Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:
const int16_t* data() const;
int16_t* mutable_data();
- data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames.
- mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_.
These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation.
This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later.
BUG=webrtc:7343
TBR=henrika
Review-Url: https://codereview.webrtc.org/2750783004
Cr-Commit-Position: refs/heads/master@{#18543}
diff --git a/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
index 87cd61c..24d0719 100644
--- a/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -223,7 +223,7 @@
if (_playoutLengthSmpls == 0) {
return false;
}
- _pcmFile.Write10MsData(audioFrame.data_,
+ _pcmFile.Write10MsData(audioFrame.data(),
audioFrame.samples_per_channel_ * audioFrame.num_channels_);
return true;
}
diff --git a/webrtc/modules/audio_coding/test/PCMFile.cc b/webrtc/modules/audio_coding/test/PCMFile.cc
index 5d2d818..03d4fa7 100644
--- a/webrtc/modules/audio_coding/test/PCMFile.cc
+++ b/webrtc/modules/audio_coding/test/PCMFile.cc
@@ -125,11 +125,13 @@
channels = 2;
}
- int32_t payload_size = (int32_t) fread(audio_frame.data_, sizeof(uint16_t),
+ int32_t payload_size = (int32_t) fread(audio_frame.mutable_data(),
+ sizeof(uint16_t),
samples_10ms_ * channels, pcm_file_);
if (payload_size < samples_10ms_ * channels) {
+ int16_t* frame_data = audio_frame.mutable_data();
for (int k = payload_size; k < samples_10ms_ * channels; k++) {
- audio_frame.data_[k] = 0;
+ frame_data[k] = 0;
}
if (auto_rewind_) {
rewind(pcm_file_);
@@ -149,19 +151,20 @@
return samples_10ms_;
}
-void PCMFile::Write10MsData(AudioFrame& audio_frame) {
+void PCMFile::Write10MsData(const AudioFrame& audio_frame) {
if (audio_frame.num_channels_ == 1) {
if (!save_stereo_) {
- if (fwrite(audio_frame.data_, sizeof(uint16_t),
+ if (fwrite(audio_frame.data(), sizeof(uint16_t),
audio_frame.samples_per_channel_, pcm_file_) !=
static_cast<size_t>(audio_frame.samples_per_channel_)) {
return;
}
} else {
+ const int16_t* frame_data = audio_frame.data();
int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_];
for (size_t k = 0; k < audio_frame.samples_per_channel_; k++) {
- stereo_audio[k << 1] = audio_frame.data_[k];
- stereo_audio[(k << 1) + 1] = audio_frame.data_[k];
+ stereo_audio[k << 1] = frame_data[k];
+ stereo_audio[(k << 1) + 1] = frame_data[k];
}
if (fwrite(stereo_audio, sizeof(int16_t),
2 * audio_frame.samples_per_channel_, pcm_file_) !=
@@ -171,7 +174,7 @@
delete[] stereo_audio;
}
} else {
- if (fwrite(audio_frame.data_, sizeof(int16_t),
+ if (fwrite(audio_frame.data(), sizeof(int16_t),
audio_frame.num_channels_ * audio_frame.samples_per_channel_,
pcm_file_) !=
static_cast<size_t>(audio_frame.num_channels_ *
@@ -181,7 +184,8 @@
}
}
-void PCMFile::Write10MsData(int16_t* playout_buffer, size_t length_smpls) {
+void PCMFile::Write10MsData(const int16_t* playout_buffer,
+ size_t length_smpls) {
if (fwrite(playout_buffer, sizeof(uint16_t), length_smpls, pcm_file_) !=
length_smpls) {
return;
diff --git a/webrtc/modules/audio_coding/test/PCMFile.h b/webrtc/modules/audio_coding/test/PCMFile.h
index b5ced0b..63ab960 100644
--- a/webrtc/modules/audio_coding/test/PCMFile.h
+++ b/webrtc/modules/audio_coding/test/PCMFile.h
@@ -33,8 +33,8 @@
int32_t Read10MsData(AudioFrame& audio_frame);
- void Write10MsData(int16_t *playout_buffer, size_t length_smpls);
- void Write10MsData(AudioFrame& audio_frame);
+ void Write10MsData(const int16_t *playout_buffer, size_t length_smpls);
+ void Write10MsData(const AudioFrame& audio_frame);
uint16_t PayloadLength10Ms() const;
int32_t SamplingFrequency() const;
diff --git a/webrtc/modules/audio_coding/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/test/TestAllCodecs.cc
index 30f0226..12fe455 100644
--- a/webrtc/modules/audio_coding/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/test/TestAllCodecs.cc
@@ -457,7 +457,7 @@
ASSERT_FALSE(muted);
// Write output speech to file.
