Chromium Code Reviews| Index: webrtc/modules/audio_coding/neteq/neteq_impl.h |
| diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h |
| index 55ba067221fcab27275fb77f9147d0a6877e239f..fec9233e9a40b5679392b63e76809e4e70a27aad 100644 |
| --- a/webrtc/modules/audio_coding/neteq/neteq_impl.h |
| +++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h |
| @@ -106,7 +106,7 @@ class NetEqImpl : public webrtc::NetEq { |
| // Returns kOK on success, or kFail in case of an error. |
| int GetAudio(size_t max_length, |
| int16_t* output_audio, |
| - int* samples_per_channel, |
| + size_t* samples_per_channel, |
| int* num_channels, |
| NetEqOutputType* type) override; |
| @@ -225,7 +225,7 @@ class NetEqImpl : public webrtc::NetEq { |
| // Returns 0 on success, otherwise an error code. |
| int GetAudioInternal(size_t max_length, |
| int16_t* output, |
| - int* samples_per_channel, |
| + size_t* samples_per_channel, |
| int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); |
| // Provides a decision to the GetAudioInternal method. The decision what to |
| @@ -318,7 +318,7 @@ class NetEqImpl : public webrtc::NetEq { |
| // |required_samples| samples. The packets are inserted into |packet_list|. |
| // Returns the number of samples that the packets in the list will produce, or |
| // -1 in case of an error. |
| - int ExtractPackets(int required_samples, PacketList* packet_list) |
| + int ExtractPackets(size_t required_samples, PacketList* packet_list) |
| EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); |
| // Resets various variables and objects to new values based on the sample rate |
| @@ -374,9 +374,9 @@ class NetEqImpl : public webrtc::NetEq { |
| Rtcp rtcp_ GUARDED_BY(crit_sect_); |
| StatisticsCalculator stats_ GUARDED_BY(crit_sect_); |
| int fs_hz_ GUARDED_BY(crit_sect_); |
| - int fs_mult_ GUARDED_BY(crit_sect_); |
| - int output_size_samples_ GUARDED_BY(crit_sect_); |
| - int decoder_frame_length_ GUARDED_BY(crit_sect_); |
| + size_t fs_mult_ GUARDED_BY(crit_sect_); |
|
hlundin-webrtc
2015/08/10 11:30:01
Not this one.
|
| + size_t output_size_samples_ GUARDED_BY(crit_sect_); |
| + size_t decoder_frame_length_ GUARDED_BY(crit_sect_); |
| Modes last_mode_ GUARDED_BY(crit_sect_); |
| rtc::scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_); |
| size_t decoded_buffer_length_ GUARDED_BY(crit_sect_); |