From f03eade8690d9914fce574afb1795f16c036bb6b Mon Sep 17 00:00:00 2001 From: Andreas Rheinhardt Date: Mon, 28 Dec 2020 17:46:44 +0100 Subject: [PATCH] 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 Signed-off-by: Andreas Rheinhardt --- 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 63ecd0aff7b..fa63353e9b3 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 462d70b3bf9..b09a542c860 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; } -- 2.39.2