diff options
Diffstat (limited to 'libavcodec/opusdec.c')
| -rw-r--r-- | libavcodec/opusdec.c | 71 |
1 files changed, 71 insertions, 0 deletions
diff --git a/libavcodec/opusdec.c b/libavcodec/opusdec.c index 5a7ba9dbb4..fb608729f1 100644 --- a/libavcodec/opusdec.c +++ b/libavcodec/opusdec.c @@ -40,7 +40,13 @@ #include "libavutil/channel_layout.h" #include "libavutil/opt.h" +#if CONFIG_SWRESAMPLE #include "libswresample/swresample.h" +#elif CONFIG_AVRESAMPLE +#include "libavresample/avresample.h" +#else +#error "swresample or avresample are required for Opus" +#endif #include "avcodec.h" #include "get_bits.h" @@ -86,9 +92,15 @@ static int opus_flush_resample(OpusStreamContext *s, int nb_samples) { int celt_size = av_audio_fifo_size(s->celt_delay); int ret, i; + +#if CONFIG_SWRESAMPLE ret = swr_convert(s->swr, (uint8_t**)s->out, nb_samples, NULL, 0); +#elif CONFIG_AVRESAMPLE + ret = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size, nb_samples, + NULL, 0, 0); +#endif if (ret < 0) return ret; else if (ret != nb_samples) { @@ -131,16 +143,26 @@ static int opus_init_resample(OpusStreamContext *s) const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay }; int ret; +#if CONFIG_SWRESAMPLE av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0); ret = swr_init(s->swr); +#elif CONFIG_AVRESAMPLE + av_opt_set_int(s->avr, "in_sample_rate", s->silk_samplerate, 0); + ret = avresample_open(s->avr); +#endif if (ret < 0) { av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n"); return ret; } +#if CONFIG_SWRESAMPLE ret = swr_convert(s->swr, NULL, 0, delayptr, silk_resample_delay[s->packet.bandwidth]); +#elif CONFIG_AVRESAMPLE + ret = avresample_convert(s->avr, NULL, 0, 0, delayptr, sizeof(delay), + silk_resample_delay[s->packet.bandwidth]); +#endif if (ret < 0) { av_log(s->avctx, AV_LOG_ERROR, "Error feeding initial silence to the resampler.\n"); @@ -184,7 +206,11 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size /* 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; @@ -198,9 +224,17 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size 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; @@ -343,6 +377,7 @@ static int opus_decode_subpacket(OpusStreamContext *s, 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; @@ -352,6 +387,17 @@ static int opus_decode_subpacket(OpusStreamContext *s, 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; @@ -375,7 +421,11 @@ static int opus_decode_subpacket(OpusStreamContext *s, 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; @@ -578,7 +628,11 @@ static av_cold void opus_decode_flush(AVCodecContext *ctx) 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])); @@ -602,7 +656,11 @@ static av_cold int opus_decode_close(AVCodecContext *avctx) s->out_dummy_allocated_size = 0; av_audio_fifo_free(s->celt_delay); +#if CONFIG_SWRESAMPLE swr_free(&s->swr); +#elif CONFIG_AVRESAMPLE + avresample_free(&s->avr); +#endif } av_freep(&c->streams); @@ -671,6 +729,7 @@ static av_cold int opus_decode_init(AVCodecContext *avctx) s->fdsp = c->fdsp; +#if CONFIG_SWRESAMPLE s->swr =swr_alloc(); if (!s->swr) goto fail; @@ -682,6 +741,18 @@ static av_cold int opus_decode_init(AVCodecContext *avctx) av_opt_set_int(s->swr, "out_channel_layout", layout, 0); av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0); av_opt_set_int(s->swr, "filter_size", 16, 0); +#elif CONFIG_AVRESAMPLE + s->avr = avresample_alloc_context(); + if (!s->avr) + goto fail; + + layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO; + av_opt_set_int(s->avr, "in_sample_fmt", avctx->sample_fmt, 0); + av_opt_set_int(s->avr, "out_sample_fmt", avctx->sample_fmt, 0); + av_opt_set_int(s->avr, "in_channel_layout", layout, 0); + av_opt_set_int(s->avr, "out_channel_layout", layout, 0); + av_opt_set_int(s->avr, "out_sample_rate", avctx->sample_rate, 0); +#endif ret = ff_silk_init(avctx, &s->silk, s->output_channels); if (ret < 0) |