diff --git a/src/manager.cpp b/src/manager.cpp
index 8f54cb90a85389139ea1566f609a00050fb1cdc6..04f4952856e488a7d9e9bf4e8251deb61976583b 100644
--- a/src/manager.cpp
+++ b/src/manager.cpp
@@ -903,7 +903,6 @@ Manager::outgoingCall(const std::string& account_id,
JAMI_DBG() << "try outgoing call to '" << to << "'"
<< " with account '" << account_id << "'";
- const auto& current_call_id(getCurrentCallId());
std::string to_cleaned = hookPreference.getNumberAddPrefix() + trim(to);
std::shared_ptr<Call> call;
@@ -939,12 +938,9 @@ Manager::answerCall(const std::string& call_id)
return false;
}
- // If ring is ringing
+ // If ringing
stopTone();
- // store the current call id
- const auto& current_call_id(getCurrentCallId());
-
try {
call->answer();
} catch (const std::runtime_error &e) {
diff --git a/src/media/audio/audio_receive_thread.h b/src/media/audio/audio_receive_thread.h
index 418efcb5b7089b6ad9e88854b5854d9a2e5999de..57af79ecbe225d1b49555683b8ef4fb048fc50bb 100644
--- a/src/media/audio/audio_receive_thread.h
+++ b/src/media/audio/audio_receive_thread.h
@@ -18,6 +18,7 @@
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
+#pragma once
#include "audiobuffer.h"
#include "media_buffer.h"
diff --git a/src/media/audio/audio_rtp_session.h b/src/media/audio/audio_rtp_session.h
index faac7acd0e5f5314653c3a8fcbb6e3f48a0fa7b7..88c2bd564be5c79816aa535e2f318a7b6033eb22 100644
--- a/src/media/audio/audio_rtp_session.h
+++ b/src/media/audio/audio_rtp_session.h
@@ -19,13 +19,11 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-#ifndef AUDIO_RTP_SESSION_H__
-#define AUDIO_RTP_SESSION_H__
+#pragma once
#include "audiobuffer.h"
#include "media_device.h"
#include "rtp_session.h"
-#include "threadloop.h"
#include <string>
#include <memory>
@@ -70,5 +68,3 @@ class AudioRtpSession : public RtpSession {
};
} // namespace jami
-
-#endif // __AUDIO_RTP_SESSION_H__
diff --git a/src/media/audio/audio_sender.h b/src/media/audio/audio_sender.h
index 2d2b123bca8adfed691c0fe4d31e09a766ba474e..10452121bcfa6bf0d4bbcb9d1247ae36b5ef89c2 100644
--- a/src/media/audio/audio_sender.h
+++ b/src/media/audio/audio_sender.h
@@ -18,6 +18,7 @@
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
+#pragma once
#include "audiobuffer.h"
#include "media_buffer.h"
diff --git a/src/media/audio/audiobuffer.h b/src/media/audio/audiobuffer.h
index ee13a2f17e44c6382811f8032b2f920a8e70b1ee..61b7119a6718c6b7c23ccae7447d3eb689ea17dd 100644
--- a/src/media/audio/audiobuffer.h
+++ b/src/media/audio/audiobuffer.h
@@ -18,8 +18,7 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-#ifndef _AUDIO_BUFFER_H
-#define _AUDIO_BUFFER_H
+#pragma once
#if __cplusplus >= 201103L
#undef isblank
@@ -376,4 +375,3 @@ class AudioBuffer {
} // namespace jami
-#endif // _AUDIO_BUFFER_H
diff --git a/src/media/audio/audiolayer.h b/src/media/audio/audiolayer.h
index 1e42a1fb303f592cf24702a7f6d9b88ab4c77a51..3e6b36e2f27deea1f52131834853b98a8c20aeca 100644
--- a/src/media/audio/audiolayer.h
+++ b/src/media/audio/audiolayer.h
@@ -20,10 +20,7 @@
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-
-#ifndef AUDIO_LAYER_H_
-#define AUDIO_LAYER_H_
-
+#pragma once
#include "ringbuffer.h"
#include "dcblocker.h"
@@ -65,265 +62,263 @@ enum class DeviceType {
class AudioLayer {
- private:
- NON_COPYABLE(AudioLayer);
-
- protected:
- enum class Status {
- Idle,
- Starting,
- Started
- };
-
- public:
-
- AudioLayer(const AudioPreference &);
- virtual ~AudioLayer();
-
- virtual std::vector<std::string> getCaptureDeviceList() const = 0;
- virtual std::vector<std::string> getPlaybackDeviceList() const = 0;
-
- virtual int getAudioDeviceIndex(const std::string& name, DeviceType type) const = 0;
- virtual std::string getAudioDeviceName(int index, DeviceType type) const = 0;
- virtual int getIndexCapture() const = 0;
- virtual int getIndexPlayback() const = 0;
- virtual int getIndexRingtone() const = 0;
-
- /**
- * Start the capture stream and prepare the playback stream.
