diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index f9cbe212f..45c0a855b 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -50,12 +50,17 @@ class AudioEncoder {
     return ret;
   }
 
-  // Returns the input sample rate in Hz, the number of input channels, and the
-  // number of 10 ms frames the encoder puts in one output packet. These are
-  // constants set at instantiation time.
+  // Return the input sample rate in Hz and the number of input channels.
+  // These are constants set at instantiation time.
   virtual int sample_rate_hz() const = 0;
   virtual int num_channels() const = 0;
-  virtual int num_10ms_frames_per_packet() const = 0;
+
+  // Returns the number of 10 ms frames the encoder will put in the next
+  // packet. This value may only change when Encode() outputs a packet; i.e.,
+  // 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;
 
  protected:
   virtual bool Encode(uint32_t timestamp,
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index ef22a2771..097e11f1b 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -48,7 +48,7 @@ int AudioEncoderPcm::sample_rate_hz() const {
 int AudioEncoderPcm::num_channels() const {
   return num_channels_;
 }
-int AudioEncoderPcm::num_10ms_frames_per_packet() const {
+int AudioEncoderPcm::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
index 8133987a9..f66829697 100644
--- a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
@@ -32,7 +32,7 @@ class AudioEncoderPcm : public AudioEncoder {
 
   virtual int sample_rate_hz() const OVERRIDE;
   virtual int num_channels() const OVERRIDE;
-  virtual int num_10ms_frames_per_packet() const OVERRIDE;
+  virtual int Num10MsFramesInNextPacket() const OVERRIDE;
 
  protected:
   virtual bool Encode(uint32_t timestamp,
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 0a3661f5b..6349b5c22 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -67,7 +67,7 @@ int AudioEncoderOpus::num_channels() const {
   return num_channels_;
 }
 
-int AudioEncoderOpus::num_10ms_frames_per_packet() const {
+int AudioEncoderOpus::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
index 7325b7e93..e2e5c73fc 100644
--- a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
@@ -32,7 +32,7 @@ class AudioEncoderOpus : public AudioEncoder {
 
   virtual int sample_rate_hz() const OVERRIDE;
   virtual int num_channels() const OVERRIDE;
-  virtual int num_10ms_frames_per_packet() const OVERRIDE;
+  virtual int Num10MsFramesInNextPacket() const OVERRIDE;
 
  protected:
   virtual bool Encode(uint32_t timestamp,
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index b6c6ba16d..5a3169611 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -142,7 +142,7 @@ class AudioDecoderTest : public ::testing::Test {
     size_t enc_len_bytes = 0;
     scoped_ptr<int16_t[]> interleaved_input(
         new int16_t[channels_ * input_len_samples]);
-    for (int i = 0; i < audio_encoder_->num_10ms_frames_per_packet(); ++i) {
+    for (int i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
       EXPECT_EQ(0u, enc_len_bytes);
 
       // Duplicate the mono input signal to however many channels the test