oysteikt/mpd
0
0
Fork 0
Code Issues Pull Requests Releases Activity

explicitly downcast

Browse Source 
Tools like "sparse" check for missing downcasts, since implicit cast
may be dangerous.  Although that does not change the compiler result,
it may make the code more readable (IMHO), because you always see when
there may be data cut off.

git-svn-id: https://svn.musicpd.org/mpd/trunk@7196 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Max Kellermann 2008-03-26 10:37:17 +00:00 committed by Eric Wong
parent 13c17c3d94
commit 66fe580642
5 changed files with 14 additions and 14 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
src/audio.c
View File

@ -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;

2
src/audioOutputs/audioOutput_alsa.c
View File

@ -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);

4
src/audioOutputs/audioOutput_oss.c
View File

@ -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;

6
src/inputPlugins/_flac_common.c
View File

@ -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));

8
src/inputPlugins/audiofile_plugin.c
View File

@ -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",