From 0578e0c2506e6fa5a2d714d82e6d5190654e6adb Mon Sep 17 00:00:00 2001
From: Tristan Matthews <tristan.matthews@savoirfairelinux.com>
Date: Tue, 30 Jul 2013 11:48:21 -0400
Subject: [PATCH] * #27693: audio: getChannel should always takes a parameter

---
 daemon/src/audio/alsa/alsalayer.cpp        | 12 +++++-----
 daemon/src/audio/audiobuffer.h             |  2 +-
 daemon/src/audio/audiorecord.cpp           | 26 +++++++++++-----------
 daemon/src/audio/gaincontrol.cpp           |  2 +-
 daemon/src/audio/noisesuppress.cpp         |  2 +-
 daemon/src/audio/pulseaudio/pulselayer.cpp |  2 +-
 daemon/src/managerimpl.cpp                 |  2 +-
 7 files changed, 24 insertions(+), 24 deletions(-)

diff --git a/daemon/src/audio/alsa/alsalayer.cpp b/daemon/src/audio/alsa/alsalayer.cpp
index 9af4251248..19ef6ba14c 100644
--- a/daemon/src/audio/alsa/alsalayer.cpp
+++ b/daemon/src/audio/alsa/alsalayer.cpp
@@ -707,7 +707,7 @@ void AlsaLayer::capture()
 
     // TODO: handle ALSA multichannel capture
     const int toGetBytes = in.samples() * sizeof(SFLAudioSample);
-    SFLAudioSample * const in_ptr = &(*in.getChannel()->begin());
+    SFLAudioSample * const in_ptr = &(*in.getChannel(0)->begin());
 
     if (read(in_ptr, toGetBytes) != toGetBytes) {
         ERROR("ALSA MIC : Couldn't read!");
@@ -748,7 +748,7 @@ void AlsaLayer::playback(int maxSamples)
         else if (file_tone && !ringtoneHandle_)
             file_tone->getNext(out, playbackGain_);
 
-        write(out.getChannel()->data(), bytesToPut, playbackHandle_);
+        write(out.getChannel(0)->data(), bytesToPut, playbackHandle_);
     } else {
         // play the regular sound samples
 
@@ -781,9 +781,9 @@ void AlsaLayer::playback(int maxSamples)
             //SFLAudioSample * const rsmpl_out_ptr = &(*rsmpl_out.begin());
             //converter_.resample(out_ptr, rsmpl_out_ptr, rsmpl_out.size(), mainBufferSampleRate, sampleRate_, samplesToGet);
             converter_.resample(out, rsmpl_out);
-            write(rsmpl_out.getChannel()->data(), outBytes, playbackHandle_);
+            write(rsmpl_out.getChannel(0)->data(), outBytes, playbackHandle_);
         } else {
-            write(out.getChannel()->data(), bytesToGet, playbackHandle_);
+            write(out.getChannel(0)->data(), bytesToGet, playbackHandle_);
         }
     }
 }
@@ -811,7 +811,7 @@ void AlsaLayer::audioCallback()
         urgentRingBuffer_.get(out, MainBuffer::DEFAULT_ID);
         out.applyGain(playbackGain_);
 
-        write(out.getChannel()->data(), samplesToGet*sizeof(SFLAudioSample), playbackHandle_);
+        write(out.getChannel(0)->data(), samplesToGet * sizeof(SFLAudioSample), playbackHandle_);
         // Consume the regular one as well (same amount of bytes)
         Manager::instance().getMainBuffer().discard(samplesToGet, MainBuffer::DEFAULT_ID);
     } else {
@@ -834,7 +834,7 @@ void AlsaLayer::audioCallback()
             file_tone->getNext(out, playbackGain_);
         }
 
-        write(out.getChannel()->data(), ringtoneAvailBytes, ringtoneHandle_);
+        write(out.getChannel(0)->data(), ringtoneAvailBytes, ringtoneHandle_);
     }
 
     // Additionally handle the mic's audio stream
diff --git a/daemon/src/audio/audiobuffer.h b/daemon/src/audio/audiobuffer.h
index 009c38f5bf..a1981f5d05 100644
--- a/daemon/src/audio/audiobuffer.h
+++ b/daemon/src/audio/audiobuffer.h
@@ -122,7 +122,7 @@ class AudioBuffer {
          * Return the data (audio samples) for a given channel number.
          * Channel data can be modified but size of individual channel vectors should not be changed manually.
          */
-        std::vector<SFLAudioSample> *getChannel(unsigned chan = 0);
+        std::vector<SFLAudioSample> *getChannel(unsigned chan);
 
