globally change all use of "frame" to refer to audio into "sample".

Generated by tools/f2s. Some hand-editing will be required in a few places to fix up comments related to timecode
and video in order to keep the legible

20 files changed, 101 insertions, 101 deletions

diff --git a/libs/backends/alsa/alsa_audiobackend.cc b/libs/backends/alsa/alsa_audiobackend.cc
index 5abb471502..292875e6f9 100644
--- a/libs/backends/alsa/alsa_audiobackend.cc
+++ b/libs/backends/alsa/alsa_audiobackend.cc
@@ -1049,13 +1049,13 @@ AlsaAudioBackend::raw_buffer_size (DataType t)
 }
 
 /* Process time */
-framepos_t
+samplepos_t
 AlsaAudioBackend::sample_time ()
 {
 	return _processed_samples;
 }
 
-framepos_t
+samplepos_t
 AlsaAudioBackend::sample_time_at_cycle_start ()
 {
 	return _processed_samples;
diff --git a/libs/backends/alsa/alsa_audiobackend.h b/libs/backends/alsa/alsa_audiobackend.h
index ad3bb41949..2cbc3b04a3 100644
--- a/libs/backends/alsa/alsa_audiobackend.h
+++ b/libs/backends/alsa/alsa_audiobackend.h
@@ -263,8 +263,8 @@ class AlsaAudioBackend : public AudioBackend {
 		size_t raw_buffer_size (DataType t);
 
 		/* Process time */
-		framepos_t sample_time ();
-		framepos_t sample_time_at_cycle_start ();
+		samplepos_t sample_time ();
+		samplepos_t sample_time_at_cycle_start ();
 		pframes_t samples_since_cycle_start ();
 
 		int create_process_thread (boost::function<void()> func);
@@ -395,7 +395,7 @@ class AlsaAudioBackend : public AudioBackend {
 		/* processing */
 		float  _dsp_load;
 		ARDOUR::DSPLoadCalculator  _dsp_load_calc;
-		framecnt_t _processed_samples;
+		samplecnt_t _processed_samples;
 		pthread_t _main_thread;
 
 		/* DLL, track main process callback timing */
diff --git a/libs/backends/alsa/alsa_slave.cc b/libs/backends/alsa/alsa_slave.cc
index e1061ccc09..8863755f7a 100644
--- a/libs/backends/alsa/alsa_slave.cc
+++ b/libs/backends/alsa/alsa_slave.cc
@@ -237,14 +237,14 @@ AlsaAudioSlave::process_thread ()
 					}
 
 					/* possible samples at end of first buffer chunk, 
-					 * incomplete audio-frame */
+					 * incomplete audio-sample */
 					uint32_t s = vec.len[0] - k * nchn;
 					assert (s < nchn);
 
 					for (c = 0; c < s; ++c) {
 						_pcmi.capt_chan (c, vec.buf[0] + k * nchn + c, 1, nchn);
 					}
-					/* cont'd audio-frame at second ringbuffer chunk */
+					/* cont'd audio-sample at second ringbuffer chunk */
 					for (; c < nchn; ++c) {
 						_pcmi.capt_chan (c, vec.buf[1] + c - s, spp - k, nchn);
 					}
diff --git a/libs/backends/coreaudio/coreaudio_backend.cc b/libs/backends/coreaudio/coreaudio_backend.cc
index c4e11ef700..0e0f33a5d9 100644
--- a/libs/backends/coreaudio/coreaudio_backend.cc
+++ b/libs/backends/coreaudio/coreaudio_backend.cc
@@ -756,13 +756,13 @@ CoreAudioBackend::raw_buffer_size (DataType t)
 }
 
 /* Process time */
-framepos_t
+samplepos_t
 CoreAudioBackend::sample_time ()
 {
 	return _processed_samples;
 }
 
-framepos_t
+samplepos_t
 CoreAudioBackend::sample_time_at_cycle_start ()
 {
 	return _processed_samples;
diff --git a/libs/backends/coreaudio/coreaudio_backend.h b/libs/backends/coreaudio/coreaudio_backend.h
index 3d7b5be860..5d8c88c3db 100644
--- a/libs/backends/coreaudio/coreaudio_backend.h
+++ b/libs/backends/coreaudio/coreaudio_backend.h
@@ -309,8 +309,8 @@ class CoreAudioBackend : public AudioBackend {
 	size_t raw_buffer_size (DataType t);
 
