int ff_audio_mix(AudioMix *am, AudioData *src)
{
int use_generic = 1;
int len = src->nb_samples;
/* determine whether to use the optimized function based on pointer and
samples alignment in both the input and output */
if (am->has_optimized_func) {
int aligned_len = FFALIGN(len, am->samples_align);
if (!(src->ptr_align % am->ptr_align) &&
src->samples_align >= aligned_len) {
len = aligned_len;
use_generic = 0;
}
}
av_dlog(am->avr, "audio_mix: %d samples - %d to %d channels (%s)\n",
src->nb_samples, am->in_channels, am->out_channels,
use_generic ? am->func_descr_generic : am->func_descr);
if (use_generic)
am->mix_generic(src->data, am->matrix, len, am->out_channels,
am->in_channels);
else
am->mix(src->data, am->matrix, len, am->out_channels, am->in_channels);
ff_audio_data_set_channels(src, am->out_channels);
return 0;
}
int ff_audio_mix(AudioMix *am, AudioData *src)
{
int use_generic = 1;
int len = src->nb_samples;
int i, j;
/* determine whether to use the optimized function based on pointer and
samples alignment in both the input and output */
if (am->has_optimized_func) {
int aligned_len = FFALIGN(len, am->samples_align);
if (!(src->ptr_align % am->ptr_align) &&
src->samples_align >= aligned_len) {
len = aligned_len;
use_generic = 0;
}
}
av_dlog(am->avr, "audio_mix: %d samples - %d to %d channels (%s)\n",
src->nb_samples, am->in_channels, am->out_channels,
use_generic ? am->func_descr_generic : am->func_descr);
if (am->in_matrix_channels && am->out_matrix_channels) {
uint8_t **data;
uint8_t *data0[AVRESAMPLE_MAX_CHANNELS];
if (am->out_matrix_channels < am->out_channels ||
am->in_matrix_channels < am->in_channels) {
for (i = 0, j = 0; i < FFMAX(am->in_channels, am->out_channels); i++) {
if (am->input_skip[i] || am->output_skip[i] || am->output_zero[i])
continue;
data0[j++] = src->data[i];
}
data = data0;
} else {
data = src->data;
}
if (use_generic)
am->mix_generic(data, am->matrix, len, am->out_matrix_channels,
am->in_matrix_channels);
else
am->mix(data, am->matrix, len, am->out_matrix_channels,
am->in_matrix_channels);
}
if (am->out_matrix_channels < am->out_channels) {
for (i = 0; i < am->out_channels; i++)
if (am->output_zero[i])
av_samples_set_silence(&src->data[i], 0, len, 1, am->fmt);
}
ff_audio_data_set_channels(src, am->out_channels);
return 0;
}
int avresample_convert(AVAudioResampleContext *avr, void **output,
int out_plane_size, int out_samples, void **input,
int in_plane_size, int in_samples)
{
AudioData input_buffer;
AudioData output_buffer;
AudioData *current_buffer;
int ret;
/* reset internal buffers */
if (avr->in_buffer) {
avr->in_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->in_buffer,
avr->in_buffer->allocated_channels);
}
if (avr->resample_out_buffer) {
avr->resample_out_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->resample_out_buffer,
avr->resample_out_buffer->allocated_channels);
}
if (avr->out_buffer) {
avr->out_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->out_buffer,
avr->out_buffer->allocated_channels);
}
av_dlog(avr, "[start conversion]\n");
/* initialize output_buffer with output data */
if (output) {
ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
avr->out_channels, out_samples,
avr->out_sample_fmt, 0, "output");
if (ret < 0)
return ret;
output_buffer.nb_samples = 0;
}
if (input) {
/* initialize input_buffer with input data */
ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
avr->in_channels, in_samples,
avr->in_sample_fmt, 1, "input");
if (ret < 0)
return ret;
current_buffer = &input_buffer;
if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
!avr->out_convert_needed && output && out_samples >= in_samples) {
/* in some rare cases we can copy input to output and upmix
directly in the output buffer */
av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
ret = ff_audio_data_copy(&output_buffer, current_buffer);
if (ret < 0)
return ret;
current_buffer = &output_buffer;
} else if (avr->mixing_needed || avr->in_convert_needed) {
/* if needed, copy or convert input to in_buffer, and downmix if
applicable */
if (avr->in_convert_needed) {
ret = ff_audio_data_realloc(avr->in_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_in, avr->in_buffer, current_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
} else {
av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
ret = ff_audio_data_copy(avr->in_buffer, current_buffer);
if (ret < 0)
return ret;
}
ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
if (avr->downmix_needed) {
av_dlog(avr, "[downmix] in_buffer\n");
