Skip to content

M
mpd
Commit 407497c4 authored by J. Alexander Treuman's avatar J. Alexander Treuman
Browse files
Options

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
Hide whitespace changes
Inline Side-by-side
Showing with 217 additions and 186 deletions
src/audioOutput.c
View file @ 407497c4
...@@ -194,11 +194,9 @@ int openAudioOutput(AudioOutput * audioOutput, AudioFormat * audioFormat) ...@@ -194,11 +194,9 @@ 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( *sizeArgPtr,
&(audioOutput->inAudioFormat), &(audioOutput->outAudioFormat));
*sizeArgPtr,
&(audioOutput->outAudioFormat));
if (size > audioOutput->convBufferLen) { if (size > audioOutput->convBufferLen) {
audioOutput->convBuffer = audioOutput->convBuffer =
......
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;
...@@ -185,198 +185,237 @@ static int pcm_getSamplerateConverter(void) ...@@ -185,198 +185,237 @@ static int pcm_getSamplerateConverter(void)
} }
#endif #endif
/* outFormat bits must be 16 and channels must be 1 or 2! */ #ifdef HAVE_LIBSAMPLERATE
void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, static int pcm_convertSampleRate(mpd_sint8 channels, mpd_uint32 inSampleRate,
size_t inSize, AudioFormat * outFormat, char *inBuffer, size_t inSize,
char *outBuffer) mpd_uint32 outSampleRate, char *outBuffer,
size_t outSize)
{ {
static char *bitConvBuffer; static SRC_STATE *state;
static int bitConvBufferLength; static SRC_DATA data;
static char *channelConvBuffer; static size_t dataInSize;
static int channelConvBufferLength; static size_t dataOutSize;
char *dataChannelConv; size_t curDataInSize;
int dataChannelLen; size_t curDataOutSize;
char *dataBitConv; double ratio;
int dataBitLen; int error;
if (!state) {
state = src_new(pcm_getSampleRateConverter(), channels, &error);
if (!state) {
ERROR("Cannot create new samplerate state: %s\n",
src_strerror(error));
return 0;
}
DEBUG("Samplerate converter initialized\n");
}
assert(outFormat->bits == 16); ratio = (double)outSampleRate / (double)inSampleRate;
assert(outFormat->channels == 2 || outFormat->channels == 1); if (ratio != data.src_ratio) {
DEBUG("Setting samplerate conversion ratio to %.2lf\n", ratio);
src_set_ratio(state, ratio);
data.src_ratio = ratio;
}
/* convert to 16 bit audio */ data.input_frames = inSize / 2 / channels;
switch (inFormat->bits) { curDataInSize = data.input_frames * sizeof(float) * channels;
case 8: if (curDataInSize > dataInSize) {
dataBitLen = inSize << 1; dataInSize = curDataInSize;
if (dataBitLen > bitConvBufferLength) { data.data_in = xrealloc(data.data_in, dataInSize);
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 */ data.output_frames = outSize / 2 / channels;
if (inFormat->channels == outFormat->channels) { curDataOutSize = data.output_frames * sizeof(float) * channels;
dataChannelConv = dataBitConv; if (curDataOutSize > dataOutSize) {
dataChannelLen = dataBitLen; dataOutSize = curDataOutSize;
} else { data.data_out = xrealloc(data.data_out, dataOutSize);
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) { src_short_to_float_array((short *)inBuffer, data.data_in,
memcpy(outBuffer, dataChannelConv, dataChannelLen); data.input_frames * channels);
} else {
#ifdef HAVE_LIBSAMPLERATE error = src_process(state, &data);
static SRC_STATE *state = NULL; if (error) {
static SRC_DATA data; ERROR("Cannot process samples: %s\n", src_strerror(error));
static size_t data_in_size, data_out_size; return 0;
int error; }
static double ratio = 0;
double newratio; src_float_to_short_array(data.data_out, (short *)outBuffer,
data.output_frames_gen * channels);
if(!state) {
state = src_new(pcm_getSamplerateConverter(), outFormat->channels, &error); return 1;
if(!state) { }
ERROR("Cannot create new samplerate state: %s\n", src_strerror(error)); #else /* !HAVE_LIBSAMPLERATE */
exit(EXIT_FAILURE); /* resampling code blatantly ripped from ESD */
} else { static int pcm_convertSampleRate(mpd_sint8 channels, mpd_uint32 inSampleRate,
DEBUG("Samplerate converter initialized\n"); char *inBuffer, size_t inSize,
} mpd_uint32 outSampleRate, char *outBuffer,
size_t outSize)
{
