On 05/10/2011 08:00 PM, Aℓex Converse wrote:
> On Tue, May 10, 2011 at 4:01 PM, Alex Converse <[email protected]> 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 <[email protected]>
> 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
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