On Tue, May 10, 2011 at 6:15 PM, Justin Ruggles
<justin.rugg...@gmail.com> wrote:
> On 05/10/2011 08:00 PM, Aℓex Converse wrote:
>
[...]
>> 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
[...]
>
>> @@ -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.
Fixed
>
>> 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..."
Fixed in 1, propagated into this patch.
>
>> @@ -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...
>
It was bad copypasta. Fixed.
[...]
>
> rest looks ok.
>
Awesome!
From 29bec80098ef42c252efc4f9bd29b116dd57b395 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 9f0599f..5388ec7 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 is 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 be 1 or 2 for mono input and 1, 2 or 6 for stereo input.\n");
+ "Resampling output channel count must be 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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