X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;ds=sidebyside;f=libavcodec%2Fresample.c;h=9e6defefdf6cb3c324fd6d4e7fa68a885b27b835;hb=612122b187d711257eecd517e4049cef3bb0b7f0;hp=7840b1679d5978e6374122921b95debd2ac9f361;hpb=90901860c21468d6e9ae437c2bacb099c7bd3acf;p=ffmpeg

diff --git a/libavcodec/resample.c b/libavcodec/resample.c
index 7840b1679d5..9e6defefdf6 100644
--- a/libavcodec/resample.c
+++ b/libavcodec/resample.c
@@ -1,6 +1,6 @@
 /*
  * samplerate conversion for both audio and video
- * Copyright (c) 2000 Fabrice Bellard.
+ * Copyright (c) 2000 Fabrice Bellard
  *
  * This file is part of FFmpeg.
  *
@@ -20,21 +20,41 @@
  */
 
 /**
- * @file resample.c
+ * @file
  * samplerate conversion for both audio and video
  */
 
 #include "avcodec.h"
+#include "audioconvert.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+
+#define MAX_CHANNELS 8
 
 struct AVResampleContext;
 
+static const char *context_to_name(void *ptr)
+{
+    return "audioresample";
+}
+
+static const AVOption options[] = {{NULL}};
+static const AVClass audioresample_context_class = {
+    "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
+};
+
 struct ReSampleContext {
     struct AVResampleContext *resample_context;
-    short *temp[2];
+    short *temp[MAX_CHANNELS];
     int temp_len;
     float ratio;
     /* channel convert */
     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
 };
 
 /* n1: number of samples */
@@ -88,61 +108,73 @@ 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]++;
+        }
     }
 }
 
 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 */
     }
 }
 
-ReSampleContext *audio_resample_init(int output_channels, int input_channels,
-                                      int output_rate, int input_rate)
+ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
+                                        int output_rate, int input_rate,
+                                        enum AVSampleFormat sample_fmt_out,
+                                        enum AVSampleFormat sample_fmt_in,
+                                        int filter_length, int log2_phase_count,
+                                        int linear, double cutoff)
 {
     ReSampleContext *s;
 
-    if ( input_channels > 2)
-      {
-        av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
+    if (input_channels > MAX_CHANNELS) {
+        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) &&
+        output_channels != input_channels) {
+        av_log(NULL, AV_LOG_ERROR,
+               "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;
+    }
 
     s = av_mallocz(sizeof(ReSampleContext));
-    if (!s)
-      {
-        av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
+    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;
 
@@ -153,16 +185,40 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
     if (s->output_channels < s->filter_channels)
         s->filter_channels = s->output_channels;
 
-/*
- * ac3 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;
+    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,
+                                                         s->sample_fmt[0], 1, NULL, 0))) {
+            av_log(s, AV_LOG_ERROR,
+                   "Cannot convert %s sample format to s16 sample format\n",
+                   av_get_sample_fmt_name(s->sample_fmt[0]));
+            av_free(s);
+            return NULL;
+        }
+    }
+
+    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
+        if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
+                                                         AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
+            av_log(s, AV_LOG_ERROR,
+                   "Cannot convert s16 sample format to %s sample format\n",
+                   av_get_sample_fmt_name(s->sample_fmt[1]));
+            av_audio_convert_free(s->convert_ctx[0]);
+            av_free(s);
+            return NULL;
+        }
+    }
 
 #define TAPS 16
-    s->resample_context= av_resample_init(output_rate, input_rate, TAPS, 10, 0, 0.8);
+    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;
 
     return s;
 }
@@ -172,9 +228,10 @@ ReSampleContext *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;
 
     if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
@@ -183,68 +240,131 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
         return nb_samples;
     }
 
+    if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
+        int istride[1] = { s->sample_size[0] };
+        int ostride[1] = { 2 };
+        const void *ibuf[1] = { input };
+        void       *obuf[1];
+        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]);
+            s->buffer_size[0] = input_size;
+            s->buffer[0] = av_malloc(s->buffer_size[0]);
+            if (!s->buffer[0]) {
+                av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
+                return 0;
+            }
+        }
+
+        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");
+            return 0;
+        }
+
+        input = s->buffer[0];
+    }
+
+    lenout= 2*s->output_channels*nb_samples * s->ratio + 16;
+
+    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
+        output_bak = output;
+
+        if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
+            av_free(s->buffer[1]);
+            s->buffer_size[1] = lenout;
+            s->buffer[1] = av_malloc(s->buffer_size[1]);
+            if (!s->buffer[1]) {
+                av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
+                return 0;
+            }
+        }
+
+        output = s->buffer[1];
+    }
+
     /* 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));
     }
 
-    /* make some zoom to avoid round pb */
-    lenout= (int)(4*nb_samples * s->ratio) + 16;
-    bufout[0]= av_malloc( lenout * sizeof(short) );
-    bufout[1]= 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));
-    } else if (s->output_channels >= 2) {
-        buftmp3[0] = bufout[0];
-        buftmp3[1] = bufout[1];
-        stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
+        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];
+        }
+        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) {
         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);
     }
 
-    for(i=0; i<s->filter_channels; i++)
+    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
+        int istride[1] = { 2 };
+        int ostride[1] = { s->sample_size[1] };
+        const void *ibuf[1] = { output };
+        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");
+            return 0;
+        }
+    }
+
+    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]);
+    av_audio_convert_free(s->convert_ctx[1]);
     av_free(s);
 }