Clean up audio passing round a bit.

9 files changed, 92 insertions, 54 deletions

diff --git a/src/lib/decoder.cc b/src/lib/decoder.cc
index 16d815a07..ee725c39d 100644
--- a/src/lib/decoder.cc
+++ b/src/lib/decoder.cc
@@ -192,12 +192,9 @@ Decoder::emit_audio (uint8_t* data, int size)
 {
 	/* Deinterleave and convert to float */
 
-	float* samples[_fs->audio_channels()];
 	int const total_samples = size / bytes_per_audio_sample();
 	int const frames = total_samples / _fs->audio_channels();
-	for (int i = 0; i < _fs->audio_channels(); ++i) {
-		samples[i] = new float[frames];
-	}
+	shared_ptr<AudioBuffers> audio (new AudioBuffers (_fs->audio_channels(), frames));
 
 	switch (audio_sample_format()) {
 	case AV_SAMPLE_FMT_S16:
@@ -211,7 +208,7 @@ Decoder::emit_audio (uint8_t* data, int size)
 			/* signed sample */
 			int const os = ou >= 0x8000 ? (- 0x10000 + ou) : ou;
 			/* float sample */
-			samples[channel][sample] = float(os) / 0x8000;
+			audio->data(channel)[sample] = float(os) / 0x8000;
 
 			++channel;
 			if (channel == _fs->audio_channels()) {
@@ -228,7 +225,7 @@ Decoder::emit_audio (uint8_t* data, int size)
 	{
 		float* p = reinterpret_cast<float*> (data);
 		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			memcpy (samples[i], p, frames * sizeof(float));
+			memcpy (audio->data(i), p, frames * sizeof(float));
 			p += frames;
 		}
 	}
@@ -243,7 +240,7 @@ Decoder::emit_audio (uint8_t* data, int size)
 		float const linear_gain = pow (10, _fs->audio_gain() / 20);
 		for (int i = 0; i < _fs->audio_channels(); ++i) {
 			for (int j = 0; j < frames; ++j) {
-				samples[i][j] *= linear_gain;
+				audio->data(i)[j] *= linear_gain;
 			}
 		}
 	}
@@ -251,11 +248,7 @@ Decoder::emit_audio (uint8_t* data, int size)
 	/* Update the number of audio frames we've pushed to the encoder */
 	_audio_frames_processed += frames;
 
-	Audio (samples, frames);
-
-	for (int i = 0; i < _fs->audio_channels(); ++i) {
-		delete[] samples[i];
-	}
+	Audio (audio);
 }
 
 /** Called by subclasses to tell the world that some video data is ready.
diff --git a/src/lib/decoder.h b/src/lib/decoder.h
index 14e602ace..85b256f5b 100644
--- a/src/lib/decoder.h
+++ b/src/lib/decoder.h
@@ -99,11 +99,8 @@ public:
 	 */
 	sigc::signal<void, boost::shared_ptr<Image>, int, boost::shared_ptr<Subtitle> > Video;
 
-	/** Emitted when some audio data is ready.
-	 *  First parameter is an array of pointers to deinterleaved, floating point sample data for each channel.
-	 *  Second parameter is the size of the data in frames (ie samples on each channel).
-	 */
-	sigc::signal<void, float**, int> Audio;
+	/** Emitted when some audio data is ready */
+	sigc::signal<void, boost::shared_ptr<AudioBuffers> > Audio;
 	
 protected:
 	/** perform a single pass at our content */
diff --git a/src/lib/encoder.h b/src/lib/encoder.h
index 5dc0804c6..3317d30d1 100644
--- a/src/lib/encoder.h
+++ b/src/lib/encoder.h
@@ -37,6 +37,7 @@ class Options;
 class Image;
 class Log;
 class Subtitle;
+class AudioBuffers;
 