ret = ff_audio_mix(avr->am, avr->in_buffer);
if (ret < 0)
return ret;
}
current_buffer = avr->in_buffer;
}
} else {
/* flush resampling buffer and/or output FIFO if input is NULL */
if (!avr->resample_needed)
return handle_buffered_output(avr, output ? &output_buffer : NULL,
NULL);
current_buffer = NULL;
}
if (avr->resample_needed) {
AudioData *resample_out;
int consumed = 0;
if (!avr->out_convert_needed && output && out_samples > 0)
resample_out = &output_buffer;
else
resample_out = avr->resample_out_buffer;
av_dlog(avr, "[resample] %s to %s\n", current_buffer->name,
resample_out->name);
ret = ff_audio_resample(avr->resample, resample_out,
current_buffer, &consumed);
if (ret < 0)
return ret;
/* if resampling did not produce any samples, just return 0 */
if (resample_out->nb_samples == 0) {
av_dlog(avr, "[end conversion]\n");
return 0;
}
current_buffer = resample_out;
}
if (avr->upmix_needed) {
av_dlog(avr, "[upmix] %s\n", current_buffer->name);
ret = ff_audio_mix(avr->am, current_buffer);
if (ret < 0)
return ret;
}
/* if we resampled or upmixed directly to output, return here */
if (current_buffer == &output_buffer) {
av_dlog(avr, "[end conversion]\n");
return current_buffer->nb_samples;
}
if (avr->out_convert_needed) {
if (output && out_samples >= current_buffer->nb_samples) {
/* convert directly to output */
av_dlog(avr, "[convert] %s to output\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
av_dlog(avr, "[end conversion]\n");
return output_buffer.nb_samples;
} else {
ret = ff_audio_data_realloc(avr->out_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
current_buffer, current_buffer->nb_samples);
if (ret < 0)
return ret;
current_buffer = avr->out_buffer;
}
}
return handle_buffered_output(avr, output ? &output_buffer : NULL,
current_buffer);
}
int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
uint8_t **output, int out_plane_size,
int out_samples, uint8_t **input,
int in_plane_size, int in_samples)
{
AudioData input_buffer;
AudioData output_buffer;
AudioData *current_buffer;
int ret, direct_output;
/* reset internal buffers */
if (avr->in_buffer) {
avr->in_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->in_buffer,
avr->in_buffer->allocated_channels);
}
if (avr->resample_out_buffer) {
avr->resample_out_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->resample_out_buffer,
avr->resample_out_buffer->allocated_channels);
}
if (avr->out_buffer) {
avr->out_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->out_buffer,
avr->out_buffer->allocated_channels);
}
av_dlog(avr, "[start conversion]\n");
/* initialize output_buffer with output data */
direct_output = output && av_audio_fifo_size(avr->out_fifo) == 0;
if (output) {
ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
avr->out_channels, out_samples,
avr->out_sample_fmt, 0, "output");
if (ret < 0)
return ret;
output_buffer.nb_samples = 0;
}
if (input) {
/* initialize input_buffer with input data */
ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
avr->in_channels, in_samples,
avr->in_sample_fmt, 1, "input");
if (ret < 0)
return ret;
current_buffer = &input_buffer;
if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
!avr->out_convert_needed && direct_output && out_samples >= in_samples) {
/* in some rare cases we can copy input to output and upmix
directly in the output buffer */
av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
ret = ff_audio_data_copy(&output_buffer, current_buffer,
avr->remap_point == REMAP_OUT_COPY ?
&avr->ch_map_info : NULL);
if (ret < 0)
return ret;
current_buffer = &output_buffer;
} else if (avr->remap_point == REMAP_OUT_COPY &&
(!direct_output || out_samples < in_samples)) {
/* if remapping channels during output copy, we may need to
* use an intermediate buffer in order to remap before adding
* samples to the output fifo */
av_dlog(avr, "[copy] %s to out_buffer\n", current_buffer->name);
ret = ff_audio_data_copy(avr->out_buffer, current_buffer,
&avr->ch_map_info);
if (ret < 0)
return ret;
current_buffer = avr->out_buffer;
} else if (avr->in_copy_needed || avr->in_convert_needed) {
/* if needed, copy or convert input to in_buffer, and downmix if
applicable */
if (avr->in_convert_needed) {
ret = ff_audio_data_realloc(avr->in_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_in, avr->in_buffer,
current_buffer);
if (ret < 0)
return ret;
} else {
av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
ret = ff_audio_data_copy(avr->in_buffer, current_buffer,
avr->remap_point == REMAP_IN_COPY ?