 	/* Process time */
-	framepos_t sample_time ();
-	framepos_t sample_time_at_cycle_start ();
+	samplepos_t sample_time ();
+	samplepos_t sample_time_at_cycle_start ();
 	pframes_t samples_since_cycle_start ();
 
 	int create_process_thread (boost::function<void()> func);
diff --git a/libs/backends/coreaudio/coreaudio_pcmio.cc b/libs/backends/coreaudio/coreaudio_pcmio.cc
index 59c054a74a..8b2f9255d3 100644
--- a/libs/backends/coreaudio/coreaudio_pcmio.cc
+++ b/libs/backends/coreaudio/coreaudio_pcmio.cc
@@ -184,11 +184,11 @@ static OSStatus render_callback_ptr (
 		AudioUnitRenderActionFlags* ioActionFlags,
 		const AudioTimeStamp* inTimeStamp,
 		UInt32 inBusNumber,
-		UInt32 inNumberFrames,
+		UInt32 inNumberSamples,
 		AudioBufferList* ioData)
 {
 	CoreAudioPCM * d = static_cast<CoreAudioPCM*> (inRefCon);
-	return d->render_callback(ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, ioData);
+	return d->render_callback(ioActionFlags, inTimeStamp, inBusNumber, inNumberSamples, ioData);
 }
 
 
@@ -752,7 +752,7 @@ static void PrintStreamDesc (AudioStreamBasicDescription *inDesc)
 	printf ("  Format ID:%.*s\n",        (int)sizeof(inDesc->mFormatID), (char*)&inDesc->mFormatID);
 	printf ("  Format Flags:%X\n",       (unsigned int)inDesc->mFormatFlags);
 	printf ("  Bytes per Packet:%d\n",   (int)inDesc->mBytesPerPacket);
-	printf ("  Frames per Packet:%d\n",  (int)inDesc->mFramesPerPacket);
+	printf ("  Samples per Packet:%d\n",  (int)inDesc->mSamplesPerPacket);
 	printf ("  Bytes per Frame:%d\n",    (int)inDesc->mBytesPerFrame);
 	printf ("  Channels per Frame:%d\n", (int)inDesc->mChannelsPerFrame);
 	printf ("  Bits per Channel:%d\n",   (int)inDesc->mBitsPerChannel);
@@ -895,7 +895,7 @@ CoreAudioPCM::pcm_start (
 		srcFormat.mFormatID = kAudioFormatLinearPCM;
 		srcFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked | kLinearPCMFormatFlagIsNonInterleaved;
 		srcFormat.mBytesPerPacket = sizeof(float);
-		srcFormat.mFramesPerPacket = 1;
+		srcFormat.mSamplesPerPacket = 1;
 		srcFormat.mBytesPerFrame = sizeof(float);
 		srcFormat.mChannelsPerFrame = chn_in;
 		srcFormat.mBitsPerChannel = 32;
@@ -903,8 +903,8 @@ CoreAudioPCM::pcm_start (
 		err = AudioUnitSetProperty(_auhal, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, AUHAL_INPUT_ELEMENT, &srcFormat, sizeof(AudioStreamBasicDescription));
 		if (err != noErr) { errorMsg="kAudioUnitProperty_StreamFormat, Output"; _state = -6; goto error; }
 
-		err = AudioUnitSetProperty(_auhal, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, AUHAL_INPUT_ELEMENT, (UInt32*)&_samples_per_period, sizeof(UInt32));
-		if (err != noErr) { errorMsg="kAudioUnitProperty_MaximumFramesPerSlice, Input"; _state = -6; goto error; }
+		err = AudioUnitSetProperty(_auhal, kAudioUnitProperty_MaximumSamplesPerSlice, kAudioUnitScope_Global, AUHAL_INPUT_ELEMENT, (UInt32*)&_samples_per_period, sizeof(UInt32));
+		if (err != noErr) { errorMsg="kAudioUnitProperty_MaximumSamplesPerSlice, Input"; _state = -6; goto error; }
 	}
 
 	if (chn_out > 0) {
@@ -912,7 +912,7 @@ CoreAudioPCM::pcm_start (
 		dstFormat.mFormatID = kAudioFormatLinearPCM;
 		dstFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked | kLinearPCMFormatFlagIsNonInterleaved;
 		dstFormat.mBytesPerPacket = sizeof(float);
-		dstFormat.mFramesPerPacket = 1;
+		dstFormat.mSamplesPerPacket = 1;
 		dstFormat.mBytesPerFrame = sizeof(float);
 		dstFormat.mChannelsPerFrame = chn_out;
 		dstFormat.mBitsPerChannel = 32;
@@ -920,8 +920,8 @@ CoreAudioPCM::pcm_start (
 		err = AudioUnitSetProperty(_auhal, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, AUHAL_OUTPUT_ELEMENT, &dstFormat, sizeof(AudioStreamBasicDescription));
 		if (err != noErr) { errorMsg="kAudioUnitProperty_StreamFormat Input"; _state = -5; goto error; }
 