 /** @class Encoder
  *  @brief Parent class for classes which can encode video and audio frames.
@@ -67,7 +68,7 @@ public:
 	 *  @param d Array of pointers to floating point sample data for each channel.
 	 *  @param s Number of frames (ie number of samples in each channel)
 	 */
-	virtual void process_audio (float** d, int s) = 0;
+	virtual void process_audio (boost::shared_ptr<const AudioBuffers>) = 0;
 
 	/** Called when a processing run has finished */
 	virtual void process_end () = 0;
diff --git a/src/lib/imagemagick_encoder.h b/src/lib/imagemagick_encoder.h
index 8e9c416ae..06f1723b1 100644
--- a/src/lib/imagemagick_encoder.h
+++ b/src/lib/imagemagick_encoder.h
@@ -38,6 +38,6 @@ public:
 
 	void process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format) {}
 	void process_video (boost::shared_ptr<Image>, int, boost::shared_ptr<Subtitle>);
-	void process_audio (float**, int) {}
+	void process_audio (boost::shared_ptr<const AudioBuffers>) {}
 	void process_end () {}
 };
diff --git a/src/lib/j2k_still_encoder.h b/src/lib/j2k_still_encoder.h
index 3c8f236ee..7e7719321 100644
--- a/src/lib/j2k_still_encoder.h
+++ b/src/lib/j2k_still_encoder.h
@@ -38,6 +38,6 @@ public:
 
 	void process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format) {}
 	void process_video (boost::shared_ptr<Image>, int, boost::shared_ptr<Subtitle>);
-	void process_audio (float**, int) {}
+	void process_audio (boost::shared_ptr<const AudioBuffers>) {}
 	void process_end () {}
 };
diff --git a/src/lib/j2k_wav_encoder.cc b/src/lib/j2k_wav_encoder.cc
index 7697a5e78..a83c283d7 100644
--- a/src/lib/j2k_wav_encoder.cc
+++ b/src/lib/j2k_wav_encoder.cc
@@ -303,13 +303,10 @@ J2KWAVEncoder::process_end ()
 #if HAVE_SWRESAMPLE	
 	if (_swr_context) {
 
-		float* out[_fs->audio_channels()];
-		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			out[i] = new float[256];
-		}
+		shared_ptr<AudioBuffers> out (new AudioBuffers (_fs->audio_channels(), 256));
 			
 		while (1) {
-			int const frames = swr_convert (_swr_context, (uint8_t **) out, 256, 0, 0);
+			int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0);
 
 			if (frames < 0) {
 				throw EncodeError ("could not run sample-rate converter");
@@ -319,11 +316,7 @@ J2KWAVEncoder::process_end ()
 				break;
 			}
 
-			write_audio (out, frames);
-		}
-
-		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			delete[] out[i];
+			write_audio (out);
 		}
 
 		swr_free (&_swr_context);
@@ -342,50 +335,43 @@ J2KWAVEncoder::process_end ()
 }
 
 void
-J2KWAVEncoder::process_audio (float** data, int frames)
+J2KWAVEncoder::process_audio (shared_ptr<const AudioBuffers> audio)
 {
-	float* resampled[_fs->audio_channels()];
+	shared_ptr<AudioBuffers> resampled;
 	
-#if HAVE_SWRESAMPLE	
+#if HAVE_SWRESAMPLE
 	/* Maybe sample-rate convert */
 	if (_swr_context) {
 
 		/* Compute the resampled frames count and add 32 for luck */
-		int const max_resampled_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate()) + 32;
+		int const max_resampled_frames = ceil (audio->frames() * _fs->target_sample_rate() / _fs->audio_sample_rate()) + 32;
 
-		/* Make a buffer to put the result in */
-		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			resampled[i] = new float[max_resampled_frames];
-		}
+		resampled.reset (new AudioBuffers (_fs->audio_channels(), max_resampled_frames));
 
 		/* Resample audio */
-		int const resampled_frames = swr_convert (_swr_context, (uint8_t **) resampled, max_resampled_frames, (uint8_t const **) data, frames);
+		int const resampled_frames = swr_convert (
+			_swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) audio->data(), audio->frames()
+			);
+		
 		if (resampled_frames < 0) {
 			throw EncodeError ("could not run sample-rate converter");
 		}
 
