On 05/10/2011 08:00 PM, Aℓex Converse wrote: > 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. > > > 0002-Allow-resampling-with-no-channel-count-change-for-up.patch > > > 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
This could be increased to something like 256 or higher if we dynamically allocate the pointers as well. Just something to consider later. 8 is fine for now. > + > 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]++; > + } > } > } ok. > @@ -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); add "is" before "unsupported" to make it a complete sentence. > 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; > } "...channel count must be 1 or 2..." > @@ -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) ); remove the extra spaces inside the (). unless you're just matching style which is then corrected by your style patch... > } > > - /* 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]); rest looks ok. Thanks, Justin _______________________________________________ libav-devel mailing list libav-devel@libav.org https://lists.libav.org/mailman/listinfo/libav-devel