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Commit 407497c4 authored by J. Alexander Treuman's avatar J. Alexander Treuman
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Split pcm_convertAudioFormat into separate functions for bitrate, channel,

and samplerate conversion. This makes the code much easier to read, and fixes a few bugs that were previously there. git-svn-id: https://svn.musicpd.org/mpd/trunk@6224 09075e82-0dd4-0310-85a5-a0d7c8717e4f
parent e6d7663b
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Showing with 191 additions and 160 deletions
src/audioOutput.c
View file @ 407497c4
...@@ -194,9 +194,7 @@ int openAudioOutput(AudioOutput * audioOutput, AudioFormat * audioFormat) ...@@ -194,9 +194,7 @@ int openAudioOutput(AudioOutput * audioOutput, AudioFormat * audioFormat)
static void convertAudioFormat(AudioOutput * audioOutput, char **chunkArgPtr, static void convertAudioFormat(AudioOutput * audioOutput, char **chunkArgPtr,
int *sizeArgPtr) int *sizeArgPtr)
{ {
int size = int size = pcm_sizeOfConvBuffer(&(audioOutput->inAudioFormat),
pcm_sizeOfOutputBufferForAudioFormatConversion(
&(audioOutput->inAudioFormat),
*sizeArgPtr, *sizeArgPtr,
&(audioOutput->outAudioFormat)); &(audioOutput->outAudioFormat));
......
src/outputBuffer.c
View file @ 407497c4
...@@ -82,13 +82,8 @@ int sendDataToOutputBuffer(OutputBuffer * cb, InputStream * inStream, ...@@ -82,13 +82,8 @@ int sendDataToOutputBuffer(OutputBuffer * cb, InputStream * inStream,
data = dataIn; data = dataIn;
datalen = dataInLen; datalen = dataInLen;
} else { } else {
datalen = datalen = pcm_sizeOfConvBuffer(&(dc->audioFormat), dataInLen,
pcm_sizeOfOutputBufferForAudioFormatConversion(& &(cb->audioFormat));
(dc->
audioFormat),
dataInLen,
&(cb->
audioFormat));
if (datalen > convBufferLen) { if (datalen > convBufferLen) {
convBuffer = xrealloc(convBuffer, datalen); convBuffer = xrealloc(convBuffer, datalen);
convBufferLen = datalen; convBufferLen = datalen;
......
src/pcm_utils.c
View file @ 407497c4
...@@ -153,7 +153,7 @@ void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1, ...@@ -153,7 +153,7 @@ void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
} }
#ifdef HAVE_LIBSAMPLERATE #ifdef HAVE_LIBSAMPLERATE
static int pcm_getSamplerateConverter(void) static int pcm_getSampleRateConverter(void)
{ {
const char *conf, *test; const char *conf, *test;
int convalgo = SRC_SINC_FASTEST; int convalgo = SRC_SINC_FASTEST;
...@@ -185,169 +185,84 @@ static int pcm_getSamplerateConverter(void) ...@@ -185,169 +185,84 @@ static int pcm_getSamplerateConverter(void)
} }
#endif #endif
/* 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)
{
static char *bitConvBuffer;
static int bitConvBufferLength;
static char *channelConvBuffer;
static int channelConvBufferLength;
char *dataChannelConv;
int dataChannelLen;
char *dataBitConv;
int dataBitLen;
assert(outFormat->bits == 16);
assert(outFormat->channels == 2 || outFormat->channels == 1);
/* convert to 16 bit audio */
switch (inFormat->bits) {
case 8:
dataBitLen = inSize << 1;
if (dataBitLen > bitConvBufferLength) {
bitConvBuffer = xrealloc(bitConvBuffer, dataBitLen);
bitConvBufferLength = dataBitLen;
}
dataBitConv = bitConvBuffer;
{
mpd_sint8 *in = (mpd_sint8 *) inBuffer;
mpd_sint16 *out = (mpd_sint16 *) dataBitConv;
int i;
for (i = 0; i < inSize; i++) {
*out++ = (*in++) << 8;
}
}
break;
case 16:
dataBitConv = inBuffer;
dataBitLen = inSize;
break;
case 24:
/* put dithering code from mp3_decode here */
default:
ERROR("only 8 or 16 bits are supported for conversion!\n");
exit(EXIT_FAILURE);
}
/* convert audio between mono and stereo */
if (inFormat->channels == outFormat->channels) {
dataChannelConv = dataBitConv;
dataChannelLen = dataBitLen;
} else {
switch (inFormat->channels) {
case 1: /* convert from 1 -> 2 channels */
dataChannelLen = (dataBitLen >> 1) << 2;
if (dataChannelLen > channelConvBufferLength) {
channelConvBuffer = xrealloc(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: /* convert from 2 -> 1 channels */
dataChannelLen = dataBitLen >> 1;
if (dataChannelLen > channelConvBufferLength) {
