[FFmpeg-cvslog] avcodec/opusdec: Move per-stream fields to OpusStreamContext
Andreas Rheinhardt
git at videolan.org
Fri Jan 1 01:07:36 EET 2021
ffmpeg | branch: master | Andreas Rheinhardt <andreas.rheinhardt at gmail.com> | Mon Dec 28 17:46:44 2020 +0100| [f03eade8690d9914fce574afb1795f16c036bb6b] | committer: Andreas Rheinhardt
avcodec/opusdec: Move per-stream fields to OpusStreamContext
Besides being more natural it also avoids allocations for separate
arrays of decoded samples/output buffers/....
Reviewed-by: Lynne <dev at lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=f03eade8690d9914fce574afb1795f16c036bb6b
---
libavcodec/opus.h | 24 +++++------
libavcodec/opusdec.c | 111 +++++++++++++++++++++++----------------------------
2 files changed, 62 insertions(+), 73 deletions(-)
diff --git a/libavcodec/opus.h b/libavcodec/opus.h
index 63ecd0aff7..fa63353e9b 100644
--- a/libavcodec/opus.h
+++ b/libavcodec/opus.h
@@ -101,6 +101,15 @@ typedef struct OpusStreamContext {
AVCodecContext *avctx;
int output_channels;
+ /* number of decoded samples for this stream */
+ int decoded_samples;
+ /* current output buffers for this stream */
+ float *out[2];
+ int out_size;
+ /* Buffer with samples from this stream for synchronizing
+ * the streams when they have different resampling delays */
+ AVAudioFifo *sync_buffer;
+
OpusRangeCoder rc;
OpusRangeCoder redundancy_rc;
SilkContext *silk;
@@ -115,9 +124,9 @@ typedef struct OpusStreamContext {
DECLARE_ALIGNED(32, float, redundancy_buf)[2][960];
float *redundancy_output[2];
- /* data buffers for the final output data */
- float *out[2];
- int out_size;
+ /* buffers for the next samples to be decoded */
+ float *cur_out[2];
+ int remaining_out_size;
float *out_dummy;
int out_dummy_allocated_size;
@@ -154,15 +163,6 @@ typedef struct OpusContext {
OpusStreamContext *streams;
int apply_phase_inv;
- /* current output buffers for each streams */
- float **out;
- int *out_size;
- /* Buffers for synchronizing the streams when they have different
- * resampling delays */
- AVAudioFifo **sync_buffers;
- /* number of decoded samples for each stream */
- int *decoded_samples;
-
int nb_streams;
int nb_stereo_streams;
diff --git a/libavcodec/opusdec.c b/libavcodec/opusdec.c
index 462d70b3bf..b09a542c86 100644
--- a/libavcodec/opusdec.c
+++ b/libavcodec/opusdec.c
@@ -87,7 +87,7 @@ static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
int celt_size = av_audio_fifo_size(s->celt_delay);
int ret, i;
ret = swr_convert(s->swr,
- (uint8_t**)s->out, nb_samples,
+ (uint8_t**)s->cur_out, nb_samples,
NULL, 0);
if (ret < 0)
return ret;
@@ -104,7 +104,7 @@ static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
}
av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
for (i = 0; i < s->output_channels; i++) {
- s->fdsp->vector_fmac_scalar(s->out[i],
+ s->fdsp->vector_fmac_scalar(s->cur_out[i],
s->celt_output[i], 1.0,
nb_samples);
}
@@ -112,15 +112,15 @@ static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
if (s->redundancy_idx) {
for (i = 0; i < s->output_channels; i++)
- opus_fade(s->out[i], s->out[i],
+ opus_fade(s->cur_out[i], s->cur_out[i],
s->redundancy_output[i] + 120 + s->redundancy_idx,
ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
s->redundancy_idx = 0;
}
- s->out[0] += nb_samples;
- s->out[1] += nb_samples;
- s->out_size -= nb_samples * sizeof(float);
+ s->cur_out[0] += nb_samples;
+ s->cur_out[1] += nb_samples;
+ s->remaining_out_size -= nb_samples * sizeof(float);
return 0;
}
@@ -199,7 +199,7 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
return samples;
}
samples = swr_convert(s->swr,
- (uint8_t**)s->out, s->packet.frame_duration,
+ (uint8_t**)s->cur_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");
