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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@lynne.ee> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This commit is contained in:
parent
794fb18369
commit
f03eade869
@ -101,6 +101,15 @@ typedef struct OpusStreamContext {
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AVCodecContext *avctx;
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int output_channels;
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/* number of decoded samples for this stream */
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int decoded_samples;
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/* current output buffers for this stream */
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float *out[2];
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int out_size;
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/* Buffer with samples from this stream for synchronizing
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* the streams when they have different resampling delays */
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AVAudioFifo *sync_buffer;
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OpusRangeCoder rc;
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OpusRangeCoder redundancy_rc;
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SilkContext *silk;
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@ -115,9 +124,9 @@ typedef struct OpusStreamContext {
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DECLARE_ALIGNED(32, float, redundancy_buf)[2][960];
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float *redundancy_output[2];
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/* data buffers for the final output data */
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float *out[2];
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int out_size;
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/* buffers for the next samples to be decoded */
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float *cur_out[2];
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int remaining_out_size;
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float *out_dummy;
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int out_dummy_allocated_size;
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@ -154,15 +163,6 @@ typedef struct OpusContext {
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OpusStreamContext *streams;
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int apply_phase_inv;
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/* current output buffers for each streams */
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float **out;
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int *out_size;
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/* Buffers for synchronizing the streams when they have different
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* resampling delays */
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AVAudioFifo **sync_buffers;
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/* number of decoded samples for each stream */
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int *decoded_samples;
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int nb_streams;
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int nb_stereo_streams;
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@ -87,7 +87,7 @@ static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
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int celt_size = av_audio_fifo_size(s->celt_delay);
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int ret, i;
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ret = swr_convert(s->swr,
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(uint8_t**)s->out, nb_samples,
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(uint8_t**)s->cur_out, nb_samples,
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NULL, 0);
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if (ret < 0)
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return ret;
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@ -104,7 +104,7 @@ static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
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}
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av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
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for (i = 0; i < s->output_channels; i++) {
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s->fdsp->vector_fmac_scalar(s->out[i],
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s->fdsp->vector_fmac_scalar(s->cur_out[i],
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s->celt_output[i], 1.0,
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nb_samples);
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}
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@ -112,15 +112,15 @@ static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
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if (s->redundancy_idx) {
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for (i = 0; i < s->output_channels; i++)
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opus_fade(s->out[i], s->out[i],
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opus_fade(s->cur_out[i], s->cur_out[i],
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s->redundancy_output[i] + 120 + s->redundancy_idx,
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ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
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s->redundancy_idx = 0;
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}
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s->out[0] += nb_samples;
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s->out[1] += nb_samples;
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s->out_size -= nb_samples * sizeof(float);
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s->cur_out[0] += nb_samples;
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s->cur_out[1] += nb_samples;
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s->remaining_out_size -= nb_samples * sizeof(float);
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return 0;
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}
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@ -199,7 +199,7 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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return samples;
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}
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samples = swr_convert(s->swr,
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(uint8_t**)s->out, s->packet.frame_duration,
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(uint8_t**)s->cur_out, s->packet.frame_duration,
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(const uint8_t**)s->silk_output, samples);
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if (samples < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
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@ -240,7 +240,7 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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/* decode the CELT frame */
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if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
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float *out_tmp[2] = { s->out[0], s->out[1] };
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float *out_tmp[2] = { s->cur_out[0], s->cur_out[1] };
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float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
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out_tmp : s->celt_output;
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int celt_output_samples = samples;
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@ -295,7 +295,7 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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if (s->redundancy_idx) {
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for (i = 0; i < s->output_channels; i++)
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opus_fade(s->out[i], s->out[i],
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opus_fade(s->cur_out[i], s->cur_out[i],
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s->redundancy_output[i] + 120 + s->redundancy_idx,
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ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
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s->redundancy_idx = 0;
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@ -308,8 +308,8 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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return ret;
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for (i = 0; i < s->output_channels; i++) {
