NetEq changes.
BUG=
R=henrik.lundin@webrtc.org, minyue@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5889 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/merge.h b/webrtc/modules/audio_coding/neteq4/merge.h
index f1f64e6..213b487 100644
--- a/webrtc/modules/audio_coding/neteq4/merge.h
+++ b/webrtc/modules/audio_coding/neteq4/merge.h
@@ -35,8 +35,8 @@
public:
Merge(int fs_hz, size_t num_channels, Expand* expand, SyncBuffer* sync_buffer)
: fs_hz_(fs_hz),
- fs_mult_(fs_hz_ / 8000),
num_channels_(num_channels),
+ fs_mult_(fs_hz_ / 8000),
timestamps_per_call_(fs_hz_ / 100),
expand_(expand),
sync_buffer_(sync_buffer),
@@ -44,6 +44,8 @@
assert(num_channels_ > 0);
}
+ virtual ~Merge() {}
+
// The main method to produce the audio data. The decoded data is supplied in
// |input|, having |input_length| samples in total for all channels
// (interleaved). The result is written to |output|. The number of channels
@@ -51,9 +53,15 @@
// de-interleaving |input|. The values in |external_mute_factor_array| (Q14)
// will be used to scale the audio, and is updated in the process. The array
// must have |num_channels_| elements.
- int Process(int16_t* input, size_t input_length,
- int16_t* external_mute_factor_array,
- AudioMultiVector* output);
+ virtual int Process(int16_t* input, size_t input_length,
+ int16_t* external_mute_factor_array,
+ AudioMultiVector* output);
+
+ virtual int RequiredFutureSamples();
+
+ protected:
+ const int fs_hz_;
+ const size_t num_channels_;
private:
static const int kMaxSampleRate = 48000;
@@ -87,9 +95,7 @@
int start_position, int input_length,
int expand_period) const;
- const int fs_hz_;
const int fs_mult_; // fs_hz_ / 8000.
- const size_t num_channels_;
const int timestamps_per_call_;
Expand* expand_;
SyncBuffer* sync_buffer_;