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00027 #include "avcodec.h"
00028 #include "audioconvert.h"
00029 #include "libavutil/opt.h"
00030 #include "libavutil/samplefmt.h"
00031
00032 #define MAX_CHANNELS 8
00033
00034 struct AVResampleContext;
00035
00036 static const char *context_to_name(void *ptr)
00037 {
00038 return "audioresample";
00039 }
00040
00041 static const AVOption options[] = {{NULL}};
00042 static const AVClass audioresample_context_class = {
00043 "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
00044 };
00045
00046 struct ReSampleContext {
00047 struct AVResampleContext *resample_context;
00048 short *temp[MAX_CHANNELS];
00049 int temp_len;
00050 float ratio;
00051
00052 int input_channels, output_channels, filter_channels;
00053 AVAudioConvert *convert_ctx[2];
00054 enum AVSampleFormat sample_fmt[2];
00055 unsigned sample_size[2];
00056 short *buffer[2];
00057 unsigned buffer_size[2];
00058 };
00059
00060
00061 static void stereo_to_mono(short *output, short *input, int n1)
00062 {
00063 short *p, *q;
00064 int n = n1;
00065
00066 p = input;
00067 q = output;
00068 while (n >= 4) {
00069 q[0] = (p[0] + p[1]) >> 1;
00070 q[1] = (p[2] + p[3]) >> 1;
00071 q[2] = (p[4] + p[5]) >> 1;
00072 q[3] = (p[6] + p[7]) >> 1;
00073 q += 4;
00074 p += 8;
00075 n -= 4;
00076 }
00077 while (n > 0) {
00078 q[0] = (p[0] + p[1]) >> 1;
00079 q++;
00080 p += 2;
00081 n--;
00082 }
00083 }
00084
00085
00086 static void mono_to_stereo(short *output, short *input, int n1)
00087 {
00088 short *p, *q;
00089 int n = n1;
00090 int v;
00091
00092 p = input;
00093 q = output;
00094 while (n >= 4) {
00095 v = p[0]; q[0] = v; q[1] = v;
00096 v = p[1]; q[2] = v; q[3] = v;
00097 v = p[2]; q[4] = v; q[5] = v;
00098 v = p[3]; q[6] = v; q[7] = v;
00099 q += 8;
00100 p += 4;
00101 n -= 4;
00102 }
00103 while (n > 0) {
00104 v = p[0]; q[0] = v; q[1] = v;
00105 q += 2;
00106 p += 1;
00107 n--;
00108 }
00109 }
00110
00111 static void deinterleave(short **output, short *input, int channels, int samples)
00112 {
00113 int i, j;
00114
00115 for (i = 0; i < samples; i++) {
00116 for (j = 0; j < channels; j++) {
00117 *output[j]++ = *input++;
00118 }
00119 }
00120 }
00121
00122 static void interleave(short *output, short **input, int channels, int samples)
00123 {
00124 int i, j;
00125
00126 for (i = 0; i < samples; i++) {
00127 for (j = 0; j < channels; j++) {
00128 *output++ = *input[j]++;
00129 }
00130 }
00131 }
00132
00133 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
00134 {
00135 int i;
00136 short l, r;
00137
00138 for (i = 0; i < n; i++) {
00139 l = *input1++;
00140 r = *input2++;
00141 *output++ = l;
00142 *output++ = (l / 2) + (r / 2);
00143 *output++ = r;
00144 *output++ = 0;
00145 *output++ = 0;
00146 *output++ = 0;
00147 }
00148 }
00149
00150 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
00151 int output_rate, int input_rate,
00152 enum AVSampleFormat sample_fmt_out,
00153 enum AVSampleFormat sample_fmt_in,
00154 int filter_length, int log2_phase_count,
00155 int linear, double cutoff)
00156 {
00157 ReSampleContext *s;
00158
00159 if (input_channels > MAX_CHANNELS) {
00160 av_log(NULL, AV_LOG_ERROR,
00161 "Resampling with input channels greater than %d is unsupported.\n",
00162 MAX_CHANNELS);
00163 return NULL;
00164 }
00165 if (output_channels != input_channels &&
00166 (input_channels > 2 ||
00167 output_channels > 2 &&
00168 !(output_channels == 6 && input_channels == 2))) {
00169 av_log(NULL, AV_LOG_ERROR,
00170 "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");
00171 return NULL;
00172 }
00173
00174 s = av_mallocz(sizeof(ReSampleContext));
00175 if (!s) {
00176 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
00177 return NULL;
00178 }
00179
00180 s->ratio = (float)output_rate / (float)input_rate;
00181
00182 s->input_channels = input_channels;
00183 s->output_channels = output_channels;
00184
00185 s->filter_channels = s->input_channels;
00186 if (s->output_channels < s->filter_channels)
00187 s->filter_channels = s->output_channels;
00188
00189 s->sample_fmt[0] = sample_fmt_in;
00190 s->sample_fmt[1] = sample_fmt_out;
00191 s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
00192 s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
00193
00194 if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
00195 if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
00196 s->sample_fmt[0], 1, NULL, 0))) {
00197 av_log(s, AV_LOG_ERROR,
00198 "Cannot convert %s sample format to s16 sample format\n",
00199 av_get_sample_fmt_name(s->sample_fmt[0]));
00200 av_free(s);
00201 return NULL;
00202 }
00203 }
00204
00205 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
00206 if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
00207 AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
00208 av_log(s, AV_LOG_ERROR,
00209 "Cannot convert s16 sample format to %s sample format\n",
00210 av_get_sample_fmt_name(s->sample_fmt[1]));
00211 av_audio_convert_free(s->convert_ctx[0]);
