From 407497c40a34ae299cca56de1bd7c92e5724c54a Mon Sep 17 00:00:00 2001
From: "J. Alexander Treuman" <jat@spatialrift.net>
Date: Tue, 22 May 2007 23:11:36 +0000
Subject: 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
---
src/audioOutput.c | 8 +-
src/outputBuffer.c | 9 +-
src/pcm_utils.c | 375 +++++++++++++++++++++++++++++------------------------
src/pcm_utils.h | 11 +-
4 files changed, 217 insertions(+), 186 deletions(-)
diff --git a/src/audioOutput.c b/src/audioOutput.c
index dd1635331..7a99dfd68 100644
--- a/src/audioOutput.c
+++ b/src/audioOutput.c
@@ -194,11 +194,9 @@ int openAudioOutput(AudioOutput * audioOutput, AudioFormat * audioFormat)
static void convertAudioFormat(AudioOutput * audioOutput, char **chunkArgPtr,
int *sizeArgPtr)
{
- int size =
- pcm_sizeOfOutputBufferForAudioFormatConversion(
- &(audioOutput->inAudioFormat),
- *sizeArgPtr,
- &(audioOutput->outAudioFormat));
+ int size = pcm_sizeOfConvBuffer(&(audioOutput->inAudioFormat),
+ *sizeArgPtr,
+ &(audioOutput->outAudioFormat));
if (size > audioOutput->convBufferLen) {
audioOutput->convBuffer =
diff --git a/src/outputBuffer.c b/src/outputBuffer.c
index 8b3a00b0e..36e15a78f 100644
--- a/src/outputBuffer.c
+++ b/src/outputBuffer.c
@@ -82,13 +82,8 @@ int sendDataToOutputBuffer(OutputBuffer * cb, InputStream * inStream,
data = dataIn;
datalen = dataInLen;
} else {
- datalen =
- pcm_sizeOfOutputBufferForAudioFormatConversion(&
- (dc->
- audioFormat),
- dataInLen,
- &(cb->
- audioFormat));
+ datalen = pcm_sizeOfConvBuffer(&(dc->audioFormat), dataInLen,
+ &(cb->audioFormat));
if (datalen > convBufferLen) {
convBuffer = xrealloc(convBuffer, datalen);
convBufferLen = datalen;
diff --git a/src/pcm_utils.c b/src/pcm_utils.c
index c05559be5..ca5d6ca5c 100644
--- a/src/pcm_utils.c
+++ b/src/pcm_utils.c
@@ -153,7 +153,7 @@ void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
}
#ifdef HAVE_LIBSAMPLERATE
-static int pcm_getSamplerateConverter(void)
+static int pcm_getSampleRateConverter(void)
{
const char *conf, *test;
int convalgo = SRC_SINC_FASTEST;
@@ -185,198 +185,237 @@ static int pcm_getSamplerateConverter(void)
}
#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)
+#ifdef HAVE_LIBSAMPLERATE
+static int pcm_convertSampleRate(mpd_sint8 channels, mpd_uint32 inSampleRate,
+ char *inBuffer, size_t inSize,
+ mpd_uint32 outSampleRate, char *outBuffer,
+ size_t outSize)
{
- static char *bitConvBuffer;
- static int bitConvBufferLength;
- static char *channelConvBuffer;
- static int channelConvBufferLength;
- char *dataChannelConv;
- int dataChannelLen;
- char *dataBitConv;
- int dataBitLen;
+ static SRC_STATE *state;
+ static SRC_DATA data;
+ static size_t dataInSize;
+ static size_t dataOutSize;
+ size_t curDataInSize;
+ size_t curDataOutSize;
+ double ratio;
+ 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);
- assert(outFormat->channels == 2 || outFormat->channels == 1);
+ ratio = (double)outSampleRate / (double)inSampleRate;
+ 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 */
- 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);
+ 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);
}
- /* 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);
- }
+ 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);
}
- if (inFormat->sampleRate == outFormat->sampleRate) {
- memcpy(outBuffer, dataChannelConv, dataChannelLen);
- } else {
-#ifdef HAVE_LIBSAMPLERATE
- static SRC_STATE *state = NULL;
- static SRC_DATA data;
- static size_t data_in_size, data_out_size;
- int error;
- static double ratio = 0;
- double newratio;
-
- if(!state) {
- state = src_new(pcm_getSamplerateConverter(), outFormat->channels, &error);
- if(!state) {
