switch to c++11 mutex and lock for MainBuffer

daemon/src/audio/audiobuffer.cpp

@@ -122,6 +122,9 @@ void AudioBuffer::applyGain(double gain)
     for (auto &channel : samples_)
         for (auto &sample : channel)
             sample *= g;
+/*    for (unsigned i=0, f=frames(), c=channels(); i < c; ++i)
+        for (unsigned j = 0; j < f; ++j)
+            samples_[i][j] *= g;*/
 }
 
 size_t AudioBuffer::interleave(SFLAudioSample* out) const
@@ -135,8 +138,8 @@ size_t AudioBuffer::interleave(SFLAudioSample* out) const
 
 size_t AudioBuffer::interleaveFloat(float* out) const
 {
-    for (unsigned i = 0; i < frames(); i++)
-        for (unsigned j = 0; j < samples_.size(); j++)
+    for (unsigned i=0, f=frames(), c=channels(); i < f; i++)
+        for (unsigned j = 0; j < c; j++)
             *out++ = (float) samples_[j][i] * .000030517578125f;
 
     return frames() * samples_.size();

daemon/src/audio/mainbuffer.cpp

@@ -31,17 +31,17 @@
 #include "mainbuffer.h"
 #include "ringbuffer.h"
 #include "sfl_types.h" // for SIZEBUF
-#include <climits>
+#include <limits>
 #include <cstring>
 #include <utility> // for std::pair
-#include "scoped_lock.h"
+//#include "scoped_lock.h"
 #include "logger.h"
 
 const char * const MainBuffer::DEFAULT_ID = "audiolayer_id";
 
 MainBuffer::MainBuffer() : ringBufferMap_(), callIDMap_(), mutex_(), internalSamplingRate_(8000)
 {
-    pthread_mutex_init(&mutex_, NULL);
+    //pthread_mutex_init(&mutex_, NULL);
 }
 
 MainBuffer::~MainBuffer()
@@ -50,7 +50,7 @@ MainBuffer::~MainBuffer()
     for (auto &item : ringBufferMap_)
         delete item.second;
 
-    pthread_mutex_destroy(&mutex_);
+    //pthread_mutex_destroy(&mutex_);
 }
 
 void MainBuffer::setInternalSamplingRate(int sr)
@@ -139,7 +139,7 @@ void MainBuffer::removeRingBuffer(const std::string &call_id)
 
 void MainBuffer::bindCallID(const std::string & call_id1, const std::string & call_id2)
 {
-    sfl::ScopedLock guard(mutex_);
+    std::lock_guard<std::mutex> guard(mutex_);
 
     createRingBuffer(call_id1);
     createCallIDSet(call_id1);
@@ -154,7 +154,7 @@ void MainBuffer::bindCallID(const std::string & call_id1, const std::string & ca
 
 void MainBuffer::bindHalfDuplexOut(const std::string & process_id, const std::string & call_id)
 {
-    sfl::ScopedLock guard(mutex_);
+    std::lock_guard<std::mutex> guard(mutex_);
 
     // This method is used only for active calls, if this call does not exist, do nothing
     if (!getRingBuffer(call_id))
@@ -167,7 +167,7 @@ void MainBuffer::bindHalfDuplexOut(const std::string & process_id, const std::st
 
 void MainBuffer::unBindCallID(const std::string & call_id1, const std::string & call_id2)
 {
-    sfl::ScopedLock guard(mutex_);
+    std::lock_guard<std::mutex> guard(mutex_);
 
     removeCallIDfromSet(call_id1, call_id2);
     removeCallIDfromSet(call_id2, call_id1);
@@ -197,7 +197,7 @@ void MainBuffer::unBindCallID(const std::string & call_id1, const std::string &
 
 void MainBuffer::unBindHalfDuplexOut(const std::string & process_id, const std::string & call_id)
 {
-    sfl::ScopedLock guard(mutex_);
+    std::lock_guard<std::mutex> guard(mutex_);
 
     removeCallIDfromSet(process_id, call_id);
 
@@ -237,7 +237,7 @@ void MainBuffer::unBindAll(const std::string & call_id)
 //void MainBuffer::putData(void *buffer, size_t toCopy, const std::string &call_id)
 void MainBuffer::putData(AudioBuffer& buffer, const std::string &call_id)
 {
-    sfl::ScopedLock guard(mutex_);
+    std::lock_guard<std::mutex> guard(mutex_);
 
     RingBuffer* ring_buffer = getRingBuffer(call_id);
 
