rbPool_->bindCallID(test_id, RingBufferPool::DEFAULT_ID);
+
+ SFLAudioSample test_sample1 = 12;
+ SFLAudioSample test_sample2 = 13;
+
+ AudioBuffer test_input1(&test_sample1, 1, AudioFormat::MONO());
+ AudioBuffer test_input2(&test_sample2, 1, AudioFormat::MONO());
+ AudioBuffer test_output(100, AudioFormat::MONO());
+
+ // get by test_id without preleminary put
+ CPPUNIT_ASSERT(rbPool_->getData(test_output, test_id) == 0);
+
+ // put by RingBufferPool::DEFAULT_ID, get by test_id
+ mainRingBuffer->put(test_input1);
+ CPPUNIT_ASSERT(rbPool_->getData(test_output, test_id) == 1);
+ CPPUNIT_ASSERT(test_sample1 == (*test_output.getChannel(0))[0]);
+
+ // get by RingBufferPool::DEFAULT_ID without preleminary put
+ CPPUNIT_ASSERT(rbPool_->getData(test_output, RingBufferPool::DEFAULT_ID) == 0);
+
+ // put by test_id, get by RingBufferPool::DEFAULT_ID
+ testRingBuffer->put(test_input2);
+ CPPUNIT_ASSERT(rbPool_->getData(test_output, RingBufferPool::DEFAULT_ID) == 1);
+ CPPUNIT_ASSERT(test_sample2 == (*test_output.getChannel(0))[0]);
+
+ rbPool_->unBindCallID(test_id, RingBufferPool::DEFAULT_ID);
+}
+
+void RingBufferPoolTest::testGetAvailableData()
+{
+ TITLE();
+ std::string test_id = "getData putData";
+ std::string false_id = "false id";
+
+ auto mainRingBuffer = rbPool_->getRingBuffer(RingBufferPool::DEFAULT_ID);
+ auto testRingBuffer = rbPool_->createRingBuffer(test_id);
+
+ rbPool_->bindCallID(test_id, RingBufferPool::DEFAULT_ID);
+
+ SFLAudioSample test_sample1 = 12;
+ SFLAudioSample test_sample2 = 13;
+
+ AudioBuffer test_input1(&test_sample1, 1, AudioFormat::MONO());
+ AudioBuffer test_input2(&test_sample2, 1, AudioFormat::MONO());
+ AudioBuffer test_output(1, AudioFormat::MONO());
+ AudioBuffer test_output_large(100, AudioFormat::MONO());
+
+ // put by RingBufferPool::DEFAULT_ID get by test_id without preleminary put
+ CPPUNIT_ASSERT(rbPool_->availableForGet(test_id) == 0);
+ CPPUNIT_ASSERT(rbPool_->getAvailableData(test_output, test_id) == 0);
+
+ // put by RingBufferPool::DEFAULT_ID, get by test_id
+ mainRingBuffer->put(test_input1);
+ CPPUNIT_ASSERT(rbPool_->availableForGet(test_id) == 1);
+
+ // get by RingBufferPool::DEFAULT_ID without preliminary input
+ CPPUNIT_ASSERT(rbPool_->availableForGet(test_id) == 0);
+ CPPUNIT_ASSERT(rbPool_->getData(test_output_large, test_id) == 0);
+
+ // put by test_id get by test_id
+ testRingBuffer->put(test_input2);
+ CPPUNIT_ASSERT(rbPool_->availableForGet(test_id) == 1);
+ CPPUNIT_ASSERT(rbPool_->getData(test_output_large, test_id) == 1);
+ CPPUNIT_ASSERT(rbPool_->availableForGet(test_id) == 0);
+ CPPUNIT_ASSERT((*test_output_large.getChannel(0))[0] == test_sample2);
+
+ // get by false id
+ CPPUNIT_ASSERT(rbPool_->getData(test_output_large, false_id) == 0);
+
+ rbPool_->unBindCallID(test_id, RingBufferPool::DEFAULT_ID);
+}
+
+void RingBufferPoolTest::testDiscardFlush()
+{
+ TITLE();
+ std::string test_id = "flush discard";
+
+ auto mainRingBuffer = rbPool_->getRingBuffer(RingBufferPool::DEFAULT_ID);
+ auto testRingBuffer = rbPool_->createRingBuffer(test_id);
+
+ rbPool_->bindCallID(test_id, RingBufferPool::DEFAULT_ID);
+
+ SFLAudioSample test_sample1 = 12;
