Fix trivial lint errors in FileRecorder and FilePlayer
Mostly, it's about replacing mutable reference arguments with pointer
arguments, and replacing C style casts with C++ style casts.
Review-Url: https://codereview.webrtc.org/2056653002
Cr-Commit-Position: refs/heads/master@{#13849}
diff --git a/webrtc/modules/utility/include/file_player.h b/webrtc/modules/utility/include/file_player.h
index 508c211..296b9ea 100644
--- a/webrtc/modules/utility/include/file_player.h
+++ b/webrtc/modules/utility/include/file_player.h
@@ -44,7 +44,7 @@
// will be set to the number of samples read (not the number of samples per
// channel).
virtual int Get10msAudioFromFile(int16_t* outBuffer,
- size_t& lengthInSamples,
+ size_t* lengthInSamples,
int frequencyInHz) = 0;
// Register callback for receiving file playing notifications.
@@ -61,7 +61,7 @@
const CodecInst* codecInst) = 0;
// Note: codecInst is used for pre-encoded files.
- virtual int32_t StartPlayingFile(InStream& sourceStream,
+ virtual int32_t StartPlayingFile(InStream* sourceStream,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
@@ -72,15 +72,37 @@
virtual bool IsPlayingFile() const = 0;
- virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0;
+ virtual int32_t GetPlayoutPosition(uint32_t* durationMs) = 0;
// Set audioCodec to the currently used audio codec.
- virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0;
+ virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0;
virtual int32_t Frequency() const = 0;
// Note: scaleFactor is in the range [0.0 - 2.0]
virtual int32_t SetAudioScaling(float scaleFactor) = 0;
+
+ // Deprecated functions. Use the functions above with the same name instead.
+ int Get10msAudioFromFile(int16_t* outBuffer,
+ size_t& lengthInSamples,
+ int frequencyInHz) {
+ return Get10msAudioFromFile(outBuffer, &lengthInSamples, frequencyInHz);
+ }
+ int32_t StartPlayingFile(InStream& sourceStream,
+ uint32_t startPosition,
+ float volumeScaling,
+ uint32_t notification,
+ uint32_t stopPosition,
+ const CodecInst* codecInst) {
+ return StartPlayingFile(&sourceStream, startPosition, volumeScaling,
+ notification, stopPosition, codecInst);
+ }
+ int32_t GetPlayoutPosition(uint32_t& durationMs) {
+ return GetPlayoutPosition(&durationMs);
+ }
+ int32_t AudioCodec(CodecInst& audioCodec) const {
+ return AudioCodec(&audioCodec);
+ }
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_