mpd_uint32 rd_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;
switch (channels) {
case 1:
while (wr_dat < nlen) {
rd_dat = wr_dat * inSampleRate / outSampleRate;
lsample = in[rd_dat++];
out[wr_dat++] = lsample;
} }
break;
case 2:
while (wr_dat < nlen) {
rd_dat = wr_dat * inSampleRate / outSampleRate;
rd_dat &= ~1;
lsample = in[rd_dat++];
rsample = in[rd_dat++];
newratio = (double)outFormat->sampleRate / (double)inFormat->sampleRate; out[wr_dat++] = lsample;
if(newratio != ratio) { out[wr_dat++] = rsample;
DEBUG("Setting samplerate conversion ratio to %.2lf\n", newratio);
src_set_ratio(state, newratio);
ratio = newratio;
} }
break;
}
data.input_frames = dataChannelLen / 2 / outFormat->channels; return 1;
data.output_frames = pcm_sizeOfOutputBufferForAudioFormatConversion(inFormat, dataChannelLen, outFormat) / 2 / outFormat->channels; }
data.src_ratio = (double)data.output_frames / (double)data.input_frames; #endif /* !HAVE_LIBSAMPLERATE */
if (data_in_size != (data.input_frames * static char *pcm_convertChannels(mpd_sint8 inChannels, char *inBuffer,
outFormat->channels)) { size_t inSize, size_t *outSize)
data_in_size = data.input_frames * outFormat->channels; {
data.data_in = xrealloc(data.data_in, data_in_size); 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);
} }
if (data_out_size != (data.output_frames * outBuffer = buf;
outFormat->channels)) {
data_out_size = data.output_frames * inSamples = inSize >> 1;
outFormat->channels; in = (mpd_sint16 *)inBuffer;
data.data_out = xrealloc(data.data_out, data_out_size); out = (mpd_sint16 *)outBuffer;
for (i = 0; i < inSamples; i++) {
*out++ = *in;
*out++ = *in++;
} }
src_short_to_float_array((short *)dataChannelConv, data.data_in, data.input_frames * outFormat->channels); break;
error = src_process(state, &data); /* convert from 2 -> 1 channels */
if(error) { case 2:
ERROR("Cannot process samples: %s\n", src_strerror(error)); *outSize = inSize >> 1;
exit(EXIT_FAILURE); 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;
} }
src_float_to_short_array(data.data_out, (short *)outBuffer, data.output_frames * outFormat->channels); break;
#else default:
/* resampling code blatantly ripped from ESD */ ERROR("only 1 or 2 channels are supported for conversion!\n");
mpd_uint32 rd_dat = 0; }
mpd_uint32 wr_dat = 0;
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) {
case 1:
while (wr_dat < nlen) {
rd_dat = wr_dat * inFormat->sampleRate /
outFormat->sampleRate;
lsample = in[rd_dat++]; return outBuffer;
}
out[wr_dat++] = lsample; static char *pcm_convertTo16bit(mpd_sint8 inBits, char *inBuffer, size_t inSize,
} size_t *outSize)
break; {
case 2: static char *buf;
while (wr_dat < nlen) { static size_t len;
rd_dat = wr_dat * inFormat->sampleRate / char *outBuffer = NULL;
outFormat->sampleRate; mpd_sint8 *in;
rd_dat &= ~1; mpd_sint16 *out;
int i;
switch (inBits) {
case 8:
*outSize = inSize << 1;
if (*outSize > len) {
len = *outSize;
buf = xrealloc(buf, len);
}
outBuffer = buf;
lsample = in[rd_dat++]; in = (mpd_sint8 *)inBuffer;
rsample = in[rd_dat++]; out = (mpd_sint16 *)outBuffer;
for (i = 0; i < inSize; i++)
*out++ = (*in++) << 8;
out[wr_dat++] = lsample; break;
out[wr_dat++] = rsample; case 16:
} *outSize = inSize;
break; outBuffer = inBuffer;
} break;
#endif case 24:
/* put dithering code from mp3_decode here */
default:
ERROR("only 8 or 16 bits are supported for conversion!\n");
} }
return; 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;
size_t outSize = inSize; size_t outSize = inSize;
......
src/pcm_utils.h
View file @ 407497c4
...@@ -26,15 +26,14 @@ ...@@ -26,15 +26,14 @@
#include <stdlib.h> #include <stdlib.h>
void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format, void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
int volume); int volume);
void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1, void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
size_t bufferSize2, AudioFormat * format, float portion1); size_t bufferSize2, AudioFormat * format, float portion1);
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
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment