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author | Thomas Jansen <mithi@mithi.net> | 2011-02-04 10:39:21 +0100 |
---|---|---|
committer | Max Kellermann <max@duempel.org> | 2011-02-09 22:42:31 +0100 |
commit | 948b8f35e6fc578c85a0cfc8e143e1325ce99043 (patch) | |
tree | 022148bcb71a123f151d05b5936bc66ca35d332a | |
parent | e776c605adeb984aae290cdff38c5f8dd0cd291e (diff) | |
download | mpd-948b8f35e6fc578c85a0cfc8e143e1325ce99043.tar.gz mpd-948b8f35e6fc578c85a0cfc8e143e1325ce99043.tar.xz mpd-948b8f35e6fc578c85a0cfc8e143e1325ce99043.zip |
general: whitespace cleanup
Remove trailing whitespace found by this command:
find -name '*.[ch]' | xargs grep "[[:space:]]$"
-rw-r--r-- | src/AudioCompress/compress.c | 28 | ||||
-rw-r--r-- | src/decoder/audiofile_decoder_plugin.c | 2 | ||||
-rw-r--r-- | src/encoder/flac_encoder.c | 6 | ||||
-rw-r--r-- | src/encoder/wave_encoder.c | 4 | ||||
-rw-r--r-- | src/mixer/winmm_mixer_plugin.c | 10 | ||||
-rw-r--r-- | src/output/httpd_internal.h | 2 | ||||
-rw-r--r-- | src/output/mvp_plugin.c | 2 | ||||
-rw-r--r-- | src/output/oss_plugin.c | 2 | ||||
-rw-r--r-- | src/pcm_byteswap.c | 2 |
9 files changed, 29 insertions, 29 deletions
diff --git a/src/AudioCompress/compress.c b/src/AudioCompress/compress.c index d5c08372c..36cdfd8dd 100644 --- a/src/AudioCompress/compress.c +++ b/src/AudioCompress/compress.c @@ -16,16 +16,16 @@ struct Compressor { //! The compressor's preferences struct CompressorConfig prefs; - + //! History of the peak values int *peaks; - + //! History of the gain values int *gain; - + //! History of clip amounts int *clipped; - + unsigned int pos; unsigned int bufsz; }; @@ -41,9 +41,9 @@ struct Compressor *Compressor_new(unsigned int history) obj->peaks = obj->gain = obj->clipped = NULL; obj->bufsz = 0; obj->pos = 0; - + Compressor_setHistory(obj, history); - + return obj; } @@ -70,7 +70,7 @@ void Compressor_setHistory(struct Compressor *obj, unsigned int history) { if (!history) history = BUCKETS; - + obj->peaks = resizeArray(obj->peaks, history, obj->bufsz); obj->gain = resizeArray(obj->gain, history, obj->bufsz); obj->clipped = resizeArray(obj->clipped, history, obj->bufsz); @@ -82,7 +82,7 @@ struct CompressorConfig *Compressor_getConfig(struct Compressor *obj) return &obj->prefs; } -void Compressor_Process_int16(struct Compressor *obj, int16_t *audio, +void Compressor_Process_int16(struct Compressor *obj, int16_t *audio, unsigned int count) { struct CompressorConfig *prefs = Compressor_getConfig(obj); @@ -97,7 +97,7 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio, int *clipped = obj->clipped + slot; unsigned int ramp = count; int delta; - + ap = audio; for (i = 0; i < count; i++) { @@ -124,15 +124,15 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio, //! Determine target gain newGain = (1 << 10)*prefs->target/peakVal; - + //! Adjust the gain with inertia from the previous gain value - newGain = (curGain*((1 << prefs->smooth) - 1) + newGain) + newGain = (curGain*((1 << prefs->smooth) - 1) + newGain) >> prefs->smooth; - + //! Make sure it's no more than the maximum gain value if (newGain > (prefs->maxgain << 10)) newGain = prefs->maxgain << 10; - + //! Make sure it's no less than 1:1 if (newGain < (1 << 10)) newGain = 1 << 10; @@ -144,7 +144,7 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio, //! Truncate the ramp time ramp = peakPos; } - + //! Record the new gain obj->gain[slot] = newGain; diff --git a/src/decoder/audiofile_decoder_plugin.c b/src/decoder/audiofile_decoder_plugin.c index 3026f3cc7..b099cf706 100644 --- a/src/decoder/audiofile_decoder_plugin.c +++ b/src/decoder/audiofile_decoder_plugin.c @@ -244,7 +244,7 @@ static const char *const audiofile_suffixes[] = { static const char *const audiofile_mime_types[] = { "audio/x-wav", "audio/x-aiff", - NULL + NULL }; const struct decoder_plugin audiofile_decoder_plugin = { diff --git a/src/encoder/flac_encoder.c b/src/encoder/flac_encoder.c index 73328fe81..c34faad00 100644 --- a/src/encoder/flac_encoder.c +++ b/src/encoder/flac_encoder.c @@ -55,7 +55,7 @@ static bool flac_encoder_configure(struct flac_encoder *encoder, const struct