Reverting to the full lsr API. Turns out the simple API needs all of the

audio at once, so it won't work for us.  The old full API code was still
heavily broken, as each call to pcm_convertSampleRate() used the same
state, even if it was processing two streams of audio.  The new code keeps
a separate state for each audio stream that's being converted.

git-svn-id: https://svn.musicpd.org/mpd/trunk@6255 09075e82-0dd4-0310-85a5-a0d7c8717e4f

1 files changed, 52 insertions, 29 deletions

diff --git a/src/pcm_utils.c b/src/pcm_utils.c
index b3c6689ae..8cc19a9d4 100644
--- a/src/pcm_utils.c
+++ b/src/pcm_utils.c
@@ -27,10 +27,6 @@
 #include <math.h>
 #include <assert.h>
 
-#ifdef HAVE_LIBSAMPLERATE
-#include <samplerate.h>
-#endif
-
 void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
                       int volume)
 {
@@ -189,47 +185,74 @@ static int pcm_getSampleRateConverter(void)
 static int pcm_convertSampleRate(mpd_sint8 channels, mpd_uint32 inSampleRate,
                                  char *inBuffer, size_t inSize,
                                  mpd_uint32 outSampleRate, char *outBuffer,
-                                 size_t outSize)
+                                 size_t outSize, ConvState *convState)
 {
 	static int convalgo = -1;
-	static SRC_DATA data;
-	static size_t dataInSize;
-	static size_t dataOutSize;
-	size_t curDataInSize;
-	size_t curDataOutSize;
+	SRC_DATA *data = &convState->data;
+	size_t dataInSize;
+	size_t dataOutSize;
 	int error;
 
 	if (convalgo < 0)
 		convalgo = pcm_getSampleRateConverter();
 
-	data.src_ratio = (double)outSampleRate / (double)inSampleRate;
+	/* (re)set the state/ratio if the in or out format changed */
+	if ((channels != convState->lastChannels) ||
+	    (inSampleRate != convState->lastInSampleRate) ||
+	    (outSampleRate != convState->lastOutSampleRate)) {
+		convState->error = 0;
+		convState->lastChannels = channels;
+		convState->lastInSampleRate = inSampleRate;
+		convState->lastOutSampleRate = outSampleRate;
+
+		if (convState->state)
+			convState->state = src_delete(convState->state);
+
+		convState->state = src_new(convalgo, channels, &error);
+		if (!convState->state) {
+			ERROR("cannot create new libsamplerate state: %s\n",
+			      src_strerror(error));
+			convState->error = 1;
+			return 0;
+		}
+		
+		data->src_ratio = (double)outSampleRate / (double)inSampleRate;
+		DEBUG("setting samplerate conversion ratio to %.2lf\n",
+		      data->src_ratio);
+		src_set_ratio(convState->state, data->src_ratio);
+	}
+
+	/* there was an error previously, and nothing has changed */
+	if (convState->error)
+		return 0;
 
-	data.input_frames = inSize / 2 / channels;
-	curDataInSize = data.input_frames * sizeof(float) * channels;
-	if (curDataInSize > dataInSize) {
-		dataInSize = curDataInSize;
-		data.data_in = xrealloc(data.data_in, dataInSize);
+	data->input_frames = inSize / 2 / channels;
+	dataInSize = data->input_frames * sizeof(float) * channels;
+	if (dataInSize > convState->dataInSize) {
+		convState->dataInSize = dataInSize;
+		data->data_in = xrealloc(data->data_in, dataInSize);
 	}
 
-	data.output_frames = outSize / 2 / channels;
-	curDataOutSize = data.output_frames * sizeof(float) * channels;
-	if (curDataOutSize > dataOutSize) {
-		dataOutSize = curDataOutSize;
-		data.data_out = xrealloc(data.data_out, dataOutSize);
+	data->output_frames = outSize / 2 / channels;
+	dataOutSize = data->output_frames * sizeof(float) * channels;
+	if (dataOutSize > convState->dataOutSize) {
+		convState->dataOutSize = dataOutSize;
+		data->data_out = xrealloc(data->data_out, dataOutSize);
 	}
 
-	src_short_to_float_array((short *)inBuffer, data.data_in,
-	                         data.input_frames * channels);
+	src_short_to_float_array((short *)inBuffer, data->data_in,
+	                         data->input_frames * channels);
 
-	error = src_simple(&data, convalgo, channels);
+	error = src_process(convState->state, data);
 	if (error) {
 		ERROR("error processing samples with libsamplerate: %s\n",
 		      src_strerror(error));
+		convState->error = 1;
 		return 0;
 	}
 
-	src_float_to_short_array(data.data_out, (short *)outBuffer,
-	                         data.output_frames_gen * channels);
+	src_float_to_short_array(data->data_out, (short *)outBuffer,
+	                         data->output_frames_gen * channels);
 
 	return 1;
 }
@@ -238,7 +261,7 @@ static int pcm_convertSampleRate(mpd_sint8 channels, mpd_uint32 inSampleRate,
 static int pcm_convertSampleRate(mpd_sint8 channels, mpd_uint32 inSampleRate,
                                  char *inBuffer, size_t inSize,
                                  mpd_uint32 outSampleRate, char *outBuffer,
-                                 size_t outSize)
+                                 size_t outSize, ConvState *convState)
 {
 	mpd_uint32 rd_dat = 0;
 	mpd_uint32 wr_dat = 0;
@@ -370,7 +393,7 @@ static char *pcm_convertTo16bit(mpd_sint8 bits, char *inBuffer, size_t inSize,
 /* outFormat bits must be 16 and channels must be 1 or 2! */
 void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer,
                             size_t inSize, AudioFormat * outFormat,
-                            char *outBuffer)
+                            char *outBuffer, ConvState *convState)
 {
 	char *buf;
 	size_t len;
@@ -397,7 +420,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer,
 		if (!pcm_convertSampleRate(outFormat->channels,
 		                           inFormat->sampleRate, buf, len,
 		                           outFormat->sampleRate, outBuffer,
-		                           outSize))
+		                           outSize, convState))
 			exit(EXIT_FAILURE);
 	}
 }