-		err = AudioUnitSetProperty(_auhal, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, AUHAL_OUTPUT_ELEMENT, (UInt32*)&_samples_per_period, sizeof(UInt32));
-		if (err != noErr) { errorMsg="kAudioUnitProperty_MaximumFramesPerSlice, Output"; _state = -5; goto error; }
+		err = AudioUnitSetProperty(_auhal, kAudioUnitProperty_MaximumSamplesPerSlice, kAudioUnitScope_Global, AUHAL_OUTPUT_ELEMENT, (UInt32*)&_samples_per_period, sizeof(UInt32));
+		if (err != noErr) { errorMsg="kAudioUnitProperty_MaximumSamplesPerSlice, Output"; _state = -5; goto error; }
 	}
 
 	/* read back stream descriptions */
@@ -1111,19 +1111,19 @@ CoreAudioPCM::render_callback (
 		AudioUnitRenderActionFlags* ioActionFlags,
 		const AudioTimeStamp* inTimeStamp,
 		UInt32 inBusNumber,
-		UInt32 inNumberFrames,
+		UInt32 inNumberSamples,
 		AudioBufferList* ioData)
 {
 	OSStatus retVal = kAudioHardwareNoError;
 
-	if (_samples_per_period < inNumberFrames) {
+	if (_samples_per_period < inNumberSamples) {
 #ifndef NDEBUG
 		printf("samples per period exceeds configured value, cycle skipped (%u < %u)\n",
-				(unsigned int)_samples_per_period, (unsigned int)inNumberFrames);
+				(unsigned int)_samples_per_period, (unsigned int)inNumberSamples);
 #endif
 		for (uint32_t i = 0; _playback_channels > 0 && i < ioData->mNumberBuffers; ++i) {
 			float* ob = (float*) ioData->mBuffers[i].mData;
-			memset(ob, 0, sizeof(float) * inNumberFrames);
+			memset(ob, 0, sizeof(float) * inNumberSamples);
 		}
 		return noErr;
 	}
@@ -1131,17 +1131,17 @@ CoreAudioPCM::render_callback (
 	assert(_playback_channels == 0 || ioData->mNumberBuffers == _playback_channels);
 
 	UInt64 cur_cycle_start = AudioGetCurrentHostTime ();
-	_cur_samples_per_period = inNumberFrames;
+	_cur_samples_per_period = inNumberSamples;
 
 	if (_capture_channels > 0) {
 		_input_audio_buffer_list->mNumberBuffers = _capture_channels;
 		for (uint32_t i = 0; i < _capture_channels; ++i) {
 			_input_audio_buffer_list->mBuffers[i].mNumberChannels = 1;
-			_input_audio_buffer_list->mBuffers[i].mDataByteSize = inNumberFrames * sizeof(float);
+			_input_audio_buffer_list->mBuffers[i].mDataByteSize = inNumberSamples * sizeof(float);
 			_input_audio_buffer_list->mBuffers[i].mData = NULL;
 		}
 
-		retVal = AudioUnitRender(_auhal, ioActionFlags, inTimeStamp, AUHAL_INPUT_ELEMENT, inNumberFrames, _input_audio_buffer_list);
+		retVal = AudioUnitRender(_auhal, ioActionFlags, inTimeStamp, AUHAL_INPUT_ELEMENT, inNumberSamples, _input_audio_buffer_list);
 	}
 
 	if (retVal != kAudioHardwareNoError) {
@@ -1162,7 +1162,7 @@ CoreAudioPCM::render_callback (
 	int rv = -1;
 
 	if (_process_callback) {
-		rv = _process_callback(_process_arg, inNumberFrames, cur_cycle_start);
+		rv = _process_callback(_process_arg, inNumberSamples, cur_cycle_start);
 	}
 
