@@ -25,12 +25,7 @@
#include <stdint.h>
-#include "libavutil/audio_fifo.h"
#include "libavutil/float_dsp.h"
-#include "libavutil/frame.h"
-#include "libavutil/mem_internal.h"
-
-#include "libswresample/swresample.h"
#include "avcodec.h"
#include "opus_rc.h"
@@ -98,51 +93,6 @@ typedef struct OpusPacket {
enum OpusBandwidth bandwidth; /**< bandwidth */
} OpusPacket;
-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;
- CeltFrame *celt;
- AVFloatDSPContext *fdsp;
-
- float silk_buf[2][960];
- float *silk_output[2];
- DECLARE_ALIGNED(32, float, celt_buf)[2][960];
- float *celt_output[2];
-
- DECLARE_ALIGNED(32, float, redundancy_buf)[2][960];
- float *redundancy_output[2];
-
- /* buffers for the next samples to be decoded */
- float *cur_out[2];
- int remaining_out_size;
-
- float *out_dummy;
- int out_dummy_allocated_size;
-
- SwrContext *swr;
- AVAudioFifo *celt_delay;
- int silk_samplerate;
- /* number of samples we still want to get from the resampler */
- int delayed_samples;
-
- OpusPacket packet;
-
- int redundancy_idx;
-} OpusStreamContext;
-
// a mapping between an opus stream and an output channel
typedef struct ChannelMap {
int stream_idx;
@@ -161,7 +111,7 @@ typedef struct ChannelMap {
typedef struct OpusContext {
AVClass *av_class;
- OpusStreamContext *streams;
+ struct OpusStreamContext *streams;
int apply_phase_inv;
int nb_streams;
@@ -38,6 +38,8 @@
#include "libavutil/attributes.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/channel_layout.h"
+#include "libavutil/frame.h"
+#include "libavutil/mem_internal.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
@@ -63,6 +65,51 @@ static const int silk_resample_delay[] = {
4, 8, 11, 11, 11
};
+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;
+ CeltFrame *celt;
+ AVFloatDSPContext *fdsp;
+
+ float silk_buf[2][960];
+ float *silk_output[2];
+ DECLARE_ALIGNED(32, float, celt_buf)[2][960];
+ float *celt_output[2];
+
+ DECLARE_ALIGNED(32, float, redundancy_buf)[2][960];
+ float *redundancy_output[2];
+
+ /* buffers for the next samples to be decoded */
+ float *cur_out[2];
+ int remaining_out_size;
+
+ float *out_dummy;
+ int out_dummy_allocated_size;
+
+ SwrContext *swr;
+ AVAudioFifo *celt_delay;
+ int silk_samplerate;
+ /* number of samples we still want to get from the resampler */
+ int delayed_samples;
+
+ OpusPacket packet;
+
+ int redundancy_idx;
+} OpusStreamContext;
+
static int get_silk_samplerate(int config)
{
if (config < 4)
Namely opusdec.c. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com> --- libavcodec/opus.h | 52 +------------------------------------------- libavcodec/opusdec.c | 47 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 48 insertions(+), 51 deletions(-)