static BOOL ffmpeg_resample_frame(AVAudioResampleContext* context,
AVFrame* in, AVFrame* out)
{
int ret;
if (!avresample_is_open(context))
{
if ((ret = avresample_config(context, out, in)) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
if ((ret = (avresample_open(context))) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
}
if ((ret = avresample_convert_frame(context, out, in)) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
return TRUE;
}
static BOOL ffmpeg_decode(AVCodecContext* dec_ctx, AVPacket* pkt,
AVFrame* frame,
AVAudioResampleContext* resampleContext,
AVFrame* resampled, wStream* out)
{
int ret;
/* send the packet with the compressed data to the decoder */
ret = avcodec_send_packet(dec_ctx, pkt);
if (ret < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error submitting the packet to the decoder %s [%d]",
err, ret);
return FALSE;
}
/* read all the output frames (in general there may be any number of them */
while (ret >= 0)
{
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return TRUE;
else if (ret < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during decoding %s [%d]", err, ret);
return FALSE;
}
if (!avresample_is_open(resampleContext))
{
if ((ret = avresample_config(resampleContext, resampled, frame)) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
if ((ret = (avresample_open(resampleContext))) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
}
if ((ret = avresample_convert_frame(resampleContext, resampled, frame)) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
{
const size_t data_size = resampled->channels * resampled->nb_samples * 2;
Stream_EnsureRemainingCapacity(out, data_size);
Stream_Write(out, resampled->data[0], data_size);
}
}
return TRUE;
}
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) {
av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
avr->in_channel_layout);
return AVERROR(EINVAL);
}
avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
avr->out_channel_layout);
return AVERROR(EINVAL);
}
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->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));
}
/* treat all mono as planar for easier comparison */
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);
/* 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 &&
!av_sample_fmt_is_planar(avr->out_sample_fmt)) {
avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
}
/* set sample format conversion parameters */
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 &&
!av_sample_fmt_is_planar(avr->out_sample_fmt);
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_dlog(avr, "remap channels during in_copy\n");
} else if (avr->in_convert_needed) {
avr->remap_point = REMAP_IN_CONVERT;
av_dlog(avr, "remap channels during in_convert\n");
} else if (avr->out_convert_needed) {
avr->remap_point = REMAP_OUT_CONVERT;
av_dlog(avr, "remap channels during out_convert\n");
} else {
avr->remap_point = REMAP_OUT_COPY;
av_dlog(avr, "remap channels during out_copy\n");
}
#ifdef DEBUG
{
int ch;
av_dlog(avr, "output map: ");
if (avr->ch_map_info.do_remap)
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.channel_map[ch]);
else
av_dlog(avr, "n/a");
av_dlog(avr, "\n");
av_dlog(avr, "copy map: ");
if (avr->ch_map_info.do_copy)
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.channel_copy[ch]);
else
av_dlog(avr, "n/a");
av_dlog(avr, "\n");
av_dlog(avr, "zero map: ");
if (avr->ch_map_info.do_zero)
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.channel_zero[ch]);
else
av_dlog(avr, "n/a");
av_dlog(avr, "\n");
av_dlog(avr, "input map: ");
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.input_map[ch]);
av_dlog(avr, "\n");
}
#endif
} else
avr->remap_point = REMAP_NONE;
/* allocate buffers */
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->in_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->resample_needed) {
avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
1024, avr->internal_sample_fmt,
"resample_out_buffer");
if (!avr->resample_out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->out_convert_needed) {
avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
avr->out_sample_fmt, "out_buffer");
if (!avr->out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
1024);
if (!avr->out_fifo) {
ret = AVERROR(ENOMEM);
goto error;
}
/* 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_rate,
avr->remap_point == REMAP_IN_CONVERT);
if (!avr->ac_in) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->out_convert_needed) {
enum AVSampleFormat src_fmt;
if (avr->in_convert_needed)
src_fmt = avr->internal_sample_fmt;
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_sample_rate,
avr->remap_point == REMAP_OUT_CONVERT);
if (!avr->ac_out) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->resample_needed) {
avr->resample = ff_audio_resample_init(avr);
if (!avr->resample) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->mixing_needed) {
avr->am = ff_audio_mix_alloc(avr);
if (!avr->am) {
ret = AVERROR(ENOMEM);
goto error;
}
}
return 0;
error:
avresample_close(avr);
return ret;
}
static int opus_decode_subpacket(OpusStreamContext *s,
const uint8_t *buf, int buf_size,
int nb_samples)
{
int output_samples = 0;
int flush_needed = 0;
int i, j, ret;
/* check if we need to flush the resampler */
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;
}
}
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;
}
avresample_close(s->avr);
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;
}
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 (!avresample_is_open(s->avr)) {
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 = 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);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
return samples;
}
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 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;
}