+		resampled->set_frames (resampled_frames);
+		
 		/* And point our variables at the resampled audio */
-		data = resampled;
-		frames = resampled_frames;
+		audio = resampled;
 	}
 #endif
 
-	write_audio (data, frames);
-
-#if HAVE_SWRESAMPLE
-	if (_swr_context) {
-		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			delete[] resampled[i];
-		}
-	}
-#endif	
+	write_audio (audio);
 }
 
 void
-J2KWAVEncoder::write_audio (float** data, int frames)
+J2KWAVEncoder::write_audio (shared_ptr<const AudioBuffers> audio) const
 {
 	for (int i = 0; i < _fs->audio_channels(); ++i) {
-		sf_write_float (_sound_files[i], data[i], frames);
+		sf_write_float (_sound_files[i], audio->data(i), audio->frames());
 	}
 }
 
diff --git a/src/lib/j2k_wav_encoder.h b/src/lib/j2k_wav_encoder.h
index 3fdefcb38..2c18a5730 100644
--- a/src/lib/j2k_wav_encoder.h
+++ b/src/lib/j2k_wav_encoder.h
@@ -39,6 +39,7 @@ class DCPVideoFrame;
 class Image;
 class Log;
 class Subtitle;
+class AudioBuffers;
 
 /** @class J2KWAVEncoder
  *  @brief An encoder which writes JPEG2000 and WAV files.
@@ -51,12 +52,12 @@ public:
 
 	void process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format);
 	void process_video (boost::shared_ptr<Image>, int, boost::shared_ptr<Subtitle>);
-	void process_audio (float**, int);
+	void process_audio (boost::shared_ptr<const AudioBuffers>);
 	void process_end ();
 
 private:
 
-	void write_audio (float** data, int frames);
+	void write_audio (boost::shared_ptr<const AudioBuffers> audio) const;
 	void encoder_thread (ServerDescription *);
 	void close_sound_files ();
 	void terminate_worker_threads ();
diff --git a/src/lib/util.cc b/src/lib/util.cc
index 3f5200ead..52a75474b 100644
--- a/src/lib/util.cc
+++ b/src/lib/util.cc
@@ -693,3 +693,39 @@ get_optional_int (multimap<string, string> const & kv, string k)
 
 	return lexical_cast<int> (i->second);
 }
+
+AudioBuffers::AudioBuffers (int channels, int frames)
+	: _channels (channels)
+	, _frames (frames)
+{
+	_data = new float*[_channels];
+	for (int i = 0; i < _channels; ++i) {
+		_data[i] = new float[frames];
+	}
+}
+
+AudioBuffers::~AudioBuffers ()
+{
+	for (int i = 0; i < _channels; ++i) {
+		delete[] _data[i];
+	}
+
+	delete[] _data;
+}
+
+float*
+AudioBuffers::data (int c) const
+{
+	assert (c >= 0 && c < _channels);
+	return _data[c];
+}
+	
+void
+AudioBuffers::set_frames (int f)
+{
+	assert (f <= _frames);
+	_frames = f;
+}
+
+	
+	
diff --git a/src/lib/util.h b/src/lib/util.h
index f2087b3f0..c98049f62 100644
--- a/src/lib/util.h
+++ b/src/lib/util.h
@@ -203,5 +203,29 @@ private:
 	int _buffer_data;
 };
 
+class AudioBuffers
+{
+public:
+	AudioBuffers (int channels, int frames);
+	~AudioBuffers ();
+
+	float** data () const {
+		return _data;
+	}
+	
+	float* data (int) const;
+
+	int frames () const {
+		return _frames;
+	}
+
+	void set_frames (int f);
+
+private:
+	int _channels;
+	int _frames;
+	float** _data;
+};
+
 #endif