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Add new methods to AudioEncoder interface

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The following three methods are added:
rtp_timestamp_rate_hz()
SetTargetBitrate()
SetProjectedPacketLossRate()

Default implementations are provided, and a few overrides are
implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new
methods to the underlying speech codec.

BUG=3926
COAUTHOR:kwiberg@webrtc.org

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34049004

Cr-Commit-Position: refs/heads/master@{#8171}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2015-01-27 18:24:45 +00:00
parent 5614cf16e7
commit 478cedc055
19 changed files with 175 additions and 30 deletions
Download Patch File Download Diff File Expand all files Collapse all files

18
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
View File

@ -74,6 +74,10 @@ int AudioEncoderCng::sample_rate_hz() const {
return sample_rate_hz_;
}
int AudioEncoderCng::rtp_timestamp_rate_hz() const {
return speech_encoder_->rtp_timestamp_rate_hz();
}
int AudioEncoderCng::num_channels() const {
return num_channels_;
}
@ -86,7 +90,17 @@ int AudioEncoderCng::Max10MsFramesInAPacket() const {
return speech_encoder_->Max10MsFramesInAPacket();
}
bool AudioEncoderCng::EncodeInternal(uint32_t timestamp,
void AudioEncoderCng::SetTargetBitrate(int bits_per_second) {
speech_encoder_->SetTargetBitrate(bits_per_second);
}
void AudioEncoderCng::SetProjectedPacketLossRate(double fraction) {
DCHECK_GE(fraction, 0.0);
DCHECK_LE(fraction, 1.0);
speech_encoder_->SetProjectedPacketLossRate(fraction);
}
bool AudioEncoderCng::EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
@ -99,7 +113,7 @@ bool AudioEncoderCng::EncodeInternal(uint32_t timestamp,
const int num_samples = sample_rate_hz() / 100 * num_channels();
if (speech_buffer_.empty()) {
CHECK_EQ(frames_in_buffer_, 0);
first_timestamp_in_buffer_ = timestamp;
first_timestamp_in_buffer_ = rtp_timestamp;
}
for (int i = 0; i < num_samples; ++i) {
speech_buffer_.push_back(audio[i]);

12
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
View File

@ -196,6 +196,18 @@ TEST_F(AudioEncoderCngTest, CheckFrameSizePropagation) {
EXPECT_EQ(17, cng_->Num10MsFramesInNextPacket());
}
TEST_F(AudioEncoderCngTest, CheckChangeBitratePropagation) {
CreateCng();
EXPECT_CALL(mock_encoder_, SetTargetBitrate(4711));
cng_->SetTargetBitrate(4711);
}
TEST_F(AudioEncoderCngTest, CheckProjectedPacketLossRatePropagation) {
CreateCng();
EXPECT_CALL(mock_encoder_, SetProjectedPacketLossRate(0.5));
cng_->SetProjectedPacketLossRate(0.5);
}
TEST_F(AudioEncoderCngTest, EncodeCallsVad) {
EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket())
.WillRepeatedly(Return(1));

5
webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
View File

@ -49,11 +49,14 @@ class AudioEncoderCng : public AudioEncoder {
virtual int sample_rate_hz() const OVERRIDE;
virtual int num_channels() const OVERRIDE;
int rtp_timestamp_rate_hz() const override;
virtual int Num10MsFramesInNextPacket() const OVERRIDE;
virtual int Max10MsFramesInAPacket() const OVERRIDE;
void SetTargetBitrate(int bits_per_second) override;
void SetProjectedPacketLossRate(double fraction) override;
protected:
virtual bool EncodeInternal(uint32_t timestamp,
virtual bool EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,