* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include "libavutil/common.h"
#include "libavutil/dict.h"
#include "libavutil/error.h"
+#include "libavutil/frame.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "avresample.h"
-#include "audio_data.h"
#include "internal.h"
+#include "audio_data.h"
+#include "audio_convert.h"
+#include "audio_mix.h"
+#include "resample.h"
int avresample_open(AVAudioResampleContext *avr)
{
int ret;
+ if (avresample_is_open(avr)) {
+ av_log(avr, AV_LOG_ERROR, "The resampling context is already open.\n");
+ return AVERROR(EINVAL);
+ }
+
/* set channel mixing parameters */
avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
avr->downmix_needed = avr->in_channels > avr->out_channels;
avr->upmix_needed = avr->out_channels > avr->in_channels ||
- avr->am->matrix ||
- (avr->out_channels == avr->in_channels &&
- avr->in_channel_layout != avr->out_channel_layout);
+ (!avr->downmix_needed && (avr->mix_matrix ||
+ avr->in_channel_layout != avr->out_channel_layout));
avr->mixing_needed = avr->downmix_needed || avr->upmix_needed;
/* set resampling parameters */
avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
avr->force_resampling;
+ /* select internal sample format if not specified by the user */
+ if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
+ (avr->mixing_needed || avr->resample_needed)) {
+ enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
+ enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
+ int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
+ av_get_bytes_per_sample(out_fmt));
+ if (max_bps <= 2) {
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
+ } else if (avr->mixing_needed) {
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
+ } else {
+ if (max_bps <= 4) {
+ if (in_fmt == AV_SAMPLE_FMT_S32P ||
+ out_fmt == AV_SAMPLE_FMT_S32P) {
+ if (in_fmt == AV_SAMPLE_FMT_FLTP ||
+ out_fmt == AV_SAMPLE_FMT_FLTP) {
+ /* if one is s32 and the other is flt, use dbl */
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
+ } else {
+ /* if one is s32 and the other is s32, s16, or u8, use s32 */
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
+ }
+ } else {
+ /* if one is flt and the other is flt, s16 or u8, use flt */
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
+ }
+ } else {
+ /* if either is dbl, use dbl */
+ avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
+ }
+ }
+ av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
+ av_get_sample_fmt_name(avr->internal_sample_fmt));
+ }
+
+ /* we may need to add an extra conversion in order to remap channels if
+ the output format is not planar */
+ if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed &&
+ !ff_sample_fmt_is_planar(avr->out_sample_fmt, avr->out_channels)) {
+ avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
+ }
+
/* set sample format conversion parameters */
- /* override user-requested internal format to avoid unexpected failures
- TODO: support more internal formats */
- if (avr->resample_needed && avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
- av_log(avr, AV_LOG_WARNING, "Using s16p as internal sample format\n");
- avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
- } else if (avr->mixing_needed &&
- avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
- avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
- av_log(avr, AV_LOG_WARNING, "Using fltp as internal sample format\n");
- avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
- }
- if (avr->in_channels == 1)
- avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
- if (avr->out_channels == 1)
- avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
- avr->in_convert_needed = (avr->resample_needed || avr->mixing_needed) &&
- avr->in_sample_fmt != avr->internal_sample_fmt;
if (avr->resample_needed || avr->mixing_needed)
+ avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt;
+ else
+ avr->in_convert_needed = avr->use_channel_map &&
+ !ff_sample_fmt_is_planar(avr->out_sample_fmt, avr->out_channels);
+
+ if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed)
avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
else
avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
+ avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed ||
+ (avr->use_channel_map && avr->resample_needed));
+
+ if (avr->use_channel_map) {
+ if (avr->in_copy_needed) {
+ avr->remap_point = REMAP_IN_COPY;
+ av_log(avr, AV_LOG_TRACE, "remap channels during in_copy\n");
+ } else if (avr->in_convert_needed) {
+ avr->remap_point = REMAP_IN_CONVERT;
+ av_log(avr, AV_LOG_TRACE, "remap channels during in_convert\n");
+ } else if (avr->out_convert_needed) {
+ avr->remap_point = REMAP_OUT_CONVERT;
+ av_log(avr, AV_LOG_TRACE, "remap channels during out_convert\n");
+ } else {
+ avr->remap_point = REMAP_OUT_COPY;
+ av_log(avr, AV_LOG_TRACE, "remap channels during out_copy\n");
+ }
+
+#ifdef DEBUG
+ {
+ int ch;
+ av_log(avr, AV_LOG_TRACE, "output map: ");
+ if (avr->ch_map_info.do_remap)
+ for (ch = 0; ch < avr->in_channels; ch++)
+ av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_map[ch]);
+ else
+ av_log(avr, AV_LOG_TRACE, "n/a");
+ av_log(avr, AV_LOG_TRACE, "\n");
+ av_log(avr, AV_LOG_TRACE, "copy map: ");
+ if (avr->ch_map_info.do_copy)
+ for (ch = 0; ch < avr->in_channels; ch++)
+ av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_copy[ch]);
+ else
+ av_log(avr, AV_LOG_TRACE, "n/a");
+ av_log(avr, AV_LOG_TRACE, "\n");
+ av_log(avr, AV_LOG_TRACE, "zero map: ");
+ if (avr->ch_map_info.do_zero)
+ for (ch = 0; ch < avr->in_channels; ch++)
+ av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_zero[ch]);
+ else
+ av_log(avr, AV_LOG_TRACE, "n/a");
+ av_log(avr, AV_LOG_TRACE, "\n");
+ av_log(avr, AV_LOG_TRACE, "input map: ");
+ for (ch = 0; ch < avr->in_channels; ch++)
+ av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.input_map[ch]);
+ av_log(avr, AV_LOG_TRACE, "\n");
+ }
+#endif
+ } else
+ avr->remap_point = REMAP_NONE;
+
/* allocate buffers */
- if (avr->mixing_needed || avr->in_convert_needed) {
+ if (avr->in_copy_needed || avr->in_convert_needed) {
avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
0, avr->internal_sample_fmt,
"in_buffer");
}
if (avr->resample_needed) {
avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
- 0, avr->internal_sample_fmt,
+ 1024, avr->internal_sample_fmt,
"resample_out_buffer");
if (!avr->resample_out_buffer) {
ret = AVERROR(EINVAL);
/* setup contexts */
if (avr->in_convert_needed) {
avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
- avr->in_sample_fmt, avr->in_channels);
+ avr->in_sample_fmt, avr->in_channels,
+ avr->in_sample_rate,
+ avr->remap_point == REMAP_IN_CONVERT);
if (!avr->ac_in) {
ret = AVERROR(ENOMEM);
goto error;
else
src_fmt = avr->in_sample_fmt;
avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
- avr->out_channels);
+ avr->out_channels,
+ avr->out_sample_rate,
+ avr->remap_point == REMAP_OUT_CONVERT);
if (!avr->ac_out) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->mixing_needed) {
- ret = ff_audio_mix_init(avr);
- if (ret < 0)
+ avr->am = ff_audio_mix_alloc(avr);
+ if (!avr->am) {
+ ret = AVERROR(ENOMEM);
goto error;
+ }
}
return 0;
return ret;
}
+int avresample_is_open(AVAudioResampleContext *avr)
+{
+ return !!avr->out_fifo;
+}
+
void avresample_close(AVAudioResampleContext *avr)
{
ff_audio_data_free(&avr->in_buffer);
ff_audio_data_free(&avr->out_buffer);
av_audio_fifo_free(avr->out_fifo);
avr->out_fifo = NULL;
- av_freep(&avr->ac_in);
- av_freep(&avr->ac_out);
+ ff_audio_convert_free(&avr->ac_in);
+ ff_audio_convert_free(&avr->ac_out);
ff_audio_resample_free(&avr->resample);
- ff_audio_mix_close(avr->am);
- return;
+ ff_audio_mix_free(&avr->am);
+ av_freep(&avr->mix_matrix);
+
+ avr->use_channel_map = 0;
}
void avresample_free(AVAudioResampleContext **avr)
if (!*avr)
return;
avresample_close(*avr);
- av_freep(&(*avr)->am);
av_opt_free(*avr);
av_freep(avr);
}
/* if there are any samples in the output FIFO or if the
user-supplied output buffer is not large enough for all samples,
we add to the output FIFO */
- av_dlog(avr, "[FIFO] add %s to out_fifo\n", converted->name);
+ av_log(avr, AV_LOG_TRACE, "[FIFO] add %s to out_fifo\n", converted->name);
ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0,
converted->nb_samples);
if (ret < 0)
/* if the user specified an output buffer, read samples from the output
FIFO to the user output */
if (output && output->allocated_samples > 0) {
- av_dlog(avr, "[FIFO] read from out_fifo to output\n");
- av_dlog(avr, "[end conversion]\n");
+ av_log(avr, AV_LOG_TRACE, "[FIFO] read from out_fifo to output\n");
+ av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
return ff_audio_data_read_from_fifo(avr->out_fifo, output,
output->allocated_samples);
}
} else if (converted) {
/* copy directly to output if it is large enough or there is not any
data in the output FIFO */
- av_dlog(avr, "[copy] %s to output\n", converted->name);
+ av_log(avr, AV_LOG_TRACE, "[copy] %s to output\n", converted->name);
output->nb_samples = 0;
- ret = ff_audio_data_copy(output, converted);
+ ret = ff_audio_data_copy(output, converted,
+ avr->remap_point == REMAP_OUT_COPY ?
+ &avr->ch_map_info : NULL);
if (ret < 0)
return ret;
- av_dlog(avr, "[end conversion]\n");
+ av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
return output->nb_samples;
}
- av_dlog(avr, "[end conversion]\n");
+ av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
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)
+int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
+ uint8_t **output, int out_plane_size,
+ int out_samples,
+ uint8_t * const *input,
+ int in_plane_size, int in_samples)
{
AudioData input_buffer;
AudioData output_buffer;
AudioData *current_buffer;
- int ret;
+ int ret, direct_output;
/* reset internal buffers */
if (avr->in_buffer) {
avr->out_buffer->allocated_channels);
}
- av_dlog(avr, "[start conversion]\n");
+ av_log(avr, AV_LOG_TRACE, "[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,
current_buffer = &input_buffer;
if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
- !avr->out_convert_needed && output && out_samples >= in_samples) {
+ !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);
+ av_log(avr, AV_LOG_TRACE, "[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->mixing_needed || avr->in_convert_needed) {
+ } 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_log(avr, AV_LOG_TRACE, "[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) {
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);
+ av_log(avr, AV_LOG_TRACE, "[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);
+ av_log(avr, AV_LOG_TRACE, "[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");
+ av_log(avr, AV_LOG_TRACE, "[downmix] in_buffer\n");
ret = ff_audio_mix(avr->am, avr->in_buffer);
if (ret < 0)
return ret;
if (avr->resample_needed) {
AudioData *resample_out;
- int consumed = 0;
- if (!avr->out_convert_needed && output && out_samples > 0)
+ 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->name,