 	_in_process = false;
@@ -1171,7 +1171,7 @@ CoreAudioPCM::render_callback (
 		// clear output
 		for (uint32_t i = 0; i < ioData->mNumberBuffers; ++i) {
 			float* ob = (float*) ioData->mBuffers[i].mData;
-			memset(ob, 0, sizeof(float) * inNumberFrames);
+			memset(ob, 0, sizeof(float) * inNumberSamples);
 		}
 	}
 	return noErr;
diff --git a/libs/backends/coreaudio/coreaudio_pcmio.h b/libs/backends/coreaudio/coreaudio_pcmio.h
index 5f9778e845..e6d82571f4 100644
--- a/libs/backends/coreaudio/coreaudio_pcmio.h
+++ b/libs/backends/coreaudio/coreaudio_pcmio.h
@@ -126,7 +126,7 @@ public:
 			AudioUnitRenderActionFlags* ioActionFlags,
 			const AudioTimeStamp* inTimeStamp,
 			UInt32 inBusNumber,
-			UInt32 inNumberFrames,
+			UInt32 inNumberSamples,
 			AudioBufferList* ioData);
 
 	void xrun_callback ();
diff --git a/libs/backends/dummy/dummy_audiobackend.cc b/libs/backends/dummy/dummy_audiobackend.cc
index 9400d03bbe..825add2c94 100644
--- a/libs/backends/dummy/dummy_audiobackend.cc
+++ b/libs/backends/dummy/dummy_audiobackend.cc
@@ -524,13 +524,13 @@ DummyAudioBackend::raw_buffer_size (DataType t)
 }
 
 /* Process time */
-framepos_t
+samplepos_t
 DummyAudioBackend::sample_time ()
 {
 	return _processed_samples;
 }
 
-framepos_t
+samplepos_t
 DummyAudioBackend::sample_time_at_cycle_start ()
 {
 	return _processed_samples;
diff --git a/libs/backends/dummy/dummy_audiobackend.h b/libs/backends/dummy/dummy_audiobackend.h
index 739a649407..4bd4078ec5 100644
--- a/libs/backends/dummy/dummy_audiobackend.h
+++ b/libs/backends/dummy/dummy_audiobackend.h
@@ -316,8 +316,8 @@ class DummyAudioBackend : public AudioBackend {
 		size_t raw_buffer_size (DataType t);
 
 		/* Process time */
-		framepos_t sample_time ();
-		framepos_t sample_time_at_cycle_start ();
+		samplepos_t sample_time ();
+		samplepos_t sample_time_at_cycle_start ();
 		pframes_t samples_since_cycle_start ();
 
 		int create_process_thread (boost::function<void()> func);
@@ -435,7 +435,7 @@ class DummyAudioBackend : public AudioBackend {
 		uint32_t _systemic_input_latency;
 		uint32_t _systemic_output_latency;
 
-		framecnt_t _processed_samples;
+		samplecnt_t _processed_samples;
 
 		pthread_t _main_thread;
 
diff --git a/libs/backends/dummy/dummy_midi_seq.h b/libs/backends/dummy/dummy_midi_seq.h
index e809a47371..8dbfb660cc 100644
--- a/libs/backends/dummy/dummy_midi_seq.h
+++ b/libs/backends/dummy/dummy_midi_seq.h
@@ -780,7 +780,7 @@ static const MIDISequence s7[] = { // channel1, nonsense
 };
 
 static const MIDISequence s8[] = { // sysex
-	{ 0.00, 10, {0xf0, 0x7f, 0x7f, 0x01, 0x01, /*hour + tc */ 0x20, /*min*/ 0x00, /*sec*/ 0x00, /*frame*/ 0x00, 0xf7} },  // 25fps, 00:00:00:00
+	{ 0.00, 10, {0xf0, 0x7f, 0x7f, 0x01, 0x01, /*hour + tc */ 0x20, /*min*/ 0x00, /*sec*/ 0x00, /*sample*/ 0x00, 0xf7} },  // 25fps, 00:00:00:00
 	// quarter frames spacing (in samples) at 25fps: SR / (25 * 4)
 	// DummyMidiPort::setup_generator uses 120bpm ie (SR / 2)
 	// hence the spacing is: SR / (25 * 4) / (SR / 2) = 0.2
diff --git a/libs/backends/jack/jack_audiobackend.cc b/libs/backends/jack/jack_audiobackend.cc
index 8b9e4af1dd..616964c3f2 100644
--- a/libs/backends/jack/jack_audiobackend.cc
+++ b/libs/backends/jack/jack_audiobackend.cc
@@ -633,14 +633,14 @@ JACKAudioBackend::transport_start ()
 }
 
 void
-JACKAudioBackend::transport_locate (framepos_t where)
+JACKAudioBackend::transport_locate (samplepos_t where)
 {
 	GET_PRIVATE_JACK_POINTER (_priv_jack);
 	jack_transport_locate (_priv_jack, where);
 }
 
