webrtc/modules/audio_coding/codecs/audio_encoder.h - Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce

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Unified Diff: webrtc/modules/audio_coding/codecs/audio_encoder.h

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master

Patch Set: Resync Created 5 years, 4 months ago

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Index: webrtc/modules/audio_coding/codecs/audio_encoder.h

diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h

index fe6fd87dfd791b53247bd26cc59fbd73a08f3187..0a40316b713025028fd67be64f9d75a08d124168 100644

--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h

+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h

@@ -90,11 +90,11 @@ class AudioEncoder {

// the encoder may vary the number of 10 ms frames from packet to packet, but

// it must decide the length of the next packet no later than when outputting

// the preceding packet.

- virtual int Num10MsFramesInNextPacket() const = 0;

+ virtual size_t Num10MsFramesInNextPacket() const = 0;

// Returns the maximum value that can be returned by

// Num10MsFramesInNextPacket().

- virtual int Max10MsFramesInAPacket() const = 0;

+ virtual size_t Max10MsFramesInAPacket() const = 0;

// Returns the current target bitrate in bits/s. The value -1 means that the

// codec adapts the target automatically, and a current target cannot be