- outfile_b_.Write10MsData(audio_frame.data_,
+ outfile_b_.Write10MsData(audio_frame.data(),
audio_frame.samples_per_channel_);
// Update loop counter
diff --git a/webrtc/modules/audio_coding/test/TestRedFec.cc b/webrtc/modules/audio_coding/test/TestRedFec.cc
index 091cc84..4ec3ed1 100644
--- a/webrtc/modules/audio_coding/test/TestRedFec.cc
+++ b/webrtc/modules/audio_coding/test/TestRedFec.cc
@@ -464,7 +464,7 @@
bool muted;
EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
ASSERT_FALSE(muted);
- _outFileB.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
+ _outFileB.Write10MsData(audioFrame.data(), audioFrame.samples_per_channel_);
}
_inFileA.Rewind();
}
diff --git a/webrtc/modules/audio_coding/test/TestStereo.cc b/webrtc/modules/audio_coding/test/TestStereo.cc
index 3d8efe0..02bc141 100644
--- a/webrtc/modules/audio_coding/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/test/TestStereo.cc
@@ -806,7 +806,7 @@
// Write output speech to file
out_file_.Write10MsData(
- audio_frame.data_,
+ audio_frame.data(),
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
}
diff --git a/webrtc/modules/audio_coding/test/delay_test.cc b/webrtc/modules/audio_coding/test/delay_test.cc
index 846ac29..ce24493 100644
--- a/webrtc/modules/audio_coding/test/delay_test.cc
+++ b/webrtc/modules/audio_coding/test/delay_test.cc
@@ -209,7 +209,7 @@
acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
RTC_DCHECK(!muted);
out_file_b_.Write10MsData(
- audio_frame.data_,
+ audio_frame.data(),
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
received_ts = channel_a2b_->LastInTimestamp();
rtc::Optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp();
diff --git a/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
index 44ef9df..4fa4e52 100644
--- a/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -147,7 +147,7 @@
receive_acm_->PlayoutData10Ms(static_cast<int>(FLAGS_output_fs_hz),
&frame_, &muted);
ASSERT_FALSE(muted);
- fwrite(frame_.data_, sizeof(frame_.data_[0]),
+ fwrite(frame_.data(), sizeof(*frame_.data()),
frame_.samples_per_channel_ * frame_.num_channels_, pcm_out_fid_);
*action |= kAudioPlayedOut;
}
diff --git a/webrtc/modules/audio_coding/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc
index a558f1c..9f5720b 100644
--- a/webrtc/modules/audio_coding/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/test/opus_test.cc
@@ -262,7 +262,7 @@
// If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
EXPECT_EQ(480,
- resampler_.Resample10Msec(audio_frame.data_,
+ resampler_.Resample10Msec(audio_frame.data(),
audio_frame.sample_rate_hz_,
48000,
channels,
@@ -347,7 +347,7 @@
// Write output speech to file.
out_file_.Write10MsData(
- audio_frame.data_,
+ audio_frame.data(),
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
// Write stand-alone speech to file.
