compress videos bigger than 10 MB before sending

1 files changed, 126 insertions, 0 deletions

diff --git a/src/main/jni/opus/silk/control_audio_bandwidth.c b/src/main/jni/opus/silk/control_audio_bandwidth.c
new file mode 100644
index 000000000..4f9bc5cbd
--- /dev/null
+++ b/src/main/jni/opus/silk/control_audio_bandwidth.c
@@ -0,0 +1,126 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "main.h"
+#include "tuning_parameters.h"
+
+/* Control internal sampling rate */
+opus_int silk_control_audio_bandwidth(
+    silk_encoder_state          *psEncC,                        /* I/O  Pointer to Silk encoder state               */
+    silk_EncControlStruct       *encControl                     /* I    Control structure                           */
+)
+{
+    opus_int   fs_kHz;
+    opus_int32 fs_Hz;
+
+    fs_kHz = psEncC->fs_kHz;
+    fs_Hz = silk_SMULBB( fs_kHz, 1000 );
+    if( fs_Hz == 0 ) {
+        /* Encoder has just been initialized */
+        fs_Hz  = silk_min( psEncC->desiredInternal_fs_Hz, psEncC->API_fs_Hz );
+        fs_kHz = silk_DIV32_16( fs_Hz, 1000 );
+    } else if( fs_Hz > psEncC->API_fs_Hz || fs_Hz > psEncC->maxInternal_fs_Hz || fs_Hz < psEncC->minInternal_fs_Hz ) {
+        /* Make sure internal rate is not higher than external rate or maximum allowed, or lower than minimum allowed */
+        fs_Hz  = psEncC->API_fs_Hz;
+        fs_Hz  = silk_min( fs_Hz, psEncC->maxInternal_fs_Hz );
+        fs_Hz  = silk_max( fs_Hz, psEncC->minInternal_fs_Hz );
+        fs_kHz = silk_DIV32_16( fs_Hz, 1000 );
+    } else {
+        /* State machine for the internal sampling rate switching */
+        if( psEncC->sLP.transition_frame_no >= TRANSITION_FRAMES ) {
+            /* Stop transition phase */
+            psEncC->sLP.mode = 0;
+        }
+        if( psEncC->allow_bandwidth_switch || encControl->opusCanSwitch ) {
+            /* Check if we should switch down */
+            if( silk_SMULBB( psEncC->fs_kHz, 1000 ) > psEncC->desiredInternal_fs_Hz )
+            {
+                /* Switch down */
+                if( psEncC->sLP.mode == 0 ) {
+                    /* New transition */
+                    psEncC->sLP.transition_frame_no = TRANSITION_FRAMES;
+
+                    /* Reset transition filter state */
+                    silk_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) );
+                }
+                if( encControl->opusCanSwitch ) {
+                    /* Stop transition phase */
+                    psEncC->sLP.mode = 0;
+
+                    /* Switch to a lower sample frequency */
+                    fs_kHz = psEncC->fs_kHz == 16 ? 12 : 8;
+                } else {
+                   if( psEncC->sLP.transition_frame_no <= 0 ) {
+                       encControl->switchReady = 1;
+                       /* Make room for redundancy */
+                       encControl->maxBits -= encControl->maxBits * 5 / ( encControl->payloadSize_ms + 5 );
+                   } else {
+                       /* Direction: down (at double speed) */
+                       psEncC->sLP.mode = -2;
+                   }
+                }
+            }
+            else
+            /* Check if we should switch up */
+            if( silk_SMULBB( psEncC->fs_kHz, 1000 ) < psEncC->desiredInternal_fs_Hz )
+            {
+                /* Switch up */
+                if( encControl->opusCanSwitch ) {
+                    /* Switch to a higher sample frequency */
+                    fs_kHz = psEncC->fs_kHz == 8 ? 12 : 16;
+
+                    /* New transition */
+                    psEncC->sLP.transition_frame_no = 0;
+
+                    /* Reset transition filter state */
+                    silk_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) );
+
+                    /* Direction: up */
+                    psEncC->sLP.mode = 1;
+                } else {
+                   if( psEncC->sLP.mode == 0 ) {
+                       encControl->switchReady = 1;
+                       /* Make room for redundancy */
+                       encControl->maxBits -= encControl->maxBits * 5 / ( encControl->payloadSize_ms + 5 );
+                   } else {
+                       /* Direction: up */
+                       psEncC->sLP.mode = 1;
+                   }
+                }
+            } else {
+               if (psEncC->sLP.mode<0)
+                  psEncC->sLP.mode = 1;
+            }
+        }
+    }
+
+    return fs_kHz;
+}