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-rw-r--r--src/audio.c8
-rw-r--r--src/audioOutputs/audioOutput_alsa.c2
-rw-r--r--src/audioOutputs/audioOutput_oss.c4
-rw-r--r--src/inputPlugins/_flac_common.c6
-rw-r--r--src/inputPlugins/audiofile_plugin.c8
5 files changed, 14 insertions, 14 deletions
diff --git a/src/audio.c b/src/audio.c
index 145e6be3c..b0c731639 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -197,7 +197,7 @@ int parseAudioConfig(AudioFormat * audioFormat, char *conf)
return -1;
}
- audioFormat->bits = strtol(test + 1, &test, 10);
+ audioFormat->bits = (mpd_sint8)strtol(test + 1, &test, 10);
if (*test != ':') {
ERROR("error parsing audio output format: %s\n", conf);
@@ -213,7 +213,7 @@ int parseAudioConfig(AudioFormat * audioFormat, char *conf)
return -1;
}
- audioFormat->channels = strtol(test + 1, &test, 10);
+ audioFormat->channels = (mpd_sint8)strtol(test + 1, &test, 10);
if (*test != '\0') {
ERROR("error parsing audio output format: %s\n", conf);
@@ -428,7 +428,7 @@ void sendMetadataToAudioDevice(MpdTag * tag)
int enableAudioDevice(int fd, int device)
{
- if (device < 0 || device >= audioOutputArraySize) {
+ if (device >= audioOutputArraySize) {
commandError(fd, ACK_ERROR_ARG, "audio output device id %i "
"doesn't exist\n", device);
return -1;
@@ -442,7 +442,7 @@ int enableAudioDevice(int fd, int device)
int disableAudioDevice(int fd, int device)
{
- if (device < 0 || device >= audioOutputArraySize) {
+ if (device >= audioOutputArraySize) {
commandError(fd, ACK_ERROR_ARG, "audio output device id %i "
"doesn't exist\n", device);
return -1;
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
index 1f90c6ee0..3da3e470d 100644
--- a/src/audioOutputs/audioOutput_alsa.c
+++ b/src/audioOutputs/audioOutput_alsa.c
@@ -215,7 +215,7 @@ configure_hw:
snd_strerror(-err));
goto fail;
}
- audioFormat->channels = channels;
+ audioFormat->channels = (mpd_sint8)channels;
err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
&sampleRate, NULL);
diff --git a/src/audioOutputs/audioOutput_oss.c b/src/audioOutputs/audioOutput_oss.c
index b2d913d6b..1051869f5 100644
--- a/src/audioOutputs/audioOutput_oss.c
+++ b/src/audioOutputs/audioOutput_oss.c
@@ -485,9 +485,9 @@ static int oss_openDevice(AudioOutput * audioOutput)
OssData *od = audioOutput->data;
AudioFormat *audioFormat = &audioOutput->outAudioFormat;
- od->channels = audioFormat->channels;
+ od->channels = (mpd_sint8)audioFormat->channels;
od->sampleRate = audioFormat->sampleRate;
- od->bits = audioFormat->bits;
+ od->bits = (mpd_sint8)audioFormat->bits;
if ((ret = oss_open(audioOutput)) < 0)
return ret;
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index 0a2adf6b7..3b351d3a7 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -66,7 +66,7 @@ static int flacFindVorbisCommentFloat(const FLAC__StreamMetadata * block,
comments[offset].entry[pos]);
tmp = p[len];
p[len] = '\0';
- *fl = atof((char *)p);
+ *fl = (float)atof((char *)p);
p[len] = tmp;
return 1;
@@ -170,9 +170,9 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
switch (block->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
- dc->audioFormat.bits = si->bits_per_sample;
+ dc->audioFormat.bits = (mpd_sint8)si->bits_per_sample;
dc->audioFormat.sampleRate = si->sample_rate;
- dc->audioFormat.channels = si->channels;
+ dc->audioFormat.channels = (mpd_sint8)si->channels;
dc->totalTime = ((float)si->total_samples) / (si->sample_rate);
getOutputAudioFormat(&(dc->audioFormat),
&(data->cb->audioFormat));
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 1213d31e5..3ca9a14c3 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -67,9 +67,9 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, 16);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
- dc->audioFormat.bits = bits;
- dc->audioFormat.sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK);
- dc->audioFormat.channels = afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
+ dc->audioFormat.bits = (mpd_uint8)bits;
+ dc->audioFormat.sampleRate = (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
+ dc->audioFormat.channels = (mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
@@ -77,7 +77,7 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
dc->totalTime =
((float)frame_count / (float)dc->audioFormat.sampleRate);
- bitRate = st.st_size * 8.0 / dc->totalTime / 1000.0 + 0.5;
+ bitRate = (mpd_uint16)(st.st_size * 8.0 / dc->totalTime / 1000.0 + 0.5);
if (dc->audioFormat.bits != 8 && dc->audioFormat.bits != 16) {
ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",