Chromium Code Reviews| Index: webrtc/common_video/include/video_bitrate_allocator.h |
| diff --git a/webrtc/modules/audio_coding/codecs/audio_format_conversion.h b/webrtc/common_video/include/video_bitrate_allocator.h |
| similarity index 50% |
| copy from webrtc/modules/audio_coding/codecs/audio_format_conversion.h |
| copy to webrtc/common_video/include/video_bitrate_allocator.h |
| index ff71282f7e2e42c8bb27a4cd23316fab15e41ff9..f8157a2afd788b6f3eea163eb39e1d036c880015 100644 |
| --- a/webrtc/modules/audio_coding/codecs/audio_format_conversion.h |
| +++ b/webrtc/common_video/include/video_bitrate_allocator.h |
| @@ -8,16 +8,23 @@ |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| -#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_ |
| -#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_ |
| +#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_BITRATE_ALLOCATOR_H_ |
| +#define WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_BITRATE_ALLOCATOR_H_ |
| #include "webrtc/common_types.h" |
| -#include "webrtc/modules/audio_coding/codecs/audio_format.h" |
| namespace webrtc { |
| -SdpAudioFormat CodecInstToSdp(const CodecInst& codec_inst); |
| +class VideoBitrateAllocator { |
| + public: |
| + VideoBitrateAllocator() {} |
| + virtual ~VideoBitrateAllocator() {} |
| + |
| + virtual BitrateAllocation GetAllocation(uint32_t total_bitrate, |
| + uint32_t framerate) = 0; |
|
stefan-webrtc
2016/11/02 10:26:34
Take the opportunity to align the framerate type h
sprang_webrtc
2016/11/02 13:28:32
Done.
|
| + virtual uint32_t GetPreferedBitrate(int frame_rate) = 0; |
|
stefan-webrtc
2016/11/02 10:26:34
Preferred
sprang_webrtc
2016/11/02 13:28:32
Done.
|
| +}; |
| } // namespace webrtc |
| -#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_ |
| +#endif // WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_BITRATE_ALLOCATOR_H_ |