diff options
Diffstat (limited to 'src')
| -rw-r--r-- | src/audio.c | 8 | ||||
| -rw-r--r-- | src/audioOutputs/audioOutput_alsa.c | 2 | ||||
| -rw-r--r-- | src/audioOutputs/audioOutput_oss.c | 4 | ||||
| -rw-r--r-- | src/inputPlugins/_flac_common.c | 6 | ||||
| -rw-r--r-- | src/inputPlugins/audiofile_plugin.c | 8 |
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", |