35 #if FF_API_AVCODEC_RESAMPLE
37 #define MAX_CHANNELS 8
39 struct AVResampleContext;
43 return "audioresample";
47 static const AVClass audioresample_context_class = {
51 struct ReSampleContext {
52 struct AVResampleContext *resample_context;
57 int input_channels, output_channels, filter_channels;
60 unsigned sample_size[2];
62 unsigned buffer_size[2];
66 static void stereo_to_mono(
short *output,
short *input,
int n1)
74 q[0] = (p[0] + p[1]) >> 1;
75 q[1] = (p[2] + p[3]) >> 1;
76 q[2] = (p[4] + p[5]) >> 1;
77 q[3] = (p[6] + p[7]) >> 1;
83 q[0] = (p[0] + p[1]) >> 1;
91 static void mono_to_stereo(
short *output,
short *input,
int n1)
100 v = p[0]; q[0] = v; q[1] = v;
101 v = p[1]; q[2] = v; q[3] = v;
102 v = p[2]; q[4] = v; q[5] = v;
103 v = p[3]; q[6] = v; q[7] = v;
109 v = p[0]; q[0] = v; q[1] = v;
116 static void deinterleave(
short **output,
short *input,
int channels,
int samples)
120 for (i = 0; i <
samples; i++) {
121 for (j = 0; j < channels; j++) {
122 *output[j]++ = *input++;
127 static void interleave(
short *output,
short **input,
int channels,
int samples)
131 for (i = 0; i <
samples; i++) {
132 for (j = 0; j < channels; j++) {
133 *output++ = *input[j]++;
138 static void ac3_5p1_mux(
short *output,
short *input1,
short *input2,
int n)
143 for (i = 0; i < n; i++) {
147 *output++ = (l / 2) + (r / 2);
155 ReSampleContext *av_audio_resample_init(
int output_channels,
int input_channels,
156 int output_rate,
int input_rate,
159 int filter_length,
int log2_phase_count,
160 int linear,
double cutoff)
166 "Resampling with input channels greater than %d is unsupported.\n",
170 if (output_channels != input_channels &&
171 (input_channels > 2 ||
172 output_channels > 2 &&
173 !(output_channels == 6 && input_channels == 2))) {
175 "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");
181 av_log(
NULL, AV_LOG_ERROR,
"Can't allocate memory for resample context.\n");
185 s->ratio = (float)output_rate / (
float)input_rate;
187 s->input_channels = input_channels;
188 s->output_channels = output_channels;
190 s->filter_channels = s->input_channels;
191 if (s->output_channels < s->filter_channels)
192 s->filter_channels = s->output_channels;
194 s->sample_fmt[0] = sample_fmt_in;
195 s->sample_fmt[1] = sample_fmt_out;
201 s->sample_fmt[0], 1,
NULL, 0))) {
203 "Cannot convert %s sample format to s16 sample format\n",
214 "Cannot convert s16 sample format to %s sample format\n",
222 s->resample_context = av_resample_init(output_rate, input_rate,
223 filter_length, log2_phase_count,
226 *(
const AVClass**)s->resample_context = &audioresample_context_class;
233 int audio_resample(ReSampleContext *s,
short *output,
short *input,
int nb_samples)
239 short *output_bak =
NULL;
242 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
244 memcpy(output, input, nb_samples * s->input_channels *
sizeof(
short));
249 int istride[1] = { s->sample_size[0] };
250 int ostride[1] = { 2 };
251 const void *ibuf[1] = { input };
253 unsigned input_size = nb_samples * s->input_channels * 2;
255 if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
257 s->buffer_size[0] = input_size;
258 s->buffer[0] =
av_malloc(s->buffer_size[0]);
260 av_log(s->resample_context, AV_LOG_ERROR,
"Could not allocate buffer\n");
265 obuf[0] = s->buffer[0];
268 ibuf, istride, nb_samples * s->input_channels) < 0) {
269 av_log(s->resample_context, AV_LOG_ERROR,
270 "Audio sample format conversion failed\n");
274 input = s->buffer[0];
277 lenout = 4 * nb_samples * s->ratio + 16;
284 if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
286 s->buffer_size[1] = out_size;
287 s->buffer[1] =
av_malloc(s->buffer_size[1]);
289 av_log(s->resample_context, AV_LOG_ERROR,
"Could not allocate buffer\n");
294 output = s->buffer[1];
298 for (i = 0; i < s->filter_channels; i++) {
299 bufin[i] =
av_malloc((nb_samples + s->temp_len) *
sizeof(
short));
300 memcpy(bufin[i], s->temp[i], s->temp_len *
sizeof(
short));
301 buftmp2[i] = bufin[i] + s->temp_len;
302 bufout[i] =
av_malloc(lenout *
sizeof(
short));
305 if (s->input_channels == 2 && s->output_channels == 1) {
307 stereo_to_mono(buftmp2[0], input, nb_samples);
308 }
else if (s->output_channels >= 2 && s->input_channels == 1) {
309 buftmp3[0] = bufout[0];
310 memcpy(buftmp2[0], input, nb_samples *
sizeof(
short));
311 }
else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
312 for (i = 0; i < s->input_channels; i++) {
313 buftmp3[i] = bufout[i];
315 deinterleave(buftmp2, input, s->input_channels, nb_samples);
318 memcpy(buftmp2[0], input, nb_samples *
sizeof(
short));
321 nb_samples += s->temp_len;
325 for (i = 0; i < s->filter_channels; i++) {
327 int is_last = i + 1 == s->filter_channels;
329 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
330 &consumed, nb_samples, lenout, is_last);
331 s->temp_len = nb_samples - consumed;
332 s->temp[i] =
av_realloc(s->temp[i], s->temp_len *
sizeof(
short));
333 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len *
sizeof(
short));
336 if (s->output_channels == 2 && s->input_channels == 1) {
337 mono_to_stereo(output, buftmp3[0], nb_samples1);
338 }
else if (s->output_channels == 6 && s->input_channels == 2) {
339 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
340 }
else if (s->output_channels == s->input_channels && s->input_channels >= 2) {
341 interleave(output, buftmp3, s->output_channels, nb_samples1);
345 int istride[1] = { 2 };
346 int ostride[1] = { s->sample_size[1] };
347 const void *ibuf[1] = { output };
348 void *obuf[1] = { output_bak };
351 ibuf, istride, nb_samples1 * s->output_channels) < 0) {
352 av_log(s->resample_context, AV_LOG_ERROR,
353 "Audio sample format conversion failed\n");
358 for (i = 0; i < s->filter_channels; i++) {
366 void audio_resample_close(ReSampleContext *s)
369 av_resample_close(s->resample_context);
370 for (i = 0; i < s->filter_channels; i++)