-framepos_t
-JACKAudioBackend::transport_frame () const
+samplepos_t
+JACKAudioBackend::transport_sample () const
 {
 	GET_PRIVATE_JACK_POINTER_RET (_priv_jack, 0);
 	return jack_get_current_transport_frame (_priv_jack);
@@ -693,14 +693,14 @@ JACKAudioBackend::get_sync_offset (pframes_t& offset) const
 	return false;
 }
 
-framepos_t
+samplepos_t
 JACKAudioBackend::sample_time ()
 {
 	GET_PRIVATE_JACK_POINTER_RET (_priv_jack, 0);
 	return jack_frame_time (_priv_jack);
 }
 
-framepos_t
+samplepos_t
 JACKAudioBackend::sample_time_at_cycle_start ()
 {
 	GET_PRIVATE_JACK_POINTER_RET (_priv_jack, 0);
@@ -1164,7 +1164,7 @@ JACKAudioBackend::set_midi_option (const string& opt)
 }
 
 bool
-JACKAudioBackend::speed_and_position (double& speed, framepos_t& position)
+JACKAudioBackend::speed_and_position (double& speed, samplepos_t& position)
 {
 	jack_position_t pos;
 	jack_transport_state_t state;
diff --git a/libs/backends/jack/jack_audiobackend.h b/libs/backends/jack/jack_audiobackend.h
index 2c67dc72fd..eb66f7ca3f 100644
--- a/libs/backends/jack/jack_audiobackend.h
+++ b/libs/backends/jack/jack_audiobackend.h
@@ -100,8 +100,8 @@ class JACKAudioBackend : public AudioBackend {
 
     float dsp_load() const;
 
-    framepos_t sample_time ();
-    framepos_t sample_time_at_cycle_start ();
+    samplepos_t sample_time ();
+    samplepos_t sample_time_at_cycle_start ();
     pframes_t samples_since_cycle_start ();
 
     size_t raw_buffer_size (DataType t);
@@ -113,9 +113,9 @@ class JACKAudioBackend : public AudioBackend {
 
     void transport_start ();
     void transport_stop ();
-    void transport_locate (framepos_t /*pos*/);
+    void transport_locate (samplepos_t /*pos*/);
     TransportState transport_state () const;
-    framepos_t transport_frame() const;
+    samplepos_t transport_sample() const;
 
     int set_time_master (bool /*yn*/);
     bool get_sync_offset (pframes_t& /*offset*/) const;
@@ -207,7 +207,7 @@ class JACKAudioBackend : public AudioBackend {
 
     /* transport sync */
 
-    bool speed_and_position (double& sp, framepos_t& pos);
+    bool speed_and_position (double& sp, samplepos_t& pos);
 
   private:
     boost::shared_ptr<JackConnection>  _jack_connection;
diff --git a/libs/backends/jack/jack_session.cc b/libs/backends/jack/jack_session.cc
index ce325321f3..b7d37a673f 100644
--- a/libs/backends/jack/jack_session.cc
+++ b/libs/backends/jack/jack_session.cc
@@ -114,14 +114,14 @@ JACKSession::timebase_callback (jack_transport_state_t /*state*/,
 {
 	Timecode::BBT_Time bbt;
 	TempoMap& tempo_map (_session->tempo_map());
-	framepos_t tf = _session->transport_frame ();
+	samplepos_t tf = _session->transport_sample ();
 
 	/* BBT info */
 
 	TempoMetric metric (tempo_map.metric_at (tf));
 
 	try {
-		bbt = tempo_map.bbt_at_frame (tf);
+		bbt = tempo_map.bbt_at_sample (tf);
 
 		pos->bar = bbt.bars;
 		pos->beat = bbt.beats;
@@ -142,7 +142,7 @@ JACKSession::timebase_callback (jack_transport_state_t /*state*/,
 
