Create media.dll / libmedia.so for the component build

media/audio/audio_util.h

 // Copyright (c) 2010 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef MEDIA_AUDIO_AUDIO_UTIL_H_
 #define MEDIA_AUDIO_AUDIO_UTIL_H_
 
 #include <vector>
 
 #include "base/basictypes.h"
+#include "media/base/media_api.h"
 
 namespace media {
 
 // For all audio functions 3 audio formats are supported:
 // 8 bits unsigned 0 to 255.
 // 16 bit signed (little endian).
 // 32 bit signed (little endian)
 
 // AdjustVolume() does a software volume adjustment of a sample buffer.
 // The samples are multiplied by the volume, which should range from
 // 0.0 (mute) to 1.0 (full volume).
 // Using software allows each audio and video to have its own volume without
 // affecting the master volume.
 // In the future the function may be used to adjust the sample format to
 // simplify hardware requirements and to support a wider variety of input
 // formats.
 // The buffer is modified in-place to avoid memory management, as this
 // function may be called in performance critical code.
-bool AdjustVolume(void* buf,
-                  size_t buflen,
-                  int channels,
-                  int bytes_per_sample,
-                  float volume);
+MEDIA_API bool AdjustVolume(void* buf,
+                            size_t buflen,
+                            int channels,
+                            int bytes_per_sample,
+                            float volume);
 
 // FoldChannels() does a software multichannel folding down to stereo.
 // Channel order is assumed to be 5.1 Dolby standard which is
 // front left, front right, center, surround left, surround right.
 // The subwoofer is ignored.
 // 6.1 adds a rear center speaker, and 7.1 has 2 rear speakers.  These
 // channels are rare and ignored.
 // After summing the channels, volume is adjusted and the samples are
 // clipped to the maximum value.
 // Volume should normally range from 0.0 (mute) to 1.0 (full volume), but
 // since clamping is performed a value of more than 1 is allowed to increase
 // volume.
 // The buffer is modified in-place to avoid memory management, as this
 // function may be called in performance critical code.
-bool FoldChannels(void* buf,
-                  size_t buflen,
-                  int channels,
-                  int bytes_per_sample,
-                  float volume);
+MEDIA_API bool FoldChannels(void* buf,
+                            size_t buflen,
+                            int channels,
+                            int bytes_per_sample,
+                            float volume);
 
 // DeinterleaveAudioChannel() takes interleaved audio buffer |source|
 // of the given |sample_fmt| and |number_of_channels| and extracts
 // |number_of_frames| data for the given |channel_index| and
 // puts it in the floating point |destination|.
 // It returns |true| on success, or |false| if the |sample_fmt| is
 // not recognized.
 bool DeinterleaveAudioChannel(void* source,
                               float* destination,
                               int channels,
                               int channel_index,
                               int bytes_per_sample,
                               size_t number_of_frames);
 
 // InterleaveFloatToInt16 scales, clips, and interleaves the planar
 // floating-point audio contained in |source| to the int16 |destination|.
 // The floating-point data is in a canonical range of -1.0 -> +1.0.
 // The size of the |source| vector determines the number of channels.
 // The |destination| buffer is assumed to be large enough to hold the
 // result. Thus it must be at least size: number_of_frames * source.size()
-void InterleaveFloatToInt16(const std::vector<float*>& source,
-                            int16* destination,
-                            size_t number_of_frames);
+MEDIA_API void InterleaveFloatToInt16(const std::vector<float*>& source,
+                                      int16* destination,
+                                      size_t number_of_frames);
 
 // Reorder PCM from AAC layout to Core Audio 5.1 layout.
 // TODO(fbarchard): Switch layout when ffmpeg is updated.
 template<class Format>
 void SwizzleCoreAudioLayout5_1(Format* b, uint32 filled) {

[scherkus (not reviewing), 2011/07/29 17:22:57]

hmm I'll move this function into a better place in a separate CL:
http://codereview.chromium.org/7533020

   static const int kNumSurroundChannels = 6;
   Format aac[kNumSurroundChannels];
   for (uint32 i = 0; i < filled; i += sizeof(aac), b += kNumSurroundChannels) {
     memcpy(aac, b, sizeof(aac));
     b[0] = aac[1];  // L
     b[1] = aac[2];  // R
     b[2] = aac[0];  // C
     b[3] = aac[5];  // LFE
     b[4] = aac[3];  // Ls
     b[5] = aac[4];  // Rs
   }
 }
 
 // Returns the default audio hardware sample-rate.
-double GetAudioHardwareSampleRate();
+MEDIA_API double GetAudioHardwareSampleRate();
 
 }  // namespace media
 
 #endif  // MEDIA_AUDIO_AUDIO_UTIL_H_