static av_cold void opus_decode_flush(AVCodecContext *ctx)
{
OpusContext *c = ctx->priv_data;
int i;
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
memset(&s->packet, 0, sizeof(s->packet));
s->delayed_samples = 0;
if (s->celt_delay)
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
#if CONFIG_SWRESAMPLE
swr_close(s->swr);
#elif CONFIG_AVRESAMPLE
avresample_close(s->avr);
#endif
av_audio_fifo_drain(c->sync_buffers[i], av_audio_fifo_size(c->sync_buffers[i]));
ff_silk_flush(s->silk);
ff_celt_flush(s->celt);
}
}
static int afilter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
LoopContext *s = ctx->priv;
int ret = 0;
if (s->ignored_samples + frame->nb_samples > s->start && s->size > 0 && s->loop != 0) {
if (s->nb_samples < s->size) {
int written = FFMIN(frame->nb_samples, s->size - s->nb_samples);
int drain = 0;
ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, written);
if (ret < 0)
return ret;
if (!s->nb_samples) {
drain = FFMAX(0, s->start - s->ignored_samples);
s->pts = frame->pts;
av_audio_fifo_drain(s->fifo, drain);
s->pts += s->start - s->ignored_samples;
}
s->nb_samples += ret - drain;
drain = frame->nb_samples - written;
if (s->nb_samples == s->size && drain > 0) {
int ret2;
ret2 = av_audio_fifo_write(s->left, (void **)frame->extended_data, frame->nb_samples);
if (ret2 < 0)
return ret2;
av_audio_fifo_drain(s->left, drain);
}
frame->nb_samples = ret;
s->pts += ret;
ret = ff_filter_frame(outlink, frame);
} else {
int nb_samples = frame->nb_samples;
av_frame_free(&frame);
ret = push_samples(ctx, nb_samples);
}
} else {
s->ignored_samples += frame->nb_samples;
frame->pts = s->pts;
s->pts += frame->nb_samples;
ret = ff_filter_frame(outlink, frame);
}
return ret;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
ShowFreqsContext *s = ctx->priv;
AVFrame *fin = NULL;
int ret = 0;
av_audio_fifo_write(s->fifo, (void **)in->extended_data, in->nb_samples);
while (av_audio_fifo_size(s->fifo) >= s->win_size) {
fin = ff_get_audio_buffer(inlink, s->win_size);
if (!fin) {
ret = AVERROR(ENOMEM);
goto fail;
}
fin->pts = s->pts;
s->pts += s->skip_samples;
ret = av_audio_fifo_peek(s->fifo, (void **)fin->extended_data, s->win_size);
if (ret < 0)
goto fail;
ret = plot_freqs(inlink, fin);
av_frame_free(&fin);
av_audio_fifo_drain(s->fifo, s->skip_samples);
if (ret < 0)
goto fail;
}
fail:
av_frame_free(&fin);
av_frame_free(&in);
return ret;
}
static av_cold void opus_decode_flush(AVCodecContext *ctx)
{
OpusContext *c = ctx->priv_data;
int i;
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
memset(&s->packet, 0, sizeof(s->packet));
s->delayed_samples = 0;
if (s->celt_delay)
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
swr_close(s->swr);
ff_silk_flush(s->silk);
ff_celt_flush(s->celt);
}
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioSurroundContext *s = ctx->priv;
av_audio_fifo_write(s->fifo, (void **)in->extended_data,
in->nb_samples);
if (s->pts == AV_NOPTS_VALUE)
s->pts = in->pts;
av_frame_free(&in);
while (av_audio_fifo_size(s->fifo) >= s->buf_size) {
AVFrame *out;
int ret;
ret = av_audio_fifo_peek(s->fifo, (void **)s->input->extended_data, s->buf_size);
if (ret < 0)
return ret;
ctx->internal->execute(ctx, fft_channel, NULL, NULL, inlink->channels);
s->filter(ctx);
out = ff_get_audio_buffer(outlink, s->hop_size);
if (!out)
return AVERROR(ENOMEM);
ctx->internal->execute(ctx, ifft_channel, out, NULL, outlink->channels);
out->pts = s->pts;
if (s->pts != AV_NOPTS_VALUE)
s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
av_audio_fifo_drain(s->fifo, s->hop_size);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
return ret;
}
return 0;
}
int avresample_read(AVAudioResampleContext *avr, void **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);
}
static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
{
int samples = s->packet.frame_duration;
int redundancy = 0;
int redundancy_size, redundancy_pos;
int ret, i, consumed;
int delayed_samples = s->delayed_samples;
ret = opus_rc_init(&s->rc, data, size);
if (ret < 0)
return ret;
/* decode the silk frame */
if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
if (!swr_is_initialized(s->swr)) {
ret = opus_init_resample(s);
if (ret < 0)
return ret;
}
samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
s->packet.stereo + 1,
silk_frame_duration_ms[s->packet.config]);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
return samples;
}
samples = swr_convert(s->swr,
(uint8_t**)s->out, s->packet.frame_duration,
(const uint8_t**)s->silk_output, samples);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
return samples;
}
av_assert2((samples & 7) == 0);
s->delayed_samples += s->packet.frame_duration - samples;
} else
ff_silk_flush(s->silk);
// decode redundancy information
consumed = opus_rc_tell(&s->rc);
if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
redundancy = opus_rc_p2model(&s->rc, 12);
else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
redundancy = 1;
if (redundancy) {
redundancy_pos = opus_rc_p2model(&s->rc, 1);
if (s->packet.mode == OPUS_MODE_HYBRID)
redundancy_size = opus_rc_unimodel(&s->rc, 256) + 2;
else
redundancy_size = size - (consumed + 7) / 8;
size -= redundancy_size;
if (size < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
return AVERROR_INVALIDDATA;
}
if (redundancy_pos) {
ret = opus_decode_redundancy(s, data + size, redundancy_size);
if (ret < 0)
return ret;
ff_celt_flush(s->celt);
}
}
/* decode the CELT frame */
if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
float *out_tmp[2] = { s->out[0], s->out[1] };
float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
out_tmp : s->celt_output;
int celt_output_samples = samples;
int delay_samples = av_audio_fifo_size(s->celt_delay);
if (delay_samples) {
if (s->packet.mode == OPUS_MODE_HYBRID) {
av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
delay_samples);
out_tmp[i] += delay_samples;
}
celt_output_samples -= delay_samples;
} else {
av_log(s->avctx, AV_LOG_WARNING,
"Spurious CELT delay samples present.\n");
av_audio_fifo_drain(s->celt_delay, delay_samples);
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_BUG;
}
}
opus_raw_init(&s->rc, data + size, size);
ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
s->packet.stereo + 1,
s->packet.frame_duration,
(s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
celt_band_end[s->packet.bandwidth]);
if (ret < 0)
return ret;
if (s->packet.mode == OPUS_MODE_HYBRID) {
int celt_delay = s->packet.frame_duration - celt_output_samples;
void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
s->celt_output[1] + celt_output_samples
};
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(out_tmp[i],
s->celt_output[i], 1.0,
celt_output_samples);
}
ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
if (ret < 0)
return ret;
}
} else
ff_celt_flush(s->celt);
if (s->redundancy_idx) {
for (i = 0; i < s->output_channels; i++)
opus_fade(s->out[i], s->out[i],
s->redundancy_output[i] + 120 + s->redundancy_idx,
ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
s->redundancy_idx = 0;
}
if (redundancy) {
if (!redundancy_pos) {
ff_celt_flush(s->celt);
ret = opus_decode_redundancy(s, data + size, redundancy_size);
if (ret < 0)
return ret;
for (i = 0; i < s->output_channels; i++) {
opus_fade(s->out[i] + samples - 120 + delayed_samples,
s->out[i] + samples - 120 + delayed_samples,
s->redundancy_output[i] + 120,
ff_celt_window2, 120 - delayed_samples);
if (delayed_samples)
s->redundancy_idx = 120 - delayed_samples;
}
} else {
for (i = 0; i < s->output_channels; i++) {
memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
opus_fade(s->out[i] + 120 + delayed_samples,
s->redundancy_output[i] + 120,
s->out[i] + 120 + delayed_samples,
ff_celt_window2, 120);
}
}
}
return samples;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AFFTFiltContext *s = ctx->priv;
const int window_size = s->window_size;
const float f = 1. / s->win_scale;
double values[VAR_VARS_NB];
AVFrame *out, *in = NULL;
int ch, n, ret, i, j, k;
int start = s->start, end = s->end;
av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples);
av_frame_free(&frame);
while (av_audio_fifo_size(s->fifo) >= window_size) {
if (!in) {
in = ff_get_audio_buffer(outlink, window_size);
if (!in)
return AVERROR(ENOMEM);
}
ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, window_size);
if (ret < 0)
break;
for (ch = 0; ch < inlink->channels; ch++) {
