Change NetEq::GetAudio to use AudioFrame
With this change, NetEq now uses AudioFrame as output type, like the
surrounding functions in ACM and VoiceEngine already do.
The computational savings is probably slim, since one memcpy is
removed while another one is added (both in AcmReceiver::GetAudio).
More simplifications and clean-up will be done in
AcmReceiver::GetAudio in future CLs.
BUG=webrtc:5607
Review URL: https://codereview.webrtc.org/1750353002
Cr-Commit-Position: refs/heads/master@{#11874}
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 9369982..f899d07 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -148,13 +148,13 @@
return kOK;
}
-int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
- size_t* samples_per_channel, size_t* num_channels,
- NetEqOutputType* type) {
+int NetEqImpl::GetAudio(AudioFrame* audio_frame, NetEqOutputType* type) {
TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
rtc::CritScope lock(&crit_sect_);
- int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
- num_channels);
+ int error = GetAudioInternal(audio_frame);
+ RTC_DCHECK_EQ(
+ audio_frame->sample_rate_hz_,
+ rtc::checked_cast<int>(audio_frame->samples_per_channel_ * 100));
if (error != 0) {
error_code_ = error;
return kFail;
@@ -162,8 +162,7 @@
if (type) {
*type = LastOutputType();
}
- last_output_sample_rate_hz_ =
- rtc::checked_cast<int>(*samples_per_channel * 100);
+ last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_;
RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
last_output_sample_rate_hz_ == 16000 ||
last_output_sample_rate_hz_ == 32000 ||
@@ -739,10 +738,7 @@
return 0;
}
-int NetEqImpl::GetAudioInternal(size_t max_length,
- int16_t* output,
- size_t* samples_per_channel,
- size_t* num_channels) {
+int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame) {
PacketList packet_list;
DtmfEvent dtmf_event;
Operations operation;
@@ -857,16 +853,18 @@
// Extract data from |sync_buffer_| to |output|.
size_t num_output_samples_per_channel = output_size_samples_;
size_t num_output_samples = output_size_samples_ * sync_buffer_->Channels();
- if (num_output_samples > max_length) {
- LOG(LS_WARNING) << "Output array is too short. " << max_length << " < " <<
- output_size_samples_ << " * " << sync_buffer_->Channels();
- num_output_samples = max_length;
- num_output_samples_per_channel = max_length / sync_buffer_->Channels();
+ if (num_output_samples > AudioFrame::kMaxDataSizeSamples) {
+ LOG(LS_WARNING) << "Output array is too short. "
+ << AudioFrame::kMaxDataSizeSamples << " < "
+ << output_size_samples_ << " * "
+ << sync_buffer_->Channels();
+ num_output_samples = AudioFrame::kMaxDataSizeSamples;
+ num_output_samples_per_channel =
+ AudioFrame::kMaxDataSizeSamples / sync_buffer_->Channels();
}
- const size_t samples_from_sync =
- sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
- output);
- *num_channels = sync_buffer_->Channels();
+ sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
+ audio_frame);
+ audio_frame->sample_rate_hz_ = fs_hz_;
if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
// The sync buffer should always contain |overlap_length| samples, but now
// too many samples have been extracted. Reinstall the |overlap_length|
@@ -877,22 +875,22 @@
sync_buffer_->set_next_index(sync_buffer_->next_index() -
missing_lookahead_samples);
}
- if (samples_from_sync != output_size_samples_) {
- LOG(LS_ERROR) << "samples_from_sync (" << samples_from_sync
+ if (audio_frame->samples_per_channel_ != output_size_samples_) {
+ LOG(LS_ERROR) << "audio_frame->samples_per_channel_ ("
+ << audio_frame->samples_per_channel_
<< ") != output_size_samples_ (" << output_size_samples_
<< ")";
// TODO(minyue): treatment of under-run, filling zeros
- memset(output, 0, num_output_samples * sizeof(int16_t));
- *samples_per_channel = output_size_samples_;
+ memset(audio_frame->data_, 0, num_output_samples * sizeof(int16_t));
return kSampleUnderrun;
}
- *samples_per_channel = output_size_samples_;
// Should always have overlap samples left in the |sync_buffer_|.
RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
if (play_dtmf) {
- return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(), output);
+ return_value =
+ DtmfOverdub(dtmf_event, sync_buffer_->Channels(), audio_frame->data_);
}
// Update the background noise parameters if last operation wrote data