@@ -248,7 +248,7 @@ void MainBuffer::putData(AudioBuffer& buffer, const std::string &call_id)
 //size_t MainBuffer::getData(void *buffer, size_t toCopy, const std::string &call_id)
 size_t MainBuffer::getData(AudioBuffer& buffer, const std::string &call_id)
 {
-    sfl::ScopedLock guard(mutex_);
+    std::lock_guard<std::mutex> guard(mutex_);
 
     CallIDSet* callid_set = getCallIDSet(call_id);
 
@@ -296,7 +296,7 @@ size_t MainBuffer::getDataByID(AudioBuffer& buffer, const std::string & call_id,
 
 size_t MainBuffer::availableForGet(const std::string &call_id)
 {
-    sfl::ScopedLock guard(mutex_);
+    std::lock_guard<std::mutex> guard(mutex_);
 
     CallIDSet* callid_set = getCallIDSet(call_id);
 
@@ -313,7 +313,7 @@ size_t MainBuffer::availableForGet(const std::string &call_id)
 
     } else {
 
-        size_t availableSamples = INT_MAX;
+        size_t availableSamples = std::numeric_limits<int>::max();
 
         for (auto &i : *callid_set) {
             const size_t nbSamples = availableForGetByID(i, call_id);
@@ -344,7 +344,7 @@ size_t MainBuffer::availableForGetByID(const std::string &call_id,
 
 size_t MainBuffer::discard(size_t toDiscard, const std::string &call_id)
 {
-    sfl::ScopedLock guard(mutex_);
+    std::lock_guard<std::mutex> guard(mutex_);
 
     CallIDSet* callid_set = getCallIDSet(call_id);
 
@@ -367,7 +367,7 @@ void MainBuffer::discardByID(size_t toDiscard, const std::string & call_id, cons
 
 void MainBuffer::flush(const std::string & call_id)
 {
-    sfl::ScopedLock guard(mutex_);
+    std::lock_guard<std::mutex> guard(mutex_);
 
     CallIDSet* callid_set = getCallIDSet(call_id);
 
@@ -396,7 +396,7 @@ void MainBuffer::flushAllBuffers()
 void MainBuffer::dumpInfo()
 {
 #if 0
-    sfl::ScopedLock guard(mutex_);
+    std::lock_guard<std::mutex> guard(mutex_);
 
     // print each call and bound call ids
     for (const auto &item_call : callIDMap_) {

daemon/src/audio/mainbuffer.h

@@ -34,7 +34,7 @@
 #include <map>
 #include <set>
 #include <string>
-#include <pthread.h>
+#include <mutex>
 
 #include "audiobuffer.h"
 #include "mainbuffer.h"
@@ -84,10 +84,8 @@ class MainBuffer {
 
         void unBindAll(const std::string &call_id);
 
-        //void putData(void *buffer, size_t toCopy, const std::string &call_id);
         void putData(AudioBuffer& buffer, const std::string &call_id);
 
-        //size_t getData(void *buffer, size_t toCopy, const std::string &call_id);
         size_t getData(AudioBuffer& buffer, const std::string &call_id);
 
         size_t availableForGet(const std::string &call_id);
@@ -126,7 +124,6 @@ class MainBuffer {
         RingBuffer* getRingBuffer(const std::string &call_id);
         const RingBuffer* getRingBuffer(const std::string & call_id) const;
 
-        //size_t getDataByID(void *buffer, size_t toCopy, const std::string &call_id, const std::string &reader_id);
         size_t getDataByID(AudioBuffer& buffer, const std::string &call_id, const std::string &reader_id);
 
         size_t availableForGetByID(const std::string &call_id, const std::string &reader_id) const;
@@ -141,7 +138,7 @@ class MainBuffer {
         typedef std::map<std::string, CallIDSet*> CallIDMap;
         CallIDMap callIDMap_;
 
-        pthread_mutex_t mutex_;
+        std::mutex mutex_;
 
         int internalSamplingRate_;
 

daemon/src/audio/opensl/opensllayer.cpp

@@ -747,7 +747,7 @@ bool OpenSLLayer::audioPlaybackFillWithToneOrRingtone(AudioBuffer &buffer)
 bool OpenSLLayer::audioPlaybackFillWithUrgent(AudioBuffer &buffer, size_t samplesToGet)
 {
     // Urgent data (dtmf, incoming call signal) come first.
-    samplesToGet = std::min(samplesToGet, buffer.frames());
+    samplesToGet = std::min(samplesToGet, BUFFER_SIZE);
     buffer.resize(samplesToGet);
     urgentRingBuffer_.get(buffer, MainBuffer::DEFAULT_ID);
     buffer.applyGain(playbackGain_);
@@ -776,7 +776,7 @@ bool OpenSLLayer::audioPlaybackFillWithVoice(AudioBuffer &buffer, size_t samples
         AudioBuffer out(buffer, false);
         out.setSampleRate(sampleRate_);
         converter_.resample(buffer, out);
-        buffer = std::move(out);
+        buffer = out;
     }
 
     return true;

daemon/src/audio/opensl/opensllayer.h

@@ -34,7 +34,6 @@
 #include <SLES/OpenSLES.h>
 #include <SLES/OpenSLES_Android.h>
 #include <vector>
-#include <atomic>
 