const float *src = (float *)in->extended_data[ch];
FFTComplex *fft_data = s->fft_data[ch];
for (n = 0; n < in->nb_samples; n++) {
fft_data[n].re = src[n] * s->window_func_lut[n];
fft_data[n].im = 0;
}
for (; n < window_size; n++) {
fft_data[n].re = 0;
fft_data[n].im = 0;
}
}
values[VAR_PTS] = s->pts;
values[VAR_SAMPLE_RATE] = inlink->sample_rate;
values[VAR_NBBINS] = window_size / 2;
values[VAR_CHANNELS] = inlink->channels;
for (ch = 0; ch < inlink->channels; ch++) {
FFTComplex *fft_data = s->fft_data[ch];
float *buf = (float *)s->buffer->extended_data[ch];
int x;
values[VAR_CHANNEL] = ch;
av_fft_permute(s->fft, fft_data);
av_fft_calc(s->fft, fft_data);
for (n = 0; n < window_size / 2; n++) {
float fr, fi;
values[VAR_BIN] = n;
fr = av_expr_eval(s->real[ch], values, s);
fi = av_expr_eval(s->imag[ch], values, s);
fft_data[n].re *= fr;
fft_data[n].im *= fi;
}
for (n = window_size / 2 + 1, x = window_size / 2 - 1; n < window_size; n++, x--) {
fft_data[n].re = fft_data[x].re;
fft_data[n].im = -fft_data[x].im;
}
av_fft_permute(s->ifft, fft_data);
av_fft_calc(s->ifft, fft_data);
start = s->start;
end = s->end;
k = end;
for (i = 0, j = start; j < k && i < window_size; i++, j++) {
buf[j] += s->fft_data[ch][i].re * f;
}
for (; i < window_size; i++, j++) {
buf[j] = s->fft_data[ch][i].re * f;
}
start += s->hop_size;
end = j;
}
s->start = start;
s->end = end;
if (start >= window_size) {
float *dst, *buf;
start -= window_size;
end -= window_size;
s->start = start;
s->end = end;
out = ff_get_audio_buffer(outlink, window_size);
if (!out) {
ret = AVERROR(ENOMEM);
break;
}
out->pts = s->pts;
s->pts += window_size;
for (ch = 0; ch < inlink->channels; ch++) {
dst = (float *)out->extended_data[ch];
buf = (float *)s->buffer->extended_data[ch];
for (n = 0; n < window_size; n++) {
dst[n] = buf[n] * (1 - s->overlap);
}
memmove(buf, buf + window_size, window_size * 4);
}
ret = ff_filter_frame(outlink, out);
if (ret < 0)
break;
}
av_audio_fifo_drain(s->fifo, s->hop_size);
}
av_frame_free(&in);
return ret;
}
static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
{
int samples = s->packet.frame_duration;
int redundancy = 0;
int redundancy_size, redundancy_pos;
int ret, i, consumed;
int delayed_samples = s->delayed_samples;
ret = ff_opus_rc_dec_init(&s->rc, data, size);
if (ret < 0)
return ret;
/* decode the silk frame */
if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
#if CONFIG_SWRESAMPLE
if (!swr_is_initialized(s->swr)) {
#elif CONFIG_AVRESAMPLE
if (!avresample_is_open(s->avr)) {
#endif
ret = opus_init_resample(s);
if (ret < 0)
return ret;
}
samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
s->packet.stereo + 1,
silk_frame_duration_ms[s->packet.config]);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
return samples;
}
#if CONFIG_SWRESAMPLE
samples = swr_convert(s->swr,
(uint8_t**)s->out, s->packet.frame_duration,
(const uint8_t**)s->silk_output, samples);
#elif CONFIG_AVRESAMPLE
samples = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size,
s->packet.frame_duration,
(uint8_t**)s->silk_output,
sizeof(s->silk_buf[0]),
samples);
#endif
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
return samples;
}
av_assert2((samples & 7) == 0);
s->delayed_samples += s->packet.frame_duration - samples;
} else
ff_silk_flush(s->silk);
// decode redundancy information
consumed = opus_rc_tell(&s->rc);
if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
redundancy = ff_opus_rc_dec_log(&s->rc, 12);
else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
redundancy = 1;
if (redundancy) {
redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1);
if (s->packet.mode == OPUS_MODE_HYBRID)
redundancy_size = ff_opus_rc_dec_uint(&s->rc, 256) + 2;
else
redundancy_size = size - (consumed + 7) / 8;
size -= redundancy_size;
if (size < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
return AVERROR_INVALIDDATA;
}
if (redundancy_pos) {
ret = opus_decode_redundancy(s, data + size, redundancy_size);
if (ret < 0)
return ret;
ff_celt_flush(s->celt);
}
}
/* decode the CELT frame */
if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
float *out_tmp[2] = { s->out[0], s->out[1] };
float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
