libavcodec.hg: resample.c comparison

comparison resample.c @ 8806:cbeaa8c0fe4f libavcodec

extend resampling API, add S16 internal conversion

author	bcoudurier
date	Wed, 11 Feb 2009 22:57:10 +0000
parents	e9d9d946f213
children	28b706b75b00

comparison

equal deleted inserted replaced

-:eda229beb608
+:cbeaa8c0fe4f
 * @file libavcodec/resample.c
 * samplerate conversion for both audio and video
 */
 #include "avcodec.h"
+#include "audioconvert.h"
+#include "opt.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 };
 struct ReSampleContext {
+const AVClass *av_class;
 struct AVResampleContext *resample_context;
 short *temp[2];
 int temp_len;
 float ratio;
 /* channel convert */
 int input_channels, output_channels, filter_channels;
+AVAudioConvert *convert_ctx[2];
+enum SampleFormat 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)
 {
 *output++ = 0;           /* right surroud */
 *output++ = 0;           /* low freq */
 }
 }
-ReSampleContext *audio_resample_init(int output_channels, int input_channels,
+ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
-int output_rate, int input_rate)
+int output_rate, int input_rate,
+enum SampleFormat sample_fmt_out,
+enum SampleFormat sample_fmt_in,
+int filter_length, int log2_phase_count,
+int linear, double cutoff)
 {
 ReSampleContext *s;
 if ( input_channels > 2)
 {
 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_format(s->sample_fmt[0])>>3;
+s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
+if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
+if (!(s->convert_ctx[0] = av_audio_convert_alloc(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",
+avcodec_get_sample_fmt_name(s->sample_fmt[0]));
+av_free(s);
+return NULL;
+}
+}
+if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
+if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
+SAMPLE_FMT_S16, 1, NULL, 0))) {
+av_log(s, AV_LOG_ERROR,
+"Cannot convert s16 sample format to %s sample format\n",
+avcodec_get_sample_fmt_name(s->sample_fmt[1]));
+av_audio_convert_free(s->convert_ctx[0]);
+av_free(s);
+return NULL;
+}
+}
 /*
 * 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, TAPS, 10, 0, 0.8);
+s->resample_context= av_resample_init(output_rate, input_rate,
+filter_length, log2_phase_count, linear, cutoff);
+s->av_class= &audioresample_context_class;
 return s;
 }
+#if LIBAVCODEC_VERSION_MAJOR < 53
+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,
+SAMPLE_FMT_S16, 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)
 {
 int i, nb_samples1;
 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) {
 /* nothing to do */
 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
 return nb_samples;
+}
+if (s->sample_fmt[0] != 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*s->sample_size[0];
+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, 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, 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] != 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, 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) );
 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= 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) {
 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 } else if (s->output_channels == 6) {
 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 }
+if (s->sample_fmt[1] != 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, 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[0]);
 av_free(bufout[1]);
 void audio_resample_close(ReSampleContext *s)
 {
 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);
 }

Mercurial > libavcodec.hg

comparison resample.c @ 8806:cbeaa8c0fe4f libavcodec