- ERROR("Cannot create new samplerate state: %s\n", src_strerror(error));
- exit(EXIT_FAILURE);
- } else {
- DEBUG("Samplerate converter initialized\n");
- }
+ src_short_to_float_array((short *)inBuffer, data.data_in,
+ data.input_frames * channels);
+
+ error = src_process(state, &data);
+ if (error) {
+ ERROR("Cannot process samples: %s\n", src_strerror(error));
+ return 0;
+ }
+
+ src_float_to_short_array(data.data_out, (short *)outBuffer,
+ data.output_frames_gen * channels);
+
+ 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 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;
- if(newratio != ratio) {
- DEBUG("Setting samplerate conversion ratio to %.2lf\n", newratio);
- src_set_ratio(state, newratio);
- ratio = newratio;
+ out[wr_dat++] = lsample;
+ out[wr_dat++] = rsample;
}
+ break;
+ }
- data.input_frames = dataChannelLen / 2 / outFormat->channels;
- data.output_frames = pcm_sizeOfOutputBufferForAudioFormatConversion(inFormat, dataChannelLen, outFormat) / 2 / outFormat->channels;
- data.src_ratio = (double)data.output_frames / (double)data.input_frames;
+ return 1;
+}
+#endif /* !HAVE_LIBSAMPLERATE */
- if (data_in_size != (data.input_frames *
- outFormat->channels)) {
- data_in_size = data.input_frames * outFormat->channels;
- data.data_in = xrealloc(data.data_in, data_in_size);
+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);
}
- if (data_out_size != (data.output_frames *
- outFormat->channels)) {
- data_out_size = data.output_frames *
- outFormat->channels;
- data.data_out = xrealloc(data.data_out, data_out_size);
+ outBuffer = buf;
+
+ inSamples = inSize >> 1;
+ in = (mpd_sint16 *)inBuffer;
+ 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);
- error = src_process(state, &data);
- if(error) {
- ERROR("Cannot process samples: %s\n", src_strerror(error));
- exit(EXIT_FAILURE);
+ 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;
}
- src_float_to_short_array(data.data_out, (short *)outBuffer, data.output_frames * outFormat->channels);
-#else
- /* resampling code blatantly ripped from ESD */
- 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;
+ break;
+ default:
+ ERROR("only 1 or 2 channels are supported for conversion!\n");
+ }
- lsample = in[rd_dat++];
+ return outBuffer;
+}
- out[wr_dat++] = lsample;
- }
- break;
- case 2:
- while (wr_dat < nlen) {
- rd_dat = wr_dat * inFormat->sampleRate /
- outFormat->sampleRate;
- rd_dat &= ~1;
+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;
- lsample = in[rd_dat++];
- rsample = in[rd_dat++];
+ in = (mpd_sint8 *)inBuffer;
+ out = (mpd_sint16 *)outBuffer;
+ for (i = 0; i < inSize; i++)
+ *out++ = (*in++) << 8;
- out[wr_dat++] = lsample;
- out[wr_dat++] = rsample;
- }
- break;
- }
-#endif
+ 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;
+ 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 inSize,
- AudioFormat * outFormat)
+size_t pcm_sizeOfConvBuffer(AudioFormat * inFormat, size_t inSize,
+ AudioFormat * outFormat)
{
const int shift = sizeof(mpd_sint16) * outFormat->channels;
size_t outSize = inSize;
diff --git a/src/pcm_utils.h b/src/pcm_utils.h
index 85ec9e3e3..5142db17b 100644
--- a/src/pcm_utils.h
+++ b/src/pcm_utils.h
@@ -26,15 +26,14 @@
#include <stdlib.h>
void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
- int volume);
+ int volume);
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
- inSize, AudioFormat * outFormat, char *outBuffer);
+ inSize, AudioFormat * outFormat, char *outBuffer);
-size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
- size_t inSize,
- AudioFormat * outFormat);
+size_t pcm_sizeOfConvBuffer(AudioFormat * inFormat, size_t inSize,
+ AudioFormat * outFormat);
#endif
--
cgit v1.2.3