add support for converting between all audio sample formats

diff --git a/src/framework/mlt_types.h b/src/framework/mlt_types.h
index f30e94db8f3da75bfd392fc4108ccfd36a2b7c26..99fd4a3ed1ae64a75c1676a473f15ec2290ee035 100644 (file)
--- a/src/framework/mlt_types.h
+++ b/src/framework/mlt_types.h
@@ -2,7 +2,7 @@
  * \file mlt_types.h
  * \brief Provides forward definitions of all public types
  *
- * Copyright (C) 2003-2009 Ushodaya Enterprises Limited
+ * Copyright (C) 2003-2012 Ushodaya Enterprises Limited
  * \author Charles Yates <charles.yates@pandora.be>
  *
  * This library is free software; you can redistribute it and/or
@@ -57,8 +57,8 @@ typedef enum
        mlt_audio_s16 = 1, /**< signed 16-bit interleaved PCM */
        mlt_audio_s32,     /**< signed 32-bit non-interleaved PCM */
        mlt_audio_float,   /**< 32-bit non-interleaved floating point */
-       mlt_audio_s32le,   /**< signed 32-bit interleaved PCM, may only used by producers */
-       mlt_audio_f32le    /**< 32-bit interleaved floating point, may only be used by producers */
+       mlt_audio_s32le,   /**< signed 32-bit interleaved PCM */
+       mlt_audio_f32le    /**< 32-bit interleaved floating point */
 }
 mlt_audio_format;
 

diff --git a/src/modules/core/filter_audioconvert.c b/src/modules/core/filter_audioconvert.c
index 6e930d7e0136ed6d8ae8d796de061765336ddd94..d1f7cb33bb416cfc4dbe5d903230b9ca366dca69 100644 (file)
--- a/src/modules/core/filter_audioconvert.c
+++ b/src/modules/core/filter_audioconvert.c
@@ -1,6 +1,6 @@
 /*
  * filter_audioconvert.c -- convert from one audio format to another
- * Copyright (C) 2009 Ushodaya Enterprises Limited
+ * Copyright (C) 2009-2012 Ushodaya Enterprises Limited
  * Author: Dan Dennedy <dan@dennedy.org>
  *
  * This library is free software; you can redistribute it and/or
@@ -81,6 +81,34 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                error = 0;
                                break;
                        }
+                       case mlt_audio_s32le:
+                       {
+                               int32_t *buffer = mlt_pool_alloc( size );
+                               int32_t *p = buffer;
+                               int16_t *q = (int16_t*) *audio;
+                               int i = samples * channels + 1;
+                               while ( --i )
+                                       *p++ = (int32_t) *q++ << 16;
+                               *audio = buffer;
+                               error = 0;
+                               break;
+                       }
+                       case mlt_audio_f32le:
+                       {
+                               float *buffer = mlt_pool_alloc( size );
+                               float *p = buffer;
+                               int16_t *q = (int16_t*) *audio;
+                               int i = samples * channels + 1;
+                               while ( --i )
+                               {
+                                       float f = (float)( *q++ ) / 32768.0;
+                                       f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
+                                       *p++ = f;
+                               }
+                               *audio = buffer;
+                               error = 0;
+                               break;
+                       }
                        default:
                                break;
                        }
@@ -113,6 +141,36 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                error = 0;
                                break;
                        }
+                       case mlt_audio_s32le:
+                       {
+                               int32_t *buffer = mlt_pool_alloc( size );
+                               int32_t *p = buffer;
+                               int32_t *q = (int32_t*) *audio;
+                               int s, c;
+                               for ( s = 0; s < samples; s++ )
+                                       for ( c = 0; c < channels; c++ )
+                                               *p++ = *( q + c * samples + s );
+                               *audio = buffer;
+                               error = 0;
+                               break;
+                       }
+                       case mlt_audio_f32le:
+                       {
+                               float *buffer = mlt_pool_alloc( size );
+                               float *p = buffer;
+                               int32_t *q = (int32_t*) *audio;
+                               int s, c;
+                               for ( s = 0; s < samples; s++ )
+                                       for ( c = 0; c < channels; c++ )
+                                       {
+                                               float f = (float)( *( q + c * samples + s ) ) / 2147483648.0;
+                                               f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
+                                               *p++ = f;
+                                       }
+                               *audio = buffer;
+                               error = 0;
+                               break;
+                       }
                        default:
                                break;
                        }
@@ -153,6 +211,36 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                error = 0;
                                break;
                        }
+                       case mlt_audio_s32le:
+                       {
+                               int32_t *buffer = mlt_pool_alloc( size );
+                               int32_t *p = buffer;
+                               float *q = (float*) *audio;
+                               int s, c;
+                               for ( s = 0; s < samples; s++ )
+                                       for ( c = 0; c < channels; c++ )
+                                       {
+                                               float f = *( q + c * samples + s );
+                                               f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
+                                               *p++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
+                                       }
+                               *audio = buffer;
+                               error = 0;
+                               break;
+                       }
+                       case mlt_audio_f32le:
+                       {
+                               float *buffer = mlt_pool_alloc( size );
+                               float *p = buffer;
+                               float *q = (float*) *audio;
+                               int s, c;
+                               for ( s = 0; s < samples; s++ )
+                                       for ( c = 0; c < channels; c++ )
+                                               *p++ = *( q + c * samples + s );
+                               *audio = buffer;
+                               error = 0;
+                               break;
+                       }
                        default:
                                break;
                        }
@@ -210,6 +298,18 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                error = 0;
                                break;
                        }
+                       case mlt_audio_f32le:
+                       {
+                               float *buffer = mlt_pool_alloc( size );
+                               float *p = buffer;
+                               int32_t *q = (int32_t*) *audio;
+                               int i = samples * channels + 1;
+                               while ( --i )
+                                       *p++ = (float)( *q++ ) / 2147483648.0;
+                               *audio = buffer;
+                               error = 0;
+                               break;
+                       }
                        default:
                                break;
                        }
@@ -273,6 +373,22 @@ static int convert_audio( mlt_frame frame, void **audio, mlt_audio_format *forma
                                error = 0;
                                break;
                        }
+                       case mlt_audio_s32le:
+                       {
+                               int32_t *buffer = mlt_pool_alloc( size );
+                               int32_t *p = buffer;
+                               float *q = (float*) *audio;
+                               int i = samples * channels + 1;
+                               while ( --i )
+                               {
+                                       float f = *q++;
+                                       f = f > 1.0 ? 1.0 : f < -1.0 ? -1.0 : f;
+                                       *p++ = ( f > 0 ? 2147483647LL : 2147483648LL ) * f;
+                               }
+                               *audio = buffer;
+                               error = 0;
+                               break;
+                       }
                        default:
                                break;
                        }