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cosmetics: Fix crazy formatting in resample.

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This commit is contained in:
Alex Converse 2011-05-10 16:58:01 -07:00 committed by Alex Converse
parent 3e00ababc4
commit ffc437c026
1 changed files with 50 additions and 47 deletions
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73
libavcodec/resample.c
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@ -39,7 +39,9 @@ static const char *context_to_name(void *ptr)
} }
static const AVOption options[] = {{NULL}}; 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 ReSampleContext {
struct AVResampleContext *resample_context; struct AVResampleContext *resample_context;
@ -131,13 +133,13 @@ static void interleave(short *output, short **input, int channels, int samples)
static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
{ {
int i; int i;
short l,r; short l, r;
for(i=0;i<n;i++) { for (i = 0; i < n; i++) {
l=*input1++; l = *input1++;
r=*input2++; r = *input2++;
*output++ = l; /* left */ *output++ = l; /* left */
*output++ = (l/2)+(r/2); /* center */ *output++ = (l / 2) + (r / 2); /* center */
*output++ = r; /* right */ *output++ = r; /* right */
*output++ = 0; /* left surround */ *output++ = 0; /* left surround */
*output++ = 0; /* right surroud */ *output++ = 0; /* right surroud */
@ -154,14 +156,13 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
{ {
ReSampleContext *s; ReSampleContext *s;
if (input_channels > MAX_CHANNELS) if (input_channels > MAX_CHANNELS) {
{
av_log(NULL, AV_LOG_ERROR, av_log(NULL, AV_LOG_ERROR,
"Resampling with input channels greater than %d is unsupported.\n", "Resampling with input channels greater than %d is unsupported.\n",
MAX_CHANNELS); MAX_CHANNELS);
return NULL; return NULL;
} }
if ( output_channels > 2 && if (output_channels > 2 &&
!(output_channels == 6 && input_channels == 2) && !(output_channels == 6 && input_channels == 2) &&
output_channels != input_channels) { output_channels != input_channels) {
av_log(NULL, AV_LOG_ERROR, av_log(NULL, AV_LOG_ERROR,
@ -170,8 +171,7 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
} }
s = av_mallocz(sizeof(ReSampleContext)); s = av_mallocz(sizeof(ReSampleContext));
if (!s) if (!s) {
{
av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n"); av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
return NULL; return NULL;
} }
@ -185,10 +185,10 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
if (s->output_channels < s->filter_channels) if (s->output_channels < s->filter_channels)
s->filter_channels = s->output_channels; s->filter_channels = s->output_channels;
s->sample_fmt [0] = sample_fmt_in; s->sample_fmt[0] = sample_fmt_in;
s->sample_fmt [1] = sample_fmt_out; 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[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_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1]) >> 3;
if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) { if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1, 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 #define TAPS 16
s->resample_context= av_resample_init(output_rate, input_rate, s->resample_context = av_resample_init(output_rate, input_rate,
filter_length, log2_phase_count, linear, cutoff); filter_length, log2_phase_count,
linear, cutoff);
*(const AVClass**)s->resample_context = &audioresample_context_class; *(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 }; int ostride[1] = { 2 };
const void *ibuf[1] = { input }; const void *ibuf[1] = { input };
void *obuf[1]; 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) { if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
av_free(s->buffer[0]); 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]; obuf[0] = s->buffer[0];
if (av_audio_convert(s->convert_ctx[0], obuf, ostride, if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
ibuf, istride, nb_samples*s->input_channels) < 0) { ibuf, istride, nb_samples * s->input_channels) < 0) {
av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n"); av_log(s->resample_context, AV_LOG_ERROR,
"Audio sample format conversion failed\n");
return 0; 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) { if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
output_bak = output; 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 */ /* XXX: move those malloc to resample init code */
for(i=0; i<s->filter_channels; i++){ for (i = 0; i < s->filter_channels; i++) {
bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short));
memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
buftmp2[i] = bufin[i] + s->temp_len; buftmp2[i] = bufin[i] + s->temp_len;
bufout[i] = av_malloc(lenout * sizeof(short)); bufout[i] = av_malloc(lenout * sizeof(short));
} }
if (s->input_channels == 2 && if (s->input_channels == 2 && s->output_channels == 1) {
s->output_channels == 1) {
buftmp3[0] = output; buftmp3[0] = output;
stereo_to_mono(buftmp2[0], input, nb_samples); stereo_to_mono(buftmp2[0], input, nb_samples);
} else if (s->output_channels >= 2 && s->input_channels == 1) { } else if (s->output_channels >= 2 && s->input_channels == 1) {
buftmp3[0] = bufout[0]; 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) { } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
for (i = 0; i < s->input_channels; i++) { for (i = 0; i < s->input_channels; i++) {
buftmp3[i] = bufout[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); deinterleave(buftmp2, input, s->input_channels, nb_samples);
} else { } else {
buftmp3[0] = output; buftmp3[0] = output;
memcpy(buftmp2[0], input, nb_samples*sizeof(short)); memcpy(buftmp2[0], input, nb_samples * sizeof(short));
} }
nb_samples += s->temp_len; nb_samples += s->temp_len;
/* resample each channel */ /* resample each channel */
nb_samples1 = 0; /* avoid warning */ nb_samples1 = 0; /* avoid warning */
for(i=0;i<s->filter_channels;i++) { for (i = 0; i < s->filter_channels; i++) {
int consumed; 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); nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
s->temp_len= nb_samples - consumed; &consumed, nb_samples, lenout, is_last);
s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short)); s->temp_len = nb_samples - consumed;
memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short)); 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) { 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 }; void *obuf[1] = { output_bak };
if (av_audio_convert(s->convert_ctx[1], obuf, ostride, if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
ibuf, istride, nb_samples1*s->output_channels) < 0) { ibuf, istride, nb_samples1 * s->output_channels) < 0) {
av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n"); av_log(s->resample_context, AV_LOG_ERROR,
"Audio sample format convertion failed\n");
return 0; return 0;
} }
} }