[FFmpeg-cvslog] avcodec/opus: Move OpusStreamContext to its only user
Andreas Rheinhardt
git at videolan.org
Wed Oct 5 03:32:07 EEST 2022
ffmpeg | branch: master | Andreas Rheinhardt <andreas.rheinhardt at outlook.com> | Mon Oct 3 20:17:40 2022 +0200| [6658028482cf3bda2a05ae7a9bf183c81f75d4bd] | committer: Andreas Rheinhardt
avcodec/opus: Move OpusStreamContext to its only user
Namely opusdec.c.
Reviewed-by: Lynne <dev at lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt at outlook.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=6658028482cf3bda2a05ae7a9bf183c81f75d4bd
---
libavcodec/opus.h | 52 +---------------------------------------------------
libavcodec/opusdec.c | 47 +++++++++++++++++++++++++++++++++++++++++++++++
2 files changed, 48 insertions(+), 51 deletions(-)
diff --git a/libavcodec/opus.h b/libavcodec/opus.h
index b73949a811..264128f09e 100644
--- a/libavcodec/opus.h
+++ b/libavcodec/opus.h
@@ -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;
diff --git a/libavcodec/opusdec.c b/libavcodec/opusdec.c
index d255486d06..87a86b6b47 100644
--- a/libavcodec/opusdec.c
+++ b/libavcodec/opusdec.c
@@ -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)
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