         /**
          * Return a pointer to the raw data in this buffer.
diff --git a/daemon/src/audio/audiorecord.cpp b/daemon/src/audio/audiorecord.cpp
index 2ed60da34a..47873fcee8 100644
--- a/daemon/src/audio/audiorecord.cpp
+++ b/daemon/src/audio/audiorecord.cpp
@@ -420,22 +420,22 @@ void AudioRecord::closeWavFile()
         WARN("Can't close file");
 }
 
-//void AudioRecord::recData(SFLAudioSample* buffer, size_t nSamples)
 void AudioRecord::recData(AudioBuffer& buffer)
 {
-    if (recordingEnabled_) {
-        if (fileHandle_ == 0) {
-            DEBUG("Can't record data, a file has not yet been opened!");
-            return;
-        }
+    if (not recordingEnabled_)
+        return;
+
+    if (fileHandle_ == 0) {
+        DEBUG("Can't record data, a file has not yet been opened!");
+        return;
+    }
 
-        size_t nSamples = buffer.samples();
+    const size_t nSamples = buffer.samples();
 
-        if (fwrite(buffer.getChannel(), sizeof(SFLAudioSample), nSamples, fileHandle_) != nSamples)
-            WARN("Could not record data! ");
-        else {
-            fflush(fileHandle_);
-            byteCounter_ += nSamples * sizeof(SFLAudioSample);
-        }
+    if (fwrite(buffer.getChannel(0), sizeof(SFLAudioSample), nSamples, fileHandle_) != nSamples) {
+        WARN("Could not record data! ");
+    } else {
+        fflush(fileHandle_);
+        byteCounter_ += nSamples * sizeof(SFLAudioSample);
     }
 }
diff --git a/daemon/src/audio/gaincontrol.cpp b/daemon/src/audio/gaincontrol.cpp
index be7a983441..8e8bc07b29 100644
--- a/daemon/src/audio/gaincontrol.cpp
+++ b/daemon/src/audio/gaincontrol.cpp
@@ -58,7 +58,7 @@ GainControl::GainControl(double sr, double target) : averager_(sr, SFL_GAIN_ATTA
 
 void GainControl::process(AudioBuffer& buf)
 {
-    process(buf.getChannel()->data(), buf.samples());
+    process(buf.getChannel(0)->data(), buf.samples());
 }
 
 void GainControl::process(SFLAudioSample *buf, int samples)
diff --git a/daemon/src/audio/noisesuppress.cpp b/daemon/src/audio/noisesuppress.cpp
index f0d5ca4307..961cb88215 100644
--- a/daemon/src/audio/noisesuppress.cpp
+++ b/daemon/src/audio/noisesuppress.cpp
@@ -64,7 +64,7 @@ NoiseSuppress::~NoiseSuppress()
 
 void NoiseSuppress::process(AudioBuffer& buff, int samples)
 {
-    SFLAudioSample* data = buff.getChannel()->data();
+    SFLAudioSample* data = buff.getChannel(0)->data();
     if (noiseState_) {
         assert(smplPerFrame_ == samples);
         speex_preprocess_run(noiseState_, data);
diff --git a/daemon/src/audio/pulseaudio/pulselayer.cpp b/daemon/src/audio/pulseaudio/pulselayer.cpp
index 8792faee6d..1fcfdc8449 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.cpp
+++ b/daemon/src/audio/pulseaudio/pulselayer.cpp
@@ -538,7 +538,7 @@ void PulseLayer::readFromMic()
     Manager::instance().getMainBuffer().putData(*out, MainBuffer::DEFAULT_ID);
 
 #ifdef RECTODISK
-    outfileResampled.write((const char *)out->getChannel(), out->samples()*sizeof(SFLAudioSample));
+    outfileResampled.write((const char *)out->getChannel(0), out->samples() * sizeof(SFLAudioSample));
 #endif
 
     if (pa_stream_drop(record_->pulseStream()) < 0)
diff --git a/daemon/src/managerimpl.cpp b/daemon/src/managerimpl.cpp
index 4dde1df152..a19019f28b 100644
--- a/daemon/src/managerimpl.cpp
+++ b/daemon/src/managerimpl.cpp
@@ -1443,7 +1443,7 @@ void ManagerImpl::playDtmf(char code)
     dtmfKey_->startTone(code);
 
     // copy the sound
-    if (dtmfKey_->generateDTMF(*buf.getChannel())) {
+    if (dtmfKey_->generateDTMF(*buf.getChannel(0))) {
         // Put buffer to urgentRingBuffer
         // put the size in bytes...
         // so size * 1 channel (mono) * sizeof (bytes for the data)
-- 
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