config_param *param, G_GNUC_UNUSED GError **error) { - encoder->compression = config_get_block_unsigned(param, + encoder->compression = config_get_block_unsigned(param, "compression", 5); return true; @@ -218,7 +218,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format, if (init_status != FLAC__STREAM_ENCODER_OK) { g_set_error(error, flac_encoder_quark(), 0, - "failed to initialize encoder: %s\n", + "failed to initialize encoder: %s\n", FLAC__StreamEncoderStateString[init_status]); flac_encoder_close(_encoder); return false; @@ -234,7 +234,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format, if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) { g_set_error(error, flac_encoder_quark(), 0, - "failed to initialize encoder: %s\n", + "failed to initialize encoder: %s\n", FLAC__StreamEncoderInitStatusString[init_status]); flac_encoder_close(_encoder); return false; diff --git a/src/encoder/wave_encoder.c b/src/encoder/wave_encoder.c index 7398b45c7..938be5e5e 100644 --- a/src/encoder/wave_encoder.c +++ b/src/encoder/wave_encoder.c @@ -58,7 +58,7 @@ wave_encoder_quark(void) } static void -fill_wave_header(struct wave_header *header, int channels, int bits, +fill_wave_header(struct wave_header *header, int channels, int bits, int freq, int block_size) { int data_size = 0x0FFFFFFF; @@ -142,7 +142,7 @@ wave_encoder_open(struct encoder *_encoder, buffer = pcm_buffer_get(&encoder->buffer, sizeof(struct wave_header) ); /* create PCM wave header in initial buffer */ - fill_wave_header((struct wave_header *) buffer, + fill_wave_header((struct wave_header *) buffer, audio_format->channels, encoder->bits, audio_format->sample_rate, diff --git a/src/mixer/winmm_mixer_plugin.c b/src/mixer/winmm_mixer_plugin.c index e1e8a7a45..5ab3e7525 100644 --- a/src/mixer/winmm_mixer_plugin.c +++ b/src/mixer/winmm_mixer_plugin.c @@ -58,11 +58,11 @@ winmm_mixer_init(void *ao, G_GNUC_UNUSED const struct config_param *param, G_GNUC_UNUSED GError **error_r) { assert(ao != NULL); - + struct winmm_mixer *wm = g_new(struct winmm_mixer, 1); mixer_init(&wm->base, &winmm_mixer_plugin); wm->output = (struct winmm_output *) ao; - + return &wm->base; } @@ -79,13 +79,13 @@ winmm_mixer_get_volume(struct mixer *mixer, GError **error_r) DWORD volume; HWAVEOUT handle = winmm_output_get_handle(wm->output); MMRESULT result = waveOutGetVolume(handle, &volume); - + if (result != MMSYSERR_NOERROR) { g_set_error(error_r, 0, winmm_mixer_quark(), "Failed to get winmm volume"); return -1; } - + return winmm_volume_decode(volume); } @@ -102,7 +102,7 @@ winmm_mixer_set_volume(struct mixer *mixer, unsigned volume, GError **error_r) "Failed to set winmm volume"); return false; } - + return true; } diff --git a/src/output/httpd_internal.h b/src/output/httpd_internal.h index fee72c07f..277e70f11 100644 --- a/src/output/httpd_internal.h +++ b/src/output/httpd_internal.h @@ -111,7 +111,7 @@ struct httpd_output { char buffer[32768]; /** - * The maximum and current number of clients connected + * The maximum and current number of clients connected * at the same time. */ guint clients_max, clients_cnt; diff --git a/src/output/mvp_plugin.c b/src/output/mvp_plugin.c index 20587f5c5..6cc8fa34e 100644 --- a/src/output/mvp_plugin.c +++ b/src/output/mvp_plugin.c @@ -17,7 +17,7 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -/* +/* * Media MVP audio output based on code from MVPMC project: * http://mvpmc.sourceforge.net/ */ diff --git a/src/output/oss_plugin.c b/src/output/oss_plugin.c index bb4164990..f5ae80e8c 100644 --- a/src/output/oss_plugin.c +++ b/src/output/oss_plugin.c @@ -347,7 +347,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format, case SUCCESS: if (!audio_valid_sample_rate(sample_rate)) break; - + audio_format->sample_rate = sample_rate; return true; diff --git a/src/pcm_byteswap.c b/src/pcm_byteswap.c index 967c574cb..6577319d4 100644 --- a/src/pcm_byteswap.c +++ b/src/pcm_byteswap.c @@ -49,7 +49,7 @@ const int16_t *pcm_byteswap_16(struct pcm_buffer *buffer, static inline uint32_t swab32(uint32_t x) { - return (x << 24) | + return (x << 24) | ((x & 0xff00) << 8) | ((x & 0xff0000) >> 8) | (x >> 24); |