 #ifdef HAVE_JACK_VIDEO_SUPPORT
 	//poke audio video ratio so Ardour can track Video Sync
-	pos->audio_frames_per_video_frame = _session->frame_rate() / _session->timecode_frames_per_second();
+	pos->audio_samples_per_video_frame = _session->sample_rate() / _session->timecode_frames_per_second();
 	pos->valid = jack_position_bits_t (pos->valid | JackAudioVideoRatio);
 #endif
 
diff --git a/libs/backends/jack/weak_libjack.h b/libs/backends/jack/weak_libjack.h
index 88365dee7e..6af3e9c351 100644
--- a/libs/backends/jack/weak_libjack.h
+++ b/libs/backends/jack/weak_libjack.h
@@ -52,9 +52,9 @@ int have_libjack(void);
 #define jack_get_client_name                WJACK_get_client_name
 #define jack_get_sample_rate                WJACK_get_sample_rate
 #define jack_get_buffer_size                WJACK_get_buffer_size
-#define jack_frames_since_cycle_start       WJACK_frames_since_cycle_start
-#define jack_frame_time                     WJACK_frame_time
-#define jack_last_frame_time                WJACK_last_frame_time
+#define jack_frames_since_cycle_start       WJACK_samples_since_cycle_start
+#define jack_frame_time                     WJACK_sample_time
+#define jack_last_frame_time                WJACK_last_sample_time
 #define jack_get_time                       WJACK_get_time
 #define jack_cpu_load                       WJACK_cpu_load
 #define jack_is_realtime                    WJACK_is_realtime
@@ -115,7 +115,7 @@ int have_libjack(void);
 #define jack_set_thread_init_callback       WJACK_set_thread_init_callback
 
 /* <jack/transport.h> */
-#define jack_get_current_transport_frame    WJACK_get_current_transport_frame
+#define jack_get_current_transport_frame    WJACK_get_current_transport_sample
 #define jack_transport_locate               WJACK_transport_locate
 #define jack_transport_start                WJACK_transport_start
 #define jack_transport_stop                 WJACK_transport_stop
diff --git a/libs/backends/portaudio/audio_utils.h b/libs/backends/portaudio/audio_utils.h
index 5e7d0a1a2a..8b7ab9d529 100644
--- a/libs/backends/portaudio/audio_utils.h
+++ b/libs/backends/portaudio/audio_utils.h
@@ -25,12 +25,12 @@ inline
 void
 deinterleave_audio_data(const float* interleaved_input,
                         float* output,
-                        uint32_t frame_count,
+                        uint32_t sample_count,
                         uint32_t channel,
                         uint32_t channel_count)
 {
 	const float* ptr = interleaved_input + channel;
-	while (frame_count-- > 0) {
+	while (sample_count-- > 0) {
 		*output++ = *ptr;
 		ptr += channel_count;
 	}
@@ -40,12 +40,12 @@ inline
 void
 interleave_audio_data(float* input,
                       float* interleaved_output,
-                      uint32_t frame_count,
+                      uint32_t sample_count,
                       uint32_t channel,
                       uint32_t channel_count)
 {
 	float* ptr = interleaved_output + channel;
-	while (frame_count-- > 0) {
+	while (sample_count-- > 0) {
 		*ptr = *input++;
 		ptr += channel_count;
 	}
diff --git a/libs/backends/portaudio/cycle_timer.h b/libs/backends/portaudio/cycle_timer.h
index 6fe665568c..b81a1ed234 100644
--- a/libs/backends/portaudio/cycle_timer.h
+++ b/libs/backends/portaudio/cycle_timer.h
@@ -25,7 +25,7 @@
 
 // Could call it FrameTimer and make it more generic
 // Could be an interface and or include clock source
-// include sample count/processed frames in iteration?
+// include sample count/processed samples in iteration?
 class CycleTimer {
 public:
 	CycleTimer ()
diff --git a/libs/backends/portaudio/portaudio_backend.cc b/libs/backends/portaudio/portaudio_backend.cc
index 46e508fba7..be43e67a62 100644
--- a/libs/backends/portaudio/portaudio_backend.cc
+++ b/libs/backends/portaudio/portaudio_backend.cc
@@ -716,7 +716,7 @@ PortAudioBackend::_start (bool for_latency_measurement)
 int
 PortAudioBackend::portaudio_callback(const void* input,
                                      void* output,
-                                     unsigned long frame_count,
+                                     unsigned long sample_count,
                                      const PaStreamCallbackTimeInfo* time_info,
                                      PaStreamCallbackFlags status_flags,
                                      void* user_data)
@@ -725,7 +725,7 @@ PortAudioBackend::portaudio_callback(const void* input,
 