00212 av_free(s);
00213 return NULL;
00214 }
00215 }
00216
00217 s->resample_context = av_resample_init(output_rate, input_rate,
00218 filter_length, log2_phase_count,
00219 linear, cutoff);
00220
00221 *(const AVClass**)s->resample_context = &audioresample_context_class;
00222
00223 return s;
00224 }
00225
00226
00227
00228 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
00229 {
00230 int i, nb_samples1;
00231 short *bufin[MAX_CHANNELS];
00232 short *bufout[MAX_CHANNELS];
00233 short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
00234 short *output_bak = NULL;
00235 int lenout;
00236
00237 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
00238
00239 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
00240 return nb_samples;
00241 }
00242
00243 if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
00244 int istride[1] = { s->sample_size[0] };
00245 int ostride[1] = { 2 };
00246 const void *ibuf[1] = { input };
00247 void *obuf[1];
00248 unsigned input_size = nb_samples * s->input_channels * 2;
00249
00250 if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
00251 av_free(s->buffer[0]);
00252 s->buffer_size[0] = input_size;
00253 s->buffer[0] = av_malloc(s->buffer_size[0]);
00254 if (!s->buffer[0]) {
00255 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
00256 return 0;
00257 }
00258 }
00259
00260 obuf[0] = s->buffer[0];
00261
00262 if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
00263 ibuf, istride, nb_samples * s->input_channels) < 0) {
00264 av_log(s->resample_context, AV_LOG_ERROR,
00265 "Audio sample format conversion failed\n");
00266 return 0;
00267 }
00268
00269 input = s->buffer[0];
00270 }
00271
00272 lenout = 4 * nb_samples * s->ratio + 16;
00273
00274 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
00275 output_bak = output;
00276
00277 if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
00278 av_free(s->buffer[1]);
00279 s->buffer_size[1] = lenout;
00280 s->buffer[1] = av_malloc(s->buffer_size[1]);
00281 if (!s->buffer[1]) {
00282 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
00283 return 0;
00284 }
00285 }
00286
00287 output = s->buffer[1];
00288 }
00289
00290
00291 for (i = 0; i < s->filter_channels; i++) {
00292 bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short));
00293 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
00294 buftmp2[i] = bufin[i] + s->temp_len;
00295 bufout[i] = av_malloc(lenout * sizeof(short));
00296 }
00297
00298 if (s->input_channels == 2 && s->output_channels == 1) {
00299 buftmp3[0] = output;
00300 stereo_to_mono(buftmp2[0], input, nb_samples);
00301 } else if (s->output_channels >= 2 && s->input_channels == 1) {
00302 buftmp3[0] = bufout[0];
00303 memcpy(buftmp2[0], input, nb_samples * sizeof(short));
00304 } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
00305 for (i = 0; i < s->input_channels; i++) {
00306 buftmp3[i] = bufout[i];
00307 }
00308 deinterleave(buftmp2, input, s->input_channels, nb_samples);
00309 } else {
00310 buftmp3[0] = output;
00311 memcpy(buftmp2[0], input, nb_samples * sizeof(short));
00312 }
00313
00314 nb_samples += s->temp_len;
00315
00316
00317 nb_samples1 = 0;
00318 for (i = 0; i < s->filter_channels; i++) {
00319 int consumed;
00320 int is_last = i + 1 == s->filter_channels;
00321
00322 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
00323 &consumed, nb_samples, lenout, is_last);
00324 s->temp_len = nb_samples - consumed;
00325 s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short));
00326 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
00327 }
00328
00329 if (s->output_channels == 2 && s->input_channels == 1) {
00330 mono_to_stereo(output, buftmp3[0], nb_samples1);
00331 } else if (s->output_channels == 6 && s->input_channels == 2) {
00332 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
00333 } else if (s->output_channels == s->input_channels && s->input_channels >= 2) {
00334 interleave(output, buftmp3, s->output_channels, nb_samples1);
00335 }
00336
00337 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
00338 int istride[1] = { 2 };
00339 int ostride[1] = { s->sample_size[1] };
00340 const void *ibuf[1] = { output };
00341 void *obuf[1] = { output_bak };
00342
00343 if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
00344 ibuf, istride, nb_samples1 * s->output_channels) < 0) {
00345 av_log(s->resample_context, AV_LOG_ERROR,
00346 "Audio sample format convertion failed\n");
00347 return 0;
00348 }
00349 }
00350
00351 for (i = 0; i < s->filter_channels; i++) {
00352 av_free(bufin[i]);
00353 av_free(bufout[i]);
00354 }
00355
00356 return nb_samples1;
00357 }
00358
00359 void audio_resample_close(ReSampleContext *s)
00360 {
00361 int i;
00362 av_resample_close(s->resample_context);
00363 for (i = 0; i < s->filter_channels; i++)
00364 av_freep(&s->temp[i]);
00365 av_freep(&s->buffer[0]);
00366 av_freep(&s->buffer[1]);
00367 av_audio_convert_free(s->convert_ctx[0]);
00368 av_audio_convert_free(s->convert_ctx[1]);
00369 av_free(s);
00370 }