@@ -240,7 +240,7 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
/* 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 *out_tmp[2] = { s->cur_out[0], s->cur_out[1] };
float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
out_tmp : s->celt_output;
int celt_output_samples = samples;
@@ -295,7 +295,7 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
if (s->redundancy_idx) {
for (i = 0; i < s->output_channels; i++)
- opus_fade(s->out[i], s->out[i],
+ opus_fade(s->cur_out[i], s->cur_out[i],
s->redundancy_output[i] + 120 + s->redundancy_idx,
ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
s->redundancy_idx = 0;
@@ -308,8 +308,8 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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,
+ opus_fade(s->cur_out[i] + samples - 120 + delayed_samples,
+ s->cur_out[i] + samples - 120 + delayed_samples,
s->redundancy_output[i] + 120,
ff_celt_window2, 120 - delayed_samples);
if (delayed_samples)
@@ -317,10 +317,10 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
}
} 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,
+ memcpy(s->cur_out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
+ opus_fade(s->cur_out[i] + 120 + delayed_samples,
s->redundancy_output[i] + 120,
- s->out[i] + 120 + delayed_samples,
+ s->cur_out[i] + 120 + delayed_samples,
ff_celt_window2, 120);
}
}
@@ -331,16 +331,15 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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;
+ s->cur_out[0] = s->out[0];
+ s->cur_out[1] = s->out[1];
+ s->remaining_out_size = s->out_size;
/* check if we need to flush the resampler */
if (swr_is_initialized(s->swr)) {
@@ -357,15 +356,16 @@ static int opus_decode_subpacket(OpusStreamContext *s,
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->cur_out[0] ||
+ (s->output_channels == 2 && !s->cur_out[1])) {
+ av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size,
+ s->remaining_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;
+ if (!s->cur_out[0])
+ s->cur_out[0] = s->out_dummy;
+ if (!s->cur_out[1])
+ s->cur_out[1] = s->out_dummy;
}
/* flush the resampler if necessary */
@@ -394,19 +394,19 @@ static int opus_decode_subpacket(OpusStreamContext *s,
return samples;
for (j = 0; j < s->output_channels; j++)
- memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
+ memset(s->cur_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);
+ s->cur_out[j] += samples;
+ s->remaining_out_size -= samples * sizeof(float);
}
finish:
- s->out[0] = s->out[1] = NULL;
- s->out_size = 0;
+ s->cur_out[0] = s->cur_out[1] = NULL;
+ s->remaining_out_size = 0;
return output_samples;
}
@@ -429,7 +429,7 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
s->out[0] =
s->out[1] = NULL;
delayed_samples = FFMAX(delayed_samples,
- s->delayed_samples + av_audio_fifo_size(c->sync_buffers[i]));
+ s->delayed_samples + av_audio_fifo_size(s->sync_buffer));
}
/* decode the header of the first sub-packet to find out the sample count */
@@ -458,17 +458,17 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
return ret;
frame->nb_samples = 0;
- memset(c->out, 0, c->nb_streams * 2 * sizeof(*c->out));
for (i = 0; i < avctx->channels; i++) {
ChannelMap *map = &c->channel_maps[i];
if (!map->copy)
- c->out[2 * map->stream_idx + map->channel_idx] = (float*)frame->extended_data[i];
+ c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
}
/* read the data from the sync buffers */
for (i = 0; i < c->nb_streams; i++) {
- float **out = c->out + 2 * i;
- int sync_size = av_audio_fifo_size(c->sync_buffers[i]);
+ OpusStreamContext *s = &c->streams[i];
+ float **out = s->out;
+ int sync_size = av_audio_fifo_size(s->sync_buffer);
float sync_dummy[32];