 	if (!pa_backend->process_callback((const float*)input,
 	                                  (float*)output,
-	                                  frame_count,
+	                                  sample_count,
 	                                  time_info,
 	                                  status_flags)) {
 		return paAbort;
@@ -737,7 +737,7 @@ PortAudioBackend::portaudio_callback(const void* input,
 bool
 PortAudioBackend::process_callback(const float* input,
                                    float* output,
-                                   uint32_t frame_count,
+                                   uint32_t sample_count,
                                    const PaStreamCallbackTimeInfo* timeInfo,
                                    PaStreamCallbackFlags statusFlags)
 {
@@ -767,7 +767,7 @@ PortAudioBackend::process_callback(const float* input,
 	}
 
 	if (!_run || _freewheel) {
-		memset(output, 0, frame_count * sizeof(float) * _system_outputs.size());
+		memset(output, 0, sample_count * sizeof(float) * _system_outputs.size());
 		return true;
 	}
 
@@ -1004,13 +1004,13 @@ PortAudioBackend::raw_buffer_size (DataType t)
 }
 
 /* Process time */
-framepos_t
+samplepos_t
 PortAudioBackend::sample_time ()
 {
 	return _processed_samples;
 }
 
-framepos_t
+samplepos_t
 PortAudioBackend::sample_time_at_cycle_start ()
 {
 	return _processed_samples;
diff --git a/libs/backends/portaudio/portaudio_backend.h b/libs/backends/portaudio/portaudio_backend.h
index 8a1624796b..336621f4ff 100644
--- a/libs/backends/portaudio/portaudio_backend.h
+++ b/libs/backends/portaudio/portaudio_backend.h
@@ -243,8 +243,8 @@ class PortAudioBackend : public AudioBackend {
 		size_t raw_buffer_size (DataType t);
 
 		/* Process time */
-		framepos_t sample_time ();
-		framepos_t sample_time_at_cycle_start ();
+		samplepos_t sample_time ();
+		samplepos_t sample_time_at_cycle_start ();
 		pframes_t samples_since_cycle_start ();
 
 		int create_process_thread (boost::function<void()> func);
@@ -342,7 +342,7 @@ class PortAudioBackend : public AudioBackend {
 
 		bool process_callback(const float* input,
 	                          float* output,
-	                          uint32_t frame_count,
+	                          uint32_t sample_count,
 	                          const PaStreamCallbackTimeInfo* timeInfo,
 	                          PaStreamCallbackFlags statusFlags);
 
@@ -407,7 +407,7 @@ class PortAudioBackend : public AudioBackend {
 
 		/* processing */
 		float  _dsp_load;
-		framecnt_t _processed_samples;
+		samplecnt_t _processed_samples;
 
 		/* blocking thread */
 		pthread_t _main_blocking_thread;
diff --git a/libs/backends/portaudio/portaudio_io.cc b/libs/backends/portaudio/portaudio_io.cc
index 9ec8f6d6fb..140de6cf97 100644
--- a/libs/backends/portaudio/portaudio_io.cc
+++ b/libs/backends/portaudio/portaudio_io.cc
@@ -170,9 +170,9 @@ PortAudioIO::available_sample_rates(int device_id, std::vector<float>& sampleRat
 #ifdef WITH_ASIO
 bool
 PortAudioIO::get_asio_buffer_properties (int device_id,
-                                         long& min_size_frames,
-                                         long& max_size_frames,
-                                         long& preferred_size_frames,
+                                         long& min_size_samples,
+                                         long& max_size_samples,
+                                         long& preferred_size_samples,
                                          long& granularity)
 {
 	// we shouldn't really need all these checks but it shouldn't hurt
@@ -191,9 +191,9 @@ PortAudioIO::get_asio_buffer_properties (int device_id,
 	}
 
 	PaError err = PaAsio_GetAvailableBufferSizes (device_id,
-	                                              &min_size_frames,
-	                                              &max_size_frames,
-	                                              &preferred_size_frames,
+	                                              &min_size_samples,
+	                                              &max_size_samples,
+	                                              &preferred_size_samples,
 	                                              &granularity);
 
 	if (err != paNoError) {
@@ -216,15 +216,15 @@ PortAudioIO::get_asio_buffer_sizes(int device_id,
                                    std::vector<uint32_t>& buffer_sizes,
                                    bool preferred_only)
 {
-	long min_size_frames = 0;
-	long max_size_frames = 0;
-	long preferred_size_frames = 0;
+	long min_size_samples = 0;
+	long max_size_samples = 0;
+	long preferred_size_samples = 0;
 	long granularity = 0;
 
 	if (!get_asio_buffer_properties (device_id,
-	                                 min_size_frames,
-	                                 max_size_frames,
-	                                 preferred_size_frames,
+	                                 min_size_samples,
+	                                 max_size_samples,
+	                                 preferred_size_samples,
 	                                 granularity)) {
 		DEBUG_AUDIO (string_compose (
 		    "Unable to get device buffer properties from device index %1\n", device_id));
@@ -232,58 +232,58 @@ PortAudioIO::get_asio_buffer_sizes(int device_id,
 	}
 
