configure shout encoding quality and audio format

git-svn-id: https://svn.musicpd.org/mpd/trunk@2307 09075e82-0dd4-0310-85a5-a0d7c8717e4f

src/audio.c

@@ -66,22 +66,27 @@ void getOutputAudioFormat(AudioFormat * inAudioFormat,
 
 void initAudioConfig() {
         char * conf = getConf()[CONF_AUDIO_OUTPUT_FORMAT];
-        char * test;
 
         if(NULL == conf) return;
 
         audio_configFormat = malloc(sizeof(AudioFormat));
 
-        memset(audio_configFormat,0,sizeof(AudioFormat));
+	if(0 != parseAudioConfig(audio_configFormat, conf)) exit(EXIT_FAILURE);
+}
 
-        audio_configFormat->sampleRate = strtol(conf,&test,10);
+int parseAudioConfig(AudioFormat * audioFormat, char * conf) {
+        char * test;
+
+        memset(audioFormat,0,sizeof(AudioFormat));
+
+        audioFormat->sampleRate = strtol(conf,&test,10);
        
         if(*test!=':') {
                 ERROR("error parsing audio output format: %s\n",conf);
-                exit(EXIT_FAILURE);
+		return -1;
         }
  
-        /*switch(audio_configFormat->sampleRate) {
+        /*switch(audioFormat->sampleRate) {
         case 48000:
         case 44100:
         case 32000:
@@ -89,47 +94,50 @@ void initAudioConfig() {
                 break;
         default:
                 ERROR("sample rate %i can not be used for audio output\n",
-                        (int)audio_configFormat->sampleRate);
+                        (int)audioFormat->sampleRate);
-                exit(EXIT_FAILURE);
+		return -1
         }*/
 
-        if(audio_configFormat->sampleRate <= 0) {
+        if(audioFormat->sampleRate <= 0) {
                 ERROR("sample rate %i is not >= 0\n",
-                                (int)audio_configFormat->sampleRate);
+                                (int)audioFormat->sampleRate);
-                exit(EXIT_FAILURE);
+		return -1;
         }
 
-        audio_configFormat->bits = strtol(test+1,&test,10);
+        audioFormat->bits = strtol(test+1,&test,10);
         
         if(*test!=':') {
                 ERROR("error parsing audio output format: %s\n",conf);
-                exit(EXIT_FAILURE);
+		return -1;
         }
 
-        switch(audio_configFormat->bits) {
+        switch(audioFormat->bits) {
         case 16:
                 break;
         default:
                 ERROR("bits %i can not be used for audio output\n",
-                        (int)audio_configFormat->bits);
+                        (int)audioFormat->bits);
-                exit(EXIT_FAILURE);
+		return -1;
         }
 
-        audio_configFormat->channels = strtol(test+1,&test,10);
+        audioFormat->channels = strtol(test+1,&test,10);
         
         if(*test!='\0') {
                 ERROR("error parsing audio output format: %s\n",conf);
-                exit(EXIT_FAILURE);
+		return -1;
         }
 
-        switch(audio_configFormat->channels) {
+        switch(audioFormat->channels) {
+	case 1:
         case 2:
                 break;
         default:
                 ERROR("channels %i can not be used for audio output\n",
-                        (int)audio_configFormat->channels);
+                        (int)audioFormat->channels);
-                exit(EXIT_FAILURE);
+		return -1;
         }
+
+	return 0;
 }
 
 void finishAudioConfig() {

src/audio.h

@@ -35,6 +35,8 @@ typedef struct _AudioFormat {
 
 void getOutputAudioFormat(AudioFormat * inFormat, AudioFormat * outFormat);
 
+int parseAudioConfig(AudioFormat * audioFormat, char * conf);
+
 void initAudioConfig();
 
 void finishAudioConfig();

src/audioOutput_shout.c

@@ -24,6 +24,7 @@
 #include "conf.h"
 #include "log.h"
 #include "sig_handlers.h"
+#include "pcm_utils.h"
 
 #include <stdlib.h>
 #include <string.h>
@@ -54,6 +55,15 @@ typedef struct _ShoutData {
 	vorbis_comment vc;
 
 	int serialno;
+
+	float quality;
+	AudioFormat outAudioFormat;
+	AudioFormat inAudioFormat;
+
+	char * convBuffer;
+	long convBufferLen;
+	/* shoud we convert the audio to a different format? */
+	int audioFormatConvert;
 } ShoutData;
 
 static ShoutData * newShoutData() {
@@ -61,6 +71,8 @@ static ShoutData * newShoutData() {
 
 	ret->shoutConn = shout_new();
 	ret->serialno = rand();
+	ret->convBuffer = NULL;
+	ret->convBufferLen = 0;
 
 	return ret;
 }
@@ -112,6 +124,16 @@ static int shout_initDriver(AudioOutput * audioOutput) {
 		exit(EXIT_FAILURE);
 	}
 
