X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fresample.c;h=1b3bb834f3b73c82c48e76f3e251025284572f26;hb=033a86f9bb6fd59ca71d4951b8e2e27cdc1b29d9;hp=86bed847c4b6ec78a7a9d4b43467deff2fac1609;hpb=4a899dd689defec022e8566465c873dd83b55cef;p=ffmpeg

diff --git a/libavcodec/resample.c b/libavcodec/resample.c
index 86bed847c4b..1b3bb834f3b 100644
--- a/libavcodec/resample.c
+++ b/libavcodec/resample.c
@@ -1,126 +1,66 @@
 /*
- * Sample rate convertion for both audio and video
- * Copyright (c) 2000 Fabrice Bellard.
+ * samplerate conversion for both audio and video
+ * Copyright (c) 2000 Fabrice Bellard
  *
- * This library is free software; you can redistribute it and/or
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This library is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 /**
- * @file resample.c
- * Sample rate convertion for both audio and video.
+ * @file
+ * samplerate conversion for both audio and video
  */
 
-#include "avcodec.h"
-#include "os_support.h"
-
-typedef struct {
-    /* fractional resampling */
-    uint32_t incr; /* fractional increment */
-    uint32_t frac;
-    int last_sample;
-    /* integer down sample */
-    int iratio;  /* integer divison ratio */
-    int icount, isum;
-    int inv;
-} ReSampleChannelContext;
+#include <string.h>
 
-struct ReSampleContext {
-    ReSampleChannelContext channel_ctx[2];
-    float ratio;
-    /* channel convert */
-    int input_channels, output_channels, filter_channels;
-};
+#include "avcodec.h"
+#include "audioconvert.h"
+#include "libavutil/opt.h"
+#include "libavutil/mem.h"
+#include "libavutil/samplefmt.h"
 
+#if FF_API_AVCODEC_RESAMPLE
 
-#define FRAC_BITS 16
-#define FRAC (1 << FRAC_BITS)
+#define MAX_CHANNELS 8
 
-static void init_mono_resample(ReSampleChannelContext *s, float ratio)
-{
-    ratio = 1.0 / ratio;
-    s->iratio = (int)floorf(ratio);
-    if (s->iratio == 0)
-        s->iratio = 1;
-    s->incr = (int)((ratio / s->iratio) * FRAC);
-    s->frac = FRAC;
-    s->last_sample = 0;
-    s->icount = s->iratio;
-    s->isum = 0;
-    s->inv = (FRAC / s->iratio);
-}
+struct AVResampleContext;
 
-/* fractional audio resampling */
-static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
+static const char *context_to_name(void *ptr)
 {
-    unsigned int frac, incr;
-    int l0, l1;
-    short *q, *p, *pend;
-
-    l0 = s->last_sample;
-    incr = s->incr;
-    frac = s->frac;
-
-    p = input;
-    pend = input + nb_samples;
-    q = output;
-
-    l1 = *p++;
-    for(;;) {
-        /* interpolate */
-        *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
-        frac = frac + s->incr;
-        while (frac >= FRAC) {
-            frac -= FRAC;
-            if (p >= pend)
-                goto the_end;
-            l0 = l1;
-            l1 = *p++;
-        }
-    }
- the_end:
-    s->last_sample = l1;
-    s->frac = frac;
-    return q - output;
+    return "audioresample";
 }
 
-static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
-{
-    short *q, *p, *pend;
-    int c, sum;
-
-    p = input;
-    pend = input + nb_samples;
-    q = output;
-
-    c = s->icount;
-    sum = s->isum;
+static const AVOption options[] = {{NULL}};
+static const AVClass audioresample_context_class = {
+    "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
+};
 
-    for(;;) {
-        sum += *p++;
-        if (--c == 0) {
-            *q++ = (sum * s->inv) >> FRAC_BITS;
-            c = s->iratio;
-            sum = 0;
-        }
-        if (p >= pend)
-            break;
-    }
-    s->isum = sum;
-    s->icount = c;
-    return q - output;
-}
+struct ReSampleContext {
+    struct AVResampleContext *resample_context;
+    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 */
 static void stereo_to_mono(short *output, short *input, int n1)
@@ -173,147 +113,267 @@ 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 int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
+static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
 {
-    short *buf1;
-    short *buftmp;
-
-    buf1= (short*)av_malloc( nb_samples * sizeof(short) );
-
-    /* first downsample by an integer factor with averaging filter */
-    if (s->iratio > 1) {
-        buftmp = buf1;
-        nb_samples = integer_downsample(s, buftmp, input, nb_samples);
-    } else {
-        buftmp = input;
-    }
-
-    /* then do a fractional resampling with linear interpolation */
-    if (s->incr != FRAC) {
-        nb_samples = fractional_resample(s, output, buftmp, nb_samples);
-    } else {
-        memcpy(output, buftmp, nb_samples * sizeof(short));
+    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 */
     }
-    av_free(buf1);
-    return nb_samples;
 }
 
-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;
-    int i;
-    
-    if (output_channels > 2 || input_channels > 2)
+
+    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 != input_channels &&
+        (input_channels  > 2 ||
+         output_channels > 2 &&
+         !(output_channels == 6 && input_channels == 2))) {
+        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)
+    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;
-    
+
     s->input_channels = input_channels;
     s->output_channels = output_channels;
-    
+
     s->filter_channels = s->input_channels;
     if (s->output_channels < s->filter_channels)
         s->filter_channels = s->output_channels;
 
-    for(i=0;i<s->filter_channels;i++) {
-        init_mono_resample(&s->channel_ctx[i], s->ratio);
+    s->sample_fmt[0]  = sample_fmt_in;
+    s->sample_fmt[1]  = sample_fmt_out;
+    s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
+    s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
+
+    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;
+        }
+    }
+
+    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;
 }
 
 /* resample audio. 'nb_samples' is the number of input samples */
 /* XXX: optimize it ! */
-/* XXX: do it with polyphase filters, since the quality here is
-   HORRIBLE. Return the number of samples available in output */
 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) {
+    if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
         /* nothing to do */
         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
         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 = 4 * nb_samples * s->ratio + 16;
+
+    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
+        int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
+                       s->output_channels;
+        output_bak = output;
+
+        if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
+            av_free(s->buffer[1]);
+            s->buffer_size[1] = out_size;
+            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 */
-    bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
-    bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
-    
-    /* make some zoom to avoid round pb */
-    lenout= (int)(nb_samples * s->ratio) + 16;
-    bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
-    bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
-
-    if (s->input_channels == 2 &&
-        s->output_channels == 1) {
-        buftmp2[0] = bufin[0];
+    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));
+    }
+
+    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) {
-        buftmp2[0] = input;
+    } else if (s->output_channels >= 2 && s->input_channels == 1) {
         buftmp3[0] = bufout[0];
-    } else if (s->output_channels == 2) {
-        buftmp2[0] = bufin[0];
-        buftmp2[1] = bufin[1];
-        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 {
-        buftmp2[0] = input;
         buftmp3[0] = output;
+        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++) {
-        nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
+    for (i = 0; i < s->filter_channels; i++) {
+        int consumed;
+        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));
     }
 
     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 && 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);
     }
 
-    av_free(bufin[0]);
-    av_free(bufin[1]);
+    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 conversion 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);
+    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);
 }
+
+#endif