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authorJ. Alexander Treuman <jat@spatialrift.net>2007-05-24 21:15:37 +0000
committerJ. Alexander Treuman <jat@spatialrift.net>2007-05-24 21:15:37 +0000
commit2814b7cfc650a73146b8e18fd0a55d54c3ec613d (patch)
treeb632095b1b73c8c310e6cdd8c92ca52807061740
parent7ba357a04e75c51f898e9418cab24d10c74ab37c (diff)
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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
-rw-r--r--src/audioOutput.c10
-rw-r--r--src/audioOutput.h2
-rw-r--r--src/outputBuffer.c3
-rw-r--r--src/outputBuffer.h2
-rw-r--r--src/pcm_utils.c81
-rw-r--r--src/pcm_utils.h22
-rw-r--r--src/playerData.c1
7 files changed, 85 insertions, 36 deletions
diff --git a/src/audioOutput.c b/src/audioOutput.c
index 7a99dfd68..ccc701451 100644
--- a/src/audioOutput.c
+++ b/src/audioOutput.c
@@ -135,6 +135,7 @@ int initAudioOutput(AudioOutput *ao, ConfigParam * param)
memset(&ao->inAudioFormat, 0, sizeof(AudioFormat));
memset(&ao->outAudioFormat, 0, sizeof(AudioFormat));
memset(&ao->reqAudioFormat, 0, sizeof(AudioFormat));
+ memset(&ao->convState, 0, sizeof(ConvState));
if (format) {
ao->convertAudioFormat = 1;
@@ -205,10 +206,11 @@ static void convertAudioFormat(AudioOutput * audioOutput, char **chunkArgPtr,
}
pcm_convertAudioFormat(&(audioOutput->inAudioFormat),
- *chunkArgPtr,
- *sizeArgPtr,
- &(audioOutput->outAudioFormat),
- audioOutput->convBuffer);
+ *chunkArgPtr,
+ *sizeArgPtr,
+ &(audioOutput->outAudioFormat),
+ audioOutput->convBuffer,
+ &audioOutput->convState);
*sizeArgPtr = size;
*chunkArgPtr = audioOutput->convBuffer;
diff --git a/src/audioOutput.h b/src/audioOutput.h
index cba445a3c..bcbe7997d 100644
--- a/src/audioOutput.h
+++ b/src/audioOutput.h
@@ -21,6 +21,7 @@
#include "../config.h"
+#include "pcm_utils.h"
#include "mpd_types.h"
#include "audio.h"
#include "tag.h"
@@ -66,6 +67,7 @@ struct _AudioOutput {
AudioFormat inAudioFormat;
AudioFormat outAudioFormat;
AudioFormat reqAudioFormat;
+ ConvState convState;
char *convBuffer;
int convBufferLen;
int sameInAndOutFormats;
diff --git a/src/outputBuffer.c b/src/outputBuffer.c
index 36e15a78f..d9cbcdc60 100644
--- a/src/outputBuffer.c
+++ b/src/outputBuffer.c
@@ -90,7 +90,8 @@ int sendDataToOutputBuffer(OutputBuffer * cb, InputStream * inStream,
}
data = convBuffer;
pcm_convertAudioFormat(&(dc->audioFormat), dataIn, dataInLen,
- &(cb->audioFormat), data);
+ &(cb->audioFormat), data,
+ &(cb->convState));
}
if (replayGainInfo && (replayGainState != REPLAYGAIN_OFF))
diff --git a/src/outputBuffer.h b/src/outputBuffer.h
index fecff91e0..f690941d4 100644
--- a/src/outputBuffer.h
+++ b/src/outputBuffer.h
@@ -19,6 +19,7 @@
#ifndef OUTPUT_BUFFER_H
#define OUTPUT_BUFFER_H
+#include "pcm_utils.h"
#include "mpd_types.h"
#include "decode.h"
#include "audio.h"
@@ -39,6 +40,7 @@ typedef struct _OutputBuffer {
mpd_sint16 volatile begin;
mpd_sint16 volatile end;
AudioFormat audioFormat;
+ ConvState convState;
MetadataChunk metadataChunks[BUFFERED_METACHUNKS];
mpd_sint8 metaChunkSet[BUFFERED_METACHUNKS];
mpd_sint8 *volatile metaChunk;
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);
}
}
diff --git a/src/pcm_utils.h b/src/pcm_utils.h
index 5142db17b..800ac27f4 100644
--- a/src/pcm_utils.h
+++ b/src/pcm_utils.h
@@ -25,14 +25,32 @@
#include <stdlib.h>
+#ifdef HAVE_LIBSAMPLERATE
+#include <samplerate.h>
+#endif
+
+typedef struct _ConvState {
+#ifdef HAVE_LIBSAMPLERATE
+ SRC_STATE *state;
+ SRC_DATA data;
+ size_t dataInSize;
+ size_t dataOutSize;
+ mpd_sint8 lastChannels;
+ mpd_sint32 lastInSampleRate;
+ mpd_sint32 lastOutSampleRate;
+ int error;
+#endif
+} ConvState;
+
void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
int volume);
void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
size_t bufferSize2, AudioFormat * format, float portion1);
-void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
- inSize, AudioFormat * outFormat, char *outBuffer);
+void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer,
+ size_t inSize, AudioFormat * outFormat,
+ char *outBuffer, ConvState *convState);
size_t pcm_sizeOfConvBuffer(AudioFormat * inFormat, size_t inSize,
AudioFormat * outFormat);
diff --git a/src/playerData.c b/src/playerData.c
index 31f27540c..0d5e6e9e4 100644
--- a/src/playerData.c
+++ b/src/playerData.c
@@ -116,6 +116,7 @@ void initPlayerData(void)
allocationSize - device_array_size;
buffer = &(playerData_pd->buffer);
+ memset(&buffer->convState, 0, sizeof(ConvState));
buffer->chunks = ((char *)playerData_pd) + sizeof(PlayerData);
buffer->chunkSize = (mpd_uint16 *) (((char *)buffer->chunks) +
buffered_chunks * CHUNK_SIZE);