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AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket

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Rename this accessor function to reflect its new, slightly changed
meaning. The reason for the change is that some codecs (iSAC) vary the
number of 10 ms frames from packet to packet, and so can't return a
truly constant value.

BUG=3926
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7556 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org 2014-10-29 08:38:50 +00:00
parent cfe3845b66
commit decd9306ae
6 changed files with 14 additions and 9 deletions
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13
webrtc/modules/audio_coding/codecs/audio_encoder.h
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@ -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,

2
webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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@ -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_;
}

2
webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
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@ -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,

2
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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@ -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_;
}

2
webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
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@ -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,

2
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
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@ -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