+	if(!getConf()[CONF_SHOUT_QUALITY]) {
+		ERROR("shout host defined but not shout quality\n");
+		exit(EXIT_FAILURE);
+	}
+
+	if(!getConf()[CONF_SHOUT_FORMAT]) {
+		ERROR("shout host defined but not shout format\n");
+		exit(EXIT_FAILURE);
+	}
+
 	host = getConf()[CONF_SHOUT_HOST];
 	passwd = getConf()[CONF_SHOUT_PASSWD];
 	user = getConf()[CONF_SHOUT_USER];
@@ -126,6 +148,20 @@ static int shout_initDriver(AudioOutput * audioOutput) {
 		exit(EXIT_FAILURE);
 	}
 
+	sd->quality = strtod(getConf()[CONF_SHOUT_QUALITY], &test);
+
+	if(*test != '\0' || sd->quality < 0.0 || sd->quality > 10.0) {
+		ERROR("shout quality \"%s\" is not a number in the range "
+				"0-10\n", getConf()[CONF_SHOUT_QUALITY]);
+		exit(EXIT_FAILURE);
+	}
+
+	if(0 != parseAudioConfig(&(sd->outAudioFormat), 
+			getConf()[CONF_SHOUT_FORMAT]) )
+	{
+		exit(EXIT_FAILURE);
+	}
+
 	if(shout_set_host(sd->shoutConn, host) !=  SHOUTERR_SUCCESS ||
 		shout_set_port(sd->shoutConn, port) != SHOUTERR_SUCCESS ||
 		shout_set_password(sd->shoutConn, passwd) != SHOUTERR_SUCCESS ||
@@ -202,8 +238,8 @@ static int shout_openDevice(AudioOutput * audioOutput,
 
 	vorbis_info_init(&(sd->vi));
 
-	if( 0 != vorbis_encode_init_vbr(&(sd->vi), audioFormat->channels,
+	if( 0 != vorbis_encode_init_vbr(&(sd->vi), sd->outAudioFormat.channels,
-			audioFormat->sampleRate, 0.5) )
+			sd->outAudioFormat.sampleRate, sd->quality) )
 	{
 		ERROR("problem seting up vorbis encoder for shout\n");
 		vorbis_info_clear(&(sd->vi));
@@ -230,19 +266,59 @@ static int shout_openDevice(AudioOutput * audioOutput,
 		write_page(sd);
 	}
 
+	memcpy(&(sd->inAudioFormat), audioFormat, sizeof(AudioFormat));
+
+	if(0 == memcmp(&(sd->inAudioFormat), &(sd->outAudioFormat), 
+			sizeof(AudioFormat))) 
+	{
+		sd->audioFormatConvert = 0;
+	}
+	else sd->audioFormatConvert = 1;
+
 	return 0;
 }
 
+static void shout_convertAudioFormat(ShoutData * sd, char ** chunkArgPtr,
+		int * sizeArgPtr)
+{
+	int size = pcm_sizeOfOutputBufferForAudioFormatConversion(
+			&(sd->inAudioFormat), *sizeArgPtr, 
+			&(sd->outAudioFormat));
+
+	if(size > sd->convBufferLen) {
+		sd->convBuffer = realloc(sd->convBuffer, size);
+		sd->convBufferLen = size;
+	}
+
+	pcm_convertAudioFormat(&(sd->inAudioFormat), *chunkArgPtr, *sizeArgPtr,
+			&(sd->outAudioFormat), sd->convBuffer);
+	
+	*sizeArgPtr = size;
+	*chunkArgPtr = sd->convBuffer;
+}
 
 static int shout_play(AudioOutput * audioOutput, char * playChunk, int size) {
 	int i,j;
 	ShoutData * sd = (ShoutData *)audioOutput->data;
+	float ** vorbbuf;
+	int samples;
+	int bytes = sd->outAudioFormat.bits/8;
 
-	float **vorbbuf = vorbis_analysis_buffer(&(sd->vd), size/4);
+	if(sd->audioFormatConvert) {
+		shout_convertAudioFormat(sd, &playChunk, &size);
+	}
 
-	for(i=0, j=0; i < size; i+=4, j++) {
+	samples = size/(bytes*sd->outAudioFormat.channels);
-		vorbbuf[0][j] = (*((mpd_sint16 *)(playChunk+i))) / 32768.0;
+
-		vorbbuf[1][j] = (*((mpd_sint16 *)(playChunk+i+2))) / 32768.0;
+	/* this is for only 16-bit audio */
+
+	vorbbuf = vorbis_analysis_buffer(&(sd->vd), samples);
+
+	for(i=0; i<samples; i++) {
+		for(j=0; j<sd->outAudioFormat.channels; j++) {
+			vorbbuf[j][i] = (*((mpd_sint16 *)playChunk)) / 32768.0;
+			playChunk += bytes;
+		}
 	}
 
 	vorbis_analysis_wrote(&(sd->vd), size/4);

src/conf.c

@@ -37,7 +37,7 @@
 
 #define CONF_COMMENT	'#'
 