+ av_log(avr, AV_LOG_TRACE, "[resample] %s to %s\n",
+ current_buffer ? current_buffer->name : "null",
resample_out->name);
ret = ff_audio_resample(avr->resample, resample_out,
- current_buffer, &consumed);
+ 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");
+ av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
return 0;
}
}
if (avr->upmix_needed) {
- av_dlog(avr, "[upmix] %s\n", current_buffer->name);
+ av_log(avr, AV_LOG_TRACE, "[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");
+ av_log(avr, AV_LOG_TRACE, "[end conversion]\n");
return current_buffer->nb_samples;
}
if (avr->out_convert_needed) {
- if (output && out_samples >= current_buffer->nb_samples) {
+ 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,
- current_buffer->nb_samples);
+ av_log(avr, AV_LOG_TRACE, "[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");
+ av_log(avr, AV_LOG_TRACE, "[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);
+ av_log(avr, AV_LOG_TRACE, "[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);
+ current_buffer);
if (ret < 0)
return ret;
current_buffer = avr->out_buffer;
}
}
- return handle_buffered_output(avr, &output_buffer, current_buffer);
+ return handle_buffered_output(avr, output ? &output_buffer : NULL,
+ current_buffer);
+}
+
+int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in)
+{
+ if (avresample_is_open(avr)) {
+ avresample_close(avr);
+ }
+
+ if (in) {
+ avr->in_channel_layout = in->channel_layout;
+ avr->in_sample_rate = in->sample_rate;
+ avr->in_sample_fmt = in->format;
+ }
+
+ if (out) {
+ avr->out_channel_layout = out->channel_layout;
+ avr->out_sample_rate = out->sample_rate;
+ avr->out_sample_fmt = out->format;
+ }
+
+ return 0;
+}
+
+static int config_changed(AVAudioResampleContext *avr,
+ AVFrame *out, AVFrame *in)
+{
+ int ret = 0;
+
+ if (in) {
+ if (avr->in_channel_layout != in->channel_layout ||
+ avr->in_sample_rate != in->sample_rate ||
+ avr->in_sample_fmt != in->format) {
+ ret |= AVERROR_INPUT_CHANGED;
+ }
+ }
+
+ if (out) {
+ if (avr->out_channel_layout != out->channel_layout ||
+ avr->out_sample_rate != out->sample_rate ||
+ avr->out_sample_fmt != out->format) {
+ ret |= AVERROR_OUTPUT_CHANGED;
+ }
+ }
+
+ return ret;
+}
+
+static inline int convert_frame(AVAudioResampleContext *avr,
+ AVFrame *out, AVFrame *in)
+{
+ int ret;
+ uint8_t **out_data = NULL, **in_data = NULL;
+ int out_linesize = 0, in_linesize = 0;
+ int out_nb_samples = 0, in_nb_samples = 0;
+
+ if (out) {
+ out_data = out->extended_data;
+ out_linesize = out->linesize[0];
+ out_nb_samples = out->nb_samples;
+ }
+
+ if (in) {
+ in_data = in->extended_data;
+ in_linesize = in->linesize[0];
+ in_nb_samples = in->nb_samples;
+ }
+
+ ret = avresample_convert(avr, out_data, out_linesize,
+ out_nb_samples,
+ in_data, in_linesize,
+ in_nb_samples);
+
+ if (ret < 0) {
+ if (out)
+ out->nb_samples = 0;
+ return ret;
+ }
+
+ if (out)
+ out->nb_samples = ret;
+
+ return 0;
+}
+
+static inline int available_samples(AVFrame *out)
+{
+ int samples;
+ int bytes_per_sample = av_get_bytes_per_sample(out->format);
+ if (!bytes_per_sample)
+ return AVERROR(EINVAL);
+
+ samples = out->linesize[0] / bytes_per_sample;
+ if (av_sample_fmt_is_planar(out->format)) {
+ return samples;
+ } else {
+ int channels = av_get_channel_layout_nb_channels(out->channel_layout);
+ return samples / channels;
+ }
+}
+
+int avresample_convert_frame(AVAudioResampleContext *avr,
+ AVFrame *out, AVFrame *in)
+{
+ int ret, setup = 0;
+
+ if (!avresample_is_open(avr)) {
+ if ((ret = avresample_config(avr, out, in)) < 0)
+ return ret;
+ if ((ret = avresample_open(avr)) < 0)
+ return ret;
+ setup = 1;
+ } else {
+ // return as is or reconfigure for input changes?
