I used indent. This is not perfect but it is good enough. I just need something that doesn't make my eyes bleed. If anyone feels more strongly about these formatting issues, I'm glad to turn this over to you but you must act quickly.
Regards, Alex
From 74e8ec80bb50f3ecaa1918155c3cb83b1783ea51 Mon Sep 17 00:00:00 2001 From: Alex Converse <[email protected]> Date: Tue, 10 May 2011 16:58:01 -0700 Subject: [PATCH 3/3] cosmetics: Fix crazy formatting in resample. --- libavcodec/resample.c | 101 +++++++++++++++++++++++++------------------------ 1 files changed, 52 insertions(+), 49 deletions(-) diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 2fe2e46..1cc7799 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 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 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) ); + 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; } } -- 1.7.3.1
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