-#define CONF_NUMBER_OF_PARAMS		42
+#define CONF_NUMBER_OF_PARAMS		43
 #define CONF_NUMBER_OF_PATHS		6
 #define CONF_NUMBER_OF_REQUIRED		5
 #define CONF_NUMBER_OF_ALLOW_CATS	1
@@ -138,7 +138,8 @@ char ** readConf(char * file) {
 		"shout_name",
 		"shout_user",
 		"shout_quality",
-		"id3v1_encoding"
+		"id3v1_encoding",
+		"shout_format"
 	};
 
 	int conf_absolutePaths[CONF_NUMBER_OF_PATHS] = {

src/conf.h

@@ -63,6 +63,7 @@
 #define CONF_SHOUT_USER				39
 #define CONF_SHOUT_QUALITY			40
 #define CONF_ID3V1_ENCODING			41
+#define CONF_SHOUT_FORMAT			42
 
 #define CONF_CAT_CHAR				"\n"
 

src/outputBuffer.c

@@ -80,7 +80,7 @@ int sendDataToOutputBuffer(OutputBuffer * cb, InputStream * inStream,
 	}
 	else {
 		datalen = pcm_sizeOfOutputBufferForAudioFormatConversion(
-				&(dc->audioFormat), dataIn, dataInLen,
+				&(dc->audioFormat), dataInLen,
 				&(cb->audioFormat));
 		if(datalen > convBufferLen) {
 			convBuffer = realloc(convBuffer,datalen);

src/pcm_utils.c

@@ -153,7 +153,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
 	int dataBitLen;
 
 	assert(outFormat->bits==16);
-	assert(outFormat->channels==2);
+	assert(outFormat->channels==2 || outFormat->channels==1);
 
 	/* converts */
 	switch(inFormat->bits) {
@@ -185,32 +185,55 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
 	}
 
 	/* converts only between 16 bit audio between mono and stereo */
-	switch(inFormat->channels) {
+	if(inFormat->channels == outFormat->channels)
-	case 1:
+	{
-		dataChannelLen = (dataBitLen >> 1) << 2;
-		if(dataChannelLen > channelConvBufferLength) {
-			channelConvBuffer = realloc(channelConvBuffer,
-					dataChannelLen);
-			channelConvBufferLength = dataChannelLen;
-		}
-		dataChannelConv = channelConvBuffer;
-		{
-			mpd_sint16 * in = (mpd_sint16 *)dataBitConv;
-			mpd_sint16 * out = (mpd_sint16 *)dataChannelConv;
-			int i, inSamples = dataBitLen >> 1;
-			for(i=0;i<inSamples;i++) {
-				*out++ = *in;
-				*out++ = *in++;
-			}
-		}
-		break;
-	case 2:
 		dataChannelConv = dataBitConv;
 		dataChannelLen = dataBitLen;
-		break;
+	}
-	default:
+	else {
-		ERROR("only 1 or 2 channels are supported for conversion!\n");
+		switch(inFormat->channels) {
-		exit(EXIT_FAILURE);
+		/* convert from 1 -> 2 channels */
+		case 1:
+			dataChannelLen = (dataBitLen >> 1) << 2;
+			if(dataChannelLen > channelConvBufferLength) {
+				channelConvBuffer = realloc(channelConvBuffer,
+						dataChannelLen);
+				channelConvBufferLength = dataChannelLen;
+			}
+			dataChannelConv = channelConvBuffer;
+			{
+				mpd_sint16 * in = (mpd_sint16 *)dataBitConv;
+				mpd_sint16 * out = (mpd_sint16 *)dataChannelConv;
+				int i, inSamples = dataBitLen >> 1;
+				for(i=0;i<inSamples;i++) {
+					*out++ = *in;
+					*out++ = *in++;
+				}
+			}
+			break;
+		/* convert from 2 -> 1 channels */
+		case 2:
+			dataChannelLen = dataBitLen >> 1;
+			if(dataChannelLen > channelConvBufferLength) {
+				channelConvBuffer = realloc(channelConvBuffer,
+						dataChannelLen);
+				channelConvBufferLength = dataChannelLen;
+			}
+			dataChannelConv = channelConvBuffer;
+			{
+				mpd_sint16 * in = (mpd_sint16 *)dataBitConv;
+				mpd_sint16 * out = (mpd_sint16 *)dataChannelConv;
+				int i, inSamples = dataBitLen >> 2;
+				for(i=0;i<inSamples;i++) {
+					*out = (*in++)/2;
+					*out++ += (*in++)/2;
+				}
+			}
+			break;
+		default:
+			ERROR("only 1 or 2 channels are supported for conversion!\n");
+			exit(EXIT_FAILURE);
+		}
 	}
 
 	if(inFormat->sampleRate == outFormat->sampleRate) {
@@ -247,7 +270,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
 }
 
 size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
-		char * inBuffer, size_t inSize, AudioFormat * outFormat)
+		size_t inSize, AudioFormat * outFormat)
 {
 	const int shift = sizeof(mpd_sint16);
 	size_t outSize = inSize;

src/pcm_utils.h

@@ -37,6 +37,6 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
                 inSize, AudioFormat * outFormat, char * outBuffer);
 
 size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
-		char * inBuffer, size_t inSize, AudioFormat * outFormat);
+		size_t inSize, AudioFormat * outFormat);
 #endif
 /* vim:set shiftwidth=8 tabstop=8 expandtab: */