+ if ((ret = config_changed(avr, out, in)))
+ return ret;
+ }
+
+ if (out) {
+ if (!out->linesize[0]) {
+ out->nb_samples = avresample_get_out_samples(avr, in->nb_samples);
+ if ((ret = av_frame_get_buffer(out, 0)) < 0) {
+ if (setup)
+ avresample_close(avr);
+ return ret;
+ }
+ } else {
+ if (!out->nb_samples)
+ out->nb_samples = available_samples(out);
+ }
+ }
+
+ return convert_frame(avr, out, in);
+}
+
+int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
+ int stride)
+{
+ int in_channels, out_channels, i, o;
+
+ if (avr->am)
+ return ff_audio_mix_get_matrix(avr->am, matrix, stride);
+
+ in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
+ out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
+
+ if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
+ out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
+ av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
+ return AVERROR(EINVAL);
+ }
+
+ if (!avr->mix_matrix) {
+ av_log(avr, AV_LOG_ERROR, "matrix is not set\n");
+ return AVERROR(EINVAL);
+ }
+
+ for (o = 0; o < out_channels; o++)
+ for (i = 0; i < in_channels; i++)
+ matrix[o * stride + i] = avr->mix_matrix[o * in_channels + i];
+
+ return 0;
+}
+
+int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
+ int stride)
+{
+ int in_channels, out_channels, i, o;
+
+ if (avr->am)
+ return ff_audio_mix_set_matrix(avr->am, matrix, stride);
+
+ in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
+ out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
+
+ if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
+ out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
+ av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
+ return AVERROR(EINVAL);
+ }
+
+ if (avr->mix_matrix)
+ av_freep(&avr->mix_matrix);
+ avr->mix_matrix = av_malloc(in_channels * out_channels *
+ sizeof(*avr->mix_matrix));
+ if (!avr->mix_matrix)
+ return AVERROR(ENOMEM);
+
+ for (o = 0; o < out_channels; o++)
+ for (i = 0; i < in_channels; i++)
+ avr->mix_matrix[o * in_channels + i] = matrix[o * stride + i];
+
+ return 0;
+}
+
+int avresample_set_channel_mapping(AVAudioResampleContext *avr,
+ const int *channel_map)
+{
+ ChannelMapInfo *info = &avr->ch_map_info;
+ int in_channels, ch, i;
+
+ in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
+ if (in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS) {
+ av_log(avr, AV_LOG_ERROR, "Invalid input channel layout\n");
+ return AVERROR(EINVAL);
+ }
+
+ memset(info, 0, sizeof(*info));
+ memset(info->input_map, -1, sizeof(info->input_map));
+
+ for (ch = 0; ch < in_channels; ch++) {
+ if (channel_map[ch] >= in_channels) {
+ av_log(avr, AV_LOG_ERROR, "Invalid channel map\n");
+ return AVERROR(EINVAL);
+ }
+ if (channel_map[ch] < 0) {
+ info->channel_zero[ch] = 1;
+ info->channel_map[ch] = -1;
+ info->do_zero = 1;
+ } else if (info->input_map[channel_map[ch]] >= 0) {
+ info->channel_copy[ch] = info->input_map[channel_map[ch]];
+ info->channel_map[ch] = -1;
+ info->do_copy = 1;
+ } else {
+ info->channel_map[ch] = channel_map[ch];
+ info->input_map[channel_map[ch]] = ch;
+ info->do_remap = 1;
+ }
+ }
+ /* Fill-in unmapped input channels with unmapped output channels.
+ This is used when remapping during conversion from interleaved to
+ planar format. */
+ for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) {
+ while (ch < in_channels && info->input_map[ch] >= 0)
+ ch++;
+ while (i < in_channels && info->channel_map[i] >= 0)
+ i++;
+ if (ch >= in_channels || i >= in_channels)
+ break;
+ info->input_map[ch] = i;
+ }
+
+ avr->use_channel_map = 1;
+ return 0;
}
int avresample_available(AVAudioResampleContext *avr)
return av_audio_fifo_size(avr->out_fifo);
}
-int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples)
+int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples)
+{
+ int64_t samples = avresample_get_delay(avr) + (int64_t)in_nb_samples;
+
+ if (avr->resample_needed) {
+ samples = av_rescale_rnd(samples,
+ avr->out_sample_rate,
+ avr->in_sample_rate,
+ AV_ROUND_UP);
+ }
+
+ samples += avresample_available(avr);
+
+ if (samples > INT_MAX)
+ return AVERROR(EINVAL);
+
+ return samples;
+}
+
+int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
{
if (!output)
return av_audio_fifo_drain(avr->out_fifo, nb_samples);
- return av_audio_fifo_read(avr->out_fifo, output, nb_samples);
+ return av_audio_fifo_read(avr->out_fifo, (void**)output, nb_samples);
}
unsigned avresample_version(void)
projects / ffmpeg / blobdiff