&avr->ch_map_info : NULL);
if (ret < 0)
return ret;
}
ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
if (avr->downmix_needed) {
av_dlog(avr, "[downmix] in_buffer\n");
ret = ff_audio_mix(avr->am, avr->in_buffer);
if (ret < 0)
return ret;
}
current_buffer = avr->in_buffer;
}
} else {
/* flush resampling buffer and/or output FIFO if input is NULL */
if (!avr->resample_needed)
return handle_buffered_output(avr, output ? &output_buffer : NULL,
NULL);
current_buffer = NULL;
}
if (avr->resample_needed) {
AudioData *resample_out;
if (!avr->out_convert_needed && direct_output && out_samples > 0)
resample_out = &output_buffer;
else
resample_out = avr->resample_out_buffer;
av_dlog(avr, "[resample] %s to %s\n",
current_buffer ? current_buffer->name : "null",
resample_out->name);
ret = ff_audio_resample(avr->resample, resample_out,
current_buffer);
if (ret < 0)
return ret;
/* if resampling did not produce any samples, just return 0 */
if (resample_out->nb_samples == 0) {
av_dlog(avr, "[end conversion]\n");
return 0;
}
current_buffer = resample_out;
}
if (avr->upmix_needed) {
av_dlog(avr, "[upmix] %s\n", current_buffer->name);
ret = ff_audio_mix(avr->am, current_buffer);
if (ret < 0)
return ret;
}
/* if we resampled or upmixed directly to output, return here */
if (current_buffer == &output_buffer) {
av_dlog(avr, "[end conversion]\n");
return current_buffer->nb_samples;
}
if (avr->out_convert_needed) {
if (direct_output && out_samples >= current_buffer->nb_samples) {
/* convert directly to output */
av_dlog(avr, "[convert] %s to output\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer);
if (ret < 0)
return ret;
av_dlog(avr, "[end conversion]\n");
return output_buffer.nb_samples;
} else {
ret = ff_audio_data_realloc(avr->out_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
current_buffer);
if (ret < 0)
return ret;
current_buffer = avr->out_buffer;
}
}
return handle_buffered_output(avr, output ? &output_buffer : NULL,
current_buffer);
}
int ff_audio_mix(AudioMix *am, AudioData *src)
{
int use_generic = 1;
int len = src->nb_samples;
int i, j;
/* determine whether to use the optimized function based on pointer and
samples alignment in both the input and output */
if (am->has_optimized_func) {
int aligned_len = FFALIGN(len, am->samples_align);
if (!(src->ptr_align % am->ptr_align) &&
src->samples_align >= aligned_len) {
len = aligned_len;
use_generic = 0;
}
}
av_dlog(am->avr, "audio_mix: %d samples - %d to %d channels (%s)\n",
src->nb_samples, am->in_channels, am->out_channels,
use_generic ? am->func_descr_generic : am->func_descr);
if (am->in_matrix_channels && am->out_matrix_channels) {
uint8_t **data;
uint8_t *data0[AVRESAMPLE_MAX_CHANNELS] = { NULL };
if (am->out_matrix_channels < am->out_channels ||
am->in_matrix_channels < am->in_channels) {
for (i = 0, j = 0; i < FFMAX(am->in_channels, am->out_channels); i++) {
if (am->input_skip[i] || am->output_skip[i] || am->output_zero[i])
continue;
data0[j++] = src->data[i];
}
data = data0;
} else {
data = src->data;
}
if (use_generic)
am->mix_generic(data, am->matrix, len, am->out_matrix_channels,
am->in_matrix_channels);
else
am->mix(data, am->matrix, len, am->out_matrix_channels,
am->in_matrix_channels);
}
if (am->out_matrix_channels < am->out_channels) {
for (i = 0; i < am->out_channels; i++)
if (am->output_zero[i])
av_samples_set_silence(&src->data[i], 0, len, 1, am->fmt);
}
ff_audio_data_set_channels(src, am->out_channels);
/* clip protection is only enabled when the internal sample format is fltp, so we don't need to check here */
if (am->clip_protection) {
for(i = 0; i < src->nb_samples; i++) {
for (j = 0; j < src->channels; j++) {
const float sample = fabs(((float **)src->data)[j][i]);
if (sample > am->clip_max) {
am->clip_max = sample;
av_log(am->avr, AV_LOG_INFO, "Clipping protection at %.3f\n", sample);
}
if (am->clip_max > 1.0f)
((float **)src->data)[j][i] /= am->clip_max;
}
}
}
return 0;
}