- * The playback starts accordingly to its threshold
- * ALSA Library API
- */
- virtual void startStream() = 0;
-
- /**
- * Stop the playback and capture streams.
- * Drops the pending frames and put the capture and playback handles to PREPARED state
- * ALSA Library API
- */
- virtual void stopStream() = 0;
-
- /**
- * Determine wether or not the audio layer is active (i.e. stream opened)
- */
- inline bool isStarted() const {
- return status_ == Status::Started;
- }
-
- template< class Rep, class Period >
- bool waitForStart(const std::chrono::duration<Rep, Period>& rel_time) const {
- std::unique_lock<std::mutex> lk(mutex_);
- startedCv_.wait_for(lk, rel_time, [&]{return isStarted();});
- return isStarted();
- }
-
- /**
- * Send a chunk of data to the hardware buffer to start the playback
- * Copy data in the urgent buffer.
- * @param buffer The buffer containing the data to be played ( ringtones )
- */
- void putUrgent(AudioBuffer& buffer);
-
- /**
- * Flush main buffer
- */
- void flushMain();
-
- /**
- * Flush urgent buffer
- */
- void flushUrgent();
-
- bool isCaptureMuted() const {
- return isCaptureMuted_;
- }
-
- /**
- * Mute capture (microphone)
- */
- void muteCapture(bool muted) {
- isCaptureMuted_ = muted;
- }
-
- bool isPlaybackMuted() const {
- return isPlaybackMuted_;
- }
-
- /**
- * Mute playback
- */
- void mutePlayback(bool muted) {
- isPlaybackMuted_ = muted;
- }
-
- bool isRingtoneMuted() const {
- return isRingtoneMuted_;
- }
- void muteRingtone(bool muted) {
- isRingtoneMuted_ = muted;
- }
-
- /**
- * Set capture stream gain (microphone)
- * Range should be [-1.0, 1.0]
- */
- void setCaptureGain(double gain) {
- captureGain_ = gain;
- }
-
- /**
- * Get capture stream gain (microphone)
- */
- double getCaptureGain() const {
- return captureGain_;
- }
-
- /**
- * Set playback stream gain (speaker)
- * Range should be [-1.0, 1.0]
- */
- void setPlaybackGain(double gain) {
- playbackGain_ = gain;
- }
-
- /**
- * Get playback stream gain (speaker)
- */
- double getPlaybackGain() const {
- return playbackGain_;
- }
-
- /**
- * Get the sample rate of the audio layer
- * @return unsigned int The sample rate
- * default: 44100 HZ
- */
- unsigned int getSampleRate() const {
- return audioFormat_.sample_rate;
- }
-
- /**
- * Get the audio format of the layer (sample rate & channel number).
- */
- AudioFormat getFormat() const {
- return audioFormat_;
- }
-
- /**
- * Emit an audio notification on incoming calls
- */
- void notifyIncomingCall();
-
- virtual void updatePreference(AudioPreference &pref, int index, DeviceType type) = 0;
-
- protected:
- /**
- * Callback to be called by derived classes when the audio output is opened.
- */
- void hardwareFormatAvailable(AudioFormat playback);
-
- /**
- * Set the input format on necessary objects.