 #include "../audiolayer.h"
 #include "logger.h"
@@ -53,8 +52,8 @@ class AudioPreference;
 
 class OpenSLThread;
 
-#define ANDROID_BUFFER_QUEUE_LENGTH 2
-#define BUFFER_SIZE 80
+#define ANDROID_BUFFER_QUEUE_LENGTH 2U
+#define BUFFER_SIZE 80U
 
 
 /**

daemon/src/audio/pulseaudio/pulselayer.cpp

@@ -109,7 +109,7 @@ PulseLayer::PulseLayer(AudioPreference &pref)
 
     pa_context_set_state_callback(context_, context_state_callback, this);
 
-    if (pa_context_connect(context_, NULL , PA_CONTEXT_NOAUTOSPAWN , NULL) < 0)
+    if (pa_context_connect(context_, nullptr , PA_CONTEXT_NOAUTOSPAWN , nullptr) < 0)
         throw std::runtime_error("Could not connect pulseaudio context to the server");
 
     pa_threaded_mainloop_lock(mainloop_);
@@ -165,7 +165,7 @@ void PulseLayer::context_state_callback(pa_context* c, void *user_data)
         case PA_CONTEXT_READY:
             DEBUG("Connection to PulseAudio server established");
             pa_threaded_mainloop_signal(pulse->mainloop_, 0);
-            pa_context_subscribe(c, mask, NULL, pulse);
+            pa_context_subscribe(c, mask, nullptr, pulse);
             pa_context_set_subscribe_callback(c, context_changed_callback, pulse);
             pulse->updateSinkList();
             pulse->updateSourceList();
@@ -190,7 +190,7 @@ void PulseLayer::updateSinkList()
     enumeratingSinks_ = true;
     pa_operation *op = pa_context_get_sink_info_list(context_, sink_input_info_callback, this);
 
-    if (op != NULL)
+    if (op != nullptr)
         pa_operation_unref(op);
 }
 
@@ -200,7 +200,7 @@ void PulseLayer::updateSourceList()
     enumeratingSources_ = true;
     pa_operation *op = pa_context_get_source_info_list(context_, source_input_info_callback, this);
 
-    if (op != NULL)
+    if (op != nullptr)
         pa_operation_unref(op);
 }
 
@@ -257,7 +257,7 @@ const PaDeviceInfos* PulseLayer::getDeviceInfos(const std::vector<PaDeviceInfos>
 {
     std::vector<PaDeviceInfos>::const_iterator dev_info = std::find_if(list.begin(), list.end(), PaDeviceInfos::nameComparator(name));
 
-    if (dev_info == list.end()) return NULL;
+    if (dev_info == list.end()) return nullptr;
 
     return &(*dev_info);
 }
@@ -304,7 +304,7 @@ void PulseLayer::createStreams(pa_context* c)
     // Create playback stream
     const PaDeviceInfos* dev_infos = getDeviceInfos(sinkList_, playbackDevice);
 
-    if (dev_infos == NULL) {
+    if (dev_infos == nullptr) {
         dev_infos = &sinkList_[0];
         DEBUG("Prefered playback device not found in device list, selecting %s instead.", dev_infos->name.c_str());
     }
@@ -317,7 +317,7 @@ void PulseLayer::createStreams(pa_context* c)
     // Create capture stream
     dev_infos = getDeviceInfos(sourceList_, captureDevice);
 
-    if (dev_infos == NULL) {
+    if (dev_infos == nullptr) {
         dev_infos = &sourceList_[0];
         DEBUG("Prefered capture device not found in device list, selecting %s instead.", dev_infos->name.c_str());
     }
@@ -330,7 +330,7 @@ void PulseLayer::createStreams(pa_context* c)
     // Create ringtone stream
     dev_infos = getDeviceInfos(sinkList_, ringtoneDevice);
 
-    if (dev_infos == NULL) {
+    if (dev_infos == nullptr) {
         dev_infos = &sinkList_[0];
         DEBUG("Prefered ringtone device not found in device list, selecting %s instead.", dev_infos->name.c_str());
     }
@@ -383,10 +383,10 @@ PulseLayer::stopStream()
     pa_threaded_mainloop_lock(mainloop_);
 
     if (playback_)
-        pa_stream_flush(playback_->pulseStream(), NULL, NULL);
+        pa_stream_flush(playback_->pulseStream(), nullptr, nullptr);
 
     if (record_)
-        pa_stream_flush(record_->pulseStream(), NULL, NULL);
+        pa_stream_flush(record_->pulseStream(), nullptr, nullptr);
 
     pa_threaded_mainloop_unlock(mainloop_);
 
