[FFmpeg-cvslog] cosmetics: Fix crazy formatting in resample.
Alex Converse
git at videolan.org
Thu May 12 04:55:07 CEST 2011
ffmpeg | branch: master | Alex Converse <aconverse at google.com> | Tue May 10 16:58:01 2011 -0700| [ffc437c026dd0e1b8e5d9114163b4e95999b95fd] | committer: Alex Converse
cosmetics: Fix crazy formatting in resample.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ffc437c026dd0e1b8e5d9114163b4e95999b95fd
---
libavcodec/resample.c | 99 +++++++++++++++++++++++++------------------------
1 files changed, 51 insertions(+), 48 deletions(-)
diff --git a/libavcodec/resample.c b/libavcodec/resample.c
index bdd32f4..0bebe1a 100644
--- a/libavcodec/resample.c
+++ b/libavcodec/resample.c
@@ -39,7 +39,9 @@ static const char *context_to_name(void *ptr)
}
static const AVOption options[] = {{NULL}};
-static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT };
+static const AVClass audioresample_context_class = {
+ "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
+};
struct ReSampleContext {
struct AVResampleContext *resample_context;
@@ -50,9 +52,9 @@ struct ReSampleContext {
int input_channels, output_channels, filter_channels;
AVAudioConvert *convert_ctx[2];
enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
- unsigned sample_size[2]; ///< size of one sample in sample_fmt
- short *buffer[2]; ///< buffers used for conversion to S16
- unsigned buffer_size[2]; ///< sizes of allocated buffers
+ unsigned sample_size[2]; ///< size of one sample in sample_fmt
+ short *buffer[2]; ///< buffers used for conversion to S16
+ unsigned buffer_size[2]; ///< sizes of allocated buffers
};
/* n1: number of samples */
@@ -131,17 +133,17 @@ static void interleave(short *output, short **input, int channels, int samples)
static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
{
int i;
- short l,r;
-
- for(i=0;i<n;i++) {
- l=*input1++;
- r=*input2++;
- *output++ = l; /* left */
- *output++ = (l/2)+(r/2); /* center */
- *output++ = r; /* right */
- *output++ = 0; /* left surround */
- *output++ = 0; /* right surroud */
- *output++ = 0; /* low freq */
+ short l, r;
+
+ for (i = 0; i < n; i++) {
+ l = *input1++;
+ r = *input2++;
+ *output++ = l; /* left */
+ *output++ = (l / 2) + (r / 2); /* center */
+ *output++ = r; /* right */
+ *output++ = 0; /* left surround */
+ *output++ = 0; /* right surroud */
+ *output++ = 0; /* low freq */
}
}
@@ -154,27 +156,25 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
{
ReSampleContext *s;
- if (input_channels > MAX_CHANNELS)
- {
+ if (input_channels > MAX_CHANNELS) {
av_log(NULL, AV_LOG_ERROR,
"Resampling with input channels greater than %d is unsupported.\n",
MAX_CHANNELS);
return NULL;
- }
- if ( output_channels > 2 &&
+ }
+ if (output_channels > 2 &&
!(output_channels == 6 && input_channels == 2) &&
- output_channels != input_channels) {
+ output_channels != input_channels) {
av_log(NULL, AV_LOG_ERROR,
"Resampling output channel count must be 1 or 2 for mono input; 1, 2 or 6 for stereo input; or N for N channel input.\n");
return NULL;
}
s = av_mallocz(sizeof(ReSampleContext));
- if (!s)
- {
+ if (!s) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
return NULL;
- }
+ }
s->ratio = (float)output_rate / (float)input_rate;
@@ -185,10 +185,10 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
if (s->output_channels < s->filter_channels)
s->filter_channels = s->output_channels;
- s->sample_fmt [0] = sample_fmt_in;
- s->sample_fmt [1] = sample_fmt_out;
- s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0])>>3;
- s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1])>>3;
+ s->sample_fmt[0] = sample_fmt_in;
+ s->sample_fmt[1] = sample_fmt_out;
+ s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0]) >> 3;
+ s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1]) >> 3;
if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
@@ -214,8 +214,9 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
}
#define TAPS 16
- s->resample_context= av_resample_init(output_rate, input_rate,
- filter_length, log2_phase_count, linear, cutoff);
+ s->resample_context = av_resample_init(output_rate, input_rate,
+ filter_length, log2_phase_count,
+ linear, cutoff);
*(const AVClass**)s->resample_context = &audioresample_context_class;
@@ -244,7 +245,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
int ostride[1] = { 2 };
const void *ibuf[1] = { input };
void *obuf[1];
- unsigned input_size = nb_samples*s->input_channels*2;
+ unsigned input_size = nb_samples * s->input_channels * 2;
if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
av_free(s->buffer[0]);
@@ -259,15 +260,16 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
obuf[0] = s->buffer[0];
if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
- ibuf, istride, nb_samples*s->input_channels) < 0) {
- av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n");
+ ibuf, istride, nb_samples * s->input_channels) < 0) {
+ av_log(s->resample_context, AV_LOG_ERROR,
+ "Audio sample format conversion failed\n");
return 0;
}
- input = s->buffer[0];
+ input = s->buffer[0];
}
- lenout= 4*nb_samples * s->ratio + 16;
+ lenout = 4 * nb_samples * s->ratio + 16;
if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
output_bak = output;
@@ -286,20 +288,19 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
}
/* XXX: move those malloc to resample init code */
- for(i=0; i<s->filter_channels; i++){
- bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
+ for (i = 0; i < s->filter_channels; i++) {
+ bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short));
memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
buftmp2[i] = bufin[i] + s->temp_len;
bufout[i] = av_malloc(lenout * sizeof(short));
}
- if (s->input_channels == 2 &&
- s->output_channels == 1) {
+ if (s->input_channels == 2 && s->output_channels == 1) {
buftmp3[0] = output;
stereo_to_mono(buftmp2[0], input, nb_samples);
} else if (s->output_channels >= 2 && s->input_channels == 1) {
buftmp3[0] = bufout[0];
- memcpy(buftmp2[0], input, nb_samples*sizeof(short));
+ memcpy(buftmp2[0], input, nb_samples * sizeof(short));
} else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
for (i = 0; i < s->input_channels; i++) {
buftmp3[i] = bufout[i];
@@ -307,21 +308,22 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
deinterleave(buftmp2, input, s->input_channels, nb_samples);
} else {
buftmp3[0] = output;
- memcpy(buftmp2[0], input, nb_samples*sizeof(short));
+ memcpy(buftmp2[0], input, nb_samples * sizeof(short));
}
nb_samples += s->temp_len;
/* resample each channel */
nb_samples1 = 0; /* avoid warning */
- for(i=0;i<s->filter_channels;i++) {
+ for (i = 0; i < s->filter_channels; i++) {
int consumed;
- int is_last= i+1 == s->filter_channels;
+ int is_last = i + 1 == s->filter_channels;
- nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
- s->temp_len= nb_samples - consumed;
- s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
- memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
+ nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
+ &consumed, nb_samples, lenout, is_last);
+ s->temp_len = nb_samples - consumed;
+ s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short));
+ memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
}
if (s->output_channels == 2 && s->input_channels == 1) {
@@ -339,8 +341,9 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
void *obuf[1] = { output_bak };
if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
- ibuf, istride, nb_samples1*s->output_channels) < 0) {
- av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n");
+ ibuf, istride, nb_samples1 * s->output_channels) < 0) {
+ av_log(s->resample_context, AV_LOG_ERROR,
+ "Audio sample format convertion failed\n");
return 0;
}
}
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