- */
- void hardwareInputFormatAvailable(AudioFormat capture);
-
- void devicesChanged();
-
- std::shared_ptr<AudioFrame> getToPlay(AudioFormat format, size_t writableSamples);
-
- std::shared_ptr<AudioFrame> getToRing(AudioFormat format, size_t writableSamples);
-
- std::shared_ptr<AudioFrame> getPlayback(AudioFormat format, size_t samples) {
- const auto& ringBuff = getToRing(format, samples);
- const auto& playBuff = getToPlay(format, samples);
- return ringBuff ? ringBuff : playBuff;
- }
-
- void flush();
-
- /**
- * True if capture is not to be used
- */
- bool isCaptureMuted_;
-
- /**
- * True if playback is not to be used
- */
- bool isPlaybackMuted_;
-
- /**
- * True if ringtone should be muted
- */
- bool isRingtoneMuted_ {false};
-
- /**
- * Gain applied to mic signal
- */
- double captureGain_;
-
- /**
- * Gain applied to playback signal
- */
- double playbackGain_;
-
- /**
- * Buffers for audio processing
- */
- AudioBuffer playbackBuffer_;
- AudioBuffer playbackResampleBuffer_;
- AudioBuffer ringtoneBuffer_;
- AudioBuffer ringtoneResampleBuffer_;
- std::unique_ptr<AudioFrameResizer> playbackQueue_;
-
- /**
- * Whether or not the audio layer stream is started
- */
- std::atomic<Status> status_ {Status::Idle};
- mutable std::condition_variable startedCv_;
-
- /**
- * Sample Rate that should be sent to the sound card
- */
- AudioFormat audioFormat_;
-
- /**
- * Sample Rate for input.
- */
- AudioFormat audioInputFormat_;
-
- /**
- * Urgent ring buffer used for ringtones
- */
- RingBuffer urgentRingBuffer_;
-
- /**
- * Lock for the entire audio layer
- */
- mutable std::mutex mutex_ {};
-
- /**
- * Remove audio offset that can be introduced by certain cheap audio device
- */
- DcBlocker dcblocker_ {};
-
- /**
- * Manage sampling rate conversion
- */
- std::unique_ptr<Resampler> resampler_;
-
- /**
- * Manage input sampling rate conversions
- */
- std::unique_ptr<Resampler> inputResampler_;
-
- private:
-
- /**
- * Time of the last incoming call notification
- */
- std::chrono::system_clock::time_point lastNotificationTime_;
+private:
+ NON_COPYABLE(AudioLayer);
+
+protected:
+ enum class Status {
+ Idle,
+ Starting,
+ Started
+ };
+
+public:
+
+ AudioLayer(const AudioPreference &);
+ virtual ~AudioLayer();
+
+ virtual std::vector<std::string> getCaptureDeviceList() const = 0;
+ virtual std::vector<std::string> getPlaybackDeviceList() const = 0;
+
+ virtual int getAudioDeviceIndex(const std::string& name, DeviceType type) const = 0;
+ virtual std::string getAudioDeviceName(int index, DeviceType type) const = 0;
+ virtual int getIndexCapture() const = 0;
+ virtual int getIndexPlayback() const = 0;
+ virtual int getIndexRingtone() const = 0;
+
+ /**
+ * Start the capture stream and prepare the playback stream.
+ * The playback starts accordingly to its threshold
+ * ALSA Library API
+ */
+ virtual void startStream() = 0;
+
+ /**
+ * Stop the playback and capture streams.
+ * Drops the pending frames and put the capture and playback handles to PREPARED state
+ * ALSA Library API
+ */
+ virtual void stopStream() = 0;
+
+ /**
+ * Determine wether or not the audio layer is active (i.e. stream opened)
+ */
+ inline bool isStarted() const {
+ return status_ == Status::Started;
+ }
+
+ template< class Rep, class Period >
+ bool waitForStart(const std::chrono::duration<Rep, Period>& rel_time) const {
+ std::unique_lock<std::mutex> lk(mutex_);
+ startedCv_.wait_for(lk, rel_time, [&]{return isStarted();});
+ return isStarted();
+ }
+
+ /**
+ * Send a chunk of data to the hardware buffer to start the playback
+ * Copy data in the urgent buffer.