@@ -433,7 +433,7 @@ void PulseLayer::writeToSpeaker()
         urgentRingBuffer_.get(linearbuff, MainBuffer::DEFAULT_ID); // retrive only the first sample_spec->channels channels
         linearbuff.applyGain(playbackGain_);
         linearbuff.interleave(data);
-        pa_stream_write(s, data, urgentBytes, NULL, 0, PA_SEEK_RELATIVE);
+        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);
         return;
@@ -449,7 +449,7 @@ void PulseLayer::writeToSpeaker()
             AudioBuffer linearbuff(writableSamples, n_channels);
             toneToPlay->getNext(linearbuff, playbackGain_); // retrive only n_channels
             linearbuff.interleave(data);
-            pa_stream_write(s, data, writableBytes, NULL, 0, PA_SEEK_RELATIVE);
+            pa_stream_write(s, data, writableBytes, nullptr, 0, PA_SEEK_RELATIVE);
         }
 
         return;
@@ -462,7 +462,7 @@ void PulseLayer::writeToSpeaker()
     if (availSamples == 0) {
         pa_stream_begin_write(s, (void**)&data, &writableBytes);
         memset(data, 0, writableBytes);
-        pa_stream_write(s, data, writableBytes, NULL, 0, PA_SEEK_RELATIVE);
+        pa_stream_write(s, data, writableBytes, nullptr, 0, PA_SEEK_RELATIVE);
         return;
     }
 
@@ -493,11 +493,11 @@ void PulseLayer::writeToSpeaker()
         converter_.resample(linearbuff, rsmpl_out);
         rsmpl_out.applyGain(playbackGain_);
         rsmpl_out.interleave(data);
-        pa_stream_write(s, data, resampledBytes, NULL, 0, PA_SEEK_RELATIVE);
+        pa_stream_write(s, data, resampledBytes, nullptr, 0, PA_SEEK_RELATIVE);
     } else {
         linearbuff.applyGain(playbackGain_);
         linearbuff.interleave(data);
-        pa_stream_write(s, data, resampledBytes, NULL, 0, PA_SEEK_RELATIVE);
+        pa_stream_write(s, data, resampledBytes, nullptr, 0, PA_SEEK_RELATIVE);
     }
 }
 
@@ -506,7 +506,7 @@ void PulseLayer::readFromMic()
     if (!record_ or !record_->isReady())
         return;
 
-    const char *data = NULL;
+    const char *data = nullptr;
     size_t bytes;
 
     const size_t sample_size = record_->sampleSize();
@@ -583,7 +583,7 @@ void PulseLayer::ringtoneToSpeaker()
         memset(data, 0, bytes);
     }
 
-    pa_stream_write(s, data, bytes, NULL, 0, PA_SEEK_RELATIVE);
+    pa_stream_write(s, data, bytes, nullptr, 0, PA_SEEK_RELATIVE);
 }
 
 void
@@ -604,7 +604,7 @@ PulseLayer::context_changed_callback(pa_context* c,
                     context->sinkList_.clear();
                     op = pa_context_get_sink_info_list(c, sink_input_info_callback, userdata);
 
-                    if (op != NULL)
+                    if (op != nullptr)
                         pa_operation_unref(op);
 
                 default:
@@ -621,7 +621,7 @@ PulseLayer::context_changed_callback(pa_context* c,
                     context->sourceList_.clear();
                     op = pa_context_get_source_info_list(c, source_input_info_callback, userdata);
 
-                    if (op != NULL)
+                    if (op != nullptr)
                         pa_operation_unref(op);
 
                 default:

daemon/src/audio/pulseaudio/pulselayer.h

@@ -48,7 +48,7 @@ class AudioStream;
 /**
  * Convenience structure to hold PulseAudio device propreties such as supported channel number etc.
  */
-typedef struct PaDeviceInfos {
+typedef struct {
         uint32_t index;
         std::string name;
         pa_sample_spec sample_spec;
@@ -150,17 +150,17 @@ class PulseLayer : public AudioLayer {
         AudioStream* ringtone_;
 
         /**
-         * Contain the list of playback devices
+         * Contains the list of playback devices
          */
         std::vector<PaDeviceInfos> sinkList_;
 
         /**
-         * Contain the list of capture devices
+         * Contains the list of capture devices