out_tmp : s->celt_output;
int celt_output_samples = samples;
int delay_samples = av_audio_fifo_size(s->celt_delay);
if (delay_samples) {
if (s->packet.mode == OPUS_MODE_HYBRID) {
av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
delay_samples);
out_tmp[i] += delay_samples;
}
celt_output_samples -= delay_samples;
} else {
av_log(s->avctx, AV_LOG_WARNING,
"Spurious CELT delay samples present.\n");
av_audio_fifo_drain(s->celt_delay, delay_samples);
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_BUG;
}
}
ff_opus_rc_dec_raw_init(&s->rc, data + size, size);
ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
s->packet.stereo + 1,
s->packet.frame_duration,
(s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
ff_celt_band_end[s->packet.bandwidth]);
if (ret < 0)
return ret;
if (s->packet.mode == OPUS_MODE_HYBRID) {
int celt_delay = s->packet.frame_duration - celt_output_samples;
void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
s->celt_output[1] + celt_output_samples };
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(out_tmp[i],
s->celt_output[i], 1.0,
celt_output_samples);
}
ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
if (ret < 0)
return ret;
}
} else
ff_celt_flush(s->celt);
if (s->redundancy_idx) {
for (i = 0; i < s->output_channels; i++)
opus_fade(s->out[i], s->out[i],
s->redundancy_output[i] + 120 + s->redundancy_idx,
ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
s->redundancy_idx = 0;
}
if (redundancy) {
if (!redundancy_pos) {
ff_celt_flush(s->celt);
ret = opus_decode_redundancy(s, data + size, redundancy_size);
if (ret < 0)
return ret;
for (i = 0; i < s->output_channels; i++) {
opus_fade(s->out[i] + samples - 120 + delayed_samples,
s->out[i] + samples - 120 + delayed_samples,
s->redundancy_output[i] + 120,
ff_celt_window2, 120 - delayed_samples);
if (delayed_samples)
s->redundancy_idx = 120 - delayed_samples;
}
} else {
for (i = 0; i < s->output_channels; i++) {
memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
opus_fade(s->out[i] + 120 + delayed_samples,
s->redundancy_output[i] + 120,
s->out[i] + 120 + delayed_samples,
ff_celt_window2, 120);
}
}
}
return samples;
}
static int opus_decode_subpacket(OpusStreamContext *s,
const uint8_t *buf, int buf_size,
float **out, int out_size,
int nb_samples)
{
int output_samples = 0;
int flush_needed = 0;
int i, j, ret;
s->out[0] = out[0];
s->out[1] = out[1];
s->out_size = out_size;
/* check if we need to flush the resampler */
#if CONFIG_SWRESAMPLE
if (swr_is_initialized(s->swr)) {
if (buf) {
int64_t cur_samplerate;
av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
} else {
flush_needed = !!s->delayed_samples;
}
}
#elif CONFIG_AVRESAMPLE
if (avresample_is_open(s->avr)) {
if (buf) {
int64_t cur_samplerate;
av_opt_get_int(s->avr, "in_sample_rate", 0, &cur_samplerate);
flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
} else {
flush_needed = !!s->delayed_samples;
}
}
#endif
if (!buf && !flush_needed)
return 0;
/* use dummy output buffers if the channel is not mapped to anything */
if (!s->out[0] ||
(s->output_channels == 2 && !s->out[1])) {
av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);
if (!s->out_dummy)
return AVERROR(ENOMEM);
if (!s->out[0])
s->out[0] = s->out_dummy;
if (!s->out[1])
s->out[1] = s->out_dummy;
}
/* flush the resampler if necessary */
if (flush_needed) {
ret = opus_flush_resample(s, s->delayed_samples);
if (ret < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
return ret;
}
#if CONFIG_SWRESAMPLE
swr_close(s->swr);
#elif CONFIG_AVRESAMPLE
avresample_close(s->avr);
#endif
output_samples += s->delayed_samples;
s->delayed_samples = 0;
if (!buf)
goto finish;
}
/* decode all the frames in the packet */
for (i = 0; i < s->packet.frame_count; i++) {
int size = s->packet.frame_size[i];
int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return samples;
for (j = 0; j < s->output_channels; j++)
memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
samples = s->packet.frame_duration;
}
output_samples += samples;
for (j = 0; j < s->output_channels; j++)
s->out[j] += samples;
s->out_size -= samples * sizeof(float);
}
finish:
s->out[0] = s->out[1] = NULL;
s->out_size = 0;
return output_samples;
}