 	DEBUG_AUDIO (string_compose ("ASIO buffer properties for device %1, "
-	                             "min_size_frames: %2, max_size_frames: %3, "
-	                             "preferred_size_frames: %4, granularity: %5\n",
+	                             "min_size_samples: %2, max_size_samples: %3, "
+	                             "preferred_size_samples: %4, granularity: %5\n",
 	                             device_id,
-	                             min_size_frames,
-	                             max_size_frames,
-	                             preferred_size_frames,
+	                             min_size_samples,
+	                             max_size_samples,
+	                             preferred_size_samples,
 	                             granularity));
 
-	bool driver_returns_one_size = (min_size_frames == max_size_frames) &&
-	                               (min_size_frames == preferred_size_frames);
+	bool driver_returns_one_size = (min_size_samples == max_size_samples) &&
+	                               (min_size_samples == preferred_size_samples);
 
 	if (preferred_only || driver_returns_one_size) {
-		buffer_sizes.push_back(preferred_size_frames);
+		buffer_sizes.push_back(preferred_size_samples);
 		return true;
 	}
 
-	long buffer_size = min_size_frames;
+	long buffer_size = min_size_samples;
 
 	// If min size and granularity are power of two then just use values that
 	// are power of 2 even if the granularity allows for more values
 	bool use_power_of_two =
-	    is_power_of_two(min_size_frames) && is_power_of_two(granularity);
+	    is_power_of_two(min_size_samples) && is_power_of_two(granularity);
 
 	if (granularity <= 0 || use_power_of_two) {
 		// driver uses buffer sizes that are power of 2
-		while (buffer_size <= max_size_frames) {
+		while (buffer_size <= max_size_samples) {
 			buffer_sizes.push_back(buffer_size);
 			buffer_size *= 2;
 		}
 	} else {
-		if (min_size_frames == max_size_frames) {
+		if (min_size_samples == max_size_samples) {
 			// The devices I have tested either return the same values for
 			// min/max/preferred and changing buffer size is intended to only be
 			// done via the control dialog or they return a range where min != max
 			// but I guess min == max could happen if a driver only supports a single
 			// buffer size
-			buffer_sizes.push_back(min_size_frames);
+			buffer_sizes.push_back(min_size_samples);
 			return true;
 		}
 
-		// If min_size_frames is not power of 2 use at most 8 of the possible
+		// If min_size_samples is not power of 2 use at most 8 of the possible
 		// buffer sizes spread evenly between min and max
 		long max_values = 8;
-		while (((max_size_frames - min_size_frames) / granularity) > max_values) {
+		while (((max_size_samples - min_size_samples) / granularity) > max_values) {
 			granularity *= 2;
 		}
 
-		while (buffer_size < max_size_frames) {
+		while (buffer_size < max_size_samples) {
 			buffer_sizes.push_back(buffer_size);
 			buffer_size += granularity;
 		}
-		buffer_sizes.push_back(max_size_frames);
+		buffer_sizes.push_back(max_size_samples);
 	}
 	return true;
 }
diff --git a/libs/backends/portaudio/portaudio_io.h b/libs/backends/portaudio/portaudio_io.h
index c67fdc1b19..34eef43186 100644
--- a/libs/backends/portaudio/portaudio_io.h
+++ b/libs/backends/portaudio/portaudio_io.h
@@ -58,9 +58,9 @@ public:
 
 #ifdef WITH_ASIO
 	bool get_asio_buffer_properties (int device_id,
-	                                 long& min_size_frames,
-	                                 long& max_size_frames,
-	                                 long& preferred_size_frames,
+	                                 long& min_size_samples,
+	                                 long& max_size_samples,
+	                                 long& preferred_size_samples,
 	                                 long& granularity);
 
 	bool get_asio_buffer_sizes(int device_id,