On Tue, May 10, 2011 at 4:01 PM, Alex Converse <aconve...@google.com> wrote: > --- > libavcodec/resample.c | 83 > ++++++++++++++++++++++++------------------------- > 1 files changed, 41 insertions(+), 42 deletions(-) >
Updated to drop an obsoleted comment.
From 5bcc724fd3a3b69c37d0419626c7b17913a09c9d Mon Sep 17 00:00:00 2001 From: Alex Converse <aconve...@google.com> Date: Tue, 10 May 2011 14:24:05 -0700 Subject: [PATCH 2/3] Allow resampling with no channel count change for up to 8 channels. --- libavcodec/resample.c | 84 ++++++++++++++++++++++++------------------------- 1 files changed, 41 insertions(+), 43 deletions(-) diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 8800098..2fe2e46 100644 --- a/libavcodec/resample.c +++ b/libavcodec/resample.c @@ -29,6 +29,8 @@ #include "libavutil/opt.h" #include "libavutil/samplefmt.h" +#define MAX_CHANNELS 8 + struct AVResampleContext; static const char *context_to_name(void *ptr) @@ -41,7 +43,7 @@ static const AVClass audioresample_context_class = { "ReSampleContext", context_ struct ReSampleContext { struct AVResampleContext *resample_context; - short *temp[2]; + short *temp[MAX_CHANNELS]; int temp_len; float ratio; /* channel convert */ @@ -104,24 +106,25 @@ static void mono_to_stereo(short *output, short *input, int n1) } } -/* XXX: should use more abstract 'N' channels system */ -static void stereo_split(short *output1, short *output2, short *input, int n) +static void deinterleave(short **output, short *input, int channels, int samples) { - int i; + int i, j; - for(i=0;i<n;i++) { - *output1++ = *input++; - *output2++ = *input++; + for (i = 0; i < samples; i++) { + for (j = 0; j < channels; j++) { + *output[j]++ = *input++; + } } } -static void stereo_mux(short *output, short *input1, short *input2, int n) +static void interleave(short *output, short **input, int channels, int samples) { - int i; + int i, j; - for(i=0;i<n;i++) { - *output++ = *input1++; - *output++ = *input2++; + for (i = 0; i < samples; i++) { + for (j = 0; j < channels; j++) { + *output++ = *input[j]++; + } } } @@ -151,14 +154,18 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, { ReSampleContext *s; - if ( input_channels > 2) + if (input_channels > MAX_CHANNELS) { - av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n"); + av_log(NULL, AV_LOG_ERROR, + "Resampling with input channels greater than %d unsupported.\n", + MAX_CHANNELS); return NULL; } - if (output_channels > 2 && !(output_channels == 6 && input_channels == 2)) { + if ( output_channels > 2 && + !(output_channels == 6 && input_channels == 2) && + output_channels != input_channels) { av_log(NULL, AV_LOG_ERROR, - "Resampling output channel count must 1 or 2 for mono input and 1, 2 or 6 for stereo input.\n"); + "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; } @@ -206,14 +213,6 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, } } -/* - * AC-3 output is the only case where filter_channels could be greater than 2. - * input channels can't be greater than 2, so resample the 2 channels and then - * expand to 6 channels after the resampling. - */ - if(s->filter_channels>2) - s->filter_channels = 2; - #define TAPS 16 s->resample_context= av_resample_init(output_rate, input_rate, filter_length, log2_phase_count, linear, cutoff); @@ -228,9 +227,9 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) { int i, nb_samples1; - short *bufin[2]; - short *bufout[2]; - short *buftmp2[2], *buftmp3[2]; + short *bufin[MAX_CHANNELS]; + short *bufout[MAX_CHANNELS]; + short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS]; short *output_bak = NULL; int lenout; @@ -291,12 +290,9 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl 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) ); } - /* make some zoom to avoid round pb */ - bufout[0]= av_malloc( lenout * sizeof(short) ); - bufout[1]= av_malloc( lenout * sizeof(short) ); - if (s->input_channels == 2 && s->output_channels == 1) { buftmp3[0] = output; @@ -304,10 +300,11 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl } else if (s->output_channels >= 2 && s->input_channels == 1) { buftmp3[0] = bufout[0]; memcpy(buftmp2[0], input, nb_samples*sizeof(short)); - } else if (s->output_channels >= 2) { - buftmp3[0] = bufout[0]; - buftmp3[1] = bufout[1]; - stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); + } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) { + for (i = 0; i < s->input_channels; i++) { + buftmp3[i] = bufout[i]; + } + deinterleave(buftmp2, input, s->input_channels, nb_samples); } else { buftmp3[0] = output; memcpy(buftmp2[0], input, nb_samples*sizeof(short)); @@ -329,10 +326,10 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl if (s->output_channels == 2 && s->input_channels == 1) { mono_to_stereo(output, buftmp3[0], nb_samples1); - } else if (s->output_channels == 2) { - stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); - } else if (s->output_channels == 6) { + } else if (s->output_channels == 6 && s->input_channels == 2) { ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); + } else if (s->output_channels == s->input_channels && s->input_channels >= 2) { + interleave(output, buftmp3, s->output_channels, nb_samples1); } if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { @@ -348,19 +345,20 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl } } - for(i=0; i<s->filter_channels; i++) + for (i = 0; i < s->filter_channels; i++) { av_free(bufin[i]); + av_free(bufout[i]); + } - av_free(bufout[0]); - av_free(bufout[1]); return nb_samples1; } void audio_resample_close(ReSampleContext *s) { + int i; av_resample_close(s->resample_context); - av_freep(&s->temp[0]); - av_freep(&s->temp[1]); + for (i = 0; i < s->filter_channels; i++) + av_freep(&s->temp[i]); av_freep(&s->buffer[0]); av_freep(&s->buffer[1]); av_audio_convert_free(s->convert_ctx[0]); -- 1.7.3.1
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