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opus_fade(s->out[i] + samples - 120 + delayed_samples,
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s->out[i] + samples - 120 + delayed_samples,
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opus_fade(s->cur_out[i] + samples - 120 + delayed_samples,
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s->cur_out[i] + samples - 120 + delayed_samples,
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s->redundancy_output[i] + 120,
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ff_celt_window2, 120 - delayed_samples);
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if (delayed_samples)
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@ -317,10 +317,10 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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}
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} else {
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for (i = 0; i < s->output_channels; i++) {
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memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
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opus_fade(s->out[i] + 120 + delayed_samples,
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memcpy(s->cur_out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
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opus_fade(s->cur_out[i] + 120 + delayed_samples,
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s->redundancy_output[i] + 120,
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s->out[i] + 120 + delayed_samples,
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s->cur_out[i] + 120 + delayed_samples,
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ff_celt_window2, 120);
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}
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}
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@ -331,16 +331,15 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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static int opus_decode_subpacket(OpusStreamContext *s,
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const uint8_t *buf, int buf_size,
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float **out, int out_size,
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int nb_samples)
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{
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int output_samples = 0;
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int flush_needed = 0;
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int i, j, ret;
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s->out[0] = out[0];
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s->out[1] = out[1];
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s->out_size = out_size;
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s->cur_out[0] = s->out[0];
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s->cur_out[1] = s->out[1];
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s->remaining_out_size = s->out_size;
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/* check if we need to flush the resampler */
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if (swr_is_initialized(s->swr)) {
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@ -357,15 +356,16 @@ static int opus_decode_subpacket(OpusStreamContext *s,
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return 0;
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/* use dummy output buffers if the channel is not mapped to anything */
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if (!s->out[0] ||
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(s->output_channels == 2 && !s->out[1])) {
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av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);
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if (!s->cur_out[0] ||
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(s->output_channels == 2 && !s->cur_out[1])) {
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av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size,
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s->remaining_out_size);
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if (!s->out_dummy)
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return AVERROR(ENOMEM);
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if (!s->out[0])
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s->out[0] = s->out_dummy;
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if (!s->out[1])
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s->out[1] = s->out_dummy;
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if (!s->cur_out[0])
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s->cur_out[0] = s->out_dummy;
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if (!s->cur_out[1])
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s->cur_out[1] = s->out_dummy;
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}
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/* flush the resampler if necessary */
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@ -394,19 +394,19 @@ static int opus_decode_subpacket(OpusStreamContext *s,
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return samples;
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for (j = 0; j < s->output_channels; j++)
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memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
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memset(s->cur_out[j], 0, s->packet.frame_duration * sizeof(float));
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samples = s->packet.frame_duration;
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}
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output_samples += samples;
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for (j = 0; j < s->output_channels; j++)
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s->out[j] += samples;
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s->out_size -= samples * sizeof(float);
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s->cur_out[j] += samples;
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s->remaining_out_size -= samples * sizeof(float);
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}
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finish:
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s->out[0] = s->out[1] = NULL;
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s->out_size = 0;
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s->cur_out[0] = s->cur_out[1] = NULL;
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s->remaining_out_size = 0;
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return output_samples;
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}
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@ -429,7 +429,7 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
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s->out[0] =
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s->out[1] = NULL;
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delayed_samples = FFMAX(delayed_samples,
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s->delayed_samples + av_audio_fifo_size(c->sync_buffers[i]));
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s->delayed_samples + av_audio_fifo_size(s->sync_buffer));
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}
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/* decode the header of the first sub-packet to find out the sample count */
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@ -458,17 +458,17 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
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return ret;
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frame->nb_samples = 0;
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memset(c->out, 0, c->nb_streams * 2 * sizeof(*c->out));
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for (i = 0; i < avctx->channels; i++) {
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ChannelMap *map = &c->channel_maps[i];
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if (!map->copy)
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c->out[2 * map->stream_idx + map->channel_idx] = (float*)frame->extended_data[i];
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c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
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}
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/* read the data from the sync buffers */
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for (i = 0; i < c->nb_streams; i++) {
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float **out = c->out + 2 * i;
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int sync_size = av_audio_fifo_size(c->sync_buffers[i]);