+ AudioBuffer test_input1(&test_sample1, 1, AudioFormat::MONO());
+
+ testRingBuffer->put(test_input1);
+ rbPool_->discard(1, RingBufferPool::DEFAULT_ID);
+
+ rbPool_->discard(1, test_id);
+
+ mainRingBuffer->put(test_input1);
+
+ rbPool_->discard(1, test_id);
+
+ rbPool_->unBindCallID(test_id, RingBufferPool::DEFAULT_ID);
+}
+
+void RingBufferPoolTest::testConference()
+{
+ TITLE();
+
+ std::string test_id1 = "participant A";
+ std::string test_id2 = "participant B";
+
+ auto mainRingBuffer = rbPool_->getRingBuffer(RingBufferPool::DEFAULT_ID);
+ auto testRingBuffer1 = rbPool_->createRingBuffer(test_id1);
+ auto testRingBuffer2 = rbPool_->createRingBuffer(test_id2);
+
+ // test bind Participant A with default
+ rbPool_->bindCallID(test_id1, RingBufferPool::DEFAULT_ID);
+
+ // test bind Participant B with default
+ rbPool_->bindCallID(test_id2, RingBufferPool::DEFAULT_ID);
+
+ // test bind Participant A with Participant B
+ rbPool_->bindCallID(test_id1, test_id2);
+
+ SFLAudioSample testint = 12;
+ AudioBuffer testbuf(&testint, 1, AudioFormat::MONO());
+
+ // put data test ring buffers
+ mainRingBuffer->put(testbuf);
+
+ // put data test ring buffers
+ testRingBuffer1->put(testbuf);
+ testRingBuffer2->put(testbuf);
+}
+
+RingBufferPoolTest::RingBufferPoolTest()
+ : CppUnit::TestCase("Audio Layer Tests") , rbPool_(new RingBufferPool)
+{}
diff --git a/daemon/test/mainbuffertest.h b/daemon/test/ringbufferpooltest.h
similarity index 83%
rename from daemon/test/mainbuffertest.h
rename to daemon/test/ringbufferpooltest.h
index ee342c118459e8493aa65993a94faae95ac1925f..68d798f5c2306fd6e6ca0fd4c05c8b2d9953d54e 100644
--- a/daemon/test/mainbuffertest.h
+++ b/daemon/test/ringbufferpooltest.h
@@ -28,8 +28,8 @@
* as that of the covered work.
*/
-#ifndef MAINBUFFER_TEST_
-#define MAINBUFFER_TEST_
+#ifndef RINGBUFFERPOOL_TEST_
+#define RINGBUFFERPOOL_TEST_
// Cppunit import
#include <cppunit/extensions/HelperMacros.h>
@@ -39,18 +39,19 @@
#include <memory>
-class MainBuffer;
+class RingBufferPool;
+
/*
* @file audiorecorderTest.cpp
* @brief Regroups unit tests related to the main buffer.
*/
-class MainBufferTest : public CppUnit::TestCase {
+class RingBufferPoolTest : public CppUnit::TestCase {
/*
* Use cppunit library macros to add unit test the factory
*/
- CPPUNIT_TEST_SUITE(MainBufferTest);
+ CPPUNIT_TEST_SUITE(RingBufferPoolTest);
CPPUNIT_TEST(testBindUnbindBuffer);
CPPUNIT_TEST(testGetPutData);
CPPUNIT_TEST(testDiscardFlush);
@@ -59,7 +60,7 @@ class MainBufferTest : public CppUnit::TestCase {
public:
- MainBufferTest();
+ RingBufferPoolTest();
void testBindUnbindBuffer();
@@ -73,11 +74,11 @@ class MainBufferTest : public CppUnit::TestCase {
private:
- std::unique_ptr<MainBuffer> mainbuffer_;
+ std::unique_ptr<RingBufferPool> rbPool_;
};
/* Register our test module */
-CPPUNIT_TEST_SUITE_NAMED_REGISTRATION(MainBufferTest, "MainBufferTest");
-CPPUNIT_TEST_SUITE_REGISTRATION(MainBufferTest);
+CPPUNIT_TEST_SUITE_NAMED_REGISTRATION(RingBufferPoolTest, "RingBufferPoolTest");
+CPPUNIT_TEST_SUITE_REGISTRATION(RingBufferPoolTest);
-#endif // MAINBUFFER_TEST_
+#endif // RINGBUFFERPOOL_TEST_