/*
- * 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 "audioconvert.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
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];
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 */
}
}
-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 ( input_channels > 2)
{
- av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
- return NULL;
+ av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
+ return NULL;
}
s = av_mallocz(sizeof(ReSampleContext));
if (!s)
{
- av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
- return NULL;
+ 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;
+ 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;
+ }
+ }
+
/*
- * ac3 output is the only case where filter_channels could be greater than 2.
+ * 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;
- s->resample_context= av_resample_init(output_rate, input_rate);
-
+#define TAPS 16
+ 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;
}
+#if FF_API_AUDIO_OLD
+ReSampleContext *audio_resample_init(int output_channels, int input_channels,
+ int output_rate, int input_rate)
+{
+ return av_audio_resample_init(output_channels, input_channels,
+ output_rate, input_rate,
+ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16,
+ TAPS, 10, 0, 0.8);
+}
+#endif
+
/* resample audio. 'nb_samples' is the number of input samples */
/* XXX: optimize it ! */
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
short *bufin[2];
short *bufout[2];
short *buftmp2[2], *buftmp3[2];
+ short *output_bak = NULL;
int lenout;
if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
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) {
+ 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]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
+ 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;
}
-
+
/* 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) );
+ bufout[0]= av_malloc( lenout * sizeof(short) );
+ bufout[1]= av_malloc( lenout * sizeof(short) );
if (s->input_channels == 2 &&
s->output_channels == 1) {
ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
}
+ 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_resample_close(s->resample_context);
av_freep(&s->temp[0]);
av_freep(&s->temp[1]);
+ 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);
}
projects / ffmpeg / blobdiff