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OpusStreamContext *s = &c->streams[i];
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float **out = s->out;
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int sync_size = av_audio_fifo_size(s->sync_buffer);
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float sync_dummy[32];
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int out_dummy = (!out[0]) | ((!out[1]) << 1);
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@ -480,7 +480,7 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
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if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy))
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return AVERROR_BUG;
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ret = av_audio_fifo_read(c->sync_buffers[i], (void**)out, sync_size);
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ret = av_audio_fifo_read(s->sync_buffer, (void**)out, sync_size);
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if (ret < 0)
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return ret;
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@ -493,7 +493,7 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
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else
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out[1] += ret;
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c->out_size[i] = frame->linesize[0] - ret * sizeof(float);
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s->out_size = frame->linesize[0] - ret * sizeof(float);
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}
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/* decode each sub-packet */
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@ -516,10 +516,10 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
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}
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ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,
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c->out + 2 * i, c->out_size[i], coded_samples);
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coded_samples);
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if (ret < 0)
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return ret;
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c->decoded_samples[i] = ret;
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s->decoded_samples = ret;
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decoded_samples = FFMIN(decoded_samples, ret);
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buf += s->packet.packet_size;
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@ -528,13 +528,14 @@ static int opus_decode_packet(AVCodecContext *avctx, void *data,
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/* buffer the extra samples */
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for (i = 0; i < c->nb_streams; i++) {
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int buffer_samples = c->decoded_samples[i] - decoded_samples;
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OpusStreamContext *s = &c->streams[i];
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int buffer_samples = s->decoded_samples - decoded_samples;
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if (buffer_samples) {
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float *buf[2] = { c->out[2 * i + 0] ? c->out[2 * i + 0] : (float*)frame->extended_data[0],
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c->out[2 * i + 1] ? c->out[2 * i + 1] : (float*)frame->extended_data[0] };
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float *buf[2] = { s->out[0] ? s->out[0] : (float*)frame->extended_data[0],
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s->out[1] ? s->out[1] : (float*)frame->extended_data[0] };
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buf[0] += decoded_samples;
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buf[1] += decoded_samples;
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ret = av_audio_fifo_write(c->sync_buffers[i], (void**)buf, buffer_samples);
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ret = av_audio_fifo_write(s->sync_buffer, (void**)buf, buffer_samples);
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if (ret < 0)
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return ret;
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}
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@ -579,7 +580,7 @@ static av_cold void opus_decode_flush(AVCodecContext *ctx)
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av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
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swr_close(s->swr);
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av_audio_fifo_drain(c->sync_buffers[i], av_audio_fifo_size(c->sync_buffers[i]));
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av_audio_fifo_drain(s->sync_buffer, av_audio_fifo_size(s->sync_buffer));
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ff_silk_flush(s->silk);
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ff_celt_flush(s->celt);
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@ -600,21 +601,13 @@ static av_cold int opus_decode_close(AVCodecContext *avctx)
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av_freep(&s->out_dummy);
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s->out_dummy_allocated_size = 0;
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av_audio_fifo_free(s->sync_buffer);
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av_audio_fifo_free(s->celt_delay);
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swr_free(&s->swr);
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}
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av_freep(&c->streams);
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if (c->sync_buffers) {
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for (i = 0; i < c->nb_streams; i++)
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av_audio_fifo_free(c->sync_buffers[i]);
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}
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av_freep(&c->sync_buffers);
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av_freep(&c->decoded_samples);
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av_freep(&c->out);
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av_freep(&c->out_size);
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c->nb_streams = 0;
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av_freep(&c->channel_maps);
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@ -644,11 +637,7 @@ static av_cold int opus_decode_init(AVCodecContext *avctx)
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/* allocate and init each independent decoder */
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c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
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c->out = av_mallocz_array(c->nb_streams, 2 * sizeof(*c->out));
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c->out_size = av_mallocz_array(c->nb_streams, sizeof(*c->out_size));
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c->sync_buffers = av_mallocz_array(c->nb_streams, sizeof(*c->sync_buffers));
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c->decoded_samples = av_mallocz_array(c->nb_streams, sizeof(*c->decoded_samples));
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if (!c->streams || !c->sync_buffers || !c->decoded_samples || !c->out || !c->out_size) {
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if (!c->streams) {
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c->nb_streams = 0;
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ret = AVERROR(ENOMEM);
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goto fail;
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@ -699,9 +688,9 @@ static av_cold int opus_decode_init(AVCodecContext *avctx)
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goto fail;
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}
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c->sync_buffers[i] = av_audio_fifo_alloc(avctx->sample_fmt,
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s->output_channels, 32);
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if (!c->sync_buffers[i]) {
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s->sync_buffer = av_audio_fifo_alloc(avctx->sample_fmt,
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s->output_channels, 32);
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if (!s->sync_buffer) {
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ret = AVERROR(ENOMEM);
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goto fail;
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}
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