int out_dummy = (!out[0]) | ((!out[1]) << 1);
@@ -480,7 +480,7 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy))
return AVERROR_BUG;
- ret = av_audio_fifo_read(c->sync_buffers[i], (void**)out, sync_size);
+ ret = av_audio_fifo_read(s->sync_buffer, (void**)out, sync_size);
if (ret < 0)
return ret;
@@ -493,7 +493,7 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
else
out[1] += ret;
- c->out_size[i] = frame->linesize[0] - ret * sizeof(float);
+ s->out_size = frame->linesize[0] - ret * sizeof(float);
}
/* decode each sub-packet */
@@ -516,10 +516,10 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
}
ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,
- c->out + 2 * i, c->out_size[i], coded_samples);
+ coded_samples);
if (ret < 0)
return ret;
- c->decoded_samples[i] = ret;
+ s->decoded_samples = ret;
decoded_samples = FFMIN(decoded_samples, ret);
buf += s->packet.packet_size;
@@ -528,13 +528,14 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
/* buffer the extra samples */
for (i = 0; i < c->nb_streams; i++) {
- int buffer_samples = c->decoded_samples[i] - decoded_samples;
+ OpusStreamContext *s = &c->streams[i];
+ int buffer_samples = s->decoded_samples - decoded_samples;
if (buffer_samples) {
- float *buf[2] = { c->out[2 * i + 0] ? c->out[2 * i + 0] : (float*)frame->extended_data[0],
- c->out[2 * i + 1] ? c->out[2 * i + 1] : (float*)frame->extended_data[0] };
+ float *buf[2] = { s->out[0] ? s->out[0] : (float*)frame->extended_data[0],
+ s->out[1] ? s->out[1] : (float*)frame->extended_data[0] };
buf[0] += decoded_samples;
buf[1] += decoded_samples;
- ret = av_audio_fifo_write(c->sync_buffers[i], (void**)buf, buffer_samples);
+ ret = av_audio_fifo_write(s->sync_buffer, (void**)buf, buffer_samples);
if (ret < 0)
return ret;
}
@@ -579,7 +580,7 @@ static av_cold void opus_decode_flush(AVCodecContext *ctx)
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
swr_close(s->swr);
- av_audio_fifo_drain(c->sync_buffers[i], av_audio_fifo_size(c->sync_buffers[i]));
+ av_audio_fifo_drain(s->sync_buffer, av_audio_fifo_size(s->sync_buffer));
ff_silk_flush(s->silk);
ff_celt_flush(s->celt);
@@ -600,21 +601,13 @@ static av_cold int opus_decode_close(AVCodecContext *avctx)
av_freep(&s->out_dummy);
s->out_dummy_allocated_size = 0;
+ av_audio_fifo_free(s->sync_buffer);
av_audio_fifo_free(s->celt_delay);
swr_free(&s->swr);
}
av_freep(&c->streams);
- if (c->sync_buffers) {
- for (i = 0; i < c->nb_streams; i++)
- av_audio_fifo_free(c->sync_buffers[i]);
- }
- av_freep(&c->sync_buffers);
- av_freep(&c->decoded_samples);
- av_freep(&c->out);
- av_freep(&c->out_size);
-
c->nb_streams = 0;
av_freep(&c->channel_maps);
@@ -644,11 +637,7 @@ static av_cold int opus_decode_init(AVCodecContext *avctx)
/* allocate and init each independent decoder */
c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
- c->out = av_mallocz_array(c->nb_streams, 2 * sizeof(*c->out));
- c->out_size = av_mallocz_array(c->nb_streams, sizeof(*c->out_size));
- c->sync_buffers = av_mallocz_array(c->nb_streams, sizeof(*c->sync_buffers));
- c->decoded_samples = av_mallocz_array(c->nb_streams, sizeof(*c->decoded_samples));
- if (!c->streams || !c->sync_buffers || !c->decoded_samples || !c->out || !c->out_size) {
+ if (!c->streams) {
c->nb_streams = 0;
ret = AVERROR(ENOMEM);
goto fail;
@@ -699,9 +688,9 @@ static av_cold int opus_decode_init(AVCodecContext *avctx)
goto fail;
}
- c->sync_buffers[i] = av_audio_fifo_alloc(avctx->sample_fmt,
- s->output_channels, 32);
- if (!c->sync_buffers[i]) {
+ s->sync_buffer = av_audio_fifo_alloc(avctx->sample_fmt,
+ s->output_channels, 32);
+ if (!s->sync_buffer) {
ret = AVERROR(ENOMEM);
goto fail;
}
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