+ * @param buffer The buffer containing the data to be played ( ringtones )
+ */
+ void putUrgent(AudioBuffer& buffer);
+
+ /**
+ * Flush main buffer
+ */
+ void flushMain();
+
+ /**
+ * Flush urgent buffer
+ */
+ void flushUrgent();
+
+ bool isCaptureMuted() const {
+ return isCaptureMuted_;
+ }
+
+ /**
+ * Mute capture (microphone)
+ */
+ void muteCapture(bool muted) {
+ isCaptureMuted_ = muted;
+ }
+
+ bool isPlaybackMuted() const {
+ return isPlaybackMuted_;
+ }
+
+ /**
+ * Mute playback
+ */
+ void mutePlayback(bool muted) {
+ isPlaybackMuted_ = muted;
+ }
+
+ bool isRingtoneMuted() const {
+ return isRingtoneMuted_;
+ }
+ void muteRingtone(bool muted) {
+ isRingtoneMuted_ = muted;
+ }
+
+ /**
+ * Set capture stream gain (microphone)
+ * Range should be [-1.0, 1.0]
+ */
+ void setCaptureGain(double gain) {
+ captureGain_ = gain;
+ }
+
+ /**
+ * Get capture stream gain (microphone)
+ */
+ double getCaptureGain() const {
+ return captureGain_;
+ }
+
+ /**
+ * Set playback stream gain (speaker)
+ * Range should be [-1.0, 1.0]
+ */
+ void setPlaybackGain(double gain) {
+ playbackGain_ = gain;
+ }
+
+ /**
+ * Get playback stream gain (speaker)
+ */
+ double getPlaybackGain() const {
+ return playbackGain_;
+ }
+
+ /**
+ * Get the sample rate of the audio layer
+ * @return unsigned int The sample rate
+ * default: 44100 HZ
+ */
+ unsigned int getSampleRate() const {
+ return audioFormat_.sample_rate;
+ }
+
+ /**
+ * Get the audio format of the layer (sample rate & channel number).
+ */
+ AudioFormat getFormat() const {
+ return audioFormat_;
+ }
+
+ /**
+ * Emit an audio notification on incoming calls
+ */
+ void notifyIncomingCall();
+
+ virtual void updatePreference(AudioPreference &pref, int index, DeviceType type) = 0;
+
+protected:
+ /**
+ * Callback to be called by derived classes when the audio output is opened.
+ */
+ void hardwareFormatAvailable(AudioFormat playback);
+
+ /**
+ * Set the input format on necessary objects.
+ */
+ void hardwareInputFormatAvailable(AudioFormat capture);
+
+ void devicesChanged();
+
+ std::shared_ptr<AudioFrame> getToPlay(AudioFormat format, size_t writableSamples);
+
+ std::shared_ptr<AudioFrame> getToRing(AudioFormat format, size_t writableSamples);
+
+ std::shared_ptr<AudioFrame> getPlayback(AudioFormat format, size_t samples) {
+ const auto& ringBuff = getToRing(format, samples);
+ const auto& playBuff = getToPlay(format, samples);
+ return ringBuff ? ringBuff : playBuff;
+ }
+
+ void flush();
+
+ /**
+ * True if capture is not to be used
+ */
+ bool isCaptureMuted_;
+
+ /**
+ * True if playback is not to be used
+ */
+ bool isPlaybackMuted_;
+
+ /**
+ * True if ringtone should be muted
+ */
+ bool isRingtoneMuted_ {false};
+
+ /**
+ * Gain applied to mic signal
+ */
+ double captureGain_;
+
+ /**
+ * Gain applied to playback signal
+ */
+ double playbackGain_;
+
+ /**
+ * Buffers for audio processing
+ */
+ AudioBuffer playbackBuffer_;
+ AudioBuffer playbackResampleBuffer_;
+ AudioBuffer ringtoneBuffer_;
+ AudioBuffer ringtoneResampleBuffer_;
+ std::unique_ptr<AudioFrameResizer> playbackQueue_;
+
+ /**
+ * Whether or not the audio layer stream is started
+ */
+ std::atomic<Status> status_ {Status::Idle};
+ mutable std::condition_variable startedCv_;
+
+ /**
+ * Sample Rate that should be sent to the sound card
+ */
+ AudioFormat audioFormat_;
+
+ /**
+ * Sample Rate for input.
