[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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