| Index: webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
|
| index 5f1b7626c99be394a67d9e8539a8d8bbedbcdaf0..07d767e778b88926e7feb7b76814843ed343f508 100644
|
| --- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
|
| +++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
|
| @@ -27,7 +27,7 @@ class AudioEncoderG722 final : public AudioEncoder {
|
|
|
| int payload_type = 9;
|
| int frame_size_ms = 20;
|
| - int num_channels = 1;
|
| + size_t num_channels = 1;
|
| };
|
|
|
| explicit AudioEncoderG722(const Config& config);
|
| @@ -36,7 +36,7 @@ class AudioEncoderG722 final : public AudioEncoder {
|
|
|
| size_t MaxEncodedBytes() const override;
|
| int SampleRateHz() const override;
|
| - int NumChannels() const override;
|
| + size_t NumChannels() const override;
|
| int RtpTimestampRateHz() const override;
|
| size_t Num10MsFramesInNextPacket() const override;
|
| size_t Max10MsFramesInAPacket() const override;
|
| @@ -59,7 +59,7 @@ class AudioEncoderG722 final : public AudioEncoder {
|
|
|
| size_t SamplesPerChannel() const;
|
|
|
| - const int num_channels_;
|
| + const size_t num_channels_;
|
| const int payload_type_;
|
| const size_t num_10ms_frames_per_packet_;
|
| size_t num_10ms_frames_buffered_;
|
|
|