+ */
+ AudioFormat audioInputFormat_;
+
+ /**
+ * Urgent ring buffer used for ringtones
+ */
+ RingBuffer urgentRingBuffer_;
+
+ /**
+ * Lock for the entire audio layer
+ */
+ mutable std::mutex mutex_ {};
+
+ /**
+ * Remove audio offset that can be introduced by certain cheap audio device
+ */
+ DcBlocker dcblocker_ {};
+
+ /**
+ * Manage sampling rate conversion
+ */
+ std::unique_ptr<Resampler> resampler_;
+
+ /**
+ * Manage input sampling rate conversions
+ */
+ std::unique_ptr<Resampler> inputResampler_;
+
+private:
+
+ /**
+ * Time of the last incoming call notification
+ */
+ std::chrono::system_clock::time_point lastNotificationTime_;
};
} // namespace jami
-
-#endif // _AUDIO_LAYER_H_
diff --git a/src/media/audio/audioloop.h b/src/media/audio/audioloop.h
index 447611697a811cbe995c1a367ab73416d65202a8..5ea5721f6b939b1156644bcf29595444dc20e316 100644
--- a/src/media/audio/audioloop.h
+++ b/src/media/audio/audioloop.h
@@ -20,7 +20,6 @@
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-
#pragma once
#include "ring_types.h"
@@ -35,58 +34,58 @@
namespace jami {
class AudioLoop {
- public:
- AudioLoop() {}
-
- AudioLoop(unsigned int sampleRate);
-
- AudioLoop& operator=(AudioLoop&& o) noexcept {
- std::swap(buffer_, o.buffer_);
- std::swap(pos_, o.pos_);
- return *this;
- }
-
- virtual ~AudioLoop();
-
- /**
- * Get the next fragment of the tone
- * the function change the intern position, and will loop
- * @param output The data buffer
- * @param nb of int16 to send
- * @param gain The gain [-1.0, 1.0]
- */
- void getNext(AudioBuffer& output, double gain);
- std::unique_ptr<AudioFrame> getNext(size_t samples = 0);
-
- void seek(double relative_position);
-
- /**
- * Reset the pointer position
- */
- void reset() {
- pos_ = 0;
- }
-
- /**
- * Accessor to the size of the buffer
- * @return unsigned int The size
- */
- size_t getSize() const {
- return buffer_->frames();
- }
- AudioFormat getFormat() const {
- return buffer_->getFormat();
- }
- protected:
- /** The data buffer */
- AudioBuffer * buffer_ {nullptr};
-
- /** current position, set to 0, when initialize */
- size_t pos_ {0};
-
- private:
- NON_COPYABLE(AudioLoop);
- virtual void onBufferFinish();
+public:
+ AudioLoop() {}
+
+ AudioLoop(unsigned int sampleRate);
+
+ AudioLoop& operator=(AudioLoop&& o) noexcept {
+ std::swap(buffer_, o.buffer_);
+ std::swap(pos_, o.pos_);
+ return *this;
+ }
+
+ virtual ~AudioLoop();
+
+ /**
+ * Get the next fragment of the tone
+ * the function change the intern position, and will loop
+ * @param output The data buffer
+ * @param nb of int16 to send
+ * @param gain The gain [-1.0, 1.0]
+ */
+ void getNext(AudioBuffer& output, double gain);
+ std::unique_ptr<AudioFrame> getNext(size_t samples = 0);
+
+ void seek(double relative_position);
+
+ /**
+ * Reset the pointer position
+ */
+ void reset() {
+ pos_ = 0;
+ }
+
+ /**
+ * Accessor to the size of the buffer
+ * @return unsigned int The size
+ */
+ size_t getSize() const {
+ return buffer_->frames();
+ }
+ AudioFormat getFormat() const {
+ return buffer_->getFormat();
+ }
+protected:
+ /** The data buffer */
+ AudioBuffer * buffer_ {nullptr};
+
+ /** current position, set to 0, when initialize */
+ size_t pos_ {0};
+
+private:
+ NON_COPYABLE(AudioLoop);
+ virtual void onBufferFinish();
};
} // namespace jami
diff --git a/src/smartools.cpp b/src/smartools.cpp
index 58ab7826219904e338b5cf90887785d817ca945a..d7bde8c3f645061838f52bc9a22d471b16c7a58f 100644
--- a/src/smartools.cpp
+++ b/src/smartools.cpp
@@ -47,12 +47,13 @@ void
Smartools::start(std::chrono::milliseconds refreshTimeMs)
{
JAMI_DBG("Start SmartInfo");
- if (auto t = std::move(task_))
- t->cancel();
- task_ = Manager::instance().scheduler().scheduleAtFixedRate([this]{
+ auto task = Manager::instance().scheduler().scheduleAtFixedRate([this]{
sendInfo();
return true;
}, refreshTimeMs);
+ task_.swap(task);
+ if (task)
+ task->cancel();
}
void