channelConvBuffer = xrealloc(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) {
memcpy(outBuffer, dataChannelConv, dataChannelLen);
} else {
#ifdef HAVE_LIBSAMPLERATE #ifdef HAVE_LIBSAMPLERATE
static SRC_STATE *state = NULL; static int pcm_convertSampleRate(mpd_sint8 channels, mpd_uint32 inSampleRate,
char *inBuffer, size_t inSize,
mpd_uint32 outSampleRate, char *outBuffer,
size_t outSize)
{
static SRC_STATE *state;
static SRC_DATA data; static SRC_DATA data;
static size_t data_in_size, data_out_size; static size_t dataInSize;
static size_t dataOutSize;
size_t curDataInSize;
size_t curDataOutSize;
double ratio;
int error; int error;
static double ratio = 0;
double newratio;
if(!state) { if (!state) {
state = src_new(pcm_getSamplerateConverter(), outFormat->channels, &error); state = src_new(pcm_getSampleRateConverter(), channels, &error);
if(!state) { if (!state) {
ERROR("Cannot create new samplerate state: %s\n", src_strerror(error)); ERROR("Cannot create new samplerate state: %s\n",
exit(EXIT_FAILURE); src_strerror(error));
} else { return 0;
DEBUG("Samplerate converter initialized\n");
} }
DEBUG("Samplerate converter initialized\n");
} }
newratio = (double)outFormat->sampleRate / (double)inFormat->sampleRate; ratio = (double)outSampleRate / (double)inSampleRate;
if(newratio != ratio) { if (ratio != data.src_ratio) {
DEBUG("Setting samplerate conversion ratio to %.2lf\n", newratio); DEBUG("Setting samplerate conversion ratio to %.2lf\n", ratio);
src_set_ratio(state, newratio); src_set_ratio(state, ratio);
ratio = newratio; data.src_ratio = ratio;
} }
data.input_frames = dataChannelLen / 2 / outFormat->channels; data.input_frames = inSize / 2 / channels;
data.output_frames = pcm_sizeOfOutputBufferForAudioFormatConversion(inFormat, dataChannelLen, outFormat) / 2 / outFormat->channels; curDataInSize = data.input_frames * sizeof(float) * channels;
data.src_ratio = (double)data.output_frames / (double)data.input_frames; if (curDataInSize > dataInSize) {
dataInSize = curDataInSize;
if (data_in_size != (data.input_frames * data.data_in = xrealloc(data.data_in, dataInSize);
outFormat->channels)) {
data_in_size = data.input_frames * outFormat->channels;
data.data_in = xrealloc(data.data_in, data_in_size);
} }
if (data_out_size != (data.output_frames *
outFormat->channels)) { data.output_frames = outSize / 2 / channels;
data_out_size = data.output_frames * curDataOutSize = data.output_frames * sizeof(float) * channels;
outFormat->channels; if (curDataOutSize > dataOutSize) {
data.data_out = xrealloc(data.data_out, data_out_size); dataOutSize = curDataOutSize;
data.data_out = xrealloc(data.data_out, dataOutSize);
} }
src_short_to_float_array((short *)dataChannelConv, data.data_in, data.input_frames * outFormat->channels); src_short_to_float_array((short *)inBuffer, data.data_in,
data.input_frames * channels);
error = src_process(state, &data); error = src_process(state, &data);
if(error) { if (error) {
ERROR("Cannot process samples: %s\n", src_strerror(error)); ERROR("Cannot process samples: %s\n", src_strerror(error));
exit(EXIT_FAILURE); return 0;
} }
src_float_to_short_array(data.data_out, (short *)outBuffer, data.output_frames * outFormat->channels); src_float_to_short_array(data.data_out, (short *)outBuffer,
#else data.output_frames_gen * channels);
/* resampling code blatantly ripped from ESD */
return 1;
}
#else /* !HAVE_LIBSAMPLERATE */
/* resampling code blatantly ripped from ESD */
static int pcm_convertSampleRate(mpd_sint8 channels, mpd_uint32 inSampleRate,
char *inBuffer, size_t inSize,
mpd_uint32 outSampleRate, char *outBuffer,
size_t outSize)
{
mpd_uint32 rd_dat = 0; mpd_uint32 rd_dat = 0;
mpd_uint32 wr_dat = 0; mpd_uint32 wr_dat = 0;
mpd_sint16 *in = (mpd_sint16 *)inBuffer;
mpd_sint16 *out = (mpd_sint16 *)outBuffer;
mpd_uint32 nlen = outSize / 2;
mpd_sint16 lsample, rsample; mpd_sint16 lsample, rsample;
mpd_sint16 *out = (mpd_sint16 *) outBuffer;
mpd_sint16 *in = (mpd_sint16 *) dataChannelConv;
mpd_uint32 nlen = pcm_sizeOfOutputBufferForAudioFormatConversion(inFormat, inSize, outFormat) / sizeof(mpd_sint16);
switch (outFormat->channels) { switch (channels) {
case 1: case 1:
while (wr_dat < nlen) { while (wr_dat < nlen) {
rd_dat = wr_dat * inFormat->sampleRate / rd_dat = wr_dat * inSampleRate / outSampleRate;
outFormat->sampleRate;
lsample = in[rd_dat++]; lsample = in[rd_dat++];
...@@ -356,8 +271,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, ...@@ -356,8 +271,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer,
break; break;
case 2: case 2:
while (wr_dat < nlen) { while (wr_dat < nlen) {
rd_dat = wr_dat * inFormat->sampleRate / rd_dat = wr_dat * inSampleRate / outSampleRate;
outFormat->sampleRate;
rd_dat &= ~1; rd_dat &= ~1;
lsample = in[rd_dat++]; lsample = in[rd_dat++];
...@@ -368,14 +282,139 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, ...@@ -368,14 +282,139 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer,
} }
break; break;
} }
#endif
return 1;
}
#endif /* !HAVE_LIBSAMPLERATE */
static char *pcm_convertChannels(mpd_sint8 inChannels, char *inBuffer,
size_t inSize, size_t *outSize)
{
static char *buf;
static size_t len;
char *outBuffer = NULL;;
mpd_sint16 *in;
mpd_sint16 *out;
int inSamples, i;
switch (inChannels) {
/* convert from 1 -> 2 channels */
case 1:
*outSize = (inSize >> 1) << 2;
if (*outSize > len) {
len = *outSize;
buf = xrealloc(buf, len);
} }
outBuffer = buf;
return; inSamples = inSize >> 1;
in = (mpd_sint16 *)inBuffer;
out = (mpd_sint16 *)outBuffer;
for (i = 0; i < inSamples; i++) {
*out++ = *in;
*out++ = *in++;
}
break;
/* convert from 2 -> 1 channels */
case 2:
*outSize = inSize >> 1;
if (*outSize > len) {
len = *outSize;
buf = xrealloc(buf, len);
}
outBuffer = buf;
inSamples = inSize >> 2;
in = (mpd_sint16 *)inBuffer;
out = (mpd_sint16 *)outBuffer;
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");
}
return outBuffer;
}
static char *pcm_convertTo16bit(mpd_sint8 inBits, char *inBuffer, size_t inSize,
size_t *outSize)
{
static char *buf;
static size_t len;
char *outBuffer = NULL;
mpd_sint8 *in;
mpd_sint16 *out;
int i;
switch (inBits) {
case 8:
*outSize = inSize << 1;
if (*outSize > len) {
len = *outSize;
buf = xrealloc(buf, len);
}
outBuffer = buf;
in = (mpd_sint8 *)inBuffer;
out = (mpd_sint16 *)outBuffer;
for (i = 0; i < inSize; i++)
*out++ = (*in++) << 8;
break;
case 16:
*outSize = inSize;
outBuffer = inBuffer;
break;
case 24:
/* put dithering code from mp3_decode here */
default:
ERROR("only 8 or 16 bits are supported for conversion!\n");
}
return outBuffer;
}
/* 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 *buf;
size_t len;
size_t outSize = pcm_sizeOfConvBuffer(inFormat, inSize, outFormat);
assert(outFormat->bits == 16);
assert(outFormat->channels == 2 || outFormat->channels == 1);
/* everything else supports 16 bit only, so convert to that first */
buf = pcm_convertTo16bit(inFormat->bits, inBuffer, inSize, &len);
if (!buf)
exit(EXIT_FAILURE);
if (inFormat->channels != outFormat->channels) {
buf = pcm_convertChannels(inFormat->channels, buf, len, &len);
if (!buf)
exit(EXIT_FAILURE);
}
if (inFormat->sampleRate == outFormat->sampleRate) {
assert(outSize >= len);
memcpy(outBuffer, buf, len);
} else {
if (!pcm_convertSampleRate(outFormat->channels,
inFormat->sampleRate, buf, len,
outFormat->sampleRate, outBuffer,
outSize))
exit(EXIT_FAILURE);
}
} }
size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat, size_t pcm_sizeOfConvBuffer(AudioFormat * inFormat, size_t inSize,
size_t inSize,
AudioFormat * outFormat) AudioFormat * outFormat)
{ {
const int shift = sizeof(mpd_sint16) * outFormat->channels; const int shift = sizeof(mpd_sint16) * outFormat->channels;
......
src/pcm_utils.h
View file @ 407497c4
...@@ -34,7 +34,6 @@ void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1, ...@@ -34,7 +34,6 @@ void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
inSize, AudioFormat * outFormat, char *outBuffer); inSize, AudioFormat * outFormat, char *outBuffer);
size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat, size_t pcm_sizeOfConvBuffer(AudioFormat * inFormat, size_t inSize,
size_t inSize,
AudioFormat * outFormat